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Subject: [rtcweb] JS friendly codec compromise?
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After reviewing the discussion on the codec mandate (particularly h.261) I
was reminded of ORBX.js.

While I have my reservations regarding ORBX, the approach (a JS friendly
codec) would be more backwards compatible than h.261 and give better
performance.  If a "submarine" patent surfaces the codec can be altered and
clients can be "updated" on the next page refresh.

I should stress the "JS friendly" aspect of the codec, "VP8 like H.264 does
not parallelize as well via WebGL as ORBX does, so a "JS shim" is not going
to compete."  - Brendan Eich

Thank you,
-Zach Lym
P.S. For those who are not familiar with ORBX, it is a codec which is
decoded in JS and transmitted over websockets.  Their timeline (
http://render.otoy.com/newsblog/?p=317) mentions a FOSS encoder sometime
this summer, funding for a pure WebGL/CL encoder, and fee-free AMI's for
video transcoding.

--047d7bd75160cb24c904f6027dd7
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<div dir=3D"ltr"><div>After reviewing the discussion on the codec mandate (=
particularly h.261) I was reminded of ORBX.js.<br><br>While I have my reser=
vations regarding ORBX, the approach (a JS friendly codec) would be more ba=
ckwards compatible than h.261 and give better performance. &nbsp;If a &quot=
;submarine&quot; patent surfaces the codec can be altered and clients can b=
e &quot;updated&quot; on the next page refresh.<br>

<br>I should stress the &quot;JS friendly&quot; aspect of the codec, &quot;=
VP8 like H.264 does not parallelize as well via WebGL as ORBX does, so a &l=
dquo;JS shim&rdquo; is not going to compete.&quot; &nbsp;- Brendan Eich<br>=
<br>Thank you,<br>

-Zach Lym<br></div>P.S. For those who are not familiar with ORBX, it is a c=
odec which is decoded
 in JS and transmitted over websockets. &nbsp;Their timeline=20
(<a href=3D"http://render.otoy.com/newsblog/?p=3D317">http://render.otoy.co=
m/newsblog/?p=3D317</a>) mentions a FOSS encoder=20
sometime this summer, funding for a pure WebGL/CL encoder, and fee-free=20
AMI&#39;s for video transcoding.&nbsp; </div>

--047d7bd75160cb24c904f6027dd7--


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* Zach Lym wrote:
>While I have my reservations regarding ORBX, the approach (a JS friendly
>codec) would be more backwards compatible than h.261 and give better
>performance.  If a "submarine" patent surfaces the codec can be altered and
>clients can be "updated" on the next page refresh.

Which aspect of "performance" do you mean here, and could you share some
references to support the claim? References to a legal analysis would be
helpful aswell.
-- 
Björn Höhrmann · mailto:bjoern@hoehrmann.de · http://bjoern.hoehrmann.de
Am Badedeich 7 · Telefon: +49(0)160/4415681 · http://www.bjoernsworld.de
25899 Dagebüll · PGP Pub. KeyID: 0xA4357E78 · http://www.websitedev.de/ 


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> Which aspect of "performance" do you mean here, and could you share som=
e
> references to support the claim? References to a legal analysis would b=
e
> helpful aswell.

ORBX claims performance on par with H.264.  We all know that's probably
marketing hype, but it's hard to imagine not being able to outperform
H.261.  Even if the working group chose to hack together a custom codec
which is worse than H.261, an upgradeable client means we can update the
spec as performance improvements are made by the browser vendors.

I know of no formal legal analysis of ORBX, but being able to update the
decoder means that any submarine patent would have to knock out every
known video encoding technique.  Software patents are generally simple
to work around, codecs may be more so but Google's due-diligence team
thought that VP8 managed to pull it off.  They paid shut-up money to
make the FUD go away but if you are really that concerned about
potential patents we could partner with Daala.  Mozilla was smart with
Opus and patented their "new methods" so they could force
cross-licensing deals with the trolls.  They are doing the same with
Daala, so the FUD around a submarine threat  against Daala is no better
or worse than with H.264.  I don't know how well Daala's encoding could
be adapted to WebCL, but we could include some patented methods of the
codec.

Come to think of it, the Daala group is looking to recreate the
development process they had for Opus.  Skype and Mumble, both VOIP
clients, were able to push updates to client software silently,
providing the Opus team with a live test-bed which they used to
benchmark their various tweaks.  They are also actively seeking "killer
features" for the codec.  H.26X designed to compress movies, not
low-motion talking heads and screencasts:
https://www.youtube.com/watch?v=3Du3Yc5Hn9bhQ


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* Zach Lym wrote:
>> Which aspect of "performance" do you mean here, and could you share some
>> references to support the claim? References to a legal analysis would be
>> helpful aswell.
>
>ORBX claims performance on par with H.264.  We all know that's probably
>marketing hype, but it's hard to imagine not being able to outperform
>H.261.  Even if the working group chose to hack together a custom codec
>which is worse than H.261, an upgradeable client means we can update the
>spec as performance improvements are made by the browser vendors.

I meant performance characteristics like, say, battery life per second
of decoded video at H.261 resolutions and GPRS bitrates, to throw in a
couple of metrics. Or something like "ORBX needs more CPU but produces
smaller files at the same quality level". Maybe ORBX is cheap to decode
but expensive to encode, while with H.261 both is cheap? Comparisons
along those lines supported by references would be useful.
-- 
Björn Höhrmann · mailto:bjoern@hoehrmann.de · http://bjoern.hoehrmann.de
Am Badedeich 7 · Telefon: +49(0)160/4415681 · http://www.bjoernsworld.de
25899 Dagebüll · PGP Pub. KeyID: 0xA4357E78 · http://www.websitedev.de/ 


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Subject: Re: [rtcweb] JS friendly codec compromise?
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I cannot find the exact analysis you appear to be looking for.  You can=20
boot up some AMI instances and check for yourself, if you like. =20
Brendan Eich apparently saw a demo of ORBX streaming to an iPhone 4s in=20
Safari.

There is some discussion of performance on this thread:
https://brendaneich.com/2013/12/orbx-js-and-related-news/#div-comment-176=
27

And there is a whitepaper detailing ORBX2 here (although they have=20
already moved on to ORBX3):
http://aws.otoy.com/docs/ORBX2_Whitepaper.pdf

However, the larger point Brendan Eich makes, and what I am trying to=20
convey here, is that eschewing a fixed bit-stream format in favor of a=20
living standard would make such comparisons moot.  Clients could even=20
send the JS needed to decode their custom format to each other.  The=20
web is based on specifying interfaces, think of this as an interface=20
for codecs.

The only reason we are having this argument is because H.264 is etched=20
into custom hardware.  The above stats show performance comparable with=20
H.264 and they chalk it up to their ability to do everything on the=20
GPU/WebGL, which you can't do with H.264 or VP8.  As long as we can=20
offload to the GPU efficiently, there is no argument to be had.





On Tue Apr  1 16:40:10 2014, Bjoern Hoehrmann wrote:
> * Zach Lym wrote:
>>> Which aspect of "performance" do you mean here, and could you share s=
ome
>>> references to support the claim? References to a legal analysis would=
 be
>>> helpful aswell.
>>
>> ORBX claims performance on par with H.264.  We all know that's probabl=
y
>> marketing hype, but it's hard to imagine not being able to outperform
>> H.261.  Even if the working group chose to hack together a custom code=
c
>> which is worse than H.261, an upgradeable client means we can update t=
he
>> spec as performance improvements are made by the browser vendors.
>
> I meant performance characteristics like, say, battery life per second
> of decoded video at H.261 resolutions and GPRS bitrates, to throw in a
> couple of metrics. Or something like "ORBX needs more CPU but produces
> smaller files at the same quality level". Maybe ORBX is cheap to decode=

> but expensive to encode, while with H.261 both is cheap? Comparisons
> along those lines supported by references would be useful.


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On 1 April 2014 18:39, Zach Lym <zachlym@indolering.com> wrote:
> However, the larger point Brendan Eich makes, and what I am trying to
> convey here, is that eschewing a fixed bit-stream format in favor of a
> living standard would make such comparisons moot.  Clients could even
> send the JS needed to decode their custom format to each other.  The
> web is based on specifying interfaces, think of this as an interface
> for codecs.

Unfortunately, a JS-implemented codec does not play well with isolated
sessions, otherwise we can't establish true private calling.  In that
context, we still need something that the browser can control.
Perhaps with the right sandbox it could work, but I think that it's a
little too early to be considering that option.  Soon, perhaps.


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My apologies, I am rather ignorant of the gritty implementation details.

Maybe ORBX.js will be of some use, I guess we will see what the summer
brings.

I'll get back to my corner of expertise,
-Zach
On Wednesday, April 2, 2014, Martin Thomson <martin.thomson@gmail.com>
wrote:

> On 1 April 2014 18:39, Zach Lym <zachlym@indolering.com <javascript:;>>
> wrote:
> > However, the larger point Brendan Eich makes, and what I am trying to
> > convey here, is that eschewing a fixed bit-stream format in favor of a
> > living standard would make such comparisons moot.  Clients could even
> > send the JS needed to decode their custom format to each other.  The
> > web is based on specifying interfaces, think of this as an interface
> > for codecs.
>
> Unfortunately, a JS-implemented codec does not play well with isolated
> sessions, otherwise we can't establish true private calling.  In that
> context, we still need something that the browser can control.
> Perhaps with the right sandbox it could work, but I think that it's a
> little too early to be considering that option.  Soon, perhaps.
>


-- 
Thank you,
-Zach Lym

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My apologies, I am rather ignorant of the gritty implementation details. =
=A0<br><br>Maybe ORBX.js<span></span> will be of some use, I guess we will =
see what the summer brings.<div><br><div>I&#39;ll get back to my corner of =
expertise,<br>
</div><div>-Zach</div><div>On Wednesday, April 2, 2014, Martin Thomson &lt;=
<a href=3D"mailto:martin.thomson@gmail.com">martin.thomson@gmail.com</a>&gt=
; wrote:<br><blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;bo=
rder-left:1px #ccc solid;padding-left:1ex">
On 1 April 2014 18:39, Zach Lym &lt;<a href=3D"javascript:;" onclick=3D"_e(=
event, &#39;cvml&#39;, &#39;zachlym@indolering.com&#39;)">zachlym@indolerin=
g.com</a>&gt; wrote:<br>
&gt; However, the larger point Brendan Eich makes, and what I am trying to<=
br>
&gt; convey here, is that eschewing a fixed bit-stream format in favor of a=
<br>
&gt; living standard would make such comparisons moot. =A0Clients could eve=
n<br>
&gt; send the JS needed to decode their custom format to each other. =A0The=
<br>
&gt; web is based on specifying interfaces, think of this as an interface<b=
r>
&gt; for codecs.<br>
<br>
Unfortunately, a JS-implemented codec does not play well with isolated<br>
sessions, otherwise we can&#39;t establish true private calling. =A0In that=
<br>
context, we still need something that the browser can control.<br>
Perhaps with the right sandbox it could work, but I think that it&#39;s a<b=
r>
little too early to be considering that option. =A0Soon, perhaps.<br>
</blockquote></div></div><br><br>-- <br>Thank you,<div>-Zach Lym</div><br>

--047d7b6d7fb416558904f611d527--


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To: "'Harald Alvestrand'" <harald@alvestrand.no>, <rtcweb@ietf.org>, "'Magnus Westerlund'" <magnus.westerlund@ericsson.com>
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Subject: Re: [rtcweb] [tram] Payload Types assignments
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Harald,=20

=20

I think my intention was misunderstood. It was not to change the =
DTLS/SCTP
structure or format of the data channel (It was to insert the traffic
bandwidth and type information between the UDP and DTLS headers.).

=20

However, having dug deeper into the data channel (and also found the
quality/priority obstacles introduced by SCTP with the p2p file transfer
(see next email), a better way to introduce the traffic info would be to =
use
a new DTLS ContentType (e.g. 24 instead of current 23) where the content
would be the unencrypted traffic info (preferable the same as in the RTP
header extension[1]) followed by the SCTP as it is today (where it today
comes directly in the DTLS as the only content).:

=20

TODAY:

Content Type: 23 (1 byte) content type 23 means =93application data=94

DTLS version:  0xfeff (2 byte)

Epoch: 0001 (2 byte)

Sequence number (e.g. 00 00 00 00 00 01) 6 byte

Length (e.g. 01 c0) 2 byte

Encrypted application data: ...... =3D the SCTP

=20

SUGGESTION WITH ADDED TRAFFIC INFO FOR LOWER LEVELS:=20

Content Type: 24 (1 byte) content type 24 meaning =93application data =
preceded
by traffic info=94

DTLS version:  0xfeff (2 byte)

Epoch: 0001 (2 byte)

Sequence number (e.g. 00 00 00 00 00 01) 6 byte

Length (e.g. 01 c0) 2 byte: length of the encrypted application data =
after
the new header extension

TRAFFIC INFO, Same as suggested for the RTP Header extension:=20

Encrypted application data: ...... ...... =3D the SCTP unchanged

=20

Shouldn=92t this be doable?

(As you see from next email, this kind of information is also required =
to
make the data channel work, separating prioritized data chunks from a =
file
transfer data channel that must NOT be prioritized.)

=20

/Karl

=20

=20

=20

=20

Fr=E5n: rtcweb [mailto:rtcweb-bounces@ietf.org] F=F6r Harald Alvestrand
Skickat: den 17 mars 2014 16:38
Till: rtcweb@ietf.org
=C4mne: Re: [rtcweb] [tram] Payload Types assignments

=20

On 03/13/2014 10:02 AM, Karl Stahl wrote:

For the =93mission to bring quality to real-time traffic over our best =
effort
Internet=94 I have started
<http://www.ietf.org/mail-archive/web/tram/current/msg00331.html>
http://www.ietf.org/mail-archive/web/tram/current/msg00331.html

[tram] [rtcweb] The way to "Interfacing to QoS", A level 3-5
IP/IETF/WebRTC-thing how to interface to lower level's QoS-stuff [1] [2]

=20

Hi Magnus,

=20

With the response just sent to P=E5l
http://www.ietf.org/mail-archive/web/rtcweb/current/msg11846.html, =
skipping
all the QoS methods and functions that really does not belong here when
creating an =93Interface to QoS=94, I only see the suggestion of =93data =
channel
information=94, i.e. also marking each data channel packet with traffic
bandwidth and type information  to be relevant and interesting. I =
checked
with one of our developers and I think the same traffic information as =
for
RTP can be transferred in data channel packets as follows:

=20

The RTP header has the following format (RFC 3550):

    0                   1                   2                   3

    0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1

   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+

   |V=3D2|P|X|  CC   |M|     PT      |       sequence number         |

   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+

   |                           timestamp                           |

   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+

   |           synchronization source (SSRC) identifier            |

   =
+=3D+=3D+=3D+=3D+=3D+=3D+=3D+=3D+=3D+=3D+=3D+=3D+=3D+=3D+=3D+=3D+=3D+=3D+=
=3D+=3D+=3D+=3D+=3D+=3D+=3D+=3D+=3D+=3D+=3D+=3D+=3D+=3D+

   |            contributing source (CSRC) identifiers             |

   |                             ....                              |

   =
+=3D+=3D+=3D+=3D+=3D+=3D+=3D+=3D+=3D+=3D+=3D+=3D+=3D+=3D+=3D+=3D+=3D+=3D+=
=3D+=3D+=3D+=3D+=3D+=3D+=3D+=3D+=3D+=3D+=3D+=3D+=3D+=3D+

   |            RFC5285 header extension ...                      |

   |     2 bytes Max Bandwidth     |   2 bytes Traffic Type        |

   |                             ....                              |

RTP Payload (often encrypted)

=20

A Data Channel header could have the following format:

    0                   1                   2                   3

    0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1

   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+

   |   Data Channel Magic Cookie   |      (sequence number)        |

   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+

   |                           timestamp                           |

   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+

   |           synchronization source (SSRC) identifier            |

   =
+=3D+=3D+=3D+=3D+=3D+=3D+=3D+=3D+=3D+=3D+=3D+=3D+=3D+=3D+=3D+=3D+=3D+=3D+=
=3D+=3D+=3D+=3D+=3D+=3D+=3D+=3D+=3D+=3D+=3D+=3D+=3D+=3D+

   |            contributing source (CSRC) identifiers             |

   |                             ....                              |

   =
+=3D+=3D+=3D+=3D+=3D+=3D+=3D+=3D+=3D+=3D+=3D+=3D+=3D+=3D+=3D+=3D+=3D+=3D+=
=3D+=3D+=3D+=3D+=3D+=3D+=3D+=3D+=3D+=3D+=3D+=3D+=3D+=3D+

   |            RFC5285 header extension ...                      |

   |     2 bytes Max Bandwidth     |   2 bytes Traffic Type        |

   |                             ....                              |

  DTLS Header 13 bytes

  SCTP

=20

The RFC5285 header extension is not in a fixed position in any of these
cases, but still trivial to find (especially when in  a TURN flow). This
could be introduced in
http://tools.ietf.org/html/draft-ietf-rtcweb-data-channel-07#section-5=20

by making room for these bytes after the UDP header, but before the DTLS
header.

=20

What do you think?



No.

We've chosen to make Data Channel a protocol within DTLS, and not use =
any
unencrypted headers for it. Neither does the data channel have any =
concept
resembling SSRC, CSRC or timestamp. Changing to an RTP-based format such =
as
the one you propose would be a drastic revision of the spec.

There's no reason to believe such a revision at this time has any =
traction
in the WG.


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<o:idmap v:ext=3D"edit" data=3D"1" />
</o:shapelayout></xml><![endif]--></head><body bgcolor=3Dwhite lang=3DSV =
link=3Dblue vlink=3Dpurple><div class=3DWordSection1><p =
class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'>Ha=
rald, <o:p></o:p></span></p><p class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'><o=
:p>&nbsp;</o:p></span></p><p class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'>I =
think my intention was misunderstood. It was not to change the DTLS/SCTP =
structure or format of the data channel (It was to insert the =
</span><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'>tr=
affic bandwidth and type information between the UDP and DTLS =
headers.).<o:p></o:p></span></p><p class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'><o=
:p>&nbsp;</o:p></span></p><p class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'>Ho=
wever, having dug deeper into the data channel (and also found the =
quality/priority obstacles introduced by SCTP with the p2p file transfer =
(see next email), a better way to introduce the traffic info would be to =
use a new DTLS ContentType (e.g. 24 instead of current 23) where the =
content would be the unencrypted traffic info (preferable the same as in =
the RTP header extension[1]) followed by the SCTP as it is today (where =
it today comes directly in the DTLS as the only =
content).:<o:p></o:p></span></p><p class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'><o=
:p>&nbsp;</o:p></span></p><p class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'>TO=
DAY:<o:p></o:p></span></p><p class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-size:12.0pt'>Content Type: 23 (1 byte) content type 23 =
means &#8220;application data&#8221;<o:p></o:p></span></p><p =
class=3DMsoNormal><span style=3D'font-size:12.0pt'>DTLS version:&nbsp; =
0xfeff (2 byte)<o:p></o:p></span></p><p class=3DMsoNormal><span =
style=3D'font-size:12.0pt'>Epoch: 0001 (2 byte)<o:p></o:p></span></p><p =
class=3DMsoNormal><span lang=3DEN-US style=3D'font-size:12.0pt'>Sequence =
number (e.g. 00 00 00 00 00 01) 6 byte</span><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Tahoma","sans-serif"'><o:p></o:p><=
/span></p><p class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-size:12.0pt'>Length (e.g. 01 c0) 2 byte</span><span =
lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Tahoma","sans-serif"'><o:p></o:p><=
/span></p><p class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-size:12.0pt'>Encrypted application data: ...... =
</span><span lang=3DEN-US style=3D'font-size:12.0pt;color:blue'>=3D the =
SCTP</span><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Tahoma","sans-serif"'><o:p></o:p><=
/span></p><p class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Tahoma","sans-serif"'>&nbsp;<o:p><=
/o:p></span></p><p class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Tahoma","sans-serif";color:blue'>S=
UGGESTION WITH ADDED TRAFFIC INFO FOR LOWER LEVELS:</span><span =
lang=3DEN-US style=3D'font-size:12.0pt'> <o:p></o:p></span></p><p =
class=3DMsoNormal><span lang=3DEN-US style=3D'font-size:12.0pt'>Content =
Type: <strong><span =
style=3D'font-family:"Calibri","sans-serif"'>24</span></strong> (1 byte) =
content type 24 meaning &#8220;application data preceded by traffic =
info&#8221;</span><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Tahoma","sans-serif"'><o:p></o:p><=
/span></p><p class=3DMsoNormal><span style=3D'font-size:12.0pt'>DTLS =
version:&nbsp; 0xfeff (2 byte)</span><span =
style=3D'font-size:10.0pt;font-family:"Tahoma","sans-serif"'><o:p></o:p><=
/span></p><p class=3DMsoNormal><span style=3D'font-size:12.0pt'>Epoch: =
0001 (2 byte)</span><span =
style=3D'font-size:10.0pt;font-family:"Tahoma","sans-serif"'><o:p></o:p><=
/span></p><p class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-size:12.0pt'>Sequence number (e.g. 00 00 00 00 00 01) 6 =
byte</span><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Tahoma","sans-serif"'><o:p></o:p><=
/span></p><p class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-size:12.0pt'>Length (e.g. 01 c0) 2 byte: length of the =
encrypted application data after the new header extension</span><span =
lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Tahoma","sans-serif"'><o:p></o:p><=
/span></p><p class=3DMsoNormal><strong><span lang=3DEN-US =
style=3D'font-size:12.0pt;font-family:"Calibri","sans-serif"'>TRAFFIC =
INFO, Same as suggested for the RTP Header extension: =
</span></strong><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Tahoma","sans-serif"'><o:p></o:p><=
/span></p><p class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-size:12.0pt'>Encrypted application data: ...... ...... =
</span><span lang=3DEN-US style=3D'font-size:12.0pt;color:blue'>=3D the =
SCTP unchanged</span><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Tahoma","sans-serif"'><o:p></o:p><=
/span></p><p class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'><o=
:p>&nbsp;</o:p></span></p><p class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'>Sh=
ouldn&#8217;t this be doable?<o:p></o:p></span></p><p =
class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'>(A=
s you see from next email, this kind of information is also required to =
make the data channel work, separating prioritized data chunks from a =
file transfer data channel that must NOT be =
prioritized.)<o:p></o:p></span></p><p class=3DMsoNormal><span =
lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'><o=
:p>&nbsp;</o:p></span></p><p class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'>/K=
arl<o:p></o:p></span></p><p class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'><o=
:p>&nbsp;</o:p></span></p><p class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'><o=
:p>&nbsp;</o:p></span></p><p class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'><o=
:p>&nbsp;</o:p></span></p><p class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'><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";color:windowt=
ext'>Fr=E5n:</span></b><span =
style=3D'font-size:10.0pt;font-family:"Tahoma","sans-serif";color:windowt=
ext'> rtcweb [mailto:rtcweb-bounces@ietf.org] <b>F=F6r </b>Harald =
Alvestrand<br><b>Skickat:</b> den 17 mars 2014 16:38<br><b>Till:</b> =
rtcweb@ietf.org<br><b>=C4mne:</b> Re: [rtcweb] [tram] Payload Types =
assignments<o:p></o:p></span></p></div></div><p =
class=3DMsoNormal><o:p>&nbsp;</o:p></p><div><p class=3DMsoNormal>On =
03/13/2014 10:02 AM, Karl Stahl wrote:<o:p></o:p></p></div><blockquote =
style=3D'margin-top:5.0pt;margin-bottom:5.0pt'><p =
class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'>Fo=
r the</span><span lang=3DEN-US =
style=3D'font-family:"Arial","sans-serif";color:blue'> </span><span =
lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'>&#=
8220;</span><span lang=3DEN-US>mission to bring quality to real-time =
traffic over our best effort Internet&#8221;</span><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'>&n=
bsp;I have started </span><span =
style=3D'font-size:10.0pt;font-family:"Tahoma","sans-serif"'><a =
href=3D"http://www.ietf.org/mail-archive/web/tram/current/msg00331.html">=
<span =
lang=3DEN-US>http://www.ietf.org/mail-archive/web/tram/current/msg00331.h=
tml</span></a></span><o:p></o:p></p><p class=3DMsoNormal><span =
lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Tahoma","sans-serif"'>[tram] =
[rtcweb] The way to &quot;Interfacing to QoS&quot;, A level 3-5 =
IP/IETF/WebRTC-thing how to interface to lower level's QoS-stuff =
</span><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Tahoma","sans-serif";color:blue'>[=
1] [2]</span><o:p></o:p></p><p class=3DMsoPlainText><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'>&n=
bsp;</span><o:p></o:p></p><p class=3DMsoPlainText><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'>Hi=
 Magnus,</span><o:p></o:p></p><p class=3DMsoPlainText><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'>&n=
bsp;</span><o:p></o:p></p><p class=3DMsoPlainText><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'>Wi=
th the response just sent to P=E5l <a =
href=3D"http://www.ietf.org/mail-archive/web/rtcweb/current/msg11846.html=
">http://www.ietf.org/mail-archive/web/rtcweb/current/msg11846.html</a>, =
skipping all the QoS methods and functions that really does not belong =
here when creating an &#8220;Interface to QoS&#8221;, I only see the =
suggestion of &#8220;</span><span lang=3DEN-US>data channel =
information</span><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'>&#=
8221;, i.e. also marking each data channel packet with traffic bandwidth =
and type information &nbsp;to be relevant and interesting. I checked =
with one of our developers and I think the same traffic information as =
for RTP can be transferred in data channel packets as =
follows:</span><o:p></o:p></p><p class=3DMsoPlainText><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'>&n=
bsp;</span><o:p></o:p></p><p class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-size:10.0pt'>The RTP header has the following format (RFC =
3550):</span><o:p></o:p></p><p class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-size:10.0pt'>&nbsp;&nbsp;&nbsp; =
0&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;=
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; =
1&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;=
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; =
2&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;=
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; 3</span><o:p></o:p></p><p =
class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-size:10.0pt'>&nbsp;&nbsp;&nbsp; 0 1 2 3 4 5 6 7 8 9 0 1 2 =
3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1</span><o:p></o:p></p><p =
class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-size:10.0pt'>&nbsp;&nbsp; =
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+</span><=
o:p></o:p></p><p class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-size:10.0pt'>&nbsp;&nbsp; |V=3D2|P|X|&nbsp; CC&nbsp;&nbsp; =
|M|&nbsp;&nbsp;&nbsp;&nbsp; PT&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; =
|&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; sequence =
number&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; =
|</span><o:p></o:p></p><p class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-size:10.0pt'>&nbsp;&nbsp; =
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+</span><=
o:p></o:p></p><p class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-size:10.0pt'>&nbsp;&nbsp; =
|&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;=
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&=
nbsp;&nbsp; =
timestamp&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbs=
p;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp=
;&nbsp;&nbsp;&nbsp; |</span><o:p></o:p></p><p class=3DMsoNormal><span =
lang=3DEN-US style=3D'font-size:10.0pt'>&nbsp;&nbsp; =
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+</span><=
o:p></o:p></p><p class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-size:10.0pt'>&nbsp;&nbsp; =
|&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; =
synchronization source (SSRC) =
identifier&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nb=
sp; |</span><o:p></o:p></p><p class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-size:10.0pt'>&nbsp;&nbsp; =
+=3D+=3D+=3D+=3D+=3D+=3D+=3D+=3D+=3D+=3D+=3D+=3D+=3D+=3D+=3D+=3D+=3D+=3D+=
=3D+=3D+=3D+=3D+=3D+=3D+=3D+=3D+=3D+=3D+=3D+=3D+=3D+=3D+</span><o:p></o:p=
></p><p class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-size:10.0pt'>&nbsp;&nbsp; =
|&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; =
contributing source (CSRC) =
identifiers&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&n=
bsp;&nbsp; |</span><o:p></o:p></p><p class=3DMsoNormal><span =
lang=3DEN-US style=3D'font-size:10.0pt'>&nbsp;&nbsp; =
|&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;=
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&=
nbsp;&nbsp;&nbsp;&nbsp; =
....&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nb=
sp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbs=
p;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; |</span><o:p></o:p></p><p =
class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-size:10.0pt'>&nbsp;&nbsp; =
+=3D+=3D+=3D+=3D+=3D+=3D+=3D+=3D+=3D+=3D+=3D+=3D+=3D+=3D+=3D+=3D+=3D+=3D+=
=3D+=3D+=3D+=3D+=3D+=3D+=3D+=3D+=3D+=3D+=3D+=3D+=3D+=3D+</span><o:p></o:p=
></p><p class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-size:10.0pt'>&nbsp;&nbsp; =
|&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; =
RFC5285 header extension =
...&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbs=
p;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; =
|</span><o:p></o:p></p><p class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-size:10.0pt'>&nbsp;&nbsp; |&nbsp;&nbsp;&nbsp;&nbsp; 2 =
bytes Max Bandwidth&nbsp;&nbsp;&nbsp;&nbsp; |&nbsp;&nbsp; 2 bytes =
Traffic Type&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; =
|</span><o:p></o:p></p><p class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-size:10.0pt'>&nbsp;&nbsp; =
|&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;=
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&=
nbsp;&nbsp;&nbsp;&nbsp; =
....&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nb=
sp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbs=
p;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; |</span><o:p></o:p></p><p =
class=3DMsoNormal><span lang=3DEN-US style=3D'font-size:10.0pt'>RTP =
Payload (often encrypted)</span><o:p></o:p></p><p =
class=3DMsoNormal><span lang=3DEN-US>&nbsp;</span><o:p></o:p></p><p =
class=3DMsoNormal><span lang=3DEN-US style=3D'font-size:10.0pt'>A Data =
Channel header could have the following format:</span><o:p></o:p></p><p =
class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-size:10.0pt'>&nbsp;&nbsp;&nbsp; =
0&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;=
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; =
1&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;=
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; =
2&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;=
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; 3</span><o:p></o:p></p><p =
class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-size:10.0pt'>&nbsp;&nbsp;&nbsp; 0 1 2 3 4 5 6 7 8 9 0 1 2 =
3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1</span><o:p></o:p></p><p =
class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-size:10.0pt'>&nbsp;&nbsp; =
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+</span><=
o:p></o:p></p><p class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-size:10.0pt'>&nbsp;&nbsp; |&nbsp;&nbsp; Data Channel Magic =
Cookie&nbsp;&nbsp; |&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;(sequence =
number)&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; =
|</span><o:p></o:p></p><p class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-size:10.0pt'>&nbsp;&nbsp; =
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+</span><=
o:p></o:p></p><p class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-size:10.0pt'>&nbsp;&nbsp; =
|&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;=
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&=
nbsp;&nbsp; =
timestamp&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbs=
p;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp=
;&nbsp;&nbsp;&nbsp; |</span><o:p></o:p></p><p class=3DMsoNormal><span =
lang=3DEN-US style=3D'font-size:10.0pt'>&nbsp;&nbsp; =
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+</span><=
o:p></o:p></p><p class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-size:10.0pt'>&nbsp;&nbsp; =
|&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; =
synchronization source (SSRC) =
identifier&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nb=
sp; |</span><o:p></o:p></p><p class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-size:10.0pt'>&nbsp;&nbsp; =
+=3D+=3D+=3D+=3D+=3D+=3D+=3D+=3D+=3D+=3D+=3D+=3D+=3D+=3D+=3D+=3D+=3D+=3D+=
=3D+=3D+=3D+=3D+=3D+=3D+=3D+=3D+=3D+=3D+=3D+=3D+=3D+=3D+</span><o:p></o:p=
></p><p class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-size:10.0pt'>&nbsp;&nbsp; =
|&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; =
contributing source (CSRC) =
identifiers&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&n=
bsp;&nbsp; |</span><o:p></o:p></p><p class=3DMsoNormal><span =
lang=3DEN-US style=3D'font-size:10.0pt'>&nbsp;&nbsp; =
|&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;=
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&=
nbsp;&nbsp;&nbsp;&nbsp; =
....&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nb=
sp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbs=
p;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; |</span><o:p></o:p></p><p =
class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-size:10.0pt'>&nbsp;&nbsp; =
+=3D+=3D+=3D+=3D+=3D+=3D+=3D+=3D+=3D+=3D+=3D+=3D+=3D+=3D+=3D+=3D+=3D+=3D+=
=3D+=3D+=3D+=3D+=3D+=3D+=3D+=3D+=3D+=3D+=3D+=3D+=3D+=3D+</span><o:p></o:p=
></p><p class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-size:10.0pt'>&nbsp;&nbsp; =
|&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; =
RFC5285 header extension =
...&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbs=
p;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; =
|</span><o:p></o:p></p><p class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-size:10.0pt'>&nbsp;&nbsp; |&nbsp;&nbsp;&nbsp;&nbsp; 2 =
bytes Max Bandwidth&nbsp;&nbsp;&nbsp;&nbsp; |&nbsp;&nbsp; 2 bytes =
Traffic Type&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; =
|</span><o:p></o:p></p><p class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-size:10.0pt'>&nbsp;&nbsp; =
|&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;=
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&=
nbsp;&nbsp;&nbsp;&nbsp; =
....&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nb=
sp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbs=
p;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; |</span><o:p></o:p></p><p =
class=3DMsoNormal><span lang=3DEN-US style=3D'font-size:10.0pt'>&nbsp; =
DTLS Header 13 bytes</span><o:p></o:p></p><p class=3DMsoNormal><span =
lang=3DEN-US style=3D'font-size:10.0pt'>&nbsp; =
SCTP</span><o:p></o:p></p><p class=3DMsoNormal><span =
lang=3DEN-US>&nbsp;</span><o:p></o:p></p><p class=3DMsoNormal><span =
lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'>Th=
e RFC5285 header extension is not in a fixed position in any of these =
cases, but still trivial to find (especially when in&nbsp; a TURN flow). =
This could be introduced in <a =
href=3D"http://tools.ietf.org/html/draft-ietf-rtcweb-data-channel-07#sect=
ion-5">http://tools.ietf.org/html/draft-ietf-rtcweb-data-channel-07#secti=
on-5</a> </span><o:p></o:p></p><p class=3DMsoPlainText><span =
lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'>by=
 making room for these bytes after the UDP header, but before the DTLS =
header.</span><o:p></o:p></p><p class=3DMsoPlainText><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'>&n=
bsp;</span><o:p></o:p></p><p class=3DMsoPlainText><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'>Wh=
at do you think?</span><o:p></o:p></p></blockquote><p class=3DMsoNormal =
style=3D'margin-bottom:12.0pt'><span =
style=3D'font-size:12.0pt;font-family:"Times New =
Roman","serif"'><br><br>No.<br><br>We've chosen to make Data Channel a =
protocol within DTLS, and not use any unencrypted headers for it. =
Neither does the data channel have any concept resembling SSRC, CSRC or =
timestamp. Changing to an RTP-based format such as the one you propose =
would be a drastic revision of the spec.<br><br>There's no reason to =
believe such a revision at this time has any traction in the =
WG.<o:p></o:p></span></p></div></body></html>
------=_NextPart_000_0099_01CF4F63.C64AA110--


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From: Ted Hardie <ted.ietf@gmail.com>
To: "rtcweb@ietf.org" <rtcweb@ietf.org>, Magnus Westerlund <magnus.westerlund@ericsson.com>,  Cullen Jennings <fluffy@cisco.com>,  =?ISO-8859-1?Q?Stefan_H=E5kansson_LK?= <stefan.lk.hakansson@ericsson.com>,  Harald Tveit Alvestrand <harald@alvestrand.no>
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Subject: [rtcweb] Joint Interim meeting May 19-21, 2014
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The upcoming joint interim meeting will be held in Washington, D.C. at the
Google offices at 1101 New York Ave NW, Washington, DC on May 19-21, 2014.
 The office is about two blocks from Metro
Centro<http://www.wmata.com/rail/station_detail.cfm?station_id=1>.
A doodle to track participation for the meeting in order to prepare badges
is here:

http://doodle.com/qewq4xvszbc6d4sn

Please complete the form by May 16th, 2014 so that the badges will be
available.  For travel arrangements, please note that the final day's
ending time, 16:30, is intended to be a hard stop.

regards,

Ted, Cullen, Magnus

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<div dir=3D"ltr"><div class=3D"gmail_default" style=3D"font-family:georgia,=
serif"><div class=3D"gmail_default"><span style=3D"font-family:tahoma,sans-=
serif">The upcoming joint interim meeting will be held in Washington, D.C. =
at the Google offices at <span style=3D"text-indent:0px;letter-spacing:norm=
al;font-variant:normal;text-align:start;font-style:normal;display:inline!im=
portant;font-weight:normal;float:none;line-height:normal;color:rgb(34,34,34=
);text-transform:none;font-size:12.8px;white-space:normal;word-spacing:0px"=
>1101 New York Ave NW,<span>=A0</span></span><span style=3D"background-colo=
r:rgb(255,255,204);color:rgb(34,34,34);font-size:12.8px;font-style:normal;f=
ont-variant:normal;font-weight:normal;letter-spacing:normal;line-height:nor=
mal;text-align:start;text-indent:0px;text-transform:none;white-space:normal=
;word-spacing:0px">Washington</span><span style=3D"text-indent:0px;letter-s=
pacing:normal;font-variant:normal;text-align:start;font-style:normal;displa=
y:inline!important;font-weight:normal;float:none;line-height:normal;color:r=
gb(34,34,34);text-transform:none;font-size:12.8px;white-space:normal;word-s=
pacing:0px">, DC on May 19-21, 2014. =A0The office is about two blocks from=
<span>=A0</span></span><a href=3D"http://www.wmata.com/rail/station_detail.=
cfm?station_id=3D1" style=3D"text-indent:0px;letter-spacing:normal;font-var=
iant:normal;text-align:start;font-style:normal;font-weight:normal;line-heig=
ht:normal;color:rgb(17,85,204);text-transform:none;font-size:12.8px;white-s=
pace:normal;word-spacing:0px" target=3D"_blank">Metro Centro</a><span style=
=3D"text-indent:0px;letter-spacing:normal;font-variant:normal;text-align:st=
art;font-style:normal;display:inline!important;font-weight:normal;float:non=
e;line-height:normal;color:rgb(34,34,34);text-transform:none;font-size:12.8=
px;white-space:normal;word-spacing:0px">.=A0 A doodle to track participatio=
n for the meeting in order to prepare badges is here:<br>

<br><a href=3D"http://doodle.com/qewq4xvszbc6d4sn">http://doodle.com/qewq4x=
vszbc6d4sn</a><br><br></span></span></div><span style=3D"text-indent:0px;le=
tter-spacing:normal;font-variant:normal;text-align:start;font-style:normal;=
display:inline!important;font-weight:normal;float:none;line-height:normal;c=
olor:rgb(34,34,34);text-transform:none;font-size:12.8px;white-space:normal;=
font-family:arial,sans-serif;word-spacing:0px"><span style=3D"font-family:t=
ahoma,sans-serif">Please complete the form by May 16th, 2014 so that the ba=
dges will be available.=A0 For travel arrangements, please note that the fi=
nal day&#39;s ending time, 16:30, is intended to be a hard stop.<br>
<br>regards,<br></span><br>Ted, Cullen, Magnus</span></div></div>

--089e010d97f81e5d4304f62563cf--


From nobody Thu Apr  3 09:31:02 2014
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To: "'Matthew Kaufman \(SKYPE\)'" <matthew.kaufman@skype.net>, "'Harald Alvestrand'" <harald@alvestrand.no>, "'Magnus Westerlund'" <magnus.westerlund@ericsson.com>
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Subject: Re: [rtcweb] Some language on "prioritization"
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The questions raised here don't have simple "local or endpoint answers".
There are overall and basic things to be made clear first.=20
Combining P2P file transfer over the rtcweb data channels with=20
prioritized traffic chunks, brings further complications to the=20
prioritization/QoS issue. This cannot be handled/done by endpoint or=20
applications without considering the transport network (The congestion=20
to handle by prioritization is in the network - NOT in the endpoints)!=20

There are below (A), (B) and (C) - to get us back to normal - then there =

is (D) to improve - to allow the network level to give us the=20
prioritization required for WebRTC critical traffic=20
(real-time and data chunks). Without (D) - an interface to network QoS=20
equipment - the network does not even know what to prioritize.

I am afraid that this will be considered "back to the drawing board"=20
rather than "some language" but I cannot see the way forward if not=20
considered.=20

Has the SCTP data channel been constructed with hopes of=20
prioritization/quality functionality that simply are not there=20
(cannot be there)? (I have browsed the (complex) SCTP spec and some=20
rtcweb drafts to get an understanding of the prioritization/QoS thoughts =

and means in this rtcweb context and the intended usage. Is there an=20
overall design/thought somewhere that I missed/am unaware of?)

File transfer - getting a larger amount of data through between =
endpoints=20
as quickly as possible - CANNOT be prioritized over IP networks! For =
file=20
transfer we normally use TCP, meaning we fill the pipe, until THE=20
ENDPOINTS see packet drop and regulate down their traffic flow. File=20
transfer over IP networks works this way and cannot be improved upon=20
("Internet would stop" if we prioritized such traffic! The available=20
bandwidth at congestion points are shared between everyone's TCP-flows=20
hitting such congestion points in the network.) File transfer's nature =
is=20
to have "infinite bandwidth demand" - the mechanism to handle this is to =

allow file transfer to "take the rest of the bandwidth" after =
prioritized=20
media and chunks have taken their needed share.=20

The improvement to make - prioritization of media traffic and data =
chunks=20
- is to allow level 3 QoS methods - not only relying on the level 4 TCP=20
backing off process. Without level 3 QoS (diffserve/DSCP, reservation or =

separate pipes) media and data chunks packets are also lost in the=20
level 4 TCP backing off process (=3Dtoday's situation over best effort=20
Internet). That is where the prioritization/QoS measures that we need =
for=20
rtcweb have to be.

SCTP in itself (instead of TLS/TCP) does not/cannot/will not improve =
things.
It can destroy though if wrongly/badly implemented or used!

>From the SCTP RFC4960: 7.  Congestion Control=20
   ...adequate resources will  be allocated to SCTP traffic=20
   to ensure prompt delivery of time-critical data ... unlikely,=20
   during normal operations, that transmissions encounter severe=20
   congestion conditions. =20

   However, SCTP must operate under adverse operational
   conditions, which can develop upon partial network failures=20
   or unexpected traffic surges.  In such situations, SCTP must=20
   follow correct congestion control steps to recover from=20
   congestion quickly in order to get data delivered as soon as=20
   possible. =20

   ... SCTP congestion control is always applied to the entire=20
   association, and not to individual streams.

Doing file transfer with the data channel will use SCTP in the "adverse=20
operational conditions, which can develop upon partial network failures=20
or unexpected traffic surges"-mode! Has that been considered when=20
constructing the data channel? Hope so...

Assuming SCTP is implemented correctly and as good as TCP for file=20
transfer (even though file transfer =3D filling the pipe mode is an=20
"adverse condition..." for SCTP). Then we must specify:


(A) Rtcweb file transfer MUST NOT use the same data channel as data=20
chunks expected to have prioritized delivery (otherwise, as per=20
above, SCTP will regulate the "the entire association" to share the=20
available bandwidth at the congestion point - self-destroying the=20
expected prioritized delivery of data chunks in the same channel).


(B) Each rtcweb file transfer MUST use its own data channel (if=20
combined they risk being regulated randomly/unfairly by SCTP itself=20
and all files will have just have one single TCP-like flow at the=20
network congestion point when sharing the available bandwidth with=20
everyone else's flows at that congestion point - That would be=20
"negative priority" compared to what we are used to in a browser.).


FURTHER, to allow the underlying network the ability to handle the=20
real-time and data chunks as least as good as "before rtcweb"=20
(Unfortunately this will make NAT-firewall less good - not sending=20
everything P2P in a single bundle/flow - I don't like it either...=20
Sorry...):

(C) File transfer using the data channel MUST NOT be bundled with=20
media and data chunk traffic in the same flow - file transfer must=20
use a separate socket creating another flow/5-tuple over the network.=20
(In reservation type of networks (e.g. mobile OTT, cable), file=20
transfer - having no bandwidth limitation - would otherwise overflow=20
the reserved bandwidth for media and chunck traffic - destroying=20
prioritized delivery of media and chunck traffic over that flow.)=20
Such file transfer data channels MUST NOT be prioritized by the=20
underlying network (applies to all IP network types, network=20
prioritization of file transfer will STOP traffic between endpoints=20
- Endpoint's TCP-like regulation has to function!). Several file=20
transfers could share such a separate flow (without specifying in=20
SDP I guess) but the underlying network must be able to separate=20
such flows (not to be prioritized) from media and chunk flows.=20

In the API, such file transfer data channel could be mapped to the=20
"below normal" classification (but would it not be better with a=20
"file transfer channel" API instead?). (And notice that without (D)=20
below, the network is not informed how to separate a file transfer=20
data channel flow from a prioritized data channel flow.)


Having (A), (B) and (C) in place we are BACK TO THE CURRENT=20
Internet prioritization/QoS level - before the rtcweb p2p data=20
channel file transfer was introduced...

Now remains the prioritization improvement required for quality=20
hungry rtcweb! Both for the real-time media and the data chuncks.


(D) The underlying transport network needs to be informed about=20
the traffic requirements.=20

- In RTP media this traffic info could be inserted by the rtcweb=20
browser in the RTP header extension [1] depending on codec usage=20
etc. The API classification could be mapped to the=20
"Prioritize X(2 bits)" suggested in [1] below.

- for the data channel, the traffic info could be inserted as=20
suggested in previous email ([2] below)=20
http://www.ietf.org/mail-archive/web/rtcweb/current/msg11973.html =20

Note that the required bandwidth (only known by the=20
browser/application) has to be passed to reservation types of=20
network. Requested prioritization from the API or DSCP bits are=20
not sufficient. The browser knows the bandwidth requirements for=20
codecs, but for data chunks, the application has the knowledge.=20
In lack of a bandwidth parameter in the data channel API, a simple=20
convention - e.g. reserve 200 kbps for data chunks could serve as=20
a compromise (or 4 bandwidths mapped to the API prioritize=20
classifications...). A modified API with the bandwidth requirement=20
spelled out would of course be better for the chuck data channel.=20
For a file transfer data channel - prioritization classification=20
does not even apply. =20


Having prioritization over the network cleared, the application=20
should use that transport properly - having media and data chunks=20
transported with prioritization (as produced/inserted), NOT queuing=20
them up, rearranged by protocols or applications and NOT involving=20
file transfer (that has to be unprioritized).=20

Using DSCP bits if available through the OS in some situations does=20
not help either, as already pointed out in the transport document.=20
For (D) above, DSCP bits could in a few situations function locally,=20
but not end-to-end, so of no use generally.=20
=20
/Karl


[1] Traffic type information to encode (recreating the=20
idea/intention of the RTP payload type (PT) header that is not=20
available for the network level anymore, by instead having the=20
browser filling the RTP header extension with relevant traffic=20
info that all IP network types can use.), as suggested already=20
October 22. Two parameters to encode=20
http://www.ietf.org/mail-archive/web/rtcweb/current/msg09129.html=20

A) The maximum bandwidth requirement: Two bytes could contain=20
everything from some bps for real-time text to Gbps for future 3D=20
supersize telepresence on a logarithmic scale.
=20
B) The quality characteristics for the stream, with the highest=20
bit set to 1, we could allocate a bit each for quality type e.g:=20
Best Effort, Audio, Video, Supplemental Video, Gaming, Data,=20
Delay Insensitive (e.g. video streaming), Minimum Delay,=20
Reliable Delivery, Prioritize X(2 bits), Variation Y(2 bits),=20
that could be combined as required to describe the stream.

And with the highest bit set to 0, there could instead be a number=20
for special usage that does not fit the general description of=20
the individual bits.

Without this information, how could e.g. a reservation type=20
network (e.g. mobile OTT and cable) reserve the bandwidth=20
required for incoming rtcweb media or data chunks from a diffserve=20
network (carrying no bandwidth requirement)?=20

=20
[2] A new DTLS ContentType (e.g. 24 instead of current 23) for the=20
rtcweb data channel, where the content would be the unencrypted=20
traffic info (preferable the same as in the RTP header extension)=20
followed by the SCTP data as it is today (where it today comes=20
directly in the DTLS as the only content).:

TODAY:
Content Type: 23 (1 byte) content type 23 means =93application data=94
DTLS version:  0xfeff (2 byte)
Epoch: 0001 (2 byte)
Sequence number (e.g. 00 00 00 00 00 01) 6 byte
Length (e.g. 01 c0) 2 byte
Encrypted application data: ...... =3D the SCTP
=20
SUGGESTION WITH ADDED TRAFFIC INFO FOR LOWER LEVELS:=20
Content Type: 24 (1 byte) content type 24 meaning =93application data =
preceded
by traffic info=94
DTLS version:  0xfeff (2 byte)
Epoch: 0001 (2 byte)
Sequence number (e.g. 00 00 00 00 00 01) 6 byte
Length (e.g. 01 c0) 2 byte: length of the encrypted application data =
after
the new header extension
TRAFFIC INFO, Same as suggested for the RTP Header extension:=20
Encrypted application data: ...... ...... =3D the SCTP unchanged


-----Ursprungligt meddelande-----
Fr=E5n: rtcweb [mailto:rtcweb-bounces@ietf.org] F=F6r Matthew Kaufman =
(SKYPE)
Skickat: den 31 mars 2014 19:50
Till: Harald Alvestrand
Kopia: rtcweb@ietf.org
=C4mne: Re: [rtcweb] Some language on "prioritization"

And if SCTP and RTP have packets to send, which goes first, why, and =
under
what circumstances?

Matthew Kaufman

(Sent from my iPhone)

> On Mar 31, 2014, at 10:48 AM, "Harald Alvestrand" =
<harald@alvestrand.no>
wrote:
>=20
>> On 03/31/2014 07:42 PM, Michael Tuexen wrote:
>>> On 31 Mar 2014, at 19:38, Martin Thomson <martin.thomson@gmail.com>
wrote:
>>>=20
>>>> On 31 March 2014 10:08, Harald Alvestrand <harald@alvestrand.no> =
wrote:
>>>> -- TODO: Specify a relative priority scheme that makes sense with=20
>>>> SCTP, with an appropriate reference.=20
>>>> [draft-ietf-tsvwg-sctp-prpolicies]
>>>> specifies a priority policy, but it's about discarding packets, not =

>>>> deciding which packets to send first, and it also makes it=20
>>>> impossible to specify time-bounded retransmission. --
>>>=20
>>> Why would SCTP need special treatment?  I can understand if there=20
>>> are particular time-sensitive control messages that need to be given =

>>> higher priority, but this is all time-sensitive.
>> A single transport connection (in this case an SCTP association) can=20
>> only use a single DSCP. So it would be OK to use the same priority=20
>> for all data channels, but things get complicated when when some data =

>> channels would have different priorities requiring different DSCP
markings.
>=20
> The SCTP protocol machine will, at some given moment in time, have=20
> packets in its send buffers that belong to multiple data channels.=20
> Only one of them can go on the wire first.
>=20
> Which packet does it choose to send first, and why?
>=20
> That's the question I'm looking for an answer to.
>=20
> If the answer has an RFC number, chapter and section, I'd be very =
happy.
> If it doesn't - perhaps we have to invent one.
> Or say that it's "implementation dependent".
>=20
> _______________________________________________
> 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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Subject: Re: [rtcweb] Some language on "prioritization"
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On 03 Apr 2014, at 18:30, Karl Stahl <karl.stahl@intertex.se> wrote:

> The questions raised here don't have simple "local or endpoint =
answers".
> There are overall and basic things to be made clear first.=20
> Combining P2P file transfer over the rtcweb data channels with=20
> prioritized traffic chunks, brings further complications to the=20
> prioritization/QoS issue. This cannot be handled/done by endpoint or=20=

> applications without considering the transport network (The congestion=20=

> to handle by prioritization is in the network - NOT in the endpoints)!=20=

>=20
> There are below (A), (B) and (C) - to get us back to normal - then =
there=20
> is (D) to improve - to allow the network level to give us the=20
> prioritization required for WebRTC critical traffic=20
> (real-time and data chunks). Without (D) - an interface to network QoS=20=

> equipment - the network does not even know what to prioritize.
>=20
> I am afraid that this will be considered "back to the drawing board"=20=

> rather than "some language" but I cannot see the way forward if not=20
> considered.=20
>=20
> Has the SCTP data channel been constructed with hopes of=20
> prioritization/quality functionality that simply are not there=20
> (cannot be there)? (I have browsed the (complex) SCTP spec and some=20
> rtcweb drafts to get an understanding of the prioritization/QoS =
thoughts=20
> and means in this rtcweb context and the intended usage. Is there an=20=

> overall design/thought somewhere that I missed/am unaware of?)
The basis idea is to use a stream scheduler to handle streams different.
Some data channels can get a higher bandwidth than others. Since the
priority stuff is still in flux in W3C, it hasn't been nailed down in
the SCTP spec. I guess, the place will be in
http://tools.ietf.org/html/draft-ietf-tsvwg-sctp-ndata-00
This document is not finalized yet.
>=20
> File transfer - getting a larger amount of data through between =
endpoints=20
> as quickly as possible - CANNOT be prioritized over IP networks! For =
file=20
> transfer we normally use TCP, meaning we fill the pipe, until THE=20
> ENDPOINTS see packet drop and regulate down their traffic flow. File=20=

> transfer over IP networks works this way and cannot be improved upon=20=

> ("Internet would stop" if we prioritized such traffic! The available=20=

> bandwidth at congestion points are shared between everyone's TCP-flows=20=

> hitting such congestion points in the network.) File transfer's nature =
is=20
> to have "infinite bandwidth demand" - the mechanism to handle this is =
to=20
> allow file transfer to "take the rest of the bandwidth" after =
prioritized=20
> media and chunks have taken their needed share.=20
>=20
> The improvement to make - prioritization of media traffic and data =
chunks=20
> - is to allow level 3 QoS methods - not only relying on the level 4 =
TCP=20
> backing off process. Without level 3 QoS (diffserve/DSCP, reservation =
or=20
> separate pipes) media and data chunks packets are also lost in the=20
> level 4 TCP backing off process (=3Dtoday's situation over best effort=20=

> Internet). That is where the prioritization/QoS measures that we need =
for=20
> rtcweb have to be.
>=20
> SCTP in itself (instead of TLS/TCP) does not/cannot/will not improve =
things.
> It can destroy though if wrongly/badly implemented or used!
>=20
>> =46rom the SCTP RFC4960: 7.  Congestion Control=20
>   ...adequate resources will  be allocated to SCTP traffic=20
>   to ensure prompt delivery of time-critical data ... unlikely,=20
>   during normal operations, that transmissions encounter severe=20
>   congestion conditions. =20
>=20
>   However, SCTP must operate under adverse operational
>   conditions, which can develop upon partial network failures=20
>   or unexpected traffic surges.  In such situations, SCTP must=20
>   follow correct congestion control steps to recover from=20
>   congestion quickly in order to get data delivered as soon as=20
>   possible. =20
>=20
>   ... SCTP congestion control is always applied to the entire=20
>   association, and not to individual streams.
>=20
> Doing file transfer with the data channel will use SCTP in the =
"adverse=20
> operational conditions, which can develop upon partial network =
failures=20
> or unexpected traffic surges"-mode! Has that been considered when=20
> constructing the data channel? Hope so...
>=20
> Assuming SCTP is implemented correctly and as good as TCP for file=20
> transfer (even though file transfer =3D filling the pipe mode is an=20
> "adverse condition..." for SCTP). Then we must specify:
>=20
>=20
> (A) Rtcweb file transfer MUST NOT use the same data channel as data=20
> chunks expected to have prioritized delivery (otherwise, as per=20
> above, SCTP will regulate the "the entire association" to share the=20
> available bandwidth at the congestion point - self-destroying the=20
> expected prioritized delivery of data chunks in the same channel).
If you use a data channel for file transfer, why would you put anything
else there? If has no sequencing constraints with the file transfer.
>=20
>=20
> (B) Each rtcweb file transfer MUST use its own data channel (if=20
Yepp. I would do that anyway...
> combined they risk being regulated randomly/unfairly by SCTP itself=20
> and all files will have just have one single TCP-like flow at the=20
> network congestion point when sharing the available bandwidth with=20
> everyone else's flows at that congestion point - That would be=20
> "negative priority" compared to what we are used to in a browser.).
>=20
>=20
> FURTHER, to allow the underlying network the ability to handle the=20
> real-time and data chunks as least as good as "before rtcweb"=20
> (Unfortunately this will make NAT-firewall less good - not sending=20
> everything P2P in a single bundle/flow - I don't like it either...=20
> Sorry...):
>=20
> (C) File transfer using the data channel MUST NOT be bundled with=20
> media and data chunk traffic in the same flow - file transfer must=20
> use a separate socket creating another flow/5-tuple over the network.=20=

> (In reservation type of networks (e.g. mobile OTT, cable), file=20
> transfer - having no bandwidth limitation - would otherwise overflow=20=

> the reserved bandwidth for media and chunck traffic - destroying=20
> prioritized delivery of media and chunck traffic over that flow.)=20
> Such file transfer data channels MUST NOT be prioritized by the=20
> underlying network (applies to all IP network types, network=20
> prioritization of file transfer will STOP traffic between endpoints=20
> - Endpoint's TCP-like regulation has to function!). Several file=20
> transfers could share such a separate flow (without specifying in=20
> SDP I guess) but the underlying network must be able to separate=20
> such flows (not to be prioritized) from media and chunk flows.=20
>=20
> In the API, such file transfer data channel could be mapped to the=20
> "below normal" classification (but would it not be better with a=20
> "file transfer channel" API instead?). (And notice that without (D)=20
> below, the network is not informed how to separate a file transfer=20
> data channel flow from a prioritized data channel flow.)
>=20
>=20
> Having (A), (B) and (C) in place we are BACK TO THE CURRENT=20
> Internet prioritization/QoS level - before the rtcweb p2p data=20
> channel file transfer was introduced...
>=20
> Now remains the prioritization improvement required for quality=20
> hungry rtcweb! Both for the real-time media and the data chuncks.
>=20
>=20
> (D) The underlying transport network needs to be informed about=20
> the traffic requirements.=20
>=20
> - In RTP media this traffic info could be inserted by the rtcweb=20
> browser in the RTP header extension [1] depending on codec usage=20
> etc. The API classification could be mapped to the=20
> "Prioritize X(2 bits)" suggested in [1] below.
>=20
> - for the data channel, the traffic info could be inserted as=20
> suggested in previous email ([2] below)=20
> http://www.ietf.org/mail-archive/web/rtcweb/current/msg11973.html =20
>=20
> Note that the required bandwidth (only known by the=20
> browser/application) has to be passed to reservation types of=20
> network. Requested prioritization from the API or DSCP bits are=20
> not sufficient. The browser knows the bandwidth requirements for=20
> codecs, but for data chunks, the application has the knowledge.=20
> In lack of a bandwidth parameter in the data channel API, a simple=20
> convention - e.g. reserve 200 kbps for data chunks could serve as=20
> a compromise (or 4 bandwidths mapped to the API prioritize=20
> classifications...). A modified API with the bandwidth requirement=20
> spelled out would of course be better for the chuck data channel.=20
> For a file transfer data channel - prioritization classification=20
> does not even apply. =20
>=20
>=20
> Having prioritization over the network cleared, the application=20
> should use that transport properly - having media and data chunks=20
> transported with prioritization (as produced/inserted), NOT queuing=20
> them up, rearranged by protocols or applications and NOT involving=20
> file transfer (that has to be unprioritized).=20
>=20
> Using DSCP bits if available through the OS in some situations does=20
> not help either, as already pointed out in the transport document.=20
> For (D) above, DSCP bits could in a few situations function locally,=20=

> but not end-to-end, so of no use generally.=20
>=20
> /Karl
>=20
>=20
> [1] Traffic type information to encode (recreating the=20
> idea/intention of the RTP payload type (PT) header that is not=20
> available for the network level anymore, by instead having the=20
> browser filling the RTP header extension with relevant traffic=20
> info that all IP network types can use.), as suggested already=20
> October 22. Two parameters to encode=20
> http://www.ietf.org/mail-archive/web/rtcweb/current/msg09129.html=20
>=20
> A) The maximum bandwidth requirement: Two bytes could contain=20
> everything from some bps for real-time text to Gbps for future 3D=20
> supersize telepresence on a logarithmic scale.
>=20
> B) The quality characteristics for the stream, with the highest=20
> bit set to 1, we could allocate a bit each for quality type e.g:=20
> Best Effort, Audio, Video, Supplemental Video, Gaming, Data,=20
> Delay Insensitive (e.g. video streaming), Minimum Delay,=20
> Reliable Delivery, Prioritize X(2 bits), Variation Y(2 bits),=20
> that could be combined as required to describe the stream.
>=20
> And with the highest bit set to 0, there could instead be a number=20
> for special usage that does not fit the general description of=20
> the individual bits.
>=20
> Without this information, how could e.g. a reservation type=20
> network (e.g. mobile OTT and cable) reserve the bandwidth=20
> required for incoming rtcweb media or data chunks from a diffserve=20
> network (carrying no bandwidth requirement)?=20
>=20
>=20
> [2] A new DTLS ContentType (e.g. 24 instead of current 23) for the=20
> rtcweb data channel, where the content would be the unencrypted=20
> traffic info (preferable the same as in the RTP header extension)=20
> followed by the SCTP data as it is today (where it today comes=20
> directly in the DTLS as the only content).:
>=20
> TODAY:
> Content Type: 23 (1 byte) content type 23 means =93application data=94
> DTLS version:  0xfeff (2 byte)
> Epoch: 0001 (2 byte)
> Sequence number (e.g. 00 00 00 00 00 01) 6 byte
> Length (e.g. 01 c0) 2 byte
> Encrypted application data: ...... =3D the SCTP
>=20
> SUGGESTION WITH ADDED TRAFFIC INFO FOR LOWER LEVELS:=20
> Content Type: 24 (1 byte) content type 24 meaning =93application data =
preceded
> by traffic info=94
> DTLS version:  0xfeff (2 byte)
> Epoch: 0001 (2 byte)
> Sequence number (e.g. 00 00 00 00 00 01) 6 byte
> Length (e.g. 01 c0) 2 byte: length of the encrypted application data =
after
> the new header extension
> TRAFFIC INFO, Same as suggested for the RTP Header extension:=20
> Encrypted application data: ...... ...... =3D the SCTP unchanged
>=20
>=20
> -----Ursprungligt meddelande-----
> Fr=E5n: rtcweb [mailto:rtcweb-bounces@ietf.org] F=F6r Matthew Kaufman =
(SKYPE)
> Skickat: den 31 mars 2014 19:50
> Till: Harald Alvestrand
> Kopia: rtcweb@ietf.org
> =C4mne: Re: [rtcweb] Some language on "prioritization"
>=20
> And if SCTP and RTP have packets to send, which goes first, why, and =
under
> what circumstances?
>=20
> Matthew Kaufman
>=20
> (Sent from my iPhone)
>=20
>> On Mar 31, 2014, at 10:48 AM, "Harald Alvestrand" =
<harald@alvestrand.no>
> wrote:
>>=20
>>> On 03/31/2014 07:42 PM, Michael Tuexen wrote:
>>>> On 31 Mar 2014, at 19:38, Martin Thomson <martin.thomson@gmail.com>
> wrote:
>>>>=20
>>>>> On 31 March 2014 10:08, Harald Alvestrand <harald@alvestrand.no> =
wrote:
>>>>> -- TODO: Specify a relative priority scheme that makes sense with=20=

>>>>> SCTP, with an appropriate reference.=20
>>>>> [draft-ietf-tsvwg-sctp-prpolicies]
>>>>> specifies a priority policy, but it's about discarding packets, =
not=20
>>>>> deciding which packets to send first, and it also makes it=20
>>>>> impossible to specify time-bounded retransmission. --
>>>>=20
>>>> Why would SCTP need special treatment?  I can understand if there=20=

>>>> are particular time-sensitive control messages that need to be =
given=20
>>>> higher priority, but this is all time-sensitive.
>>> A single transport connection (in this case an SCTP association) can=20=

>>> only use a single DSCP. So it would be OK to use the same priority=20=

>>> for all data channels, but things get complicated when when some =
data=20
>>> channels would have different priorities requiring different DSCP
> markings.
>>=20
>> The SCTP protocol machine will, at some given moment in time, have=20
>> packets in its send buffers that belong to multiple data channels.=20
>> Only one of them can go on the wire first.
>>=20
>> Which packet does it choose to send first, and why?
>>=20
>> That's the question I'm looking for an answer to.
>>=20
>> If the answer has an RFC number, chapter and section, I'd be very =
happy.
>> If it doesn't - perhaps we have to invent one.
>> Or say that it's "implementation dependent".
>>=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
>=20
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb
>=20


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Date: Thu, 3 Apr 2014 12:13:33 -0700
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Subject: Re: [rtcweb] Transport -03, bundling question (Re: Comments on draft-ietf-rtcweb-transports-02)
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--001a11c258f0b7e70504f6283327
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Harald said:
"My memory was faulty; I had thought that the same conclusion had applied
for TCP ICE candidates as for TURN ICE candidates. Version -04 will have
this fixed."

[BA]  My memory was similarly faulty. But assuming the problem is with my
memory as opposed to the minutes, the decision to require support for ICE
TCP brings to mind other questions such as:

a. Is support for RFC 4571 (framing of RTP/RTCP over connection oriented
transport)  also mandated?  With respect to JSEP, what about RFC 4145
(TCP-based media transport in SDP )?
b. Are there any implications for transport of DTLS/SRTP key management?
 (e.g. if ICE TCP is needed because UDP transport is blocked, wouldn't that
also cause the DTLS/SRTP key exchange to fail?)
c. Are there any implications for the data channel?  (e.g. if UDP transport
is blocked, wouldn't SCTP/DTLS/UDP also not work?)


On Mon, Mar 31, 2014 at 8:08 AM, Harald Alvestrand <harald@alvestrand.no>wrote:

>  On 03/31/2014 03:54 PM, Rauschenbach, Uwe (NSN - DE/Munich) wrote:
>
> Hi Harald,
>
> The decision in London to make ICE-TCP a "MUST" has not been implemented in the -03 draft
> (I have noticed you have reflected the related TURN-TCP discussion already).
>
> What are your plans for addressing the ICE-TCP decision?
>
>
> Thanks!
>
> Checking the minutes, I find:
>
>
> Chairs called a hum between the alternatives:
>
> 1) TCP ICE Candidates are MUST implement
> 2) TCP ICE Candidates are SHOULD/MAY implement
> 3) TCP ICE Candidates will not be discussed in document.
>
> The hum indicated very strong support for 1).
>
>
> My memory was faulty; I had thought that the same conclusion had applied
> for TCP ICE candidates as for TURN ICE candidates. Version -04 will have
> this fixed.
>
>
>
> Kind regards,
> Uwe
>
>
>
>  -----Original Message-----
> From: rtcweb [mailto:rtcweb-bounces@ietf.org <rtcweb-bounces@ietf.org>] On Behalf Of ext Harald
> Alvestrand
> Sent: Monday, March 31, 2014 3:40 PM
> To: Magnus Westerlund; rtcweb@ietf.org; Harald Alvestrand
> Subject: [rtcweb] Transport -03, bundling question (Re: Comments on
> draft-ietf-rtcweb-transports-02)
>
> Version -03 is now published. I hope you like it!
>
> New outstanding item:
>
> I added requirements for implementations to be able to generate both
> fully bundled (one 5-tuple for everything) and fully unbundled (one
> 5-tuple for each flow) configurations, and for implementations to be
> able to tolerate being hit with any combination of bundling schemes.
>
> Is there a need to specify at MUST, SHOULD or MAY levels other
> combinations?
>
> Harald
>
> _______________________________________________
> rtcweb mailing listrtcweb@ietf.orghttps://www.ietf.org/mailman/listinfo/rtcweb
>
>
>
> --
> Surveillance is pervasive. Go Dark.
>
>
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb
>
>

--001a11c258f0b7e70504f6283327
Content-Type: text/html; charset=ISO-8859-1
Content-Transfer-Encoding: quoted-printable

<div dir=3D"ltr">Harald said:=A0<div>&quot;<span style=3D"font-family:arial=
,sans-serif;font-size:13px">My memory was faulty; I had thought that the sa=
me conclusion had applied for TCP ICE candidates as for TURN ICE candidates=
. Version -04 will have this fixed.&quot;</span></div>

<div><span style=3D"font-family:arial,sans-serif;font-size:13px"><br></span=
></div><div><span style=3D"font-family:arial,sans-serif;font-size:13px">[BA=
] =A0My memory was similarly faulty. But assuming the problem is with my me=
mory as opposed to the minutes, the decision to require support for ICE TCP=
 brings to mind other questions such as:</span></div>

<div><span style=3D"font-family:arial,sans-serif;font-size:13px"><br></span=
></div><div><span style=3D"font-family:arial,sans-serif;font-size:13px">a. =
Is</span><span style=3D"font-family:arial,sans-serif;font-size:13px">=A0sup=
port for RFC 4571 (framing of RTP/RTCP over connection oriented transport) =
=A0also mandated? =A0With respect to JSEP, what about RFC 4145 (TCP-based m=
edia transport in SDP )?=A0</span></div>

<div><span style=3D"font-family:arial,sans-serif;font-size:13px">b. Are the=
re any implications for transport of DTLS/SRTP key management? =A0(e.g. if =
ICE TCP is needed because UDP transport is blocked, wouldn&#39;t that also =
cause the DTLS/SRTP key exchange to fail?)</span></div>

<div><span style=3D"font-family:arial,sans-serif;font-size:13px">c. Are the=
re any implications for the data channel? =A0(e.g. if UDP transport is bloc=
ked, wouldn&#39;t SCTP/DTLS/UDP also not work?)</span></div></div><div clas=
s=3D"gmail_extra">

<br><br><div class=3D"gmail_quote">On Mon, Mar 31, 2014 at 8:08 AM, Harald =
Alvestrand <span dir=3D"ltr">&lt;<a href=3D"mailto:harald@alvestrand.no" ta=
rget=3D"_blank">harald@alvestrand.no</a>&gt;</span> wrote:<br><blockquote c=
lass=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1px #ccc solid;=
padding-left:1ex">


 =20
   =20
 =20
  <div bgcolor=3D"#FFFFFF" text=3D"#000000"><div class=3D"">
    <div>On 03/31/2014 03:54 PM, Rauschenbach,
      Uwe (NSN - DE/Munich) wrote:<br>
    </div>
    <blockquote type=3D"cite">
      <pre>Hi Harald,

The decision in London to make ICE-TCP a &quot;MUST&quot; has not been impl=
emented in the -03 draft
(I have noticed you have reflected the related TURN-TCP discussion already)=
.

What are your plans for addressing the ICE-TCP decision?</pre>
    </blockquote>
    <br></div>
    Thanks!<br>
    <br>
    Checking the minutes, I find:<br>
    <br>
   =20
    <pre style=3D"line-height:normal;text-indent:0px;letter-spacing:normal;=
text-align:start;font-variant:normal;text-transform:none;font-style:normal;=
white-space:pre-wrap;word-wrap:break-word;font-weight:normal;word-spacing:0=
px">

Chairs called a hum between the alternatives:

1) TCP ICE Candidates are MUST implement
2) TCP ICE Candidates are SHOULD/MAY implement
3) TCP ICE Candidates will not be discussed in document.=20

The hum indicated very strong support for 1).=20
</pre>
    <br>
    My memory was faulty; I had thought that the same conclusion had
    applied for TCP ICE candidates as for TURN ICE candidates. Version
    -04 will have this fixed.<div class=3D""><br>
    <br>
    <blockquote type=3D"cite">
      <pre>
Kind regards,
Uwe=20


</pre>
      <blockquote type=3D"cite">
        <pre>-----Original Message-----
From: rtcweb [<a href=3D"mailto:rtcweb-bounces@ietf.org" target=3D"_blank">=
mailto:rtcweb-bounces@ietf.org</a>] On Behalf Of ext Harald
Alvestrand
Sent: Monday, March 31, 2014 3:40 PM
To: Magnus Westerlund; <a href=3D"mailto:rtcweb@ietf.org" target=3D"_blank"=
>rtcweb@ietf.org</a>; Harald Alvestrand
Subject: [rtcweb] Transport -03, bundling question (Re: Comments on
draft-ietf-rtcweb-transports-02)

Version -03 is now published. I hope you like it!

New outstanding item:

I added requirements for implementations to be able to generate both
fully bundled (one 5-tuple for everything) and fully unbundled (one
5-tuple for each flow) configurations, and for implementations to be
able to tolerate being hit with any combination of bundling schemes.

Is there a need to specify at MUST, SHOULD or MAY levels other
combinations?

Harald

_______________________________________________
rtcweb mailing list
<a href=3D"mailto:rtcweb@ietf.org" target=3D"_blank">rtcweb@ietf.org</a>
<a href=3D"https://www.ietf.org/mailman/listinfo/rtcweb" target=3D"_blank">=
https://www.ietf.org/mailman/listinfo/rtcweb</a>
</pre>
      </blockquote>
    </blockquote>
    <br>
    <br>
    </div><span class=3D"HOEnZb"><font color=3D"#888888"><pre cols=3D"72">-=
-=20
Surveillance is pervasive. Go Dark.
</pre>
  </font></span></div>

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

--001a11c258f0b7e70504f6283327--


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On 3 April 2014 12:13, Bernard Aboba <bernard.aboba@gmail.com> wrote:
> "My memory was faulty; I had thought that the same conclusion had applied
> for TCP ICE candidates as for TURN ICE candidates. Version -04 will have
> this fixed."
>
> [BA]  My memory was similarly faulty. But assuming the problem is with my
> memory as opposed to the minutes, the decision to require support for ICE
> TCP brings to mind other questions such as:

That makes three with faulty memories.  But perhaps that's my
prejudice at play.  The marginal benefit remains...tiny relative to
the cost.


From nobody Thu Apr  3 13:45:32 2014
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--047d7bfd0bdeb6aad504f6297a56
Content-Type: text/plain; charset=ISO-8859-1

Martin said:

"The marginal benefit remains...tiny relative to the cost"

[BA] This is particularly true if you consider *all* of the additional work
items required to enable the usage scenario.  It is one thing to support
SRTP/SRTCP over RFC 4571 framing, but without also  encapsulating DTLS/SRTP
over RFC 4571 framing, how do you reliably obtain the SRTP/SRTCP master
keys?  Similarly, do we also run SCTP/DTLS data channel over RFC 4571
framing?   In my experience, without a great deal of care, running one
reliable transport over another can works poorly if packet loss is
encountered (e.g. the time-out periods interfere with one another).


On Thu, Apr 3, 2014 at 1:25 PM, Martin Thomson <martin.thomson@gmail.com>wrote:

> On 3 April 2014 12:13, Bernard Aboba <bernard.aboba@gmail.com> wrote:
> > "My memory was faulty; I had thought that the same conclusion had applied
> > for TCP ICE candidates as for TURN ICE candidates. Version -04 will have
> > this fixed."
> >
> > [BA]  My memory was similarly faulty. But assuming the problem is with my
> > memory as opposed to the minutes, the decision to require support for ICE
> > TCP brings to mind other questions such as:
>
> That makes three with faulty memories.  But perhaps that's my
> prejudice at play.  The marginal benefit remains...tiny relative to
> the cost.
>

--047d7bfd0bdeb6aad504f6297a56
Content-Type: text/html; charset=ISO-8859-1
Content-Transfer-Encoding: quoted-printable

<div dir=3D"ltr">Martin said:=A0<div><br></div><div>&quot;<span style=3D"fo=
nt-family:arial,sans-serif;font-size:13px">The marginal benefit remains...t=
iny relative to=A0</span><span style=3D"font-family:arial,sans-serif;font-s=
ize:13px">the cost&quot;</span></div>

<div><span style=3D"font-family:arial,sans-serif;font-size:13px"><br></span=
></div><div><span style=3D"font-family:arial,sans-serif;font-size:13px">[BA=
] This is particularly true if you consider *all* of the additional work it=
ems required to enable the usage scenario. =A0It is one thing to support SR=
TP/SRTCP over RFC 4571 framing, but without also =A0encapsulating DTLS/SRTP=
 over RFC 4571 framing, how do you reliably obtain the SRTP/SRTCP master ke=
ys? =A0Similarly, do we also run SCTP/DTLS data channel over RFC 4571 frami=
ng? =A0 In my experience, without a great deal of care, running one reliabl=
e transport over another can works poorly if packet loss is encountered (e.=
g. the time-out periods interfere with one another). =A0=A0</span></div>

</div><div class=3D"gmail_extra"><br><br><div class=3D"gmail_quote">On Thu,=
 Apr 3, 2014 at 1:25 PM, Martin Thomson <span dir=3D"ltr">&lt;<a href=3D"ma=
ilto:martin.thomson@gmail.com" target=3D"_blank">martin.thomson@gmail.com</=
a>&gt;</span> wrote:<br>

<blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1p=
x #ccc solid;padding-left:1ex"><div class=3D"">On 3 April 2014 12:13, Berna=
rd Aboba &lt;<a href=3D"mailto:bernard.aboba@gmail.com">bernard.aboba@gmail=
.com</a>&gt; wrote:<br>


&gt; &quot;My memory was faulty; I had thought that the same conclusion had=
 applied<br>
&gt; for TCP ICE candidates as for TURN ICE candidates. Version -04 will ha=
ve<br>
&gt; this fixed.&quot;<br>
&gt;<br>
&gt; [BA] =A0My memory was similarly faulty. But assuming the problem is wi=
th my<br>
&gt; memory as opposed to the minutes, the decision to require support for =
ICE<br>
&gt; TCP brings to mind other questions such as:<br>
<br>
</div>That makes three with faulty memories. =A0But perhaps that&#39;s my<b=
r>
prejudice at play. =A0The marginal benefit remains...tiny relative to<br>
the cost.<br>
</blockquote></div><br></div>

--047d7bfd0bdeb6aad504f6297a56--


From nobody Thu Apr  3 17:14:32 2014
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From: Martin Thomson <martin.thomson@gmail.com>
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Subject: [rtcweb] Asking TLS for help with media isolation
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As I described briefly at the last meeting, ensuring that media is
isolated from the application or web site is a key part of addressing
our security goals.

The key part of that is making sure that any media that is isolated on
the sending side of RTCPeerConnection remains isolated when it reaches
the other side.

As I noted, this is also necessary in order to ensure the integrity of
the same origin model.  In that model, cross origin media is required
to be inaccessible to content, and as it stands RTCPeerConnection
could be used to work around those restrictions (implementations can
implement other protections, as Firefox already does).

The alternatives as I see them (and I hope that this is sufficiently
exhaustive) are:

 1. ask the TLS working group for a TLS-based solution
 2. build something into the session signaling (i.e., new SDP bits)
 3. give up on the idea

I prefer 1 for reasons already outlined.

I propose that we formally request the TLS working group recommend or
define a mechanism whereby we can signal in the TLS handshake that the
session contents are to be protected from access (where the
description of that protection may need to our responsibility).

I have pointed to draft-thomson-tls-acp as a potential solution here,
but others have noted that ALPN tokens could be used.  I expect that
TLS are capable of making the right decision here.


From nobody Thu Apr  3 18:51:36 2014
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From: Watson Ladd <watsonbladd@gmail.com>
To: Martin Thomson <martin.thomson@gmail.com>
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Subject: Re: [rtcweb] Asking TLS for help with media isolation
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On Thu, Apr 3, 2014 at 5:14 PM, Martin Thomson <martin.thomson@gmail.com> wrote:
> As I described briefly at the last meeting, ensuring that media is
> isolated from the application or web site is a key part of addressing
> our security goals.
>
> The key part of that is making sure that any media that is isolated on
> the sending side of RTCPeerConnection remains isolated when it reaches
> the other side.
>
> As I noted, this is also necessary in order to ensure the integrity of
> the same origin model.  In that model, cross origin media is required
> to be inaccessible to content, and as it stands RTCPeerConnection
> could be used to work around those restrictions (implementations can
> implement other protections, as Firefox already does).
>
> The alternatives as I see them (and I hope that this is sufficiently
> exhaustive) are:
>
>  1. ask the TLS working group for a TLS-based solution
>  2. build something into the session signaling (i.e., new SDP bits)
>  3. give up on the idea
>
> I prefer 1 for reasons already outlined.

Putting on my TLS hat, TLS already lets you send data across the
network securely. Why does this bit need to be treated differently
from all others?

Sincerely,
Watson Ladd


-- 
"Those who would give up Essential Liberty to purchase a little
Temporary Safety deserve neither  Liberty nor Safety."
-- Benjamin Franklin


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Subject: Re: [rtcweb] Asking TLS for help with media isolation
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--089e013d1a60e3debe04f62e4bae
Content-Type: text/plain; charset=ISO-8859-1

Martin said:

"I have pointed to draft-thomson-tls-acp as a potential solution here,
but others have noted that ALPN tokens could be used."

Watson said:

"Putting on my TLS hat, TLS already lets you send data across the
network securely. Why does this bit need to be treated differently
from all others?

[BA] As Martin indicates, the desire for isolation needs to be communicated
to ensure that remote media is not misused.  With either of the TLS
approaches that Martin has suggested,  the desire for isolation is
communicated directly between the peers.   Having this occur E2E via media,
not hop-by-hop via signaling avoids the risk of a MITM preventing isolation
from being negotiated.

However once the desire for isolation is communicated E2E (either via ACP
or ALPN tokens), there is nothing in the SRTP traffic (keyed by DTLS/SRTP)
that indicates that the traffic is to be isolated.


On Thu, Apr 3, 2014 at 6:51 PM, Watson Ladd <watsonbladd@gmail.com> wrote:

> On Thu, Apr 3, 2014 at 5:14 PM, Martin Thomson <martin.thomson@gmail.com>
> wrote:
> > As I described briefly at the last meeting, ensuring that media is
> > isolated from the application or web site is a key part of addressing
> > our security goals.
> >
> > The key part of that is making sure that any media that is isolated on
> > the sending side of RTCPeerConnection remains isolated when it reaches
> > the other side.
> >
> > As I noted, this is also necessary in order to ensure the integrity of
> > the same origin model.  In that model, cross origin media is required
> > to be inaccessible to content, and as it stands RTCPeerConnection
> > could be used to work around those restrictions (implementations can
> > implement other protections, as Firefox already does).
> >
> > The alternatives as I see them (and I hope that this is sufficiently
> > exhaustive) are:
> >
> >  1. ask the TLS working group for a TLS-based solution
> >  2. build something into the session signaling (i.e., new SDP bits)
> >  3. give up on the idea
> >
> > I prefer 1 for reasons already outlined.
>
> Putting on my TLS hat, TLS already lets you send data across the
> network securely. Why does this bit need to be treated differently
> from all others?
>
> Sincerely,
> Watson Ladd
>
>
> --
> "Those who would give up Essential Liberty to purchase a little
> Temporary Safety deserve neither  Liberty nor Safety."
> -- Benjamin Franklin
>
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb
>

--089e013d1a60e3debe04f62e4bae
Content-Type: text/html; charset=ISO-8859-1
Content-Transfer-Encoding: quoted-printable

<div dir=3D"ltr">Martin said:=A0<div><br></div><div>&quot;<span style=3D"fo=
nt-family:arial,sans-serif;font-size:13px">I have pointed to draft-thomson-=
tls-acp as a potential solution here,</span></div><div><span style=3D"font-=
family:arial,sans-serif;font-size:13px">but others have noted that ALPN tok=
ens could be used.&quot;</span><span style=3D"font-family:arial,sans-serif;=
font-size:13px">=A0</span></div>

<div><br></div><div>Watson said:=A0<div><br></div><div>&quot;<span style=3D=
"font-family:arial,sans-serif;font-size:13px">Putting on my TLS hat, TLS al=
ready lets you send data across the</span></div><span style=3D"font-family:=
arial,sans-serif;font-size:13px">network securely. Why does this bit need t=
o be treated differently</span><br style=3D"font-family:arial,sans-serif;fo=
nt-size:13px">

<span style=3D"font-family:arial,sans-serif;font-size:13px">from all others=
?</span><br style=3D"font-family:arial,sans-serif;font-size:13px"><br style=
=3D"font-family:arial,sans-serif;font-size:13px"><font face=3D"arial, sans-=
serif">[BA] As Martin indicates, the desire for isolation needs to be commu=
nicated to ensure that remote media is not misused. =A0</font><span style=
=3D"font-family:arial,sans-serif">With either of the TLS approaches that Ma=
rtin has suggested, =A0the desire for isolation is communicated directly be=
tween the peers. =A0 Having this occur E2E via media, not hop-by-hop via si=
gnaling avoids the risk of a MITM preventing isolation from being negotiate=
d. =A0</span></div>

<div><span style=3D"font-family:arial,sans-serif"><br></span></div><div><sp=
an style=3D"font-family:arial,sans-serif">However once the desire for isola=
tion is communicated E2E (either via ACP or ALPN tokens), there is nothing =
in the SRTP traffic (keyed by DTLS/SRTP) that indicates that the traffic is=
 to be isolated. =A0</span></div>

</div><div class=3D"gmail_extra"><br><br><div class=3D"gmail_quote">On Thu,=
 Apr 3, 2014 at 6:51 PM, Watson Ladd <span dir=3D"ltr">&lt;<a href=3D"mailt=
o:watsonbladd@gmail.com" target=3D"_blank">watsonbladd@gmail.com</a>&gt;</s=
pan> wrote:<br>

<blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1p=
x #ccc solid;padding-left:1ex"><div class=3D"">On Thu, Apr 3, 2014 at 5:14 =
PM, Martin Thomson &lt;<a href=3D"mailto:martin.thomson@gmail.com">martin.t=
homson@gmail.com</a>&gt; wrote:<br>


&gt; As I described briefly at the last meeting, ensuring that media is<br>
&gt; isolated from the application or web site is a key part of addressing<=
br>
&gt; our security goals.<br>
&gt;<br>
&gt; The key part of that is making sure that any media that is isolated on=
<br>
&gt; the sending side of RTCPeerConnection remains isolated when it reaches=
<br>
&gt; the other side.<br>
&gt;<br>
&gt; As I noted, this is also necessary in order to ensure the integrity of=
<br>
&gt; the same origin model. =A0In that model, cross origin media is require=
d<br>
&gt; to be inaccessible to content, and as it stands RTCPeerConnection<br>
&gt; could be used to work around those restrictions (implementations can<b=
r>
&gt; implement other protections, as Firefox already does).<br>
&gt;<br>
&gt; The alternatives as I see them (and I hope that this is sufficiently<b=
r>
&gt; exhaustive) are:<br>
&gt;<br>
&gt; =A01. ask the TLS working group for a TLS-based solution<br>
&gt; =A02. build something into the session signaling (i.e., new SDP bits)<=
br>
&gt; =A03. give up on the idea<br>
&gt;<br>
&gt; I prefer 1 for reasons already outlined.<br>
<br>
</div>Putting on my TLS hat, TLS already lets you send data across the<br>
network securely. Why does this bit need to be treated differently<br>
from all others?<br>
<br>
Sincerely,<br>
Watson Ladd<br>
<span class=3D"HOEnZb"><font color=3D"#888888"><br>
<br>
--<br>
&quot;Those who would give up Essential Liberty to purchase a little<br>
Temporary Safety deserve neither =A0Liberty nor Safety.&quot;<br>
-- Benjamin Franklin<br>
</font></span><div class=3D"HOEnZb"><div class=3D"h5"><br>
_______________________________________________<br>
rtcweb mailing list<br>
<a href=3D"mailto:rtcweb@ietf.org">rtcweb@ietf.org</a><br>
<a href=3D"https://www.ietf.org/mailman/listinfo/rtcweb" target=3D"_blank">=
https://www.ietf.org/mailman/listinfo/rtcweb</a><br>
</div></div></blockquote></div><br></div>

--089e013d1a60e3debe04f62e4bae--


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On Thu, Apr 3, 2014 at 7:29 PM, Bernard Aboba <bernard.aboba@gmail.com> wrote:
> Martin said:
>
> "I have pointed to draft-thomson-tls-acp as a potential solution here,
> but others have noted that ALPN tokens could be used."
>
> Watson said:
>
> "Putting on my TLS hat, TLS already lets you send data across the
> network securely. Why does this bit need to be treated differently
> from all others?
>
> [BA] As Martin indicates, the desire for isolation needs to be communicated
> to ensure that remote media is not misused.  With either of the TLS
> approaches that Martin has suggested,  the desire for isolation is
> communicated directly between the peers.   Having this occur E2E via media,
> not hop-by-hop via signaling avoids the risk of a MITM preventing isolation
> from being negotiated.

The MITM can kill any TLS connection containing the extension. My
understanding is that signalling data is immutable, hence the need to
ask the browser to generate it.

>
> However once the desire for isolation is communicated E2E (either via ACP or
> ALPN tokens), there is nothing in the SRTP traffic (keyed by DTLS/SRTP) that
> indicates that the traffic is to be isolated.

I don't see why the isolation status cannot be included as an
extension to SRTP. You aren't asking TLS to make extensions for video
resolution and codec after all.

Sincerely,
Watson Ladd

-- 
"Those who would give up Essential Liberty to purchase a little
Temporary Safety deserve neither  Liberty nor Safety."
-- Benjamin Franklin


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From: Bernard Aboba <bernard.aboba@gmail.com>
Date: Thu, 3 Apr 2014 20:21:20 -0700
To: Watson Ladd <watsonbladd@gmail.com>
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Subject: Re: [rtcweb] Asking TLS for help with media isolation
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> On Apr 3, 2014, at 19:49, Watson Ladd <watsonbladd@gmail.com> wrote:
>=20
> I don't see why the isolation status cannot be included as an
> extension to SRTP. You aren't asking TLS to make extensions for video
> resolution and codec after all.

[BA] The isolation request could be carried in an RTP header extension until=
, for example, a RR was obtained by the sender confirming it was received.  H=
owever, RTP extensions are optional and the sender wouldn't have confirmatio=
n from the receiver via the media plane that the isolation request was honor=
ed. So the TLS approach provides better semantics.=


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On Thu, Apr 3, 2014 at 8:21 PM, Bernard Aboba <bernard.aboba@gmail.com> wro=
te:
>
>
>> On Apr 3, 2014, at 19:49, Watson Ladd <watsonbladd@gmail.com> wrote:
>>
>> I don't see why the isolation status cannot be included as an
>> extension to SRTP. You aren't asking TLS to make extensions for video
>> resolution and codec after all.
>
> [BA] The isolation request could be carried in an RTP header extension un=
til, for example, a RR was obtained by the sender confirming it was receive=
d.  However, RTP extensions are optional and the sender wouldn't have confi=
rmation from the receiver via the media plane that the isolation request wa=
s honored. So the TLS approach provides better semantics.

I feel this isn't worth changing TLS for, because you don't change
transports for each application because the application didn't do it
right the first time. But we have ALPN, so maybe a tag on that could
work. The right fix IMHO, is clearly to fix RTP to send this data and
keep the transport layer clean.

Sincerely,
Watson Ladd


From nobody Thu Apr  3 20:26:09 2014
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Perhaps some more context is necessary.

(D)TLS provides us good protection against network attackers.  The
problem is that this protection ends at the browser, and our security
story doesn't.  Media streams in the browser are an extension of those
that traverse the network, and we need something of a gentlemen's
agreement between browsers to be reached.  That agreement says that
your browser and mine will agree between themselves to protect media
from the application that is running in their sandbox: all that nasty
JavaScript that came from a site (or sites) that we might not want to
include in our conversation.

On 3 April 2014 19:49, Watson Ladd <watsonbladd@gmail.com> wrote:
> The MITM can kill any TLS connection containing the extension. My
> understanding is that signalling data is immutable, hence the need to
> ask the browser to generate it.

Signaling is totally mutable, which is a big problem.  Once an
application gets it, there's a lot that can be changed between peers.

>> However once the desire for isolation is communicated E2E (either via ACP or
>> ALPN tokens), there is nothing in the SRTP traffic (keyed by DTLS/SRTP) that
>> indicates that the traffic is to be isolated.
>
> I don't see why the isolation status cannot be included as an
> extension to SRTP. You aren't asking TLS to make extensions for video
> resolution and codec after all.

That's a good question, and it's an option I should have added to the
list.  The problem there is that nothing in RTP is reliable, and this
signal needs to be reliable.  Combine that with the fact that the
extent of an RTP session is essentially unknowable, and you end up
having to put an RTP header extension on every packet.  RTP header
extensions have other drawback as well, though I suspect most of those
aren't applicable in this context.

With the TLS handshake, you either get a marked session, or you don't
get a session at all.  That's a big advantage with this approach.  And
it costs far fewer bytes.


From nobody Fri Apr  4 01:51:59 2014
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I assume this is the scheduler envisioned in -ndata:


      SCTP_SS_PRIORITY:  Scheduling with different priorities is used.
         Streams having a higher priority will be scheduled first and
         when multiple streams have the same priority, the default
         scheduling should be used for them.  The priority can be
         assigned with the sctp_stream_value struct.  The higher the
         assigned value, the lower the priority, that is the default
         value 0 is the highest priority and therefore the default
         scheduling will be used if no priorities have been assigned.


This sounds like a "strict" scheduler, in that higher priority queues
will starve out lower priority ones completely. I remember having the
discussion at an IETF meeting about whether we wanted a "strict"
scheduler or a "weighted round robin" scheduler for this, but I wouldn't
trust my memory with what the conclusion was.

Was the conclusion that we should do "strict" scheduling? If so, it may
be best to make that consistent across the board - I had written in a
"weighted" scheduler for media into the prioritization text that I
started this thread with, but I think there's value to consistency.

(Note: -ndata has SS_PRIORITY in one place and SS_PRIO / SS_PRIO_INTER
in another place. Is there a subtlety here I'm not seeing?)




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<html>
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    <meta content="text/html; charset=windows-1252"
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    I assume this is the scheduler envisioned in -ndata:<br>
    <br>
    <meta http-equiv="content-type" content="text/html;
      charset=windows-1252">
    <pre class="newpage" style="font-size: 1em; margin-top: 0px; margin-bottom: 0px; page-break-before: always; color: rgb(0, 0, 0); font-style: normal; font-variant: normal; font-weight: normal; letter-spacing: normal; line-height: normal; orphans: auto; text-align: start; text-indent: 0px; text-transform: none; widows: auto; word-spacing: 0px; -webkit-text-stroke-width: 0px;">

      SCTP_SS_PRIORITY:  Scheduling with different priorities is used.
         Streams having a higher priority will be scheduled first and
         when multiple streams have the same priority, the default
         scheduling should be used for them.  The priority can be
         assigned with the sctp_stream_value struct.  The higher the
         assigned value, the lower the priority, that is the default
         value 0 is the highest priority and therefore the default
         scheduling will be used if no priorities have been assigned.

</pre>
    <br class="Apple-interchange-newline">
    This sounds like a "strict" scheduler, in that higher priority
    queues will starve out lower priority ones completely. I remember
    having the discussion at an IETF meeting about whether we wanted a
    "strict" scheduler or a "weighted round robin" scheduler for this,
    but I wouldn't trust my memory with what the conclusion was.<br>
    <br>
    Was the conclusion that we should do "strict" scheduling? If so, it
    may be best to make that consistent across the board - I had written
    in a "weighted" scheduler for media into the prioritization text
    that I started this thread with, but I think there's value to
    consistency.<br>
    <br>
    (Note: -ndata has SS_PRIORITY in one place and SS_PRIO /
    SS_PRIO_INTER in another place. Is there a subtlety here I'm not
    seeing?)<br>
    <br>
    <br>
    <br>
  </body>
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Subject: Re: [rtcweb] Transport -03, bundling question (Re: Comments on draft-ietf-rtcweb-transports-02)
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On 04/03/2014 10:44 PM, Bernard Aboba wrote:
> Martin said: 
>
> "The marginal benefit remains...tiny relative to the cost"
>
> [BA] This is particularly true if you consider *all* of the additional
> work items required to enable the usage scenario.  It is one thing to
> support SRTP/SRTCP over RFC 4571 framing, but without also
>  encapsulating DTLS/SRTP over RFC 4571 framing, how do you reliably
> obtain the SRTP/SRTCP master keys?  Similarly, do we also run
> SCTP/DTLS data channel over RFC 4571 framing?   In my experience,
> without a great deal of care, running one reliable transport over
> another can works poorly if packet loss is encountered (e.g. the
> time-out periods interfere with one another).  

One result of the London discussion was that I read the TCP TURN
candidate specification, and found that it most explicitly states that
you use 4571 framing for DTLS.

Since it would be very silly to have two types of TCP candidates with
different encapsulation strategies, I wrote in 4571 framing for DTLS too.

"The absence of alternatives clears the mind marvellously".

RFC 4571 is a framing protocol without support for resynchronization. So
it depends on getting whole frames through - something that seems to be
hard to achieve with POSIX TCP send() semantics. A simple implementation
would use blocking writes, in which case packet loss at the higher level
could not occur; another implementation strategy would use non-blocking
mode and drop packets from a higher level buffer if the TCP transmission
buffer filled up, leading to the two-level packet loss issue mentioned.

Which implementation strategy would people expect to use?

-- 
Surveillance is pervasive. Go Dark.


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<html>
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    <meta content="text/html; charset=ISO-8859-1"
      http-equiv="Content-Type">
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  <body bgcolor="#FFFFFF" text="#000000">
    <div class="moz-cite-prefix">On 04/03/2014 10:44 PM, Bernard Aboba
      wrote:<br>
    </div>
    <blockquote
cite="mid:CAOW+2dt+YdsJLJ8dFmRsaKpOpQQXd1ohX0u_r8sJjvifk_kB3w@mail.gmail.com"
      type="cite">
      <div dir="ltr">Martin said:&nbsp;
        <div><br>
        </div>
        <div>"<span style="font-family:arial,sans-serif;font-size:13px">The
            marginal benefit remains...tiny relative to&nbsp;</span><span
            style="font-family:arial,sans-serif;font-size:13px">the
            cost"</span></div>
        <div><span style="font-family:arial,sans-serif;font-size:13px"><br>
          </span></div>
        <div><span style="font-family:arial,sans-serif;font-size:13px">[BA]
            This is particularly true if you consider *all* of the
            additional work items required to enable the usage scenario.
            &nbsp;It is one thing to support SRTP/SRTCP over RFC 4571
            framing, but without also &nbsp;encapsulating DTLS/SRTP over RFC
            4571 framing, how do you reliably obtain the SRTP/SRTCP
            master keys? &nbsp;Similarly, do we also run SCTP/DTLS data
            channel over RFC 4571 framing? &nbsp; In my experience, without a
            great deal of care, running one reliable transport over
            another can works poorly if packet loss is encountered (e.g.
            the time-out periods interfere with one another). &nbsp; <br>
          </span></div>
      </div>
    </blockquote>
    <br>
    One result of the London discussion was that I read the TCP TURN
    candidate specification, and found that it most explicitly states
    that you use 4571 framing for DTLS.<br>
    <br>
    Since it would be very silly to have two types of TCP candidates
    with different encapsulation strategies, I wrote in 4571 framing for
    DTLS too.<br>
    <br>
    "The absence of alternatives clears the mind marvellously".<br>
    <br>
    RFC 4571 is a framing protocol without support for
    resynchronization. So it depends on getting whole frames through -
    something that seems to be hard to achieve with POSIX TCP send()
    semantics. A simple implementation would use blocking writes, in
    which case packet loss at the higher level could not occur; another
    implementation strategy would use non-blocking mode and drop packets
    from a higher level buffer if the TCP transmission buffer filled up,
    leading to the two-level packet loss issue mentioned.<br>
    <br>
    Which implementation strategy would people expect to use?<br>
    <pre class="moz-signature" cols="72">-- 
Surveillance is pervasive. Go Dark.
</pre>
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WG,
(As author)

Based on the discussion we authors have drafted a proposal for changing
the CNAME creation usage to be created unique between PeerConnections.
Our draft text is the below one. Please review. If we receive no
feedback we will include this in the next version of the draft that we
hope to be able to produce next week.

Section 4.1:

   o  Support for multiple synchronisation contexts.  Participants that
      send multiple simultaneous RTP packet streams do so as part of a
      single synchronisation context, using a single RTCP CNAME for all
      streams and allowing receivers to play the streams out in a
      synchronised manner.  Receivers MUST support reception of multiple
      RTCP CNAMEs in an RTP session, and hence multiple synchronisation
      contexts, to support RTP middleboxes, and future evolution needing
      multiple CNAMEs in an endpoint.  See also Section 4.9.


Section 4.9:

4.9.  Generation of the RTCP Canonical Name (CNAME)

   The RTCP Canonical Name (CNAME) provides a persistent transport-level
   identifier for an RTP end-point.  While the Synchronisation Source
   (SSRC) identifier for an RTP end-point can change if a collision is
   detected, or when the RTP application is restarted, its RTCP CNAME is
   meant to stay unchanged, so that RTP end-points can be uniquely
   identified and associated with their RTP packet streams within a set
   of related RTP sessions.  For proper functionality, each RTP end-
   point needs to have at least one unique RTCP CNAME value.  From an
   RTP perspective an end-point can have multiple CNAMEs, as the CNAME
   also identifies a particular synchronisation context, i.e. all SSRC
   associated with a CNAME share a common reference clock, and if an
   end-point have SSRCs associated with different reference clocks it
   will need to use multiple CNAMEs.  This ought not be common, and if
   possible reference clocks ought to be mapped to each other and one
   chosen to be used with RTP and RTCP.  Taking the discussion in
   Section 11 into account a regular WebRTC end-point MUST NOT use more
   than a single CNAME in the RTP sessions belonging to single
   RTCPeerConnection.  RTP middleboxes MAY send RTP packet streams
   identified by more than one CNAME, to allow them to avoid having to
   resynchronize media from multiple different end-points part of a
   multi-party RTP session.

   The RTP specification [RFC3550] includes guidelines for choosing a
   unique RTP CNAME, but these are not sufficient in the presence of NAT
   devices.  In addition, long-term persistent identifiers can be
   problematic from a privacy viewpoint (Section 13).  Accordingly,
   support for generating a short-term persistent RTCP CNAMEs following
   [RFC7022] is REQUIRED to be implemented and RECOMMENDED to be used.
   Unless a persistent CNAME is explicitly configured, a unique CNAME(s)
   MUST be generated for each synchronization context an end-point has
   within each RTCPeerConnection, i.e. normally one across all the RTP
   sessions within the scope of the RTCPeerConnection.

      There are no known reasons for WebRTC end-points that aren't RTP
      middleboxes or servers to configure a persistent identifiers,
      these considerations are limited to RTP middleboxes.  Thus, such
      configuration options SHOULD NOT be present in most WebRTC end-
      point implementations to prevent misconfiguration or malicious
      configuration that allows tracking or linking of a user.

   An WebRTC end-point MUST support reception of any CNAME that matches
   the syntax limitations specified by the RTP specification [RFC3550]
   and cannot assume that any CNAME will be chosen according to the form
   suggested above.

Section 11:

   The same MediaStreamTrack can also be included in multiple
   MediaStreams, thus multiple sets of MediaStreams can implicitly need
   to use the same synchronisation base.  To ensure that this works in
   all cases, and don't forces a end-point to change synchronisation
   base and CNAME in the middle of a ongoing delivery of any packet
   streams, which would cause media disruption; all MediaStreamTracks
   and their associated SSRCs originating from the same end-point needs
   to be sent using the same CNAME within one RTCPeerConnection.  This
   is motivating the strong recommendation in Section 4.9 to only use a
   single CNAME.

      Note: It is important that the same CNAME is not used in different
      communication session contexts or origins, as that could enable
      tracking of a user and its device usage of different services.
      See Section 4.4.1 of Security Considerations for WebRTC
      [I-D.ietf-rtcweb-security] for further discussion.

      The requirement on using the same CNAME for all SSRCs that
      originates from the same end-point, does not require middleboxes
      that forwards traffic from multiple end-points to only use a
      single CNAME.

   The above will currently force a WebRTC end-point that receives an
   MediaStreamTrack on one RTCPeerConnection and adds it as an outgoing
   on any RTCPeerConnection to perform resynchronisation of the stream.
   This, as the sending party needs to change the CNAME, which implies
   that it has to use a locally available system clock as timebase for
   the synchronisation.  Thus, the relative relation between the
   timebase of the incoming stream and the system sending out needs to
   defined.  This relation also needs monitoring for clock drift and
   likely adjustments of the synchronisation.  The sending entity is
   also responsible for congestion control for its the sent streams.  In
   cases of packet loss the loss of incoming data also needs to be
   handled.  This leads to the observation that the method that is least
   likely to cause issues or interruptions in the outgoing source packet
   stream is a model of full decoding, including repair etc followed by
   encoding of the media again into the outgoing packet stream.
   Optimisations of this method is clearly possible and implementation
   specific.

   A WebRTC end-point MUST support receiving multiple MediaStreamTracks,
   where each of different MediaStreamTracks (and their sets of
   associated packet streams) uses different CNAMEs.  However,
   MediaStreamTracks that are received with different CNAMEs have no
   defined synchronisation.

      Note: The motivation for supporting reception of multiple CNAMEs
      are to allow for forward compatibility with any future changes
      that enables more efficient stream handling when end-points relay/
      forward streams.  It also ensures that end-points can interoperate
      with certain types of multi-stream middleboxes or end-points that
      are not WebRTC.

-- 

Magnus Westerlund

----------------------------------------------------------------------
Services, Media and Network features, Ericsson Research EAB/TXM
----------------------------------------------------------------------
Ericsson AB                 | Phone  +46 10 7148287
Färögatan 6                 | Mobile +46 73 0949079
SE-164 80 Stockholm, Sweden | mailto: magnus.westerlund@ericsson.com
----------------------------------------------------------------------


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Subject: [rtcweb] Draft proposal for updating Multiparty topologies in draft-ietf-rtcweb-rtp-usage
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WG,
(As Author)

Colin and I have been working on resolving terminology usage and in
general giving the draft a polish over before the WG last call. One
section that has gotten quite some attention from us is the below one.
The changes are significantly from an editorial stand point. However,
the intended content has not been intended to be changed, although
clarified and better motivated. So please review it, we intended to
include this in the upcoming draft update. Feedback appreciated.

5.1.  Conferencing Extensions and Topologies

   RTP is a protocol that inherently supports group communication.
   Groups can be implemented by having each endpoint sending its RTP
   packet streams to an RTP middlebox that redistributes the traffic, by
   using a mesh of unicast RTP packet streams between endpoints, or by
   using an IP multicast group to distribute the RTP packet streams.
   These topologies can be implemented in a number of ways as discussed
   in [I-D.ietf-avtcore-rtp-topologies-update].

   While the use of IP multicast groups is popular in IPTV systems, the
   topologies based on RTP middleboxes are dominant in interactive video
   conferencing environments.  Topologies based on a mesh of unicast
   transport-layer flows to create a common RTP session have not seen
   widespread deployment.  Accordingly, WebRTC implementations are not
   expected to support topologies based on IP multicast groups.  WebRTC
   implementations are also not expected to support mesh-based
   topologies, such as a point-to-multipoint mesh configured as a single
   RTP session (Topo-Mesh in the terminology of
   [I-D.ietf-avtcore-rtp-topologies-update]).  However, a point-to-
   multipoint mesh constructed using several RTP sessions, in the WebRTC
   context using independent RTCPeerConnections can be expected to be
   utilised by WebRTC applications.

   WebRTC implementations of RTP endpoints implemented according to this
   memo are expected to support all the topologies described in
   [I-D.ietf-avtcore-rtp-topologies-update] where the RTP endpoints send
   and receive unicast RTP packet streams to some peer device, provided
   that peer can participate in performing congestion control on the RTP
   packet streams.  The peer device could be another RTP endpoint, or it
   could be an RTP middlebox that redistributes the RTP packet streams
   to other RTP endpoints.  This limitation means that some of the RTP
   middlebox-based topologies are not suitable for use in the WebRTC
   environment.  Specifically:

   o  Video switching MCUs (Topo-Video-switch-MCU) SHOULD NOT be used,
      since they make the use of RTCP for congestion control and quality
      of service reports problematic (see Section 3.6.2 of
      [I-D.ietf-avtcore-rtp-topologies-update]).

   o  Content modifying MCUs with RTCP termination (Topo-RTCP-
      terminating-MCU) SHOULD NOT be used since they break RTP loop
      detection, and prevent receivers from identifying active senders
      (see section 3.8 of [I-D.ietf-avtcore-rtp-topologies-update]).

   o  The Relay-Transport Translator (Topo-PtM-Trn-Translator) topology
      SHOULD NOT be used because its safe use requires a point to
      multipoint congestion control algorithm or RTP circuit breaker,
      which has not yet been standardised.

   The RTP extensions described in Section 5.1.1 to Section 5.1.6 are
   designed to be used with centralised conferencing, where an RTP
   middlebox (e.g., a conference bridge) receives a participant's RTP
   packet streams and distributes them to the other participants.  These
   extensions are not necessary for interoperability; an RTP end-point
   that does not implement these extensions will work correctly, but
   might offer poor performance.  Support for the listed extensions will
   greatly improve the quality of experience and, to provide a
   reasonable baseline quality, some of these extensions are mandatory
   to be supported by WebRTC end-points.

   The RTCP conferencing extensions are defined in Extended RTP Profile
   for Real-time Transport Control Protocol (RTCP)-Based Feedback (RTP/
   AVPF) [RFC4585] and the "Codec Control Messages in the RTP Audio-
   Visual Profile with Feedback (AVPF)" (CCM) [RFC5104] and are fully
   usable by the Secure variant of this profile (RTP/SAVPF) [RFC5124].


Cheers

Magnus Westerlund

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Cc: Colin Perkins <csp@csperkins.org>
Subject: [rtcweb] Text proposal for change regarding Audio level security in draft-ietf-rtcweb-rtp-usage
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WG,
(As Author)

There was some discussion at the meeting and afterwards regarding the
encryption of the audio level header extensions. This made us authors
review the text on this. We have clarified the text to require
implementation of header extension encryption and recommend usage,
unless explicitly requested to turn it off. See text below from Section
5.2.2., 5.2.3 and Section 13. We appreciate feedback on this change. We
plan to include this in the next draft update.


5.2.2.  Client-to-Mixer Audio Level

   The Client to Mixer Audio Level extension [RFC6464] is an RTP header
   extension used by a client to inform a mixer about the level of audio
   activity in the packet to which the header is attached.  This enables
   an RTP middlebox to make mixing or selection decisions without
   decoding or detailed inspection of the payload, reducing the
   complexity in some types of mixer.  It can also save decoding
   resources in receivers, which can choose to decode only the most
   relevant RTP packet streams based on audio activity levels.

   The Client-to-Mixer Audio Level [RFC6464] extension is RECOMMENDED to
   be implemented.  If the header extension is implemented, it is
   REQUIRED that implementations is capable of encrypting the header
   extension according to [RFC6904] since the information contained in
   these header extensions can be considered sensitive.  It is further
   RECOMMENDED that this encryption is used, unless the encryption has
   been explicitly requested to be turned off through API or signalling.

5.2.3.  Mixer-to-Client Audio Level

   The Mixer to Client Audio Level header extension [RFC6465] provides
   the client with the audio level of the different sources mixed into a
   common mix by a RTP mixer.  This enables a user interface to indicate
   the relative activity level of each session participant, rather than
   just being included or not based on the CSRC field.  This is a pure
   optimisations of non critical functions, and is hence OPTIONAL to
   implement.  If the header extension is implemented, it is REQUIRED
   that implementations are capable of encrypting the header extension
   according to [RFC6904] since the information contained in these
   header extensions can be considered sensitive.  It is further
   RECOMMENDED that this encryption is used, unless the encryption has
   been explicitly requested to be turned off through API or signalling.

Section 13:

   The guidelines in [RFC6562] apply when using variable bit rate (VBR)
   audio codecs such as Opus (see Section 4.3 for discussion of mandated
   audio codecs).  These guidelines in [RFC6562] also apply, but are of
   lesser importance, when using the client-to-mixer audio level header
   extensions (Section 5.2.2) or the mixer-to-client audio level header
   extensions (Section 5.2.3).  The use of the encryption of the header
   extensions are RECOMMENDED, unless there are known reasons, like
   middleboxes or third party monitoring that will greatly benefit from
   the information, and this has been expressed using API or signalling.
   If further evidence are produced to show that information leakage is
   significant from audio level indications, then use of encryption
   needs to be mandated at that time.


Cheers

Magnus Westerlund

----------------------------------------------------------------------
Services, Media and Network features, Ericsson Research EAB/TXM
----------------------------------------------------------------------
Ericsson AB                 | Phone  +46 10 7148287
Färögatan 6                 | Mobile +46 73 0949079
SE-164 80 Stockholm, Sweden | mailto: magnus.westerlund@ericsson.com
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From nobody Fri Apr  4 03:48:48 2014
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On 04 Apr 2014, at 10:51, Harald Alvestrand <harald@alvestrand.no> =
wrote:

> I assume this is the scheduler envisioned in -ndata:
>=20
>=20
>       SCTP_SS_PRIORITY:  Scheduling with different priorities is used.
>          Streams having a higher priority will be scheduled first and
>          when multiple streams have the same priority, the default
>          scheduling should be used for them.  The priority can be
>          assigned with the sctp_stream_value struct.  The higher the
>          assigned value, the lower the priority, that is the default
>          value 0 is the highest priority and therefore the default
>          scheduling will be used if no priorities have been assigned.
>=20
>=20
>=20
> This sounds like a "strict" scheduler, in that higher priority queues =
will starve out lower priority ones completely. I remember having the =
discussion at an IETF meeting about whether we wanted a "strict" =
scheduler or a "weighted round robin" scheduler for this, but I wouldn't =
trust my memory with what the conclusion was.
It is.
>=20
> Was the conclusion that we should do "strict" scheduling? If so, it =
may be best to make that consistent across the board - I had written in =
a "weighted" scheduler for media into the prioritization text that I =
started this thread with, but I think there's value to consistency.
No, in an IETF WG meeting we discussed this a while ago and the result =
was a non-strict
priority, weighted fair queueing. This scheduling discipline in not =
there yet.
>=20
> (Note: -ndata has SS_PRIORITY in one place and SS_PRIO / SS_PRIO_INTER =
in another place. Is there a subtlety here I'm not seeing?)
Yes, there is a difference. It is about interleaving. You can use =
SS_PRIO
with or without NDATA, SS_PRIO_INTER only with NDATA.
This document still needs some work... Even the constant names might =
change...

Best regards
Michael
>=20
>=20
>=20


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From: "Jeremy Laurenson (jlaurens)" <jlaurens@cisco.com>
To: Harald Alvestrand <harald@alvestrand.no>
Thread-Topic: [rtcweb] Some language on "prioritization"
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Do we really want the default (lazy coder) behavior to be highest priority =
in this case?

I assume the priority is browser-wide and so this could disrupt other app's=
 streams?

J


On Apr 4, 2014, at 4:52 AM, "Harald Alvestrand" <harald@alvestrand.no<mailt=
o:harald@alvestrand.no>> wrote:

I assume this is the scheduler envisioned in -ndata:



      SCTP_SS_PRIORITY:  Scheduling with different priorities is used.
         Streams having a higher priority will be scheduled first and
         when multiple streams have the same priority, the default
         scheduling should be used for them.  The priority can be
         assigned with the sctp_stream_value struct.  The higher the
         assigned value, the lower the priority, that is the default
         value 0 is the highest priority and therefore the default
         scheduling will be used if no priorities have been assigned.



This sounds like a "strict" scheduler, in that higher priority queues will =
starve out lower priority ones completely. I remember having the discussion=
 at an IETF meeting about whether we wanted a "strict" scheduler or a "weig=
hted round robin" scheduler for this, but I wouldn't trust my memory with w=
hat the conclusion was.

Was the conclusion that we should do "strict" scheduling? If so, it may be =
best to make that consistent across the board - I had written in a "weighte=
d" scheduler for media into the prioritization text that I started this thr=
ead with, but I think there's value to consistency.

(Note: -ndata has SS_PRIORITY in one place and SS_PRIO / SS_PRIO_INTER in a=
nother place. Is there a subtlety here I'm not seeing?)



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

--_000_CC8CDB3B530D40FAA3F491177C64A6BFciscocom_
Content-Type: text/html; charset="us-ascii"
Content-Transfer-Encoding: quoted-printable

<html>
<head>
<meta http-equiv=3D"Content-Type" content=3D"text/html; charset=3Dus-ascii"=
>
</head>
<body dir=3D"auto">
<div>Do we really want the default (lazy coder) behavior to be highest prio=
rity in this case?</div>
<div><br>
</div>
<div>I assume the priority is browser-wide and so this could disrupt other =
app's streams?<br>
<br>
J<br>
<blockquote type=3D"cite"><br>
</blockquote>
</div>
<div><br>
On Apr 4, 2014, at 4:52 AM, &quot;Harald Alvestrand&quot; &lt;<a href=3D"ma=
ilto:harald@alvestrand.no">harald@alvestrand.no</a>&gt; wrote:<br>
<br>
</div>
<blockquote type=3D"cite">
<div>I assume this is the scheduler envisioned in -ndata:<br>
<br>
<pre class=3D"newpage" style=3D"font-size: 1em; margin-top: 0px; margin-bot=
tom: 0px; page-break-before: always; color: rgb(0, 0, 0); font-style: norma=
l; font-variant: normal; font-weight: normal; letter-spacing: normal; line-=
height: normal; orphans: auto; text-align: start; text-indent: 0px; text-tr=
ansform: none; widows: auto; word-spacing: 0px; -webkit-text-stroke-width: =
0px;">
      SCTP_SS_PRIORITY:  Scheduling with different priorities is used.
         Streams having a higher priority will be scheduled first and
         when multiple streams have the same priority, the default
         scheduling should be used for them.  The priority can be
         assigned with the sctp_stream_value struct.  The higher the
         assigned value, the lower the priority, that is the default
         value 0 is the highest priority and therefore the default
         scheduling will be used if no priorities have been assigned.

</pre>
<br class=3D"Apple-interchange-newline">
This sounds like a &quot;strict&quot; scheduler, in that higher priority qu=
eues will starve out lower priority ones completely. I remember having the =
discussion at an IETF meeting about whether we wanted a &quot;strict&quot; =
scheduler or a &quot;weighted round robin&quot; scheduler for
 this, but I wouldn't trust my memory with what the conclusion was.<br>
<br>
Was the conclusion that we should do &quot;strict&quot; scheduling? If so, =
it may be best to make that consistent across the board - I had written in =
a &quot;weighted&quot; scheduler for media into the prioritization text tha=
t I started this thread with, but I think there's value
 to consistency.<br>
<br>
(Note: -ndata has SS_PRIORITY in one place and SS_PRIO / SS_PRIO_INTER in a=
nother place. Is there a subtlety here I'm not seeing?)<br>
<br>
<br>
<br>
</div>
</blockquote>
<blockquote type=3D"cite">
<div><span>_______________________________________________</span><br>
<span>rtcweb mailing list</span><br>
<span><a href=3D"mailto:rtcweb@ietf.org">rtcweb@ietf.org</a></span><br>
<span><a href=3D"https://www.ietf.org/mailman/listinfo/rtcweb">https://www.=
ietf.org/mailman/listinfo/rtcweb</a></span><br>
</div>
</blockquote>
</body>
</html>

--_000_CC8CDB3B530D40FAA3F491177C64A6BFciscocom_--


From nobody Fri Apr  4 05:49:55 2014
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References: <5339A120.3040909@alvestrand.no> <CABkgnnVUHUx+3wY3Dsi=UvNkUw_Es1apeMSonq7DtEg_3UKRNg@mail.gmail.com> <FBA84C78-FE8E-4FEF-8AD3-CAF24C57E512@lurchi.franken.de>, <5339AA58.9070301@alvestrand.no> <834D5209-5EEA-4001-B8ED-3835FC4D05FB@skype.net> <00af01cf4f59$fa617b90$ef2472b0$@stahl@intertex.se> <CB16B8F0-DDC2-4404-A81F-1B3101647DE9@lurchi.franken.de>, <533E729F.4000302@alvestrand.no> <CC8CDB3B-530D-40FA-A3F4-91177C64A6BF@cisco.com>
To: "Jeremy Laurenson (jlaurens)" <jlaurens@cisco.com>
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Subject: Re: [rtcweb] Some language on "prioritization"
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On 04 Apr 2014, at 13:55, Jeremy Laurenson (jlaurens) =
<jlaurens@cisco.com> wrote:

> Do we really want the default (lazy coder) behavior to be highest =
priority in this case?
Harald is referring to the SCTP socket API description. This won't be =
exposed
to the JS user. What will be exposed there needs to be defined by W3C =
and
will be mapped to the API of the SCTP implementation.
>=20
> I assume the priority is browser-wide and so this could disrupt other =
app's streams?
The priories described below are only relevant for the streams of an =
SCTP
association. It is just a scheduler which deals with the streams of an =
SCTP
association.

Best regards
Michael
>=20
> J
>>=20
>=20
> On Apr 4, 2014, at 4:52 AM, "Harald Alvestrand" <harald@alvestrand.no> =
wrote:
>=20
>> I assume this is the scheduler envisioned in -ndata:
>>=20
>>       SCTP_SS_PRIORITY:  Scheduling with different priorities is =
used.
>>          Streams having a higher priority will be scheduled first and
>>          when multiple streams have the same priority, the default
>>          scheduling should be used for them.  The priority can be
>>          assigned with the sctp_stream_value struct.  The higher the
>>          assigned value, the lower the priority, that is the default
>>          value 0 is the highest priority and therefore the default
>>          scheduling will be used if no priorities have been assigned.
>>=20
>>=20
>>=20
>> This sounds like a "strict" scheduler, in that higher priority queues =
will starve out lower priority ones completely. I remember having the =
discussion at an IETF meeting about whether we wanted a "strict" =
scheduler or a "weighted round robin" scheduler for  this, but I =
wouldn't trust my memory with what the conclusion was.
>>=20
>> Was the conclusion that we should do "strict" scheduling? If so, it =
may be best to make that consistent across the board - I had written in =
a "weighted" scheduler for media into the prioritization text that I =
started this thread with, but I think there's value to consistency.
>>=20
>> (Note: -ndata has SS_PRIORITY in one place and SS_PRIO / =
SS_PRIO_INTER in another place. Is there a subtlety here I'm not =
seeing?)
>>=20
>>=20
>>=20
>> _______________________________________________
>> rtcweb mailing list
>> rtcweb@ietf.org
>> https://www.ietf.org/mailman/listinfo/rtcweb


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From: Martin Thomson <martin.thomson@gmail.com>
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On 4 April 2014 02:02, Harald Alvestrand <harald@alvestrand.no> wrote:
> Which implementation strategy would people expect to use?

I think that for TURN, we were planning on using non-blocking writes
and discard packets that didn't make the cut.  This means that a
blocked congestion window would manifest as packet loss to upper
layers.

That way, at least congestion indications make it through to SCTP in
some fashion.  It's certainly not ideal, but I'm not sure that there
is a good solution in this space (I do know some people who claim to
have solved the problem of TCP congestion control within a TCP
congestion controlled envelope, maybe you can ask them.


From nobody Fri Apr  4 08:59:50 2014
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Subject: Re: [rtcweb] Text proposal for CNAME in draft-ietf-rtcweb-rtp-usage
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That's a lot of text.

In the interests of maintaining privacy (specifically unlinkability),
or the ability to do so, applications should be able to request that a
given RTCPeerConnection use a different CNAME to other
RTCPeerConnection instances in the same session context.  Otherwise, a
site operating a service would be unable to provide callers with
anonymity or pseudonymity toward different remote peers.

That's my minimum ask, but even that isn't ideal.

Better yet would be to be privacy protecting by default.  To avoid
issues where CNAME needs to be consistent, allow sites to link
RTCPeerConnections to allow for consistent CNAME use across
"sessions".

I haven't thought it through, but I can't think of a reason off the
top of my head not to allow sites to specify a CNAME directly.  After
all, a site is perfectly able to use signaling to link sessions, so
we're not giving anything away.  A random value could be chosen in the
absence of an explicit one.

I realize that this does - to some extent - compromise the purity and
integrity of your RTP usage.  I think that it's worth it.  The text
you've provided gives good context for that.  It just needs a
different conclusion.

Nits:

   This ought not be common, and if
   possible reference clocks ought to be mapped to each other and one
   chosen to be used with RTP and RTCP.

I can't parse this.

   Accordingly,
   support for generating a short-term persistent RTCP CNAMEs following
   [RFC7022] is REQUIRED to be implemented and RECOMMENDED to be used.

That's a strange statement.  How will you know if someone has
implemented 7022 if they aren't using it?  Seems like a SHOULD to me.


On 4 April 2014 02:09, Magnus Westerlund <magnus.westerlund@ericsson.com> w=
rote:
> WG,
> (As author)
>
> Based on the discussion we authors have drafted a proposal for changing
> the CNAME creation usage to be created unique between PeerConnections.
> Our draft text is the below one. Please review. If we receive no
> feedback we will include this in the next version of the draft that we
> hope to be able to produce next week.
>
> Section 4.1:
>
>    o  Support for multiple synchronisation contexts.  Participants that
>       send multiple simultaneous RTP packet streams do so as part of a
>       single synchronisation context, using a single RTCP CNAME for all
>       streams and allowing receivers to play the streams out in a
>       synchronised manner.  Receivers MUST support reception of multiple
>       RTCP CNAMEs in an RTP session, and hence multiple synchronisation
>       contexts, to support RTP middleboxes, and future evolution needing
>       multiple CNAMEs in an endpoint.  See also Section 4.9.
>
>
> Section 4.9:
>
> 4.9.  Generation of the RTCP Canonical Name (CNAME)
>
>    The RTCP Canonical Name (CNAME) provides a persistent transport-level
>    identifier for an RTP end-point.  While the Synchronisation Source
>    (SSRC) identifier for an RTP end-point can change if a collision is
>    detected, or when the RTP application is restarted, its RTCP CNAME is
>    meant to stay unchanged, so that RTP end-points can be uniquely
>    identified and associated with their RTP packet streams within a set
>    of related RTP sessions.  For proper functionality, each RTP end-
>    point needs to have at least one unique RTCP CNAME value.  From an
>    RTP perspective an end-point can have multiple CNAMEs, as the CNAME
>    also identifies a particular synchronisation context, i.e. all SSRC
>    associated with a CNAME share a common reference clock, and if an
>    end-point have SSRCs associated with different reference clocks it
>    will need to use multiple CNAMEs.  This ought not be common, and if
>    possible reference clocks ought to be mapped to each other and one
>    chosen to be used with RTP and RTCP.  Taking the discussion in
>    Section 11 into account a regular WebRTC end-point MUST NOT use more
>    than a single CNAME in the RTP sessions belonging to single
>    RTCPeerConnection.  RTP middleboxes MAY send RTP packet streams
>    identified by more than one CNAME, to allow them to avoid having to
>    resynchronize media from multiple different end-points part of a
>    multi-party RTP session.
>
>    The RTP specification [RFC3550] includes guidelines for choosing a
>    unique RTP CNAME, but these are not sufficient in the presence of NAT
>    devices.  In addition, long-term persistent identifiers can be
>    problematic from a privacy viewpoint (Section 13).  Accordingly,
>    support for generating a short-term persistent RTCP CNAMEs following
>    [RFC7022] is REQUIRED to be implemented and RECOMMENDED to be used.
>    Unless a persistent CNAME is explicitly configured, a unique CNAME(s)
>    MUST be generated for each synchronization context an end-point has
>    within each RTCPeerConnection, i.e. normally one across all the RTP
>    sessions within the scope of the RTCPeerConnection.
>
>       There are no known reasons for WebRTC end-points that aren't RTP
>       middleboxes or servers to configure a persistent identifiers,
>       these considerations are limited to RTP middleboxes.  Thus, such
>       configuration options SHOULD NOT be present in most WebRTC end-
>       point implementations to prevent misconfiguration or malicious
>       configuration that allows tracking or linking of a user.
>
>    An WebRTC end-point MUST support reception of any CNAME that matches
>    the syntax limitations specified by the RTP specification [RFC3550]
>    and cannot assume that any CNAME will be chosen according to the form
>    suggested above.
>
> Section 11:
>
>    The same MediaStreamTrack can also be included in multiple
>    MediaStreams, thus multiple sets of MediaStreams can implicitly need
>    to use the same synchronisation base.  To ensure that this works in
>    all cases, and don't forces a end-point to change synchronisation
>    base and CNAME in the middle of a ongoing delivery of any packet
>    streams, which would cause media disruption; all MediaStreamTracks
>    and their associated SSRCs originating from the same end-point needs
>    to be sent using the same CNAME within one RTCPeerConnection.  This
>    is motivating the strong recommendation in Section 4.9 to only use a
>    single CNAME.
>
>       Note: It is important that the same CNAME is not used in different
>       communication session contexts or origins, as that could enable
>       tracking of a user and its device usage of different services.
>       See Section 4.4.1 of Security Considerations for WebRTC
>       [I-D.ietf-rtcweb-security] for further discussion.
>
>       The requirement on using the same CNAME for all SSRCs that
>       originates from the same end-point, does not require middleboxes
>       that forwards traffic from multiple end-points to only use a
>       single CNAME.
>
>    The above will currently force a WebRTC end-point that receives an
>    MediaStreamTrack on one RTCPeerConnection and adds it as an outgoing
>    on any RTCPeerConnection to perform resynchronisation of the stream.
>    This, as the sending party needs to change the CNAME, which implies
>    that it has to use a locally available system clock as timebase for
>    the synchronisation.  Thus, the relative relation between the
>    timebase of the incoming stream and the system sending out needs to
>    defined.  This relation also needs monitoring for clock drift and
>    likely adjustments of the synchronisation.  The sending entity is
>    also responsible for congestion control for its the sent streams.  In
>    cases of packet loss the loss of incoming data also needs to be
>    handled.  This leads to the observation that the method that is least
>    likely to cause issues or interruptions in the outgoing source packet
>    stream is a model of full decoding, including repair etc followed by
>    encoding of the media again into the outgoing packet stream.
>    Optimisations of this method is clearly possible and implementation
>    specific.
>
>    A WebRTC end-point MUST support receiving multiple MediaStreamTracks,
>    where each of different MediaStreamTracks (and their sets of
>    associated packet streams) uses different CNAMEs.  However,
>    MediaStreamTracks that are received with different CNAMEs have no
>    defined synchronisation.
>
>       Note: The motivation for supporting reception of multiple CNAMEs
>       are to allow for forward compatibility with any future changes
>       that enables more efficient stream handling when end-points relay/
>       forward streams.  It also ensures that end-points can interoperate
>       with certain types of multi-stream middleboxes or end-points that
>       are not WebRTC.
>
> --
>
> Magnus Westerlund
>
> ----------------------------------------------------------------------
> Services, Media and Network features, Ericsson Research EAB/TXM
> ----------------------------------------------------------------------
> 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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From: Colin Perkins <csp@csperkins.org>
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Subject: Re: [rtcweb] Text proposal for CNAME in draft-ietf-rtcweb-rtp-usage
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Martin,

I=92m perhaps misunderstanding something, or we may not have phrased the =
text clearly, but what you say you want generally matches my =
understanding of what the draft provides. Can you expand on what parts =
of the text you think are problematic?

Colin




On 4 Apr 2014, at 16:59, Martin Thomson <martin.thomson@gmail.com> =
wrote:
> That's a lot of text.
>=20
> In the interests of maintaining privacy (specifically unlinkability),
> or the ability to do so, applications should be able to request that a
> given RTCPeerConnection use a different CNAME to other
> RTCPeerConnection instances in the same session context.  Otherwise, a
> site operating a service would be unable to provide callers with
> anonymity or pseudonymity toward different remote peers.
>=20
> That's my minimum ask, but even that isn't ideal.
>=20
> Better yet would be to be privacy protecting by default.  To avoid
> issues where CNAME needs to be consistent, allow sites to link
> RTCPeerConnections to allow for consistent CNAME use across
> "sessions".
>=20
> I haven't thought it through, but I can't think of a reason off the
> top of my head not to allow sites to specify a CNAME directly.  After
> all, a site is perfectly able to use signaling to link sessions, so
> we're not giving anything away.  A random value could be chosen in the
> absence of an explicit one.
>=20
> I realize that this does - to some extent - compromise the purity and
> integrity of your RTP usage.  I think that it's worth it.  The text
> you've provided gives good context for that.  It just needs a
> different conclusion.
>=20
> Nits:
>=20
>   This ought not be common, and if
>   possible reference clocks ought to be mapped to each other and one
>   chosen to be used with RTP and RTCP.
>=20
> I can't parse this.
>=20
>   Accordingly,
>   support for generating a short-term persistent RTCP CNAMEs following
>   [RFC7022] is REQUIRED to be implemented and RECOMMENDED to be used.
>=20
> That's a strange statement.  How will you know if someone has
> implemented 7022 if they aren't using it?  Seems like a SHOULD to me.
>=20
>=20
> On 4 April 2014 02:09, Magnus Westerlund =
<magnus.westerlund@ericsson.com> wrote:
>> WG,
>> (As author)
>>=20
>> Based on the discussion we authors have drafted a proposal for =
changing
>> the CNAME creation usage to be created unique between =
PeerConnections.
>> Our draft text is the below one. Please review. If we receive no
>> feedback we will include this in the next version of the draft that =
we
>> hope to be able to produce next week.
>>=20
>> Section 4.1:
>>=20
>>   o  Support for multiple synchronisation contexts.  Participants =
that
>>      send multiple simultaneous RTP packet streams do so as part of a
>>      single synchronisation context, using a single RTCP CNAME for =
all
>>      streams and allowing receivers to play the streams out in a
>>      synchronised manner.  Receivers MUST support reception of =
multiple
>>      RTCP CNAMEs in an RTP session, and hence multiple =
synchronisation
>>      contexts, to support RTP middleboxes, and future evolution =
needing
>>      multiple CNAMEs in an endpoint.  See also Section 4.9.
>>=20
>>=20
>> Section 4.9:
>>=20
>> 4.9.  Generation of the RTCP Canonical Name (CNAME)
>>=20
>>   The RTCP Canonical Name (CNAME) provides a persistent =
transport-level
>>   identifier for an RTP end-point.  While the Synchronisation Source
>>   (SSRC) identifier for an RTP end-point can change if a collision is
>>   detected, or when the RTP application is restarted, its RTCP CNAME =
is
>>   meant to stay unchanged, so that RTP end-points can be uniquely
>>   identified and associated with their RTP packet streams within a =
set
>>   of related RTP sessions.  For proper functionality, each RTP end-
>>   point needs to have at least one unique RTCP CNAME value.  =46rom =
an
>>   RTP perspective an end-point can have multiple CNAMEs, as the CNAME
>>   also identifies a particular synchronisation context, i.e. all SSRC
>>   associated with a CNAME share a common reference clock, and if an
>>   end-point have SSRCs associated with different reference clocks it
>>   will need to use multiple CNAMEs.  This ought not be common, and if
>>   possible reference clocks ought to be mapped to each other and one
>>   chosen to be used with RTP and RTCP.  Taking the discussion in
>>   Section 11 into account a regular WebRTC end-point MUST NOT use =
more
>>   than a single CNAME in the RTP sessions belonging to single
>>   RTCPeerConnection.  RTP middleboxes MAY send RTP packet streams
>>   identified by more than one CNAME, to allow them to avoid having to
>>   resynchronize media from multiple different end-points part of a
>>   multi-party RTP session.
>>=20
>>   The RTP specification [RFC3550] includes guidelines for choosing a
>>   unique RTP CNAME, but these are not sufficient in the presence of =
NAT
>>   devices.  In addition, long-term persistent identifiers can be
>>   problematic from a privacy viewpoint (Section 13).  Accordingly,
>>   support for generating a short-term persistent RTCP CNAMEs =
following
>>   [RFC7022] is REQUIRED to be implemented and RECOMMENDED to be used.
>>   Unless a persistent CNAME is explicitly configured, a unique =
CNAME(s)
>>   MUST be generated for each synchronization context an end-point has
>>   within each RTCPeerConnection, i.e. normally one across all the RTP
>>   sessions within the scope of the RTCPeerConnection.
>>=20
>>      There are no known reasons for WebRTC end-points that aren't RTP
>>      middleboxes or servers to configure a persistent identifiers,
>>      these considerations are limited to RTP middleboxes.  Thus, such
>>      configuration options SHOULD NOT be present in most WebRTC end-
>>      point implementations to prevent misconfiguration or malicious
>>      configuration that allows tracking or linking of a user.
>>=20
>>   An WebRTC end-point MUST support reception of any CNAME that =
matches
>>   the syntax limitations specified by the RTP specification [RFC3550]
>>   and cannot assume that any CNAME will be chosen according to the =
form
>>   suggested above.
>>=20
>> Section 11:
>>=20
>>   The same MediaStreamTrack can also be included in multiple
>>   MediaStreams, thus multiple sets of MediaStreams can implicitly =
need
>>   to use the same synchronisation base.  To ensure that this works in
>>   all cases, and don't forces a end-point to change synchronisation
>>   base and CNAME in the middle of a ongoing delivery of any packet
>>   streams, which would cause media disruption; all MediaStreamTracks
>>   and their associated SSRCs originating from the same end-point =
needs
>>   to be sent using the same CNAME within one RTCPeerConnection.  This
>>   is motivating the strong recommendation in Section 4.9 to only use =
a
>>   single CNAME.
>>=20
>>      Note: It is important that the same CNAME is not used in =
different
>>      communication session contexts or origins, as that could enable
>>      tracking of a user and its device usage of different services.
>>      See Section 4.4.1 of Security Considerations for WebRTC
>>      [I-D.ietf-rtcweb-security] for further discussion.
>>=20
>>      The requirement on using the same CNAME for all SSRCs that
>>      originates from the same end-point, does not require middleboxes
>>      that forwards traffic from multiple end-points to only use a
>>      single CNAME.
>>=20
>>   The above will currently force a WebRTC end-point that receives an
>>   MediaStreamTrack on one RTCPeerConnection and adds it as an =
outgoing
>>   on any RTCPeerConnection to perform resynchronisation of the =
stream.
>>   This, as the sending party needs to change the CNAME, which implies
>>   that it has to use a locally available system clock as timebase for
>>   the synchronisation.  Thus, the relative relation between the
>>   timebase of the incoming stream and the system sending out needs to
>>   defined.  This relation also needs monitoring for clock drift and
>>   likely adjustments of the synchronisation.  The sending entity is
>>   also responsible for congestion control for its the sent streams.  =
In
>>   cases of packet loss the loss of incoming data also needs to be
>>   handled.  This leads to the observation that the method that is =
least
>>   likely to cause issues or interruptions in the outgoing source =
packet
>>   stream is a model of full decoding, including repair etc followed =
by
>>   encoding of the media again into the outgoing packet stream.
>>   Optimisations of this method is clearly possible and implementation
>>   specific.
>>=20
>>   A WebRTC end-point MUST support receiving multiple =
MediaStreamTracks,
>>   where each of different MediaStreamTracks (and their sets of
>>   associated packet streams) uses different CNAMEs.  However,
>>   MediaStreamTracks that are received with different CNAMEs have no
>>   defined synchronisation.
>>=20
>>      Note: The motivation for supporting reception of multiple CNAMEs
>>      are to allow for forward compatibility with any future changes
>>      that enables more efficient stream handling when end-points =
relay/
>>      forward streams.  It also ensures that end-points can =
interoperate
>>      with certain types of multi-stream middleboxes or end-points =
that
>>      are not WebRTC.
>>=20
>> --
>>=20
>> Magnus Westerlund
>>=20
>> =
----------------------------------------------------------------------
>> Services, Media and Network features, Ericsson Research EAB/TXM
>> =
----------------------------------------------------------------------
>> Ericsson AB                 | Phone  +46 10 7148287
>> F=E4r=F6gatan 6                 | Mobile +46 73 0949079
>> SE-164 80 Stockholm, Sweden | mailto: magnus.westerlund@ericsson.com
>> =
----------------------------------------------------------------------
>>=20
>> _______________________________________________
>> rtcweb mailing list
>> rtcweb@ietf.org
>> https://www.ietf.org/mailman/listinfo/rtcweb



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




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On 4 April 2014 09:34, Colin Perkins <csp@csperkins.org> wrote:
> I=E2=80=99m perhaps misunderstanding something, or we may not have phrase=
d the text clearly, but what you say you want generally matches my understa=
nding of what the draft provides. Can you expand on what parts of the text =
you think are problematic?

The first paragraph of Section 11 seems directly contrary to what I outline=
d.

I don't know how you interpret this, but I read the following text as
saying the complete opposite of my suggestion to allow setting of
CNAME:

      There are no known reasons for WebRTC end-points that aren't RTP
      middleboxes or servers to configure a persistent identifiers,
      these considerations are limited to RTP middleboxes.  Thus, such
      configuration options SHOULD NOT be present in most WebRTC end-
      point implementations to prevent misconfiguration or malicious
      configuration that allows tracking or linking of a user.

Now that I've gone back and re-read again, I think that this text is
possibly confused, and certainly confusing, with respect to use of the
various contexts in play.

 * The text uses RTP end-point, which I infer corresponds to the
(local) end of the ICE component in use (i.e., the selected transport
5-tuple).  An RTP end-point is required to have a unique CNAME, but
that's not consistent with unbundled (SDP negotiated) sessions (i.e.,
what corresponds to a single unbundled RTCPeerConnection), which I
assume will want to use a consistent CNAME if at all possible.

 * We also have guidance regarding RTCPeerConnection.

 * And there is also guidance regarding the browser session,
presumably scoped to a given origin [RFC6454].

 * I'll list RTP session for completeness, though I don't think it's a
useful context to use in any case.

I think that the advice could be much clearer, once all the context is
properly established:

  1. MUST use the same CNAME for an RTP end-point, unless different
SSRCs are on a different time base (i.e., the uncommon case)

  2. MUST use different CNAME for different RTCPeerConnection
instances, following RFC 7022, unless the using application explicitly
provides a CNAME.

The latter point covers the origin case adequately.  Servers and
middleboxes can be given a blanket exemption if you like, allowing for
the fact that they should be doing all the normal and proper RTP
things.


From nobody Fri Apr  4 13:42:17 2014
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I'm confused by this, in at least a couple of respects:

- IIUC this isolation is a browser thing. If the other end doesn't
   have a browser then what does it mean?

- Isn't the isolation supposed to be on a per-stream basis?
   (Or is it really on a peer-connection basis.) If it is per-stream
   then how does signaling it on a per-TLS basis work?

- The use of DTLS to do SRTP keying is a sort of hack, since it
   isn't about the DTLS payload at all. I would think that whatever
   new you introduce into TLS for this would most naturally apply
   to the DTLS payload (the data channels).

	Thanks,
	Paul

On 4/3/14 8:14 PM, Martin Thomson wrote:
> As I described briefly at the last meeting, ensuring that media is
> isolated from the application or web site is a key part of addressing
> our security goals.
>
> The key part of that is making sure that any media that is isolated on
> the sending side of RTCPeerConnection remains isolated when it reaches
> the other side.
>
> As I noted, this is also necessary in order to ensure the integrity of
> the same origin model.  In that model, cross origin media is required
> to be inaccessible to content, and as it stands RTCPeerConnection
> could be used to work around those restrictions (implementations can
> implement other protections, as Firefox already does).
>
> The alternatives as I see them (and I hope that this is sufficiently
> exhaustive) are:
>
>   1. ask the TLS working group for a TLS-based solution
>   2. build something into the session signaling (i.e., new SDP bits)
>   3. give up on the idea
>
> I prefer 1 for reasons already outlined.
>
> I propose that we formally request the TLS working group recommend or
> define a mechanism whereby we can signal in the TLS handshake that the
> session contents are to be protected from access (where the
> description of that protection may need to our responsibility).
>
> I have pointed to draft-thomson-tls-acp as a potential solution here,
> but others have noted that ALPN tokens could be used.  I expect that
> TLS are capable of making the right decision here.
>
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb
>


From nobody Fri Apr  4 13:50:56 2014
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Fair questions...

On 4 April 2014 13:42, Paul Kyzivat <pkyzivat@alum.mit.edu> wrote:
> - IIUC this isolation is a browser thing. If the other end doesn't
>   have a browser then what does it mean?

Yes, this is specific to a browser.  A non-browser peer would probably
respect this isolation property without any effort at all.  After all,
few non-browser applications permit the running of arbitrary,
remotely-source code in the way that a browser does.

> - Isn't the isolation supposed to be on a per-stream basis?
>   (Or is it really on a peer-connection basis.) If it is per-stream
>   then how does signaling it on a per-TLS basis work?

The question of scope is an important one.  The isolation property is,
at least at the most abstract layer, applicable to a MediaStreamTrack.
 I believe that the right approach is to broaden this scope to
RTCPeerConnection, where the identity controls exist.  This means that
you won't be able to mix isolated and non-isolated streams in the same
RTCPeerConnection.

> - The use of DTLS to do SRTP keying is a sort of hack, since it
>   isn't about the DTLS payload at all. I would think that whatever
>   new you introduce into TLS for this would most naturally apply
>   to the DTLS payload (the data channels).

This is true only in the strictest academic or abstract sense.  We are
already taking things from the DTLS handshake and using those things
to make assertions about the properties of the SRTP.  Certificates.

Since you raise data channels, I'll note that we have no way to use
data channels in an isolated fashion.  Thus, I conclude that if we
intend to make this property consistent, we should not be using data
channels in the same RTCPeerConnection as media streams.  Of course,
I'm open to taking the pragmatists approach too, unless we think that
there is a use case for - say, file transfer - where it makes sense
for data to be isolated in the same way I'm proposing that media be.


From nobody Sun Apr  6 11:37:20 2014
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On 4/4/14 4:50 PM, Martin Thomson wrote:
> Fair questions...
>
> On 4 April 2014 13:42, Paul Kyzivat <pkyzivat@alum.mit.edu> wrote:
>> - IIUC this isolation is a browser thing. If the other end doesn't
>>    have a browser then what does it mean?
>
> Yes, this is specific to a browser.  A non-browser peer would probably
> respect this isolation property without any effort at all.  After all,
> few non-browser applications permit the running of arbitrary,
> remotely-source code in the way that a browser does.

But ISTM there would need to be a different set of limitations for 
non-browsers. For instance, to not echo the stream back to the source. 
And not to store the stream in a place that the sender can retrieve.

Maybe it is possible to define a set of rules, but I don't have a vision 
of what that would be.

>> - Isn't the isolation supposed to be on a per-stream basis?
>>    (Or is it really on a peer-connection basis.) If it is per-stream
>>    then how does signaling it on a per-TLS basis work?
>
> The question of scope is an important one.  The isolation property is,
> at least at the most abstract layer, applicable to a MediaStreamTrack.
>   I believe that the right approach is to broaden this scope to
> RTCPeerConnection, where the identity controls exist.  This means that
> you won't be able to mix isolated and non-isolated streams in the same
> RTCPeerConnection.

I'll wait to see what others have to say about this.
I haven't been able to wrap my head around it.

>> - The use of DTLS to do SRTP keying is a sort of hack, since it
>>    isn't about the DTLS payload at all. I would think that whatever
>>    new you introduce into TLS for this would most naturally apply
>>    to the DTLS payload (the data channels).
>
> This is true only in the strictest academic or abstract sense.  We are
> already taking things from the DTLS handshake and using those things
> to make assertions about the properties of the SRTP.  Certificates.
>
> Since you raise data channels, I'll note that we have no way to use
> data channels in an isolated fashion.  Thus, I conclude that if we
> intend to make this property consistent, we should not be using data
> channels in the same RTCPeerConnection as media streams.  Of course,
> I'm open to taking the pragmatists approach too, unless we think that
> there is a use case for - say, file transfer - where it makes sense
> for data to be isolated in the same way I'm proposing that media be.

Again I'm interested to hear what others have to say about this.

	Thanks,
	Paul



From nobody Sun Apr  6 19:42:30 2014
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Subject: Re: [rtcweb] Draft proposal for updating Multiparty topologies in draft-ietf-rtcweb-rtp-usage
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Hello Magnus,

Please see my comments marked CNG below.

Regards, Christian

On 4/04/2014 8:24 PM, Magnus Westerlund wrote:
> WG,
> (As Author)
>
> Colin and I have been working on resolving terminology usage and in
> general giving the draft a polish over before the WG last call. One
> section that has gotten quite some attention from us is the below one.
> The changes are significantly from an editorial stand point. However,
> the intended content has not been intended to be changed, although
> clarified and better motivated. So please review it, we intended to
> include this in the upcoming draft update. Feedback appreciated.
>
> 5.1.  Conferencing Extensions and Topologies
>
>     RTP is a protocol that inherently supports group communication.
>     Groups can be implemented by having each endpoint sending its RTP
>     packet streams to an RTP middlebox that redistributes the traffic, by
>     using a mesh of unicast RTP packet streams between endpoints, or by
>     using an IP multicast group to distribute the RTP packet streams.
>     These topologies can be implemented in a number of ways as discussed
>     in [I-D.ietf-avtcore-rtp-topologies-update].
>
>     While the use of IP multicast groups is popular in IPTV systems, the
>     topologies based on RTP middleboxes are dominant in interactive video
>     conferencing environments.  Topologies based on a mesh of unicast
>     transport-layer flows to create a common RTP session have not seen
>     widespread deployment.  Accordingly, WebRTC implementations are not
>     expected to support topologies based on IP multicast groups.  WebRTC
>     implementations are also not expected to support mesh-based
>     topologies, such as a point-to-multipoint mesh configured as a single
>     RTP session (Topo-Mesh in the terminology of
>     [I-D.ietf-avtcore-rtp-topologies-update]).  However, a point-to-
>     multipoint mesh constructed using several RTP sessions, in the WebRTC
>     context using independent RTCPeerConnections can be expected to be
>     utilised by WebRTC applications.
>
>     WebRTC implementations of RTP endpoints implemented according to this
>     memo are expected to support all the topologies described in
>     [I-D.ietf-avtcore-rtp-topologies-update] where the RTP endpoints send
>     and receive unicast RTP packet streams to some peer device, provided
>     that peer can participate in performing congestion control on the RTP
>     packet streams.  The peer device could be another RTP endpoint, or it
>     could be an RTP middlebox that redistributes the RTP packet streams
>     to other RTP endpoints.  This limitation means that some of the RTP
>     middlebox-based topologies are not suitable for use in the WebRTC
>     environment.  Specifically:
>
>     o  Video switching MCUs (Topo-Video-switch-MCU) SHOULD NOT be used,
>        since they make the use of RTCP for congestion control and quality
>        of service reports problematic (see Section 3.6.2 of
>        [I-D.ietf-avtcore-rtp-topologies-update]).
[CNG] Is 3.6.2 the right reference? clause 3.8 talks about 
Topo-Video-switch-MCU and the reports problem.
>
>     o  Content modifying MCUs with RTCP termination (Topo-RTCP-
>        terminating-MCU) SHOULD NOT be used since they break RTP loop
>        detection, and prevent receivers from identifying active senders
>        (see section 3.8 of [I-D.ietf-avtcore-rtp-topologies-update]).
[CNG] Is 3.8 the right reference? clause 3.9 talks about loop detection 
for Topo-RTCP-
       terminating-MCU.
>
>     o  The Relay-Transport Translator (Topo-PtM-Trn-Translator) topology
>        SHOULD NOT be used because its safe use requires a point to
>        multipoint congestion control algorithm or RTP circuit breaker,
>        which has not yet been standardised.
>
>     The RTP extensions described in Section 5.1.1 to Section 5.1.6 are
>     designed to be used with centralised conferencing, where an RTP
>     middlebox (e.g., a conference bridge) receives a participant's RTP
>     packet streams and distributes them to the other participants.  These
>     extensions are not necessary for interoperability; an RTP end-point
>     that does not implement these extensions will work correctly, but
>     might offer poor performance.  Support for the listed extensions will
>     greatly improve the quality of experience and, to provide a
>     reasonable baseline quality, some of these extensions are mandatory
>     to be supported by WebRTC end-points.
>
>     The RTCP conferencing extensions are defined in Extended RTP Profile
>     for Real-time Transport Control Protocol (RTCP)-Based Feedback (RTP/
>     AVPF) [RFC4585] and the "Codec Control Messages in the RTP Audio-
>     Visual Profile with Feedback (AVPF)" (CCM) [RFC5104] and are fully
>     usable by the Secure variant of this profile (RTP/SAVPF) [RFC5124].
>
>
> Cheers
>
> Magnus Westerlund
>
> ----------------------------------------------------------------------
> Services, Media and Network features, Ericsson Research EAB/TXM
> ----------------------------------------------------------------------
> Ericsson AB                 | Phone  +46 10 7148287
> Färögatan 6                 | Mobile +46 73 0949079
> SE-164 80 Stockholm, Sweden | mailto: magnus.westerlund@ericsson.com
> ----------------------------------------------------------------------
>
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb
>


From nobody Mon Apr  7 01:03:11 2014
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Subject: [rtcweb] Isolating data channels (Re: Asking TLS for help with media isolation)
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On 04/04/2014 10:50 PM, Martin Thomson wrote:
> Since you raise data channels, I'll note that we have no way to use 
> data channels in an isolated fashion. Thus, I conclude that if we 
> intend to make this property consistent, we should not be using data 
> channels in the same RTCPeerConnection as media streams. Of course, 
> I'm open to taking the pragmatists approach too, unless we think that 
> there is a use case for - say, file transfer - where it makes sense 
> for data to be isolated in the same way I'm proposing that media be. 

Focusing on the "should we ever isolate data channels" question:

If we ever define a protocol carried *over* data channels that is 
interpreted in the browser, not in the Javascript, and exposes a 
different API to Javascript than the datachannel API, this might make sense.

In that case, I think we should investigate whether it's possible to 
make use of the "protocol" field (put some reserved values into it 
saying "for protocols that look like *this*, Javascript can't set 
them?), it's the logical place to say "this is datachannels, but with 
some extra rules".

Wild suggestion: if you want per-track isolation properties, open up a 
data channel with a protocol called '*WebRTCIsolationInfo' and use it to 
send information about the isolation status of each track, thereby also 
providing a working example for the rule 'all data channels that have 
protocols starting with "*" are for browser internal usage'.....


From nobody Mon Apr  7 01:08:15 2014
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From: Christer Holmberg <christer.holmberg@ericsson.com>
To: "rtcweb@ietf.org" <rtcweb@ietf.org>
Thread-Topic: [MMUSIC] I-D Action: draft-ietf-mmusic-sdp-bundle-negotiation-06.txt
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Subject: [rtcweb] FW: [MMUSIC] I-D Action: draft-ietf-mmusic-sdp-bundle-negotiation-06.txt
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FYI,

To avoid cross-posting, if you have any comments on the document, please po=
st them on the MMUSIC list.

Regards,

Christer

-----Original Message-----
From: mmusic [mailto:mmusic-bounces@ietf.org] On Behalf Of internet-drafts@=
ietf.org
Sent: 7. huhtikuuta 2014 10:56
To: i-d-announce@ietf.org
Cc: mmusic@ietf.org
Subject: [MMUSIC] I-D Action: draft-ietf-mmusic-sdp-bundle-negotiation-06.t=
xt


A New Internet-Draft is available from the on-line Internet-Drafts director=
ies.
 This draft is a work item of the Multiparty Multimedia Session Control Wor=
king Group of the IETF.

        Title           : Multiplexing Negotiation Using Session Descriptio=
n Protocol (SDP) Port Numbers
        Authors         : Christer Holmberg
                          Harald Tveit Alvestrand
                          Cullen Jennings
	Filename        : draft-ietf-mmusic-sdp-bundle-negotiation-06.txt
	Pages           : 38
	Date            : 2014-04-07

Abstract:
   This specification defines a new SDP Grouping Framework extension,
   "BUNDLE", that can be used with the Session Description Protocol
   (SDP) Offer/Answer mechanism to negotiate the usage of bundled media,
   which refers to the usage of a single 5-tuple for sending and
   receiving media associated with multiple SDP media descriptions ("m=3D"
   lines).

   This specification also updates sections 5.1, 8.1 and 8.2 of RFC
   3264, in order to allow an answerer to in an SDP answer assign a non-
   zero port value to an "m=3D" line, even if the offerer in the
   associated SDP offer had assigned a zero port value to the "m=3D" line.


The IETF datatracker status page for this draft is:
https://datatracker.ietf.org/doc/draft-ietf-mmusic-sdp-bundle-negotiation/

There's also a htmlized version available at:
http://tools.ietf.org/html/draft-ietf-mmusic-sdp-bundle-negotiation-06

A diff from the previous version is available at:
http://www.ietf.org/rfcdiff?url2=3Ddraft-ietf-mmusic-sdp-bundle-negotiation=
-06


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

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

_______________________________________________
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From nobody Mon Apr  7 01:12:40 2014
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Thanks for the update!

It does seem reasonably clear now - clear enough that I found one point 
that I violently disagree with (if I'm interpreting it correctly). See 
below.

I don't have any negative comments on the rest of the section.

On 04/04/2014 11:24 AM, Magnus Westerlund wrote:
> WG,
> (As Author)
>
> Colin and I have been working on resolving terminology usage and in
> general giving the draft a polish over before the WG last call. One
> section that has gotten quite some attention from us is the below one.
> The changes are significantly from an editorial stand point. However,
> the intended content has not been intended to be changed, although
> clarified and better motivated. So please review it, we intended to
> include this in the upcoming draft update. Feedback appreciated.
>
> 5.1.  Conferencing Extensions and Topologies
>
>     RTP is a protocol that inherently supports group communication.
>     Groups can be implemented by having each endpoint sending its RTP
>     packet streams to an RTP middlebox that redistributes the traffic, by
>     using a mesh of unicast RTP packet streams between endpoints, or by
>     using an IP multicast group to distribute the RTP packet streams.
>     These topologies can be implemented in a number of ways as discussed
>     in [I-D.ietf-avtcore-rtp-topologies-update].
>
>     While the use of IP multicast groups is popular in IPTV systems, the
>     topologies based on RTP middleboxes are dominant in interactive video
>     conferencing environments.  Topologies based on a mesh of unicast
>     transport-layer flows to create a common RTP session have not seen
>     widespread deployment.  Accordingly, WebRTC implementations are not
>     expected to support topologies based on IP multicast groups.  WebRTC
>     implementations are also not expected to support mesh-based
>     topologies, such as a point-to-multipoint mesh configured as a single
>     RTP session (Topo-Mesh in the terminology of
>     [I-D.ietf-avtcore-rtp-topologies-update]).  However, a point-to-
>     multipoint mesh constructed using several RTP sessions, in the WebRTC
>     context using independent RTCPeerConnections can be expected to be
>     utilised by WebRTC applications.
>
>     WebRTC implementations of RTP endpoints implemented according to this
>     memo are expected to support all the topologies described in
>     [I-D.ietf-avtcore-rtp-topologies-update] where the RTP endpoints send
>     and receive unicast RTP packet streams to some peer device, provided
>     that peer can participate in performing congestion control on the RTP
>     packet streams.  The peer device could be another RTP endpoint, or it
>     could be an RTP middlebox that redistributes the RTP packet streams
>     to other RTP endpoints.  This limitation means that some of the RTP
>     middlebox-based topologies are not suitable for use in the WebRTC
>     environment.  Specifically:
>
>     o  Video switching MCUs (Topo-Video-switch-MCU) SHOULD NOT be used,
>        since they make the use of RTCP for congestion control and quality
>        of service reports problematic (see Section 3.6.2 of
>        [I-D.ietf-avtcore-rtp-topologies-update]).
>
>     o  Content modifying MCUs with RTCP termination (Topo-RTCP-
>        terminating-MCU) SHOULD NOT be used since they break RTP loop
>        detection, and prevent receivers from identifying active senders
>        (see section 3.8 of [I-D.ietf-avtcore-rtp-topologies-update]).

In my (strong) opinion: This recommendation is wrong.

All central conferencing units that use individual RTP sessions with 
clients fall into this category. This includes, I believe, every single 
multiuser video conferencing system based on WebRTC in existence (and 
there are many of them).

I believe the recommendation should be:

* Content modifying MCUs with RTCP termination 
(Topo-RTCP-terminating-MCU) MAY be used. Note that in this scenario, RTP 
loop detection and identification of active senders is the resposibility 
of the application; since the clients are isolated from each other at 
the RTP layer, RTP cannot assist with these functions.

>
>     o  The Relay-Transport Translator (Topo-PtM-Trn-Translator) topology
>        SHOULD NOT be used because its safe use requires a point to
>        multipoint congestion control algorithm or RTP circuit breaker,
>        which has not yet been standardised.
>
>     The RTP extensions described in Section 5.1.1 to Section 5.1.6 are
>     designed to be used with centralised conferencing, where an RTP
>     middlebox (e.g., a conference bridge) receives a participant's RTP
>     packet streams and distributes them to the other participants.  These
>     extensions are not necessary for interoperability; an RTP end-point
>     that does not implement these extensions will work correctly, but
>     might offer poor performance.  Support for the listed extensions will
>     greatly improve the quality of experience and, to provide a
>     reasonable baseline quality, some of these extensions are mandatory
>     to be supported by WebRTC end-points.
>
>     The RTCP conferencing extensions are defined in Extended RTP Profile
>     for Real-time Transport Control Protocol (RTCP)-Based Feedback (RTP/
>     AVPF) [RFC4585] and the "Codec Control Messages in the RTP Audio-
>     Visual Profile with Feedback (AVPF)" (CCM) [RFC5104] and are fully
>     usable by the Secure variant of this profile (RTP/SAVPF) [RFC5124].
>
>
> Cheers
>
> Magnus Westerlund
>
> ----------------------------------------------------------------------
> Services, Media and Network features, Ericsson Research EAB/TXM
> ----------------------------------------------------------------------
> Ericsson AB                 | Phone  +46 10 7148287
> Färögatan 6                 | Mobile +46 73 0949079
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> ----------------------------------------------------------------------
>
> _______________________________________________
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> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb


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Subject: Re: [rtcweb] Text proposal for CNAME in draft-ietf-rtcweb-rtp-usage
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On 2014-04-04 19:37, Martin Thomson wrote:
> On 4 April 2014 09:34, Colin Perkins <csp@csperkins.org> wrote:
>> Iâ€™m perhaps misunderstanding something, or we may not have phrased the text clearly, but what you say you want generally matches my understanding of what the draft provides. Can you expand on what parts of the text you think are problematic?
> 
> The first paragraph of Section 11 seems directly contrary to what I outlined.
> 
> I don't know how you interpret this, but I read the following text as
> saying the complete opposite of my suggestion to allow setting of
> CNAME:
> 
>       There are no known reasons for WebRTC end-points that aren't RTP
>       middleboxes or servers to configure a persistent identifiers,
>       these considerations are limited to RTP middleboxes.  Thus, such
>       configuration options SHOULD NOT be present in most WebRTC end-
>       point implementations to prevent misconfiguration or malicious
>       configuration that allows tracking or linking of a user.
> 
> Now that I've gone back and re-read again, I think that this text is
> possibly confused, and certainly confusing, with respect to use of the
> various contexts in play.
> 
>  * The text uses RTP end-point, which I infer corresponds to the
> (local) end of the ICE component in use (i.e., the selected transport
> 5-tuple).  An RTP end-point is required to have a unique CNAME, but
> that's not consistent with unbundled (SDP negotiated) sessions (i.e.,
> what corresponds to a single unbundled RTCPeerConnection), which I
> assume will want to use a consistent CNAME if at all possible.
> 
>  * We also have guidance regarding RTCPeerConnection.
> 
>  * And there is also guidance regarding the browser session,
> presumably scoped to a given origin [RFC6454].
> 
>  * I'll list RTP session for completeness, though I don't think it's a
> useful context to use in any case.
> 
> I think that the advice could be much clearer, once all the context is
> properly established:
> 
>   1. MUST use the same CNAME for an RTP end-point, unless different
> SSRCs are on a different time base (i.e., the uncommon case)
> 
>   2. MUST use different CNAME for different RTCPeerConnection
> instances, following RFC 7022, unless the using application explicitly
> provides a CNAME.

This is the intent of the text provided. So, I guess it is further work
on how things are formulated that are needed, not the intended content.
In fact regarding the first we have been stricter than that, the reason
is to avoid the synchronization issues that arises from creating
MediaStreams from MediaStreamTracks with different time-bases. That
issue would disappear if MediaStream are abandoned in the API, and then
I would prefer your notice of exception when handling multiple timebases
locally.

> 
> The latter point covers the origin case adequately.  Servers and
> middleboxes can be given a blanket exemption if you like, allowing for
> the fact that they should be doing all the normal and proper RTP
> things.

I think a important aspect from a middlebox why multiple CNAMEs is to be
expected and not necessariyl follow the default rules, is that they may
not be controlled as normal WebRTC endpoint, i.e. no normal JS API.

Cheers

Magnus Westerlund

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----------------------------------------------------------------------
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On 2014-04-07 10:12, Harald Alvestrand wrote:
> Thanks for the update!
> 
> It does seem reasonably clear now - clear enough that I found one point
> that I violently disagree with (if I'm interpreting it correctly). See
> below.
> 
> I don't have any negative comments on the rest of the section.
> 
> On 04/04/2014 11:24 AM, Magnus Westerlund wrote:
>> WG,
>> (As Author)
>>
>> Colin and I have been working on resolving terminology usage and in
>> general giving the draft a polish over before the WG last call. One
>> section that has gotten quite some attention from us is the below one.
>> The changes are significantly from an editorial stand point. However,
>> the intended content has not been intended to be changed, although
>> clarified and better motivated. So please review it, we intended to
>> include this in the upcoming draft update. Feedback appreciated.
>>
>> 5.1.  Conferencing Extensions and Topologies
>>
>>     RTP is a protocol that inherently supports group communication.
>>     Groups can be implemented by having each endpoint sending its RTP
>>     packet streams to an RTP middlebox that redistributes the traffic, by
>>     using a mesh of unicast RTP packet streams between endpoints, or by
>>     using an IP multicast group to distribute the RTP packet streams.
>>     These topologies can be implemented in a number of ways as discussed
>>     in [I-D.ietf-avtcore-rtp-topologies-update].
>>
>>     While the use of IP multicast groups is popular in IPTV systems, the
>>     topologies based on RTP middleboxes are dominant in interactive video
>>     conferencing environments.  Topologies based on a mesh of unicast
>>     transport-layer flows to create a common RTP session have not seen
>>     widespread deployment.  Accordingly, WebRTC implementations are not
>>     expected to support topologies based on IP multicast groups.  WebRTC
>>     implementations are also not expected to support mesh-based
>>     topologies, such as a point-to-multipoint mesh configured as a single
>>     RTP session (Topo-Mesh in the terminology of
>>     [I-D.ietf-avtcore-rtp-topologies-update]).  However, a point-to-
>>     multipoint mesh constructed using several RTP sessions, in the WebRTC
>>     context using independent RTCPeerConnections can be expected to be
>>     utilised by WebRTC applications.
>>
>>     WebRTC implementations of RTP endpoints implemented according to this
>>     memo are expected to support all the topologies described in
>>     [I-D.ietf-avtcore-rtp-topologies-update] where the RTP endpoints send
>>     and receive unicast RTP packet streams to some peer device, provided
>>     that peer can participate in performing congestion control on the RTP
>>     packet streams.  The peer device could be another RTP endpoint, or it
>>     could be an RTP middlebox that redistributes the RTP packet streams
>>     to other RTP endpoints.  This limitation means that some of the RTP
>>     middlebox-based topologies are not suitable for use in the WebRTC
>>     environment.  Specifically:
>>
>>     o  Video switching MCUs (Topo-Video-switch-MCU) SHOULD NOT be used,
>>        since they make the use of RTCP for congestion control and quality
>>        of service reports problematic (see Section 3.6.2 of
>>        [I-D.ietf-avtcore-rtp-topologies-update]).
>>
>>     o  Content modifying MCUs with RTCP termination (Topo-RTCP-
>>        terminating-MCU) SHOULD NOT be used since they break RTP loop
>>        detection, and prevent receivers from identifying active senders
>>        (see section 3.8 of [I-D.ietf-avtcore-rtp-topologies-update]).
> 
> In my (strong) opinion: This recommendation is wrong.
> 
> All central conferencing units that use individual RTP sessions with
> clients fall into this category. This includes, I believe, every single
> multiuser video conferencing system based on WebRTC in existence (and
> there are many of them).

Do they? The point of this topology is that it terminates the source
identification and hides when it is performing any type of mixing or
switching operation. You can perform this with a back to back RTP
session style without breaking this property. Just by correctly linking
information between the sessions. Which means that the individual RTP
session is from the identification part inseparable from an SFM or
classic RTP mixer. Only the SSRC level loop detection gets impacted.

I do understand that people have implemented it this way. Implementation
are not necessarily equivalent to recommended practice.

> 
> I believe the recommendation should be:
> 
> * Content modifying MCUs with RTCP termination
> (Topo-RTCP-terminating-MCU) MAY be used. Note that in this scenario, RTP
> loop detection and identification of active senders is the resposibility
> of the application; since the clients are isolated from each other at
> the RTP layer, RTP cannot assist with these functions.
> 

If this topology was without issues, then one would simply remove the
bullet, and let it fall under the general clause. However, it clearly
needs that warning. Which results in this topology being both positively
and negatively treated in regards to the other "you may encounter this
topology".

Cheers

Magnus Westerlund

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Subject: Re: [rtcweb] Draft proposal for updating Multiparty topologies in draft-ietf-rtcweb-rtp-usage
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Thanks Christian,

You are correct, these section references points to the wrong sections,
will address.

/Magnus

On 2014-04-07 04:42, Christian Groves wrote:
> Hello Magnus,
> 
> Please see my comments marked CNG below.
> 
> Regards, Christian
> 
> On 4/04/2014 8:24 PM, Magnus Westerlund wrote:
>> WG,
>> (As Author)
>>
>> Colin and I have been working on resolving terminology usage and in
>> general giving the draft a polish over before the WG last call. One
>> section that has gotten quite some attention from us is the below one.
>> The changes are significantly from an editorial stand point. However,
>> the intended content has not been intended to be changed, although
>> clarified and better motivated. So please review it, we intended to
>> include this in the upcoming draft update. Feedback appreciated.
>>
>> 5.1.  Conferencing Extensions and Topologies
>>
>>     RTP is a protocol that inherently supports group communication.
>>     Groups can be implemented by having each endpoint sending its RTP
>>     packet streams to an RTP middlebox that redistributes the traffic, by
>>     using a mesh of unicast RTP packet streams between endpoints, or by
>>     using an IP multicast group to distribute the RTP packet streams.
>>     These topologies can be implemented in a number of ways as discussed
>>     in [I-D.ietf-avtcore-rtp-topologies-update].
>>
>>     While the use of IP multicast groups is popular in IPTV systems, the
>>     topologies based on RTP middleboxes are dominant in interactive video
>>     conferencing environments.  Topologies based on a mesh of unicast
>>     transport-layer flows to create a common RTP session have not seen
>>     widespread deployment.  Accordingly, WebRTC implementations are not
>>     expected to support topologies based on IP multicast groups.  WebRTC
>>     implementations are also not expected to support mesh-based
>>     topologies, such as a point-to-multipoint mesh configured as a single
>>     RTP session (Topo-Mesh in the terminology of
>>     [I-D.ietf-avtcore-rtp-topologies-update]).  However, a point-to-
>>     multipoint mesh constructed using several RTP sessions, in the WebRTC
>>     context using independent RTCPeerConnections can be expected to be
>>     utilised by WebRTC applications.
>>
>>     WebRTC implementations of RTP endpoints implemented according to this
>>     memo are expected to support all the topologies described in
>>     [I-D.ietf-avtcore-rtp-topologies-update] where the RTP endpoints send
>>     and receive unicast RTP packet streams to some peer device, provided
>>     that peer can participate in performing congestion control on the RTP
>>     packet streams.  The peer device could be another RTP endpoint, or it
>>     could be an RTP middlebox that redistributes the RTP packet streams
>>     to other RTP endpoints.  This limitation means that some of the RTP
>>     middlebox-based topologies are not suitable for use in the WebRTC
>>     environment.  Specifically:
>>
>>     o  Video switching MCUs (Topo-Video-switch-MCU) SHOULD NOT be used,
>>        since they make the use of RTCP for congestion control and quality
>>        of service reports problematic (see Section 3.6.2 of
>>        [I-D.ietf-avtcore-rtp-topologies-update]).
> [CNG] Is 3.6.2 the right reference? clause 3.8 talks about
> Topo-Video-switch-MCU and the reports problem.
>>
>>     o  Content modifying MCUs with RTCP termination (Topo-RTCP-
>>        terminating-MCU) SHOULD NOT be used since they break RTP loop
>>        detection, and prevent receivers from identifying active senders
>>        (see section 3.8 of [I-D.ietf-avtcore-rtp-topologies-update]).
> [CNG] Is 3.8 the right reference? clause 3.9 talks about loop detection
> for Topo-RTCP-
>       terminating-MCU.
>>
>>     o  The Relay-Transport Translator (Topo-PtM-Trn-Translator) topology
>>        SHOULD NOT be used because its safe use requires a point to
>>        multipoint congestion control algorithm or RTP circuit breaker,
>>        which has not yet been standardised.
>>
>>     The RTP extensions described in Section 5.1.1 to Section 5.1.6 are
>>     designed to be used with centralised conferencing, where an RTP
>>     middlebox (e.g., a conference bridge) receives a participant's RTP
>>     packet streams and distributes them to the other participants.  These
>>     extensions are not necessary for interoperability; an RTP end-point
>>     that does not implement these extensions will work correctly, but
>>     might offer poor performance.  Support for the listed extensions will
>>     greatly improve the quality of experience and, to provide a
>>     reasonable baseline quality, some of these extensions are mandatory
>>     to be supported by WebRTC end-points.
>>
>>     The RTCP conferencing extensions are defined in Extended RTP Profile
>>     for Real-time Transport Control Protocol (RTCP)-Based Feedback (RTP/
>>     AVPF) [RFC4585] and the "Codec Control Messages in the RTP Audio-
>>     Visual Profile with Feedback (AVPF)" (CCM) [RFC5104] and are fully
>>     usable by the Secure variant of this profile (RTP/SAVPF) [RFC5124].
>>
>>
>> Cheers
>>
>> Magnus Westerlund
>>
>> ----------------------------------------------------------------------
>> Services, Media and Network features, Ericsson Research EAB/TXM
>> ----------------------------------------------------------------------
>> Ericsson AB                 | Phone  +46 10 7148287
>> Färögatan 6                 | Mobile +46 73 0949079
>> SE-164 80 Stockholm, Sweden | mailto: magnus.westerlund@ericsson.com
>> ----------------------------------------------------------------------
>>
>> _______________________________________________
>> rtcweb mailing list
>> rtcweb@ietf.org
>> https://www.ietf.org/mailman/listinfo/rtcweb
>>
> 
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb
> 
> 


-- 

Magnus Westerlund

----------------------------------------------------------------------
Services, Media and Network features, Ericsson Research EAB/TXM
----------------------------------------------------------------------
Ericsson AB                 | Phone  +46 10 7148287
Färögatan 6                 | Mobile +46 73 0949079
SE-164 80 Stockholm, Sweden | mailto: magnus.westerlund@ericsson.com
----------------------------------------------------------------------


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On 04/07/2014 01:15 PM, Magnus Westerlund wrote:
> On 2014-04-07 10:12, Harald Alvestrand wrote:
>> Thanks for the update!
>>
>> It does seem reasonably clear now - clear enough that I found one point
>> that I violently disagree with (if I'm interpreting it correctly). See
>> below.
>>
>> I don't have any negative comments on the rest of the section.
>>
>> On 04/04/2014 11:24 AM, Magnus Westerlund wrote:
>>> WG,
>>> (As Author)
>>>
>>> Colin and I have been working on resolving terminology usage and in
>>> general giving the draft a polish over before the WG last call. One
>>> section that has gotten quite some attention from us is the below one.
>>> The changes are significantly from an editorial stand point. However,
>>> the intended content has not been intended to be changed, although
>>> clarified and better motivated. So please review it, we intended to
>>> include this in the upcoming draft update. Feedback appreciated.
>>>
>>> 5.1.  Conferencing Extensions and Topologies
>>>
>>>      RTP is a protocol that inherently supports group communication.
>>>      Groups can be implemented by having each endpoint sending its RTP
>>>      packet streams to an RTP middlebox that redistributes the traffic, by
>>>      using a mesh of unicast RTP packet streams between endpoints, or by
>>>      using an IP multicast group to distribute the RTP packet streams.
>>>      These topologies can be implemented in a number of ways as discussed
>>>      in [I-D.ietf-avtcore-rtp-topologies-update].
>>>
>>>      While the use of IP multicast groups is popular in IPTV systems, the
>>>      topologies based on RTP middleboxes are dominant in interactive video
>>>      conferencing environments.  Topologies based on a mesh of unicast
>>>      transport-layer flows to create a common RTP session have not seen
>>>      widespread deployment.  Accordingly, WebRTC implementations are not
>>>      expected to support topologies based on IP multicast groups.  WebRTC
>>>      implementations are also not expected to support mesh-based
>>>      topologies, such as a point-to-multipoint mesh configured as a single
>>>      RTP session (Topo-Mesh in the terminology of
>>>      [I-D.ietf-avtcore-rtp-topologies-update]).  However, a point-to-
>>>      multipoint mesh constructed using several RTP sessions, in the WebRTC
>>>      context using independent RTCPeerConnections can be expected to be
>>>      utilised by WebRTC applications.
>>>
>>>      WebRTC implementations of RTP endpoints implemented according to this
>>>      memo are expected to support all the topologies described in
>>>      [I-D.ietf-avtcore-rtp-topologies-update] where the RTP endpoints send
>>>      and receive unicast RTP packet streams to some peer device, provided
>>>      that peer can participate in performing congestion control on the RTP
>>>      packet streams.  The peer device could be another RTP endpoint, or it
>>>      could be an RTP middlebox that redistributes the RTP packet streams
>>>      to other RTP endpoints.  This limitation means that some of the RTP
>>>      middlebox-based topologies are not suitable for use in the WebRTC
>>>      environment.  Specifically:
>>>
>>>      o  Video switching MCUs (Topo-Video-switch-MCU) SHOULD NOT be used,
>>>         since they make the use of RTCP for congestion control and quality
>>>         of service reports problematic (see Section 3.6.2 of
>>>         [I-D.ietf-avtcore-rtp-topologies-update]).
>>>
>>>      o  Content modifying MCUs with RTCP termination (Topo-RTCP-
>>>         terminating-MCU) SHOULD NOT be used since they break RTP loop
>>>         detection, and prevent receivers from identifying active senders
>>>         (see section 3.8 of [I-D.ietf-avtcore-rtp-topologies-update]).
>> In my (strong) opinion: This recommendation is wrong.
>>
>> All central conferencing units that use individual RTP sessions with
>> clients fall into this category. This includes, I believe, every single
>> multiuser video conferencing system based on WebRTC in existence (and
>> there are many of them).
> Do they? The point of this topology is that it terminates the source
> identification and hides when it is performing any type of mixing or
> switching operation. You can perform this with a back to back RTP
> session style without breaking this property. Just by correctly linking
> information between the sessions.

Is there a definition for "correctly linking information"?
In particular, are you thinking of mappings that preserve SSRC?
Because that's not the ones I'm thinking of.

>   Which means that the individual RTP
> session is from the identification part inseparable from an SFM or
> classic RTP mixer. Only the SSRC level loop detection gets impacted.
>
> I do understand that people have implemented it this way. Implementation
> are not necessarily equivalent to recommended practice.

But when there's one common practice, and no recommended practice, that 
might be a gentle hint that there's a reason why people are doing it - 
indeed, it might be worth making it a recommended practice.
>
>> I believe the recommendation should be:
>>
>> * Content modifying MCUs with RTCP termination
>> (Topo-RTCP-terminating-MCU) MAY be used. Note that in this scenario, RTP
>> loop detection and identification of active senders is the resposibility
>> of the application; since the clients are isolated from each other at
>> the RTP layer, RTP cannot assist with these functions.
>>
> If this topology was without issues, then one would simply remove the
> bullet, and let it fall under the general clause. However, it clearly
> needs that warning. Which results in this topology being both positively
> and negatively treated in regards to the other "you may encounter this
> topology".

It might need that warning - I'm not sure it's a real issue; loop 
detection is only an issue when you have potential loops, and in a 
single-mixer architecture, that potential is minimal (each endpoint 
knows perfectly well the difference between what it's sourcing and what 
it's receiving). Identification of active senders can be an 
application-layer thing.

Anyway, I stand by my statement: SHOULD NOT is the wrong thing to say.


>
> Cheers
>
> Magnus Westerlund
>
> ----------------------------------------------------------------------
> Services, Media and Network features, Ericsson Research EAB/TXM
> ----------------------------------------------------------------------
> Ericsson AB                 | Phone  +46 10 7148287
> Färögatan 6                 | Mobile +46 73 0949079
> SE-164 80 Stockholm, Sweden | mailto: magnus.westerlund@ericsson.com
> ----------------------------------------------------------------------
>


From nobody Mon Apr  7 09:32:30 2014
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Subject: Re: [rtcweb] Asking TLS for help with media isolation
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On 6 April 2014 11:37, Paul Kyzivat <pkyzivat@alum.mit.edu> wrote:
> But ISTM there would need to be a different set of limitations for
> non-browsers. For instance, to not echo the stream back to the source. And
> not to store the stream in a place that the sender can retrieve.

Yes.  At the most abstract level, the requirements are the same.
Don't pass the media on to someone else.


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Subject: Re: [rtcweb] Isolating data channels (Re: Asking TLS for help with media isolation)
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The subject is largely speculative, but when it comes to solutions,
the problem will not be a lack of options, but an inability to choose
a "right" one.

On 7 April 2014 01:02, Harald Alvestrand <harald@alvestrand.no> wrote:
> Wild suggestion: if you want per-track isolation properties, open up a data
> channel with a protocol called '*WebRTCIsolationInfo' and use it to send
> information about the isolation status of each track, thereby also providing
> a working example for the rule 'all data channels that have protocols
> starting with "*" are for browser internal usage'.....

I assume that you are talking PPID here.  Given that it's off limits
for JavaScript currently, then it does provide an opportunity for this
communication.  The problem there is that you need to spin up data
channels, even if the application has no need of them.  That's a
fairly high cost.


From nobody Mon Apr  7 10:07:57 2014
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From: "Matthew Kaufman (SKYPE)" <matthew.kaufman@skype.net>
To: Martin Thomson <martin.thomson@gmail.com>, Harald Alvestrand <harald@alvestrand.no>
Thread-Topic: [rtcweb] Isolating data channels (Re: Asking TLS for help with media isolation)
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Subject: Re: [rtcweb] Isolating data channels (Re: Asking TLS for help with media isolation)
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> From: rtcweb [mailto:rtcweb-bounces@ietf.org] On Behalf Of Martin
> Thomson
>...  The problem there is that you need to spin up data
> channels, even if the application has no need of them.  That's a fairly h=
igh
> cost.

But only because the working group chose poorly for how data channels are t=
ransported.

Matthew Kaufman


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On 7 April 2014 10:07, Matthew Kaufman (SKYPE)
<matthew.kaufman@skype.net> wrote:
> the working group chose poorly

Yes, thanks.

That seems to be a pattern.  It's just not always obvious immediately.

Sometimes it doesn't matter.  Especially when you haven't noticed yet.


From nobody Mon Apr  7 10:21:02 2014
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Subject: Re: [rtcweb] Isolating data channels (Re: Asking TLS for help with media isolation)
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On 04/07/2014 07:01 PM, Martin Thomson wrote:
> The subject is largely speculative, but when it comes to solutions,
> the problem will not be a lack of options, but an inability to choose
> a "right" one.
>
> On 7 April 2014 01:02, Harald Alvestrand <harald@alvestrand.no> wrote:
>> Wild suggestion: if you want per-track isolation properties, open up a data
>> channel with a protocol called '*WebRTCIsolationInfo' and use it to send
>> information about the isolation status of each track, thereby also providing
>> a working example for the rule 'all data channels that have protocols
>> starting with "*" are for browser internal usage'.....
> I assume that you are talking PPID here.  Given that it's off limits
> for JavaScript currently, then it does provide an opportunity for this
> communication.  The problem there is that you need to spin up data
> channels, even if the application has no need of them.  That's a
> fairly high cost.
I was actually thinking "protocol" as in the string that goes into the 
datachannel setup packets. PPIDs would work too for separating 
browser-to-brower from app-to-app, but I wasn't thinking of them.

Yes, data channels do cost something to set up. But we're already paying 
the DTLS tax in order to set up the keying, so it's "just" another 
request/response. How many extra round trips does the SCTP setup add?

(Would need the response in order to make sure the respondent gave the 
promise to obey isolation, I think.)

(and to Matthew: At least we wouldn't have *yet* another congestion 
context to manage, which would be the case with a separate TCP 
connection. There are always tradeoffs.)


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On 7 April 2014 10:20, Harald Alvestrand <harald@alvestrand.no> wrote:
> I was actually thinking "protocol" as in the string that goes into the
> datachannel setup packets. PPIDs would work too for separating
> browser-to-brower from app-to-app, but I wasn't thinking of them.

That would require carving out a space to use right now.  That
impinges to much on application autonomy for my liking.

> Yes, data channels do cost something to set up. But we're already paying the
> DTLS tax in order to set up the keying, so it's "just" another
> request/response. How many extra round trips does the SCTP setup add?

2.

> (and to Matthew: At least we wouldn't have *yet* another congestion context to
> manage, which would be the case with a separate TCP connection. There are
> always tradeoffs.)

I'm certain that Matthew wasn't talking about TCP.


From nobody Mon Apr  7 10:28:16 2014
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From: "Matthew Kaufman (SKYPE)" <matthew.kaufman@skype.net>
To: Harald Alvestrand <harald@alvestrand.no>, Martin Thomson <martin.thomson@gmail.com>
Thread-Topic: [rtcweb] Isolating data channels (Re: Asking TLS for help with media isolation)
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Subject: Re: [rtcweb] Isolating data channels (Re: Asking TLS for help with media isolation)
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From: rtcweb [mailto:rtcweb-bounces@ietf.org] On Behalf Of Harald
> Alvestrand
>...
> (and to Matthew: At least we wouldn't have *yet* another congestion
> context to manage, which would be the case with a separate TCP connection=
.
> There are always tradeoffs.)

If it was me (and at one time, it was) I would use a protocol that allows f=
or multiplexing and prioritization of media and data channels over the same=
 secure session with shared congestion state. Over such a protocol, opening=
 another data stream for this purpose could be done immediately without eve=
n a round trip.

RFC 7016 documents such an approach.

Matthew Kaufman


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On 07 Apr 2014, at 19:23, Martin Thomson <martin.thomson@gmail.com> =
wrote:

> On 7 April 2014 10:20, Harald Alvestrand <harald@alvestrand.no> wrote:
>> I was actually thinking "protocol" as in the string that goes into =
the
>> datachannel setup packets. PPIDs would work too for separating
>> browser-to-brower from app-to-app, but I wasn't thinking of them.
>=20
> That would require carving out a space to use right now.  That
> impinges to much on application autonomy for my liking.
>=20
>> Yes, data channels do cost something to set up. But we're already =
paying the
>> DTLS tax in order to set up the keying, so it's "just" another
>> request/response. How many extra round trips does the SCTP setup add?
>=20
> 2.
Right now: Correct. But one could optimise the initial handshake...

Best regards
Michael
>=20
>> (and to Matthew: At least we wouldn't have *yet* another congestion =
context to
>> manage, which would be the case with a separate TCP connection. There =
are
>> always tradeoffs.)
>=20
> I'm certain that Matthew wasn't talking about TCP.
>=20
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb
>=20


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--089e01493cb4a94dd404f677a65f
Content-Type: text/plain; charset=ISO-8859-1

Martin said:

"With the TLS handshake, you either get a marked session, or you don't
get a session at all.  That's a big advantage with this approach.  And
it costs far fewer bytes."

[BA] Agreed.  To be clear, the "marked session" we are talking about is the
DTLS session.  All DTLS/SRTP sessions subsequently occurring within that
DTLS marked session inherit the "isolation" property, correct?

The implication here is that not only do media sharing the same DTLS
session (e.g. audio and video multiplexed on the same port) share the
"isolation" property, but even if audio and video are not multiplexed, if
the same DTLS session is used by subsequent DTLS/SRTP sessions, then the
"isolation" property is also shared.

>From RFC 5764, Section 3:

   RTP and RTCP traffic MAY be multiplexed on a single UDP port
   [RFC5761 <http://tools.ietf.org/html/rfc5761>].  In this case, both
RTP and RTCP packets may be sent over
   the same DTLS-SRTP session, halving the number of DTLS-SRTP sessions
   needed.  This improves the cryptographic performance of DTLS, but may
   cause problems when RTCP and RTP are subject to different network
   treatment (e.g., for bandwidth reservation or scheduling reasons).

   Between a single pair of participants, there may be multiple media
   sessions.  There MUST be a separate DTLS-SRTP session for each
   distinct pair of source and destination ports used by a media session
   (though the sessions can share a single DTLS session and hence
   amortize the initial public key handshake!).







On Thu, Apr 3, 2014 at 8:25 PM, Martin Thomson <martin.thomson@gmail.com>wrote:

> Perhaps some more context is necessary.
>
> (D)TLS provides us good protection against network attackers.  The
> problem is that this protection ends at the browser, and our security
> story doesn't.  Media streams in the browser are an extension of those
> that traverse the network, and we need something of a gentlemen's
> agreement between browsers to be reached.  That agreement says that
> your browser and mine will agree between themselves to protect media
> from the application that is running in their sandbox: all that nasty
> JavaScript that came from a site (or sites) that we might not want to
> include in our conversation.
>
> On 3 April 2014 19:49, Watson Ladd <watsonbladd@gmail.com> wrote:
> > The MITM can kill any TLS connection containing the extension. My
> > understanding is that signalling data is immutable, hence the need to
> > ask the browser to generate it.
>
> Signaling is totally mutable, which is a big problem.  Once an
> application gets it, there's a lot that can be changed between peers.
>
> >> However once the desire for isolation is communicated E2E (either via
> ACP or
> >> ALPN tokens), there is nothing in the SRTP traffic (keyed by DTLS/SRTP)
> that
> >> indicates that the traffic is to be isolated.
> >
> > I don't see why the isolation status cannot be included as an
> > extension to SRTP. You aren't asking TLS to make extensions for video
> > resolution and codec after all.
>
> That's a good question, and it's an option I should have added to the
> list.  The problem there is that nothing in RTP is reliable, and this
> signal needs to be reliable.  Combine that with the fact that the
> extent of an RTP session is essentially unknowable, and you end up
> having to put an RTP header extension on every packet.  RTP header
> extensions have other drawback as well, though I suspect most of those
> aren't applicable in this context.
>
> With the TLS handshake, you either get a marked session, or you don't
> get a session at all.  That's a big advantage with this approach.  And
> it costs far fewer bytes.
>

--089e01493cb4a94dd404f677a65f
Content-Type: text/html; charset=ISO-8859-1
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<div dir=3D"ltr">Martin said:=A0<div><br></div><div>&quot;<span style=3D"fo=
nt-family:arial,sans-serif;font-size:13px">With the TLS handshake, you eith=
er get a marked session, or you don&#39;t</span></div><span style=3D"font-f=
amily:arial,sans-serif;font-size:13px">get a session at all. =A0That&#39;s =
a big advantage with this approach. =A0And</span><br style=3D"font-family:a=
rial,sans-serif;font-size:13px">

<div><span style=3D"font-family:arial,sans-serif;font-size:13px">it costs f=
ar fewer bytes.</span>&quot;</div><div><br></div><div>[BA] Agreed. =A0To be=
 clear, the &quot;marked session&quot; we are talking about is the DTLS ses=
sion. =A0All DTLS/SRTP sessions subsequently occurring within that DTLS mar=
ked session inherit the &quot;isolation&quot; property, correct?=A0</div>

<div><br></div><div>The implication here is that not only do media sharing =
the same DTLS session (e.g. audio and video multiplexed on the same port) s=
hare the &quot;isolation&quot; property, but even if audio and video are no=
t multiplexed, if the same DTLS session is used by subsequent DTLS/SRTP ses=
sions, then the &quot;isolation&quot; property is also shared.=A0</div>

<div><br></div><div>From RFC 5764, Section 3:=A0</div><div><br></div><div><=
pre class=3D"" style=3D"font-size:1em;margin-top:0px;margin-bottom:0px;colo=
r:rgb(0,0,0)">   RTP and RTCP traffic MAY be multiplexed on a single UDP po=
rt
   [<a href=3D"http://tools.ietf.org/html/rfc5761" title=3D"&quot;Multiplex=
ing RTP Data and Control Packets on a Single Port&quot;">RFC5761</a>].  In =
this case, both RTP and RTCP packets may be sent over
   the same DTLS-SRTP session, halving the number of DTLS-SRTP sessions
   needed.  This improves the cryptographic performance of DTLS, but may
   cause problems when RTCP and RTP are subject to different network
   treatment (e.g., for bandwidth reservation or scheduling reasons).

   Between a single pair of participants, there may be multiple media
   sessions.  There MUST be a separate DTLS-SRTP session for each
   distinct pair of source and destination ports used by a media session
   (though the sessions can share a single DTLS session and hence
   amortize the initial public key handshake!).</pre></div><div><br></div><=
div><br></div><div><br></div><div><br></div></div><div class=3D"gmail_extra=
"><br><br><div class=3D"gmail_quote">On Thu, Apr 3, 2014 at 8:25 PM, Martin=
 Thomson <span dir=3D"ltr">&lt;<a href=3D"mailto:martin.thomson@gmail.com" =
target=3D"_blank">martin.thomson@gmail.com</a>&gt;</span> wrote:<br>

<blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1p=
x #ccc solid;padding-left:1ex">Perhaps some more context is necessary.<br>
<br>
(D)TLS provides us good protection against network attackers. =A0The<br>
problem is that this protection ends at the browser, and our security<br>
story doesn&#39;t. =A0Media streams in the browser are an extension of thos=
e<br>
that traverse the network, and we need something of a gentlemen&#39;s<br>
agreement between browsers to be reached. =A0That agreement says that<br>
your browser and mine will agree between themselves to protect media<br>
from the application that is running in their sandbox: all that nasty<br>
JavaScript that came from a site (or sites) that we might not want to<br>
include in our conversation.<br>
<div class=3D""><br>
On 3 April 2014 19:49, Watson Ladd &lt;<a href=3D"mailto:watsonbladd@gmail.=
com">watsonbladd@gmail.com</a>&gt; wrote:<br>
&gt; The MITM can kill any TLS connection containing the extension. My<br>
&gt; understanding is that signalling data is immutable, hence the need to<=
br>
&gt; ask the browser to generate it.<br>
<br>
</div>Signaling is totally mutable, which is a big problem. =A0Once an<br>
application gets it, there&#39;s a lot that can be changed between peers.<b=
r>
<div class=3D""><br>
&gt;&gt; However once the desire for isolation is communicated E2E (either =
via ACP or<br>
&gt;&gt; ALPN tokens), there is nothing in the SRTP traffic (keyed by DTLS/=
SRTP) that<br>
&gt;&gt; indicates that the traffic is to be isolated.<br>
&gt;<br>
&gt; I don&#39;t see why the isolation status cannot be included as an<br>
&gt; extension to SRTP. You aren&#39;t asking TLS to make extensions for vi=
deo<br>
&gt; resolution and codec after all.<br>
<br>
</div>That&#39;s a good question, and it&#39;s an option I should have adde=
d to the<br>
list. =A0The problem there is that nothing in RTP is reliable, and this<br>
signal needs to be reliable. =A0Combine that with the fact that the<br>
extent of an RTP session is essentially unknowable, and you end up<br>
having to put an RTP header extension on every packet. =A0RTP header<br>
extensions have other drawback as well, though I suspect most of those<br>
aren&#39;t applicable in this context.<br>
<br>
With the TLS handshake, you either get a marked session, or you don&#39;t<b=
r>
get a session at all. =A0That&#39;s a big advantage with this approach. =A0=
And<br>
it costs far fewer bytes.<br>
</blockquote></div><br></div>

--089e01493cb4a94dd404f677a65f--


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Subject: Re: [rtcweb] Asking TLS for help with media isolation
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On 7 April 2014 11:00, Bernard Aboba <bernard.aboba@gmail.com> wrote:
> The implication here is that not only do media sharing the same DTLS session
> (e.g. audio and video multiplexed on the same port) share the "isolation"
> property, but even if audio and video are not multiplexed, if the same DTLS
> session is used by subsequent DTLS/SRTP sessions, then the "isolation"
> property is also shared.

The usual behaviour is to use the same 5-tuple for sending SRTP as the
one used for the DTLS handshake.  However, if the extracted keys from
the DTLS session is used to key different SRTP flows on other
5-tuples, then I suppose that the conditions attached to the DTLS
session would have to apply to those separate flows too.  That means
isolation, authentication, and whatever else we bind in.

I'm not sure that I'd do that, given the risks of things like SSRC
reuse, but maybe I'm misunderstanding the question.


From nobody Mon Apr  7 11:20:24 2014
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Subject: [rtcweb] Review of draft-ietf-rtcweb-stun-consent-freshness
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--001a1130c446a8703a04f677eab4
Content-Type: text/plain; charset=ISO-8859-1

Howdy,

The chairs recently asked for a review
draft-ietf-rtcweb-stun-consent-freshness; Oleg was kind enough to do one.
Below is the review.

regards,

Ted


On Fri, Apr 4, 2014 at 12:40 AM, Oleg Moskalenko <mom040267@gmail.com>wrote:
Hi Ted

I went through the document and I have two things to comment:

1) This document defines a "voluntary" pattern of the browser behavior.
Nothing stops the determined attacker from taking the WebRTC code and
creating a malicious client application that ignores all proposed
connectivity checks. May be it is worth mentioning in the "Security
Considerations" section.

2) I have a feeling that the document is written with somewhat optimistic
idea about the modern IP network qualities. The proposed timeouts are
probably too small. I am hearing from our TURN server users that in modern
Wi Fi public networks that's common to observe a freeze the IP traffic for
several seconds. After that "freeze" the connectivity is restored. The
users do not want the connection to be broken during that time - they want
the video screen frozen, temporary. I had to make adjustments to the TURN
server in our recent versions so that it does not disconnects the sessions
too quickly under those conditions (when TCP is used). I have a feeling
that you may have the same complains that the browser stops transmission in
public Wi Fi networks too quickly. I'd suggest to review the wording of the
proposal (like re-transmission after 500 ms and 15 secs timeout) to make it
more tolerant for the bad IP networks (which are surprisingly are rather
common).

Overall, I think that this proposal is very useful.

Best regards,
Oleg

--001a1130c446a8703a04f677eab4
Content-Type: text/html; charset=ISO-8859-1
Content-Transfer-Encoding: quoted-printable

<div dir=3D"ltr"><div class=3D"gmail_default" style=3D"font-family:georgia,=
serif">Howdy,<br><br></div><div class=3D"gmail_default" style=3D"font-famil=
y:georgia,serif">The chairs recently asked for a review draft-ietf-rtcweb-s=
tun-consent-freshness; Oleg was kind enough to do one.=A0 Below is the revi=
ew.<br>
<br>regards,<br><br>Ted<br></div><div class=3D"gmail_default" style=3D"font=
-family:georgia,serif"><br><br>On Fri, Apr 4, 2014 at 12:40 AM, Oleg Moskal=
enko <span dir=3D"ltr">&lt;<a href=3D"mailto:mom040267@gmail.com" target=3D=
"_blank">mom040267@gmail.com</a>&gt;</span> wrote:<br>


<div><div><div><div><div>Hi Ted<br><br></div>I went through the document an=
d I have two things to comment:<br>

<br></div>1) This document defines a &quot;voluntary&quot; pattern of the b=
rowser=20
behavior. Nothing stops the determined attacker from taking the WebRTC=20
code and creating a malicious client application that ignores all=20
proposed connectivity checks. May be it is worth mentioning in the=20
&quot;Security Considerations&quot; section.<br>


<br></div>2) I have a feeling that the document is written with somewhat
 optimistic idea about the modern IP network qualities. The proposed=20
timeouts are probably too small. I am hearing from our TURN server users
 that in modern Wi Fi public networks that&#39;s common to observe a freeze=
=20
the IP traffic for several seconds. After that &quot;freeze&quot; the conne=
ctivity
 is restored. The users do not want the connection to be broken during=20
that time - they want the video screen frozen, temporary. I had to make=20
adjustments to the TURN server in our recent versions so that it does=20
not disconnects the sessions too quickly under those conditions (when=20
TCP is used). I have a feeling that you may have the same complains that
 the browser stops transmission in public Wi Fi networks too quickly.=20
I&#39;d suggest to review the wording of the proposal (like re-transmission=
=20
after 500 ms and 15 secs timeout) to make it more tolerant for the bad=20
IP networks (which are surprisingly are rather common).<br>


<br></div>Overall, I think that this proposal is very useful.<br><br></div>=
Best regards,<br>Oleg</div></div>

--001a1130c446a8703a04f677eab4--


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Subject: Re: [rtcweb] Isolating data channels (Re: Asking TLS for help with media isolation)
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On 4/7/14 1:27 PM, Matthew Kaufman (SKYPE) wrote:
>
> From: rtcweb [mailto:rtcweb-bounces@ietf.org] On Behalf Of Harald
>> Alvestrand
>> ...
>> (and to Matthew: At least we wouldn't have *yet* another congestion
>> context to manage, which would be the case with a separate TCP connection.
>> There are always tradeoffs.)
>
> If it was me (and at one time, it was) I would use a protocol that allows for multiplexing and prioritization of media and data channels over the same secure session with shared congestion state. Over such a protocol, opening another data stream for this purpose could be done immediately without even a round trip.
>
> RFC 7016 documents such an approach.

I see that the title starts with "Adobe's". Sigh.

If you want to entertain something other than existing solutions for 
media, why not simply run the RTP media streams over SCTP along with the 
data channels?

	Thanks,
	Paul


From nobody Mon Apr  7 12:28:19 2014
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From: "Matthew Kaufman (SKYPE)" <matthew.kaufman@skype.net>
To: Paul Kyzivat <pkyzivat@alum.mit.edu>, "rtcweb@ietf.org" <rtcweb@ietf.org>
Thread-Topic: [rtcweb] Isolating data channels (Re: Asking TLS for help with media isolation)
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Subject: Re: [rtcweb] Isolating data channels (Re: Asking TLS for help with media isolation)
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-----Original Message-----
> From: rtcweb [mailto:rtcweb-bounces@ietf.org] On Behalf Of Paul Kyzivat
> Sent: Monday, April 7, 2014 12:22 PM
> To: rtcweb@ietf.org
> Subject: Re: [rtcweb] Isolating data channels (Re: Asking TLS for help wi=
th
> media isolation)
>=20
> On 4/7/14 1:27 PM, Matthew Kaufman (SKYPE) wrote:
> >
> > From: rtcweb [mailto:rtcweb-bounces@ietf.org] On Behalf Of Harald
> >> Alvestrand
> >> ...
> >> (and to Matthew: At least we wouldn't have *yet* another congestion
> >> context to manage, which would be the case with a separate TCP
> connection.
> >> There are always tradeoffs.)
> >
> > If it was me (and at one time, it was) I would use a protocol that allo=
ws for
> multiplexing and prioritization of media and data channels over the same
> secure session with shared congestion state. Over such a protocol, openin=
g
> another data stream for this purpose could be done immediately without
> even a round trip.
> >
> > RFC 7016 documents such an approach.
>=20
> I see that the title starts with "Adobe's". Sigh.

Yes, but there's little stopping many of these ideas from becoming the basi=
s for an IETF-developed transport protocol that actually meets the needs we=
 have here.
=20
> If you want to entertain something other than existing solutions for medi=
a,
> why not simply run the RTP media streams over SCTP along with the data
> channels?

Except for some critical shortcomings of SCTP (which are being fixed), this=
 isn't actually a terrible idea.

Matthew Kaufman


From nobody Mon Apr  7 12:32:21 2014
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From: Mary Barnes <mary.ietf.barnes@gmail.com>
To: "Matthew Kaufman (SKYPE)" <matthew.kaufman@skype.net>
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Subject: Re: [rtcweb] Isolating data channels (Re: Asking TLS for help with media isolation)
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--001a11c38e7059564204f678ec9c
Content-Type: text/plain; charset=ISO-8859-1

On Mon, Apr 7, 2014 at 2:27 PM, Matthew Kaufman (SKYPE) <
matthew.kaufman@skype.net> wrote:

> -----Original Message-----
> > From: rtcweb [mailto:rtcweb-bounces@ietf.org] On Behalf Of Paul Kyzivat
> > Sent: Monday, April 7, 2014 12:22 PM
> > To: rtcweb@ietf.org
> > Subject: Re: [rtcweb] Isolating data channels (Re: Asking TLS for help
> with
> > media isolation)
> >
> > On 4/7/14 1:27 PM, Matthew Kaufman (SKYPE) wrote:
> > >
> > > From: rtcweb [mailto:rtcweb-bounces@ietf.org] On Behalf Of Harald
> > >> Alvestrand
> > >> ...
> > >> (and to Matthew: At least we wouldn't have *yet* another congestion
> > >> context to manage, which would be the case with a separate TCP
> > connection.
> > >> There are always tradeoffs.)
> > >
> > > If it was me (and at one time, it was) I would use a protocol that
> allows for
> > multiplexing and prioritization of media and data channels over the same
> > secure session with shared congestion state. Over such a protocol,
> opening
> > another data stream for this purpose could be done immediately without
> > even a round trip.
> > >
> > > RFC 7016 documents such an approach.
> >
> > I see that the title starts with "Adobe's". Sigh.
>
> Yes, but there's little stopping many of these ideas from becoming the
> basis for an IETF-developed transport protocol that actually meets the
> needs we have here.
>

[MB] Except, maybe, IPR:

https://datatracker.ietf.org/ipr/search/?option=document_search&id=draft-thornburgh-adobe-rtmfp
[/MB]

> > If you want to entertain something other than existing solutions for
> media,
> > why not simply run the RTP media streams over SCTP along with the data
> > channels?
>
> Except for some critical shortcomings of SCTP (which are being fixed),
> this isn't actually a terrible idea.
>
> Matthew Kaufman
>
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb
>

--001a11c38e7059564204f678ec9c
Content-Type: text/html; charset=ISO-8859-1
Content-Transfer-Encoding: quoted-printable

<div dir=3D"ltr"><br><div class=3D"gmail_extra"><br><br><div class=3D"gmail=
_quote">On Mon, Apr 7, 2014 at 2:27 PM, Matthew Kaufman (SKYPE) <span dir=
=3D"ltr">&lt;<a href=3D"mailto:matthew.kaufman@skype.net" target=3D"_blank"=
>matthew.kaufman@skype.net</a>&gt;</span> wrote:<br>
<blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1p=
x #ccc solid;padding-left:1ex"><div class=3D"">-----Original Message-----<b=
r>
&gt; From: rtcweb [mailto:<a href=3D"mailto:rtcweb-bounces@ietf.org">rtcweb=
-bounces@ietf.org</a>] On Behalf Of Paul Kyzivat<br>
&gt; Sent: Monday, April 7, 2014 12:22 PM<br>
&gt; To: <a href=3D"mailto:rtcweb@ietf.org">rtcweb@ietf.org</a><br>
&gt; Subject: Re: [rtcweb] Isolating data channels (Re: Asking TLS for help=
 with<br>
&gt; media isolation)<br>
&gt;<br>
&gt; On 4/7/14 1:27 PM, Matthew Kaufman (SKYPE) wrote:<br>
&gt; &gt;<br>
&gt; &gt; From: rtcweb [mailto:<a href=3D"mailto:rtcweb-bounces@ietf.org">r=
tcweb-bounces@ietf.org</a>] On Behalf Of Harald<br>
&gt; &gt;&gt; Alvestrand<br>
&gt; &gt;&gt; ...<br>
&gt; &gt;&gt; (and to Matthew: At least we wouldn&#39;t have *yet* another =
congestion<br>
&gt; &gt;&gt; context to manage, which would be the case with a separate TC=
P<br>
&gt; connection.<br>
&gt; &gt;&gt; There are always tradeoffs.)<br>
&gt; &gt;<br>
&gt; &gt; If it was me (and at one time, it was) I would use a protocol tha=
t allows for<br>
&gt; multiplexing and prioritization of media and data channels over the sa=
me<br>
&gt; secure session with shared congestion state. Over such a protocol, ope=
ning<br>
&gt; another data stream for this purpose could be done immediately without=
<br>
&gt; even a round trip.<br>
&gt; &gt;<br>
&gt; &gt; RFC 7016 documents such an approach.<br>
&gt;<br>
&gt; I see that the title starts with &quot;Adobe&#39;s&quot;. Sigh.<br>
<br>
</div>Yes, but there&#39;s little stopping many of these ideas from becomin=
g the basis for an IETF-developed transport protocol that actually meets th=
e needs we have here.<br></blockquote><div><br></div><div>[MB] Except, mayb=
e, IPR:=A0</div>
<div>=A0<a href=3D"https://datatracker.ietf.org/ipr/search/?option=3Ddocume=
nt_search&amp;id=3Ddraft-thornburgh-adobe-rtmfp">https://datatracker.ietf.o=
rg/ipr/search/?option=3Ddocument_search&amp;id=3Ddraft-thornburgh-adobe-rtm=
fp</a></div>
<div>[/MB]</div><blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8e=
x;border-left:1px #ccc solid;padding-left:1ex"><div class=3D"">
&gt; If you want to entertain something other than existing solutions for m=
edia,<br>
&gt; why not simply run the RTP media streams over SCTP along with the data=
<br>
&gt; channels?<br>
<br>
</div>Except for some critical shortcomings of SCTP (which are being fixed)=
, this isn&#39;t actually a terrible idea.<br>
<span class=3D"HOEnZb"><font color=3D"#888888"><br>
Matthew Kaufman<br>
</font></span><div class=3D"HOEnZb"><div class=3D"h5"><br>
_______________________________________________<br>
rtcweb mailing list<br>
<a href=3D"mailto:rtcweb@ietf.org">rtcweb@ietf.org</a><br>
<a href=3D"https://www.ietf.org/mailman/listinfo/rtcweb" target=3D"_blank">=
https://www.ietf.org/mailman/listinfo/rtcweb</a><br>
</div></div></blockquote></div><br></div></div>

--001a11c38e7059564204f678ec9c--


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From: "Matthew Kaufman (SKYPE)" <matthew.kaufman@skype.net>
To: Mary Barnes <mary.ietf.barnes@gmail.com>
Thread-Topic: [rtcweb] Isolating data channels (Re: Asking TLS for help with media isolation)
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Subject: Re: [rtcweb] Isolating data channels (Re: Asking TLS for help with media isolation)
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From: rtcweb [mailto:rtcweb-bounces@ietf.org] On Behalf Of Paul Kyzivat
> Sent: Monday, April 7, 2014 12:22 PM
> To: rtcweb@ietf.org
> Subject: Re: [rtcweb] Isolating data channels (Re: Asking TLS for help wi=
th
> media isolation)
> > RFC 7016 documents such an approach.
>
> I see that the title starts with "Adobe's". Sigh.
Yes, but there's little stopping many of these ideas from becoming the basi=
s for an IETF-developed transport protocol that actually meets the needs we=
 have here.

[MB] Except, maybe, IPR:=A0
=A0https://datatracker.ietf.org/ipr/search/?option=3Ddocument_search&id=3Dd=
raft-thornburgh-adobe-rtmfp
[/MB]

Conveniently, RTMFP was preceded by an open-source project which had no pat=
ents filed. so to the extent the "ideas" are present there (and many of the=
m are), that's easily worked around, if one needed a starting place.

But my point wasn't to start discussing the merits of the specific approach=
 described in that particular RFC... rather to note that the cobbled-togeth=
er collection of choices that have been made, specifically gluing SCTP on t=
o the side of RTCWEB as a way of transporting data, is highly suboptimal an=
d that other solutions do exist.

In fact, simply allowing the data we need to be sent unsolicited as another=
 SRTP stream would be an improvement over doing 2 round trips of SCTP setup=
 for the problem space we are discussing in this thread.

Matthew Kaufman




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From: Michael Thornburgh <mthornbu@adobe.com>
To: Mary Barnes <mary.ietf.barnes@gmail.com>, "Matthew Kaufman (SKYPE)" <matthew.kaufman@skype.net>
Thread-Topic: [rtcweb] Isolating data channels (Re: Asking TLS for help with media isolation)
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Subject: Re: [rtcweb] Isolating data channels (Re: Asking TLS for help with media isolation)
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> From: rtcweb [mailto:rtcweb-bounces@ietf.org] On Behalf Of Mary Barnes
> Sent: Monday, April 07, 2014 12:32 PM
>=20
> On Mon, Apr 7, 2014 at 2:27 PM, Matthew Kaufman (SKYPE) <matthew.kaufman@=
skype.net> wrote:
>=20
> 	-----Original Message-----
> 	> From: rtcweb [mailto:rtcweb-bounces@ietf.org] On Behalf Of Paul Kyziva=
t
> 	> Sent: Monday, April 7, 2014 12:22 PM
> 	> To: rtcweb@ietf.org
> 	> Subject: Re: [rtcweb] Isolating data channels (Re: Asking TLS for help=
 with
> 	> media isolation)
> 	>
> 	> On 4/7/14 1:27 PM, Matthew Kaufman (SKYPE) wrote:
> 	> >
> 	> > From: rtcweb [mailto:rtcweb-bounces@ietf.org] On Behalf Of Harald
> 	> >> Alvestrand
> 	> >> ...
> 	> >> (and to Matthew: At least we wouldn't have *yet* another congestion
> 	> >> context to manage, which would be the case with a separate TCP
> 	> connection.
> 	> >> There are always tradeoffs.)
> 	> >
> 	> > If it was me (and at one time, it was) I would use a protocol that a=
llows for
> 	> multiplexing and prioritization of media and data channels over the sa=
me
> 	> secure session with shared congestion state. Over such a protocol, ope=
ning
> 	> another data stream for this purpose could be done immediately without
> 	> even a round trip.
> 	> >
> 	> > RFC 7016 documents such an approach.
> 	>
> 	> I see that the title starts with "Adobe's". Sigh.
>=20
> 	Yes, but there's little stopping many of these ideas from becoming the b=
asis for an IETF-
> developed transport protocol that actually meets the needs we have here.
>=20
> [MB] Except, maybe, IPR:
>  https://datatracker.ietf.org/ipr/search/?option=3Ddocument_search&id=3Dd=
raft-thornburgh-adobe-rtmfp
> [/MB]

we would love for RTMFP to become the basis for an IETF-developed standard =
transport protocol.  and as far as using RTMFP as-is, note section VI of th=
e (most recent) above-referenced IPR disclosure: "Royalty-Free, Reasonable =
and Non-Discriminatory License to All Implementers."  we think it's pretty =
good already and published it (among other reasons) in the hopes that other=
s would find new uses for it.

> 	> If you want to entertain something other than existing solutions for m=
edia,
> 	> why not simply run the RTP media streams over SCTP along with the data
> 	> channels?
>=20
> 	Except for some critical shortcomings of SCTP (which are being fixed), t=
his isn't actually a
> terrible idea.

except for the critical shortcomings of SCTP (for real-time communications)=
 that aren't being fixed.

> 	Matthew Kaufman

-michael thornburgh


From nobody Tue Apr  8 04:42:38 2014
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From: Christer Holmberg <christer.holmberg@ericsson.com>
To: Martin Thomson <martin.thomson@gmail.com>, Paul Kyzivat <pkyzivat@alum.mit.edu>
Thread-Topic: [rtcweb] Asking TLS for help with media isolation
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Subject: Re: [rtcweb] Asking TLS for help with media isolation
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Hi,

>> But ISTM there would need to be a different set of limitations for=20
>> non-browsers. For instance, to not echo the stream back to the source.=20
>> And not to store the stream in a place that the sender can retrieve.
>
>Yes.  At the most abstract level, the requirements are the same.
>Don't pass the media on to someone else.

How would that work with an intermediary mixer, transcoder, etc?

Regards,

Christer


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Hi,
Inline
Roni

> -----Original Message-----
> From: rtcweb [mailto:rtcweb-bounces@ietf.org] On Behalf Of Magnus
> Westerlund
> Sent: 07 April, 2014 2:15 PM
> To: Harald Alvestrand; rtcweb@ietf.org
> Subject: Re: [rtcweb] Draft proposal for updating Multiparty =
topologies in
draft-
> ietf-rtcweb-rtp-usage
>=20
> On 2014-04-07 10:12, Harald Alvestrand wrote:
> > Thanks for the update!
> >
> > It does seem reasonably clear now - clear enough that I found one
> > point that I violently disagree with (if I'm interpreting it
> > correctly). See below.
> >
> > I don't have any negative comments on the rest of the section.
> >
> > On 04/04/2014 11:24 AM, Magnus Westerlund wrote:
> >> WG,
> >> (As Author)
> >>
> >> Colin and I have been working on resolving terminology usage and in
> >> general giving the draft a polish over before the WG last call. One
> >> section that has gotten quite some attention from us is the below =
one.
> >> The changes are significantly from an editorial stand point. =
However,
> >> the intended content has not been intended to be changed, although
> >> clarified and better motivated. So please review it, we intended to
> >> include this in the upcoming draft update. Feedback appreciated.
> >>
> >> 5.1.  Conferencing Extensions and Topologies
> >>
> >>     RTP is a protocol that inherently supports group communication.
> >>     Groups can be implemented by having each endpoint sending its =
RTP
> >>     packet streams to an RTP middlebox that redistributes the =
traffic,
by
> >>     using a mesh of unicast RTP packet streams between endpoints, =
or by
> >>     using an IP multicast group to distribute the RTP packet =
streams.
> >>     These topologies can be implemented in a number of ways as
discussed
> >>     in [I-D.ietf-avtcore-rtp-topologies-update].
> >>
> >>     While the use of IP multicast groups is popular in IPTV =
systems,
the
> >>     topologies based on RTP middleboxes are dominant in interactive
video
> >>     conferencing environments.  Topologies based on a mesh of =
unicast
> >>     transport-layer flows to create a common RTP session have not =
seen
> >>     widespread deployment.  Accordingly, WebRTC implementations are =
not
> >>     expected to support topologies based on IP multicast groups.
WebRTC
> >>     implementations are also not expected to support mesh-based
> >>     topologies, such as a point-to-multipoint mesh configured as a
single
> >>     RTP session (Topo-Mesh in the terminology of
> >>     [I-D.ietf-avtcore-rtp-topologies-update]).  However, a =
point-to-
> >>     multipoint mesh constructed using several RTP sessions, in the
WebRTC
> >>     context using independent RTCPeerConnections can be expected to =
be
> >>     utilised by WebRTC applications.
> >>
> >>     WebRTC implementations of RTP endpoints implemented according =
to
> this
> >>     memo are expected to support all the topologies described in
> >>     [I-D.ietf-avtcore-rtp-topologies-update] where the RTP =
endpoints
send
> >>     and receive unicast RTP packet streams to some peer device,
provided
> >>     that peer can participate in performing congestion control on =
the
RTP
> >>     packet streams.  The peer device could be another RTP endpoint, =
or
it
> >>     could be an RTP middlebox that redistributes the RTP packet =
streams
> >>     to other RTP endpoints.  This limitation means that some of the =
RTP
> >>     middlebox-based topologies are not suitable for use in the =
WebRTC
> >>     environment.  Specifically:
> >>
> >>     o  Video switching MCUs (Topo-Video-switch-MCU) SHOULD NOT be =
used,
> >>        since they make the use of RTCP for congestion control and
quality
> >>        of service reports problematic (see Section 3.6.2 of
> >>        [I-D.ietf-avtcore-rtp-topologies-update]).
> >>
> >>     o  Content modifying MCUs with RTCP termination (Topo-RTCP-
> >>        terminating-MCU) SHOULD NOT be used since they break RTP =
loop
> >>        detection, and prevent receivers from identifying active =
senders
> >>        (see section 3.8 of =
[I-D.ietf-avtcore-rtp-topologies-update]).
> >
> > In my (strong) opinion: This recommendation is wrong.
> >
> > All central conferencing units that use individual RTP sessions with
> > clients fall into this category. This includes, I believe, every
> > single multiuser video conferencing system based on WebRTC in
> > existence (and there are many of them).
>=20
> Do they? The point of this topology is that it terminates the source
> identification and hides when it is performing any type of mixing or
switching
> operation. You can perform this with a back to back RTP session style
without
> breaking this property. Just by correctly linking information between =
the
> sessions. Which means that the individual RTP session is from the
identification
> part inseparable from an SFM or classic RTP mixer. Only the SSRC level
loop
> detection gets impacted.
>=20
> I do understand that people have implemented it this way. =
Implementation
are
> not necessarily equivalent to recommended practice.
>=20
[Roni Even] In this topology the source identification is done by the
application layer and not the RTP layer (using for example conference =
event
package or just the UI). So I think that "MAY" is strong enough and not
"SHOULD NOT". I agree with Harald that it is widely used.

> >
> > I believe the recommendation should be:
> >
> > * Content modifying MCUs with RTCP termination
> > (Topo-RTCP-terminating-MCU) MAY be used. Note that in this scenario,
> > RTP loop detection and identification of active senders is the
> > resposibility of the application; since the clients are isolated =
from
> > each other at the RTP layer, RTP cannot assist with these functions.
> >
>=20
> If this topology was without issues, then one would simply remove the
bullet,
> and let it fall under the general clause. However, it clearly needs =
that
warning.
> Which results in this topology being both positively and negatively
treated in
> regards to the other "you may encounter this topology".
>=20
> Cheers
>=20
> Magnus Westerlund
>=20
> ----------------------------------------------------------------------
> Services, Media and Network features, Ericsson Research EAB/TXM
> ----------------------------------------------------------------------
> Ericsson AB                 | Phone  +46 10 7148287
> F=E4r=F6gatan 6                 | Mobile +46 73 0949079
> SE-164 80 Stockholm, Sweden | mailto: magnus.westerlund@ericsson.com
> ----------------------------------------------------------------------
>=20
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb


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From: "Roni Even" <ron.even.tlv@gmail.com>
To: "'Magnus Westerlund'" <magnus.westerlund@ericsson.com>, <rtcweb@ietf.org>
References: <533E7A50.5040909@ericsson.com>
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Date: Tue, 8 Apr 2014 18:13:19 +0300
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Subject: Re: [rtcweb] Draft proposal for updating Multiparty topologies in draft-ietf-rtcweb-rtp-usage
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Hi,
Comment on video switching in-line
Roni

> -----Original Message-----
> From: rtcweb [mailto:rtcweb-bounces@ietf.org] On Behalf Of Magnus
> Westerlund
> Sent: 04 April, 2014 12:25 PM
> To: rtcweb@ietf.org
> Cc: Colin Perkins
> Subject: [rtcweb] Draft proposal for updating Multiparty topologies in
draft-
> ietf-rtcweb-rtp-usage
>=20
> WG,
> (As Author)
>=20
> Colin and I have been working on resolving terminology usage and in
general
> giving the draft a polish over before the WG last call. One section =
that
has
> gotten quite some attention from us is the below one.
> The changes are significantly from an editorial stand point. However, =
the
> intended content has not been intended to be changed, although =
clarified
and
> better motivated. So please review it, we intended to include this in =
the
> upcoming draft update. Feedback appreciated.
>=20
> 5.1.  Conferencing Extensions and Topologies
>=20
>    RTP is a protocol that inherently supports group communication.
>    Groups can be implemented by having each endpoint sending its RTP
>    packet streams to an RTP middlebox that redistributes the traffic, =
by
>    using a mesh of unicast RTP packet streams between endpoints, or by
>    using an IP multicast group to distribute the RTP packet streams.
>    These topologies can be implemented in a number of ways as =
discussed
>    in [I-D.ietf-avtcore-rtp-topologies-update].
>=20
>    While the use of IP multicast groups is popular in IPTV systems, =
the
>    topologies based on RTP middleboxes are dominant in interactive =
video
>    conferencing environments.  Topologies based on a mesh of unicast
>    transport-layer flows to create a common RTP session have not seen
>    widespread deployment.  Accordingly, WebRTC implementations are not
>    expected to support topologies based on IP multicast groups.  =
WebRTC
>    implementations are also not expected to support mesh-based
>    topologies, such as a point-to-multipoint mesh configured as a =
single
>    RTP session (Topo-Mesh in the terminology of
>    [I-D.ietf-avtcore-rtp-topologies-update]).  However, a point-to-
>    multipoint mesh constructed using several RTP sessions, in the =
WebRTC
>    context using independent RTCPeerConnections can be expected to be
>    utilised by WebRTC applications.
>=20
>    WebRTC implementations of RTP endpoints implemented according to =
this
>    memo are expected to support all the topologies described in
>    [I-D.ietf-avtcore-rtp-topologies-update] where the RTP endpoints =
send
>    and receive unicast RTP packet streams to some peer device, =
provided
>    that peer can participate in performing congestion control on the =
RTP
>    packet streams.  The peer device could be another RTP endpoint, or =
it
>    could be an RTP middlebox that redistributes the RTP packet streams
>    to other RTP endpoints.  This limitation means that some of the RTP
>    middlebox-based topologies are not suitable for use in the WebRTC
>    environment.  Specifically:
>=20
>    o  Video switching MCUs (Topo-Video-switch-MCU) SHOULD NOT be used,
>       since they make the use of RTCP for congestion control and =
quality
>       of service reports problematic (see Section 3.6.2 of
>       [I-D.ietf-avtcore-rtp-topologies-update]).
[Roni Even] I think that SHOULD NOT is too strong here better use MAY be
used, this is a simple common mode of centralized video conferencing =
being
used. If you are talking about the RTP layer the MCU gets all the RTCP
reports. The problem of loop detection may occur when cascading MCUs and =
the
avtcore topology should point it out. I do not see it as a major problem
since the video switching is managed by the application layer who should
prevent such situations.

>=20
>    o  Content modifying MCUs with RTCP termination (Topo-RTCP-
>       terminating-MCU) SHOULD NOT be used since they break RTP loop
>       detection, and prevent receivers from identifying active senders
>       (see section 3.8 of [I-D.ietf-avtcore-rtp-topologies-update]).
>=20
>    o  The Relay-Transport Translator (Topo-PtM-Trn-Translator) =
topology
>       SHOULD NOT be used because its safe use requires a point to
>       multipoint congestion control algorithm or RTP circuit breaker,
>       which has not yet been standardised.
>=20
>    The RTP extensions described in Section 5.1.1 to Section 5.1.6 are
>    designed to be used with centralised conferencing, where an RTP
>    middlebox (e.g., a conference bridge) receives a participant's RTP
>    packet streams and distributes them to the other participants.  =
These
>    extensions are not necessary for interoperability; an RTP end-point
>    that does not implement these extensions will work correctly, but
>    might offer poor performance.  Support for the listed extensions =
will
>    greatly improve the quality of experience and, to provide a
>    reasonable baseline quality, some of these extensions are mandatory
>    to be supported by WebRTC end-points.
>=20
>    The RTCP conferencing extensions are defined in Extended RTP =
Profile
>    for Real-time Transport Control Protocol (RTCP)-Based Feedback =
(RTP/
>    AVPF) [RFC4585] and the "Codec Control Messages in the RTP Audio-
>    Visual Profile with Feedback (AVPF)" (CCM) [RFC5104] and are fully
>    usable by the Secure variant of this profile (RTP/SAVPF) [RFC5124].
>=20
>=20
> Cheers
>=20
> Magnus Westerlund
>=20
> ----------------------------------------------------------------------
> Services, Media and Network features, Ericsson Research EAB/TXM
> ----------------------------------------------------------------------
> Ericsson AB                 | Phone  +46 10 7148287
> F=E4r=F6gatan 6                 | Mobile +46 73 0949079
> SE-164 80 Stockholm, Sweden | mailto: magnus.westerlund@ericsson.com
> ----------------------------------------------------------------------
>=20
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb


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References: <CABkgnnWWuU63Vd=gw+wrh2ADgVYtQzhoRzRE1sv5azJE=MhWDg@mail.gmail.com> <533F191D.8050109@alum.mit.edu> <CABkgnnVht5EmJ7a2LDh50ivjUdoTpJ8GannQKReBSJbVGQGmgA@mail.gmail.com> <53419ED4.8020102@alum.mit.edu> <CABkgnnVjZ51bt5WQ1uvHHUz-4xFzpXQGhuMqxeMpOqJ1d+hQiA@mail.gmail.com> <7594FB04B1934943A5C02806D1A2204B1D2B26CB@ESESSMB209.ericsson.se>
To: Christer Holmberg <christer.holmberg@ericsson.com>
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Cc: "rtcweb@ietf.org" <rtcweb@ietf.org>
Subject: Re: [rtcweb] Asking TLS for help with media isolation
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On Apr 8, 2014, at 4:42 AM, Christer Holmberg =
<christer.holmberg@ericsson.com> wrote:

> Hi,
>=20
>>> But ISTM there would need to be a different set of limitations for=20=

>>> non-browsers. For instance, to not echo the stream back to the =
source.=20
>>> And not to store the stream in a place that the sender can retrieve.
>>=20
>> Yes.  At the most abstract level, the requirements are the same.
>> Don't pass the media on to someone else.
>=20
> How would that work with an intermediary mixer, transcoder, etc?

Or voicemail or an audio conference.

-d



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From: Bernard Aboba <bernard.aboba@gmail.com>
Date: Tue, 8 Apr 2014 09:50:55 -0700
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Subject: Re: [rtcweb] Asking TLS for help with media isolation
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Christer said:

"How would that work with an intermediary mixer, transcoder, etc."

[BA] I'm not sure that the concept of "isolation" makes sense for those
intermediaries (or to voicemail or an audio/video conference, for that
matter).   While in a point-to-point call it might be useful, in a
conference the whole point is to have audio/video sent to multiple parties,
and recording is commonplace.  The problem is that from a protocol point of
view the cases are not easily distinguishable -- and so if the browser
insists on "isolation" then one wonders what will happen if the conference
bridge/video MCU/voicemail system refuses to negotiate it.   Refusing to
send media would not be a desirable outcome.


On Tue, Apr 8, 2014 at 4:42 AM, Christer Holmberg <
christer.holmberg@ericsson.com> wrote:

> Hi,
>
> >> But ISTM there would need to be a different set of limitations for
> >> non-browsers. For instance, to not echo the stream back to the source.
> >> And not to store the stream in a place that the sender can retrieve.
> >
> >Yes.  At the most abstract level, the requirements are the same.
> >Don't pass the media on to someone else.
>
> How would that work with an intermediary mixer, transcoder, etc?
>
> Regards,
>
> Christer
>
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb
>

--047d7bb03f9ad780cf04f68aca01
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<div dir=3D"ltr"><div>Christer said: </div><div><br></div><div>&quot;How wo=
uld that work with an intermediary mixer, transcoder, etc.&quot;</div><div>=
<br></div><div>[BA] I&#39;m not sure that the concept of &quot;isolation&qu=
ot; makes sense for those intermediaries (or to voicemail or an audio/video=
 conference, for that matter).=A0=A0 While in a point-to-point call it migh=
t be useful, in a conference the whole point is to have audio/video sent to=
 multiple parties, and recording is commonplace.=A0 The problem is that fro=
m a protocol point of view the cases are not=A0easily distinguishable -- an=
d so if the browser insists on &quot;isolation&quot; then one wonders what =
will happen if the conference bridge/video MCU/voicemail system refuses to =
negotiate it.=A0=A0 Refusing to send media would not be a desirable outcome=
. </div>

</div><div class=3D"gmail_extra"><br><br><div class=3D"gmail_quote">On Tue,=
 Apr 8, 2014 at 4:42 AM, Christer Holmberg <span dir=3D"ltr">&lt;<a href=3D=
"mailto:christer.holmberg@ericsson.com" target=3D"_blank">christer.holmberg=
@ericsson.com</a>&gt;</span> wrote:<br>

<blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1p=
x #ccc solid;padding-left:1ex">Hi,<br>
<div><br>
&gt;&gt; But ISTM there would need to be a different set of limitations for=
<br>
&gt;&gt; non-browsers. For instance, to not echo the stream back to the sou=
rce.<br>
&gt;&gt; And not to store the stream in a place that the sender can retriev=
e.<br>
&gt;<br>
&gt;Yes. =A0At the most abstract level, the requirements are the same.<br>
&gt;Don&#39;t pass the media on to someone else.<br>
<br>
</div>How would that work with an intermediary mixer, transcoder, etc?<br>
<br>
Regards,<br>
<br>
Christer<br>
<div class=3D"HOEnZb"><div class=3D"h5"><br>
_______________________________________________<br>
rtcweb mailing list<br>
<a href=3D"mailto:rtcweb@ietf.org">rtcweb@ietf.org</a><br>
<a href=3D"https://www.ietf.org/mailman/listinfo/rtcweb" target=3D"_blank">=
https://www.ietf.org/mailman/listinfo/rtcweb</a><br>
</div></div></blockquote></div><br></div>

--047d7bb03f9ad780cf04f68aca01--


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On 8 April 2014 09:50, Bernard Aboba <bernard.aboba@gmail.com> wrote:
> [BA] I'm not sure that the concept of "isolation" makes sense for those
> intermediaries (or to voicemail or an audio/video conference, for that
> matter).   While in a point-to-point call it might be useful, in a
> conference the whole point is to have audio/video sent to multiple parties,
> and recording is commonplace.  The problem is that from a protocol point of
> view the cases are not easily distinguishable -- and so if the browser
> insists on "isolation" then one wonders what will happen if the conference
> bridge/video MCU/voicemail system refuses to negotiate it.   Refusing to
> send media would not be a desirable outcome.

I think that for this, it's perfectly reasonable to use identity, but
not stream isolation.  With isolation, if the peer does not agree to
comply, then the session fails to complete.

The authenticated party here is an MCU (or bridge, or voicemail,
etc...).  Rather than sending to "anindividual@example.org", media is
sent to "mcu@example.com".  Is it reasonable for that MCU to forward
media to other, unspecified entities?  Clearly it can, but should it?

(Not having thought it through completely, a voicemail box could
conceivably work.  I think that I'd want to use a different identity
for it though.)


From nobody Tue Apr  8 12:41:27 2014
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On Apr 8, 2014, at 11:24 AM, Martin Thomson <martin.thomson@gmail.com> =
wrote:

> On 8 April 2014 09:50, Bernard Aboba <bernard.aboba@gmail.com> wrote:
>> [BA] I'm not sure that the concept of "isolation" makes sense for =
those
>> intermediaries (or to voicemail or an audio/video conference, for =
that
>> matter).   While in a point-to-point call it might be useful, in a
>> conference the whole point is to have audio/video sent to multiple =
parties,
>> and recording is commonplace.  The problem is that from a protocol =
point of
>> view the cases are not easily distinguishable -- and so if the =
browser
>> insists on "isolation" then one wonders what will happen if the =
conference
>> bridge/video MCU/voicemail system refuses to negotiate it.   Refusing =
to
>> send media would not be a desirable outcome.
>=20
> I think that for this, it's perfectly reasonable to use identity, but
> not stream isolation.  With isolation, if the peer does not agree to
> comply, then the session fails to complete.
>=20
> The authenticated party here is an MCU (or bridge, or voicemail,
> etc...).  Rather than sending to "anindividual@example.org", media is
> sent to "mcu@example.com".  Is it reasonable for that MCU to forward
> media to other, unspecified entities?  Clearly it can, but should it?
>=20
> (Not having thought it through completely, a voicemail box could
> conceivably work.  I think that I'd want to use a different identity
> for it though.)

For voicemail, there was a proposal in 2009 for doing what amounted to =
object security, http://tools.ietf.org/html/draft-naslund-srtp-saf.  =
That document's last update was in 2011 but I don't know if it evolved =
into a product or died out or what.

-d


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

        Title           : WebRTC Data Channels
        Authors         : Randell Jesup
                          Salvatore Loreto
                          Michael Tuexen
	Filename        : draft-ietf-rtcweb-data-channel-08.txt
	Pages           : 15
	Date            : 2014-04-09

Abstract:
   The Real-Time Communication in WEB-browsers 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 specifies the non-(S)RTP media data transport aspects of the
   WebRTC framework.  It provides an architectural overview of how the
   Stream Control Transmission Protocol (SCTP) is used in the WebRTC
   context as a generic transport service allowing WEB-browsers to
   exchange generic data from peer to peer.


The 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-08

A diff from the previous version is available at:
http://www.ietf.org/rfcdiff?url2=draft-ietf-rtcweb-data-channel-08


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

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ftp://ftp.ietf.org/internet-drafts/


From nobody Wed Apr  9 03:03:35 2014
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A New Internet-Draft is available from the on-line Internet-Drafts directories.
 This draft is a work item of the Real-Time Communication in WEB-browsers Working Group of the IETF.

        Title           : WebRTC Data Channel Establishment Protocol
        Authors         : Randell Jesup
                          Salvatore Loreto
                          Michael Tuexen
	Filename        : draft-ietf-rtcweb-data-protocol-04.txt
	Pages           : 12
	Date            : 2014-04-09

Abstract:
   The Web Real-Time Communication (WebRTC) working group is charged to
   provide protocols to support for direct interactive rich
   communication using audio, video, and data between two peers' web-
   browsers.  This document specifies a simple protocol for establishing
   symmetric data channels between the peers.  It uses a two way
   handshake and allows sending of user data without waiting for the
   handshake to complete.


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

There's also a htmlized version available at:
http://tools.ietf.org/html/draft-ietf-rtcweb-data-protocol-04

A diff from the previous version is available at:
http://www.ietf.org/rfcdiff?url2=draft-ietf-rtcweb-data-protocol-04


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From nobody Wed Apr  9 04:00:58 2014
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Archived-At: http://mailarchive.ietf.org/arch/msg/rtcweb/REgXw1MDM1RmCB73iGIlMdgK1BE
Subject: Re: [rtcweb] I-D Action: draft-ietf-rtcweb-data-channel-08.txt
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On 9 Apr 2014, at 11:02, Internet-Drafts@ietf.org wrote:
> A New Internet-Draft is available from the on-line Internet-Drafts =
directories.
> This draft is a work item of the Real-Time Communication in =
WEB-browsers Working Group of the IETF.
>=20
>        Title           : WebRTC Data Channels
>        Authors         : Randell Jesup
>                          Salvatore Loreto
>                          Michael Tuexen
> 	Filename        : draft-ietf-rtcweb-data-channel-08.txt
> 	Pages           : 15
> 	Date            : 2014-04-09
>=20
> Abstract:
>   The Real-Time Communication in WEB-browsers 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 specifies the non-(S)RTP media data transport aspects of =
the
>   WebRTC framework.  It provides an architectural overview of how the
>   Stream Control Transmission Protocol (SCTP) is used in the WebRTC
>   context as a generic transport service allowing WEB-browsers to
>   exchange generic data from peer to peer.

This talks about =93(S)RTP=94 throughout, but the rtp-usage draft =
requires that SRTP be used for WebRTC, and disallows plain RTP. I think =
this draft could be simplified by changing =93(S)RTP=94 to =93SRTP=94 =
throughout.

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




From nobody Wed Apr  9 04:23:08 2014
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From: Christer Holmberg <christer.holmberg@ericsson.com>
To: Colin Perkins <csp@csperkins.org>, "rtcweb@ietf.org" <rtcweb@ietf.org>
Thread-Topic: [rtcweb] I-D Action: draft-ietf-rtcweb-data-channel-08.txt
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Hi,

The WebRTC Data Channel will be used also for non-WebRTC applications (it i=
s e.g. base for the CLUE Data Channel), where the must-use-SRTP rule does n=
ot necessarily apply.

Having said that, I DO think that the draft text need to be more generic. W=
ebRTC can be described as a use-case, but it shall be clear that the mechan=
ism can also be used in other environments.

Regards.

Christer

-----Original Message-----
From: rtcweb [mailto:rtcweb-bounces@ietf.org] On Behalf Of Colin Perkins
Sent: 9. huhtikuuta 2014 14:01
To: rtcweb@ietf.org
Subject: Re: [rtcweb] I-D Action: draft-ietf-rtcweb-data-channel-08.txt

On 9 Apr 2014, at 11:02, Internet-Drafts@ietf.org wrote:
> A New Internet-Draft is available from the on-line Internet-Drafts direct=
ories.
> This draft is a work item of the Real-Time Communication in WEB-browsers =
Working Group of the IETF.
>=20
>        Title           : WebRTC Data Channels
>        Authors         : Randell Jesup
>                          Salvatore Loreto
>                          Michael Tuexen
> 	Filename        : draft-ietf-rtcweb-data-channel-08.txt
> 	Pages           : 15
> 	Date            : 2014-04-09
>=20
> Abstract:
>   The Real-Time Communication in WEB-browsers 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 specifies the non-(S)RTP media data transport aspects of the
>   WebRTC framework.  It provides an architectural overview of how the
>   Stream Control Transmission Protocol (SCTP) is used in the WebRTC
>   context as a generic transport service allowing WEB-browsers to
>   exchange generic data from peer to peer.

This talks about "(S)RTP" throughout, but the rtp-usage draft requires that=
 SRTP be used for WebRTC, and disallows plain RTP. I think this draft could=
 be simplified by changing "(S)RTP" to "SRTP" throughout.

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



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


From nobody Wed Apr  9 04:42:02 2014
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From: Barry Dingle <btdingle@gmail.com>
Date: Wed, 9 Apr 2014 21:41:25 +1000
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--bcaec53f35f5c66cf704f69a9563
Content-Type: text/plain; charset=UTF-8

Hi,

If "WebRTC Data Channel" is to be used for other use-cases than just
WebRTC, then the Title needs to change.  But what do we call it?

Cheers,
Barry Dingle
Australia


On Wed, Apr 9, 2014 at 9:22 PM, Christer Holmberg <
christer.holmberg@ericsson.com> wrote:

> Hi,
>
> The WebRTC Data Channel will be used also for non-WebRTC applications (it
> is e.g. base for the CLUE Data Channel), where the must-use-SRTP rule does
> not necessarily apply.
>
> Having said that, I DO think that the draft text need to be more generic.
> WebRTC can be described as a use-case, but it shall be clear that the
> mechanism can also be used in other environments.
>
> Regards.
>
> Christer
>
> -----Original Message-----
> From: rtcweb [mailto:rtcweb-bounces@ietf.org] On Behalf Of Colin Perkins
> Sent: 9. huhtikuuta 2014 14:01
> To: rtcweb@ietf.org
> Subject: Re: [rtcweb] I-D Action: draft-ietf-rtcweb-data-channel-08.txt
>
> On 9 Apr 2014, at 11:02, Internet-Drafts@ietf.org wrote:
> > A New Internet-Draft is available from the on-line Internet-Drafts
> directories.
> > This draft is a work item of the Real-Time Communication in WEB-browsers
> Working Group of the IETF.
> >
> >        Title           : WebRTC Data Channels
> >        Authors         : Randell Jesup
> >                          Salvatore Loreto
> >                          Michael Tuexen
> >       Filename        : draft-ietf-rtcweb-data-channel-08.txt
> >       Pages           : 15
> >       Date            : 2014-04-09
> >
> > Abstract:
> >   The Real-Time Communication in WEB-browsers 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 specifies the non-(S)RTP media data transport aspects of the
> >   WebRTC framework.  It provides an architectural overview of how the
> >   Stream Control Transmission Protocol (SCTP) is used in the WebRTC
> >   context as a generic transport service allowing WEB-browsers to
> >   exchange generic data from peer to peer.
>
> This talks about "(S)RTP" throughout, but the rtp-usage draft requires
> that SRTP be used for WebRTC, and disallows plain RTP. I think this draft
> could be simplified by changing "(S)RTP" to "SRTP" throughout.
>
> --
> Colin Perkins
> http://csperkins.org/
>
>
>
> _______________________________________________
> 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
>

--bcaec53f35f5c66cf704f69a9563
Content-Type: text/html; charset=UTF-8
Content-Transfer-Encoding: quoted-printable

<div dir=3D"ltr"><div>Hi,<br><br></div>If &quot;WebRTC Data Channel&quot; i=
s to be used for other use-cases than just WebRTC, then the Title needs to =
change.=C2=A0 But what do we call it? <br><div class=3D"gmail_extra"><br cl=
ear=3D"all">

<div><div dir=3D"ltr">Cheers,<br>Barry Dingle<br>Australia</div></div>
<br><br><div class=3D"gmail_quote">On Wed, Apr 9, 2014 at 9:22 PM, Christer=
 Holmberg <span dir=3D"ltr">&lt;<a href=3D"mailto:christer.holmberg@ericsso=
n.com" target=3D"_blank">christer.holmberg@ericsson.com</a>&gt;</span> wrot=
e:<br>

<blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1p=
x #ccc solid;padding-left:1ex">Hi,<br>
<br>
The WebRTC Data Channel will be used also for non-WebRTC applications (it i=
s e.g. base for the CLUE Data Channel), where the must-use-SRTP rule does n=
ot necessarily apply.<br>
<br>
Having said that, I DO think that the draft text need to be more generic. W=
ebRTC can be described as a use-case, but it shall be clear that the mechan=
ism can also be used in other environments.<br>
<br>
Regards.<br>
<span class=3D"HOEnZb"><font color=3D"#888888"><br>
Christer<br>
</font></span><div class=3D"HOEnZb"><div class=3D"h5"><br>
-----Original Message-----<br>
From: rtcweb [mailto:<a href=3D"mailto:rtcweb-bounces@ietf.org">rtcweb-boun=
ces@ietf.org</a>] On Behalf Of Colin Perkins<br>
Sent: 9. huhtikuuta 2014 14:01<br>
To: <a href=3D"mailto:rtcweb@ietf.org">rtcweb@ietf.org</a><br>
Subject: Re: [rtcweb] I-D Action: draft-ietf-rtcweb-data-channel-08.txt<br>
<br>
On 9 Apr 2014, at 11:02, <a href=3D"mailto:Internet-Drafts@ietf.org">Intern=
et-Drafts@ietf.org</a> wrote:<br>
&gt; A New Internet-Draft is available from the on-line Internet-Drafts dir=
ectories.<br>
&gt; This draft is a work item of the Real-Time Communication in WEB-browse=
rs Working Group of the IETF.<br>
&gt;<br>
&gt; =C2=A0 =C2=A0 =C2=A0 =C2=A0Title =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 : =
WebRTC Data Channels<br>
&gt; =C2=A0 =C2=A0 =C2=A0 =C2=A0Authors =C2=A0 =C2=A0 =C2=A0 =C2=A0 : Rande=
ll Jesup<br>
&gt; =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=A0Salvatore Loreto<br>
&gt; =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=A0Michael Tuexen<br>
&gt; =C2=A0 =C2=A0 =C2=A0 Filename =C2=A0 =C2=A0 =C2=A0 =C2=A0: draft-ietf-=
rtcweb-data-channel-08.txt<br>
&gt; =C2=A0 =C2=A0 =C2=A0 Pages =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 : 15<br>
&gt; =C2=A0 =C2=A0 =C2=A0 Date =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0: 2=
014-04-09<br>
&gt;<br>
&gt; Abstract:<br>
&gt; =C2=A0 The Real-Time Communication in WEB-browsers working group is ch=
arged<br>
&gt; =C2=A0 to provide protocol support for direct interactive rich communi=
cation<br>
&gt; =C2=A0 using audio, video, and data between two peers&#39; web-browser=
s. =C2=A0This<br>
&gt; =C2=A0 document specifies the non-(S)RTP media data transport aspects =
of the<br>
&gt; =C2=A0 WebRTC framework. =C2=A0It provides an architectural overview o=
f how the<br>
&gt; =C2=A0 Stream Control Transmission Protocol (SCTP) is used in the WebR=
TC<br>
&gt; =C2=A0 context as a generic transport service allowing WEB-browsers to=
<br>
&gt; =C2=A0 exchange generic data from peer to peer.<br>
<br>
This talks about &quot;(S)RTP&quot; throughout, but the rtp-usage draft req=
uires that SRTP be used for WebRTC, and disallows plain RTP. I think this d=
raft could be simplified by changing &quot;(S)RTP&quot; to &quot;SRTP&quot;=
 throughout.<br>


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

--bcaec53f35f5c66cf704f69a9563--


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From: Barry Dingle <btdingle@gmail.com>
Date: Wed, 9 Apr 2014 22:17:37 +1000
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Subject: Re: [rtcweb] I-D Action: draft-ietf-rtcweb-data-protocol-04.txt
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Hi,

The first sentence of the Abstract in the WebRTC Data Channel Establishment
Protocol incorrectly refers to the "WebRTC working group. (The wording in
the second line also needs editing.)

I suggest copying the first sentence of the Abstract from WebRTC Data
Channels (-08 version) so it reads -

The Real-Time Communication in WEB-browsers working group is charged
   to provide protocol support for direct interactive rich communication
   using audio, video, and data between two peers' web-browsers.



Cheers,
/Barry Dingle
Australia


On Wed, Apr 9, 2014 at 8:03 PM, <internet-drafts@ietf.org> wrote:

>
> A New Internet-Draft is available from the on-line Internet-Drafts
> directories.
>  This draft is a work item of the Real-Time Communication in WEB-browsers
> Working Group of the IETF.
>
>         Title           : WebRTC Data Channel Establishment Protocol
>         Authors         : Randell Jesup
>                           Salvatore Loreto
>                           Michael Tuexen
>         Filename        : draft-ietf-rtcweb-data-protocol-04.txt
>         Pages           : 12
>         Date            : 2014-04-09
>
> Abstract:
>    The Web Real-Time Communication (WebRTC) working group is charged to
>    provide protocols to support for direct interactive rich
>    communication using audio, video, and data between two peers' web-
>    browsers.  This document specifies a simple protocol for establishing
>    symmetric data channels between the peers.  It uses a two way
>    handshake and allows sending of user data without waiting for the
>    handshake to complete.
>
>
> The IETF datatracker status page for this draft is:
> https://datatracker.ietf.org/doc/draft-ietf-rtcweb-data-protocol/
>
> There's also a htmlized version available at:
> http://tools.ietf.org/html/draft-ietf-rtcweb-data-protocol-04
>
> A diff from the previous version is available at:
> http://www.ietf.org/rfcdiff?url2=draft-ietf-rtcweb-data-protocol-04
>
>
> Please note that it may take a couple of minutes from the time of
> submission
> until the htmlized version and diff are available at tools.ietf.org.
>
> Internet-Drafts are also available by anonymous FTP at:
> ftp://ftp.ietf.org/internet-drafts/
>
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb
>

--f46d0434c0ca4bca1104f69b17aa
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<div dir=3D"ltr"><div>Hi,<br><br>The first sentence of the Abstract in the =
WebRTC Data Channel Establishment Protocol incorrectly refers to the &quot;=
WebRTC working group. (The wording in the second line also needs editing.)<=
br>

<br></div>I suggest copying the first sentence of the Abstract from WebRTC =
Data Channels (-08 version) so it reads - <br><br><pre>The Real-Time Commun=
ication in WEB-browsers working group is charged
   to provide protocol support for direct interactive rich communication
   using audio, video, and data between two peers&#39; web-browsers.</pre><=
br><div class=3D"gmail_extra"><br clear=3D"all"><div><div dir=3D"ltr">Cheer=
s,<br>/Barry Dingle<br></div><div>Australia<br><br></div></div><br><div cla=
ss=3D"gmail_quote">

On Wed, Apr 9, 2014 at 8:03 PM,  <span dir=3D"ltr">&lt;<a href=3D"mailto:in=
ternet-drafts@ietf.org" target=3D"_blank">internet-drafts@ietf.org</a>&gt;<=
/span> wrote:<br><blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8=
ex;border-left:1px #ccc solid;padding-left:1ex">

<br>
A New Internet-Draft is available from the on-line Internet-Drafts director=
ies.<br>
=C2=A0This draft is a work item of the Real-Time Communication in WEB-brows=
ers Working Group of the IETF.<br>
<br>
=C2=A0 =C2=A0 =C2=A0 =C2=A0 Title =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 : WebR=
TC Data Channel Establishment Protocol<br>
=C2=A0 =C2=A0 =C2=A0 =C2=A0 Authors =C2=A0 =C2=A0 =C2=A0 =C2=A0 : Randell J=
esup<br>
=C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=
=A0 =C2=A0 =C2=A0 Salvatore Loreto<br>
=C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=
=A0 =C2=A0 =C2=A0 Michael Tuexen<br>
=C2=A0 =C2=A0 =C2=A0 =C2=A0 Filename =C2=A0 =C2=A0 =C2=A0 =C2=A0: draft-iet=
f-rtcweb-data-protocol-04.txt<br>
=C2=A0 =C2=A0 =C2=A0 =C2=A0 Pages =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 : 12<b=
r>
=C2=A0 =C2=A0 =C2=A0 =C2=A0 Date =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0:=
 2014-04-09<br>
<br>
Abstract:<br>
=C2=A0 =C2=A0The Web Real-Time Communication (WebRTC) working group is char=
ged to<br>
=C2=A0 =C2=A0provide protocols to support for direct interactive rich<br>
=C2=A0 =C2=A0communication using audio, video, and data between two peers&#=
39; web-<br>
=C2=A0 =C2=A0browsers. =C2=A0This document specifies a simple protocol for =
establishing<br>
=C2=A0 =C2=A0symmetric data channels between the peers. =C2=A0It uses a two=
 way<br>
=C2=A0 =C2=A0handshake and allows sending of user data without waiting for =
the<br>
=C2=A0 =C2=A0handshake to complete.<br>
<br>
<br>
The IETF datatracker status page for this draft is:<br>
<a href=3D"https://datatracker.ietf.org/doc/draft-ietf-rtcweb-data-protocol=
/" target=3D"_blank">https://datatracker.ietf.org/doc/draft-ietf-rtcweb-dat=
a-protocol/</a><br>
<br>
There&#39;s also a htmlized version available at:<br>
<a href=3D"http://tools.ietf.org/html/draft-ietf-rtcweb-data-protocol-04" t=
arget=3D"_blank">http://tools.ietf.org/html/draft-ietf-rtcweb-data-protocol=
-04</a><br>
<br>
A diff from the previous version is available at:<br>
<a href=3D"http://www.ietf.org/rfcdiff?url2=3Ddraft-ietf-rtcweb-data-protoc=
ol-04" target=3D"_blank">http://www.ietf.org/rfcdiff?url2=3Ddraft-ietf-rtcw=
eb-data-protocol-04</a><br>
<br>
<br>
Please note that it may take a couple of minutes from the time of submissio=
n<br>
until the htmlized version and diff are available at <a href=3D"http://tool=
s.ietf.org" target=3D"_blank">tools.ietf.org</a>.<br>
<br>
Internet-Drafts are also available by anonymous FTP at:<br>
<a href=3D"ftp://ftp.ietf.org/internet-drafts/" target=3D"_blank">ftp://ftp=
.ietf.org/internet-drafts/</a><br>
<br>
_______________________________________________<br>
rtcweb mailing list<br>
<a href=3D"mailto:rtcweb@ietf.org">rtcweb@ietf.org</a><br>
<a href=3D"https://www.ietf.org/mailman/listinfo/rtcweb" target=3D"_blank">=
https://www.ietf.org/mailman/listinfo/rtcweb</a><br>
</blockquote></div><br></div></div>

--f46d0434c0ca4bca1104f69b17aa--


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References: <20140409100258.9712.74771.idtracker@ietfa.amsl.com> <F09BCD44-1060-4DCB-A796-7A31F1C634DE@csperkins.org> <7594FB04B1934943A5C02806D1A2204B1D2B7AAB@ESESSMB209.ericsson.se> <CAN=GVAvLNyw6m0_edBOSJhq_tAy0duCjKHtq4ed+rLwW5_U35A@mail.gmail.com> <7594FB04B1934943A5C02806D1A2204B1D2B7B0E@ESESSMB209.ericsson.se>
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SCTP Data Channel, or simply Data Channel.

On 4/9/14 7:46 AM, Christer Holmberg wrote:
> Hi,
>
>> If "WebRTC Data Channel" is to be used for other use-cases than just WebRTC, then the Title needs to change.  But what do we call it?
>
> I think we had this discussion already earlier, and people suggested different new names.
>
> Regards,
>
> Christer
>
>
>
> On Wed, Apr 9, 2014 at 9:22 PM, Christer Holmberg <christer.holmberg@ericsson.com> wrote:
> Hi,
>
> The WebRTC Data Channel will be used also for non-WebRTC applications (it is e.g. base for the CLUE Data Channel), where the must-use-SRTP rule does not necessarily apply.
>
> Having said that, I DO think that the draft text need to be more generic. WebRTC can be described as a use-case, but it shall be clear that the mechanism can also be used in other environments.
>
> Regards.
>
> Christer
>
> -----Original Message-----
> From: rtcweb [mailto:rtcweb-bounces@ietf.org] On Behalf Of Colin Perkins
> Sent: 9. huhtikuuta 2014 14:01
> To: rtcweb@ietf.org
> Subject: Re: [rtcweb] I-D Action: draft-ietf-rtcweb-data-channel-08.txt
>
> On 9 Apr 2014, at 11:02, Internet-Drafts@ietf.org wrote:
>> A New Internet-Draft is available from the on-line Internet-Drafts directories.
>> This draft is a work item of the Real-Time Communication in WEB-browsers Working Group of the IETF.
>>
>>         Title           : WebRTC Data Channels
>>         Authors         : Randell Jesup
>>                           Salvatore Loreto
>>                           Michael Tuexen
>>        Filename        : draft-ietf-rtcweb-data-channel-08.txt
>>        Pages           : 15
>>        Date            : 2014-04-09
>>
>> Abstract:
>>    The Real-Time Communication in WEB-browsers 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 specifies the non-(S)RTP media data transport aspects of the
>>    WebRTC framework.  It provides an architectural overview of how the
>>    Stream Control Transmission Protocol (SCTP) is used in the WebRTC
>>    context as a generic transport service allowing WEB-browsers to
>>    exchange generic data from peer to peer.
>
> This talks about "(S)RTP" throughout, but the rtp-usage draft requires that SRTP be used for WebRTC, and disallows plain RTP. I think this draft could be simplified by changing "(S)RTP" to "SRTP" throughout.
>
> --
> Colin Perkins
> http://csperkins.org/
>
>
>
> _______________________________________________
> 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
>


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From: Barry Dingle <btdingle@gmail.com>
Date: Thu, 10 Apr 2014 00:19:52 +1000
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--047d7bb03f9c7b20a104f69ccca7
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Data Channel Establishment Protocol *Editorials* -

Section 4 - 2nd paragraph -

Change

"The set of consistent properties includes "   to the following

"The set of consistent properties includes : "


Section 4 - 3rd para (after the 6 dot points), 2nd sentence reads as
follows -

"The side wanting to open a data channel selects an SCTP stream
identifier for which the
   corresponding incoming and outgoing SCTP stream is unused and sends a
   DATA_CHANNEL_OPEN message on this outgoing SCTP stream. "

This is confusing. I suggest it be simplified to -

"The side wanting to open a data channel selects an unused SCTP stream
identifier   and sends a
   DATA_CHANNEL_OPEN message on this outgoing SCTP stream. "


Cheers,
/Barry Dingle
Australia


On Wed, Apr 9, 2014 at 8:03 PM, <internet-drafts@ietf.org> wrote:

>
> A New Internet-Draft is available from the on-line Internet-Drafts
> directories.
>  This draft is a work item of the Real-Time Communication in WEB-browsers
> Working Group of the IETF.
>
>         Title           : WebRTC Data Channel Establishment Protocol
>         Authors         : Randell Jesup
>                           Salvatore Loreto
>                           Michael Tuexen
>         Filename        : draft-ietf-rtcweb-data-protocol-04.txt
>         Pages           : 12
>         Date            : 2014-04-09
>
> Abstract:
>    The Web Real-Time Communication (WebRTC) working group is charged to
>    provide protocols to support for direct interactive rich
>    communication using audio, video, and data between two peers' web-
>    browsers.  This document specifies a simple protocol for establishing
>    symmetric data channels between the peers.  It uses a two way
>    handshake and allows sending of user data without waiting for the
>    handshake to complete.
>
>
> The IETF datatracker status page for this draft is:
> https://datatracker.ietf.org/doc/draft-ietf-rtcweb-data-protocol/
>
> There's also a htmlized version available at:
> http://tools.ietf.org/html/draft-ietf-rtcweb-data-protocol-04
>
> A diff from the previous version is available at:
> http://www.ietf.org/rfcdiff?url2=draft-ietf-rtcweb-data-protocol-04
>
>
> Please note that it may take a couple of minutes from the time of
> submission
> until the htmlized version and diff are available at tools.ietf.org.
>
> Internet-Drafts are also available by anonymous FTP at:
> ftp://ftp.ietf.org/internet-drafts/
>
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb
>

--047d7bb03f9c7b20a104f69ccca7
Content-Type: text/html; charset=UTF-8
Content-Transfer-Encoding: quoted-printable

<div dir=3D"ltr"><div><div><div>Data Channel Establishment Protocol <u>Edit=
orials</u> - <br><br></div>Section 4 - 2nd paragraph - <br><br></div><div>C=
hange=C2=A0 <br><pre class=3D"">&quot;The set of consistent properties incl=
udes &quot;   to the following <br>

<br>&quot;The set of consistent properties includes : &quot;   </pre></div>=
<div><br></div>Section 4 - 3rd para (after the 6 dot points), 2nd sentence =
reads as follows -=C2=A0 <br></div><pre class=3D"">&quot;The side wanting t=
o open a data channel selects an SCTP stream identifier for which the
   corresponding incoming and outgoing SCTP stream is unused and sends a
   DATA_CHANNEL_OPEN message on this outgoing SCTP stream. &quot;</pre>This=
 is confusing. I suggest it be simplified to - <br><pre class=3D"">&quot;Th=
e side wanting to open a data channel selects an <span style=3D"color:rgb(0=
,0,255)">unused</span> SCTP stream identifier   and sends a
   DATA_CHANNEL_OPEN message on this outgoing SCTP stream. &quot;</pre><div=
 class=3D"gmail_extra"><br clear=3D"all"><div><div dir=3D"ltr">Cheers,<br>/=
Barry Dingle<br></div><div>Australia<br></div></div>
<br><br><div class=3D"gmail_quote">On Wed, Apr 9, 2014 at 8:03 PM,  <span d=
ir=3D"ltr">&lt;<a href=3D"mailto:internet-drafts@ietf.org" target=3D"_blank=
">internet-drafts@ietf.org</a>&gt;</span> wrote:<br><blockquote class=3D"gm=
ail_quote" style=3D"margin:0px 0px 0px 0.8ex;border-left:1px solid rgb(204,=
204,204);padding-left:1ex">

<br>
A New Internet-Draft is available from the on-line Internet-Drafts director=
ies.<br>
=C2=A0This draft is a work item of the Real-Time Communication in WEB-brows=
ers Working Group of the IETF.<br>
<br>
=C2=A0 =C2=A0 =C2=A0 =C2=A0 Title =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 : WebR=
TC Data Channel Establishment Protocol<br>
=C2=A0 =C2=A0 =C2=A0 =C2=A0 Authors =C2=A0 =C2=A0 =C2=A0 =C2=A0 : Randell J=
esup<br>
=C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=
=A0 =C2=A0 =C2=A0 Salvatore Loreto<br>
=C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=
=A0 =C2=A0 =C2=A0 Michael Tuexen<br>
=C2=A0 =C2=A0 =C2=A0 =C2=A0 Filename =C2=A0 =C2=A0 =C2=A0 =C2=A0: draft-iet=
f-rtcweb-data-protocol-04.txt<br>
=C2=A0 =C2=A0 =C2=A0 =C2=A0 Pages =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 : 12<b=
r>
=C2=A0 =C2=A0 =C2=A0 =C2=A0 Date =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0 =C2=A0:=
 2014-04-09<br>
<br>
Abstract:<br>
=C2=A0 =C2=A0The Web Real-Time Communication (WebRTC) working group is char=
ged to<br>
=C2=A0 =C2=A0provide protocols to support for direct interactive rich<br>
=C2=A0 =C2=A0communication using audio, video, and data between two peers&#=
39; web-<br>
=C2=A0 =C2=A0browsers. =C2=A0This document specifies a simple protocol for =
establishing<br>
=C2=A0 =C2=A0symmetric data channels between the peers. =C2=A0It uses a two=
 way<br>
=C2=A0 =C2=A0handshake and allows sending of user data without waiting for =
the<br>
=C2=A0 =C2=A0handshake to complete.<br>
<br>
<br>
The IETF datatracker status page for this draft is:<br>
<a href=3D"https://datatracker.ietf.org/doc/draft-ietf-rtcweb-data-protocol=
/" target=3D"_blank">https://datatracker.ietf.org/doc/draft-ietf-rtcweb-dat=
a-protocol/</a><br>
<br>
There&#39;s also a htmlized version available at:<br>
<a href=3D"http://tools.ietf.org/html/draft-ietf-rtcweb-data-protocol-04" t=
arget=3D"_blank">http://tools.ietf.org/html/draft-ietf-rtcweb-data-protocol=
-04</a><br>
<br>
A diff from the previous version is available at:<br>
<a href=3D"http://www.ietf.org/rfcdiff?url2=3Ddraft-ietf-rtcweb-data-protoc=
ol-04" target=3D"_blank">http://www.ietf.org/rfcdiff?url2=3Ddraft-ietf-rtcw=
eb-data-protocol-04</a><br>
<br>
<br>
Please note that it may take a couple of minutes from the time of submissio=
n<br>
until the htmlized version and diff are available at <a href=3D"http://tool=
s.ietf.org" target=3D"_blank">tools.ietf.org</a>.<br>
<br>
Internet-Drafts are also available by anonymous FTP at:<br>
<a href=3D"ftp://ftp.ietf.org/internet-drafts/" target=3D"_blank">ftp://ftp=
.ietf.org/internet-drafts/</a><br>
<br>
_______________________________________________<br>
rtcweb mailing list<br>
<a href=3D"mailto:rtcweb@ietf.org">rtcweb@ietf.org</a><br>
<a href=3D"https://www.ietf.org/mailman/listinfo/rtcweb" target=3D"_blank">=
https://www.ietf.org/mailman/listinfo/rtcweb</a><br>
</blockquote></div><br></div></div>

--047d7bb03f9c7b20a104f69ccca7--


From nobody Wed Apr  9 07:20:50 2014
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On 09 Apr 2014, at 13:00, Colin Perkins <csp@csperkins.org> wrote:

> On 9 Apr 2014, at 11:02, Internet-Drafts@ietf.org wrote:
>> A New Internet-Draft is available from the on-line Internet-Drafts =
directories.
>> This draft is a work item of the Real-Time Communication in =
WEB-browsers Working Group of the IETF.
>>=20
>>       Title           : WebRTC Data Channels
>>       Authors         : Randell Jesup
>>                         Salvatore Loreto
>>                         Michael Tuexen
>> 	Filename        : draft-ietf-rtcweb-data-channel-08.txt
>> 	Pages           : 15
>> 	Date            : 2014-04-09
>>=20
>> Abstract:
>>  The Real-Time Communication in WEB-browsers 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 specifies the non-(S)RTP media data transport aspects of =
the
>>  WebRTC framework.  It provides an architectural overview of how the
>>  Stream Control Transmission Protocol (SCTP) is used in the WebRTC
>>  context as a generic transport service allowing WEB-browsers to
>>  exchange generic data from peer to peer.
>=20
> This talks about =93(S)RTP=94 throughout, but the rtp-usage draft =
requires that SRTP be used for WebRTC, and disallows plain RTP. I think =
this draft could be simplified by changing =93(S)RTP=94 to =93SRTP=94 =
throughout.
Hi Colin,

The (S)RTP notion goes back to a comment from Magnus. If I remember it =
correctly he considers
SRTP a profile of RTP. Since I don't wanted to just use RTP, I ended up =
with (S)RTP based
on a discussion with Magnus.

However, I'm fine with changing it to SRTP...

Magnus: Any comment?

Best regards
Michael

>=20
> --=20
> Colin Perkins
> http://csperkins.org/
>=20
>=20
>=20
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb
>=20


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On 9 Apr 2014, at 15:20, Michael Tuexen =
<Michael.Tuexen@lurchi.franken.de> wrote:
> On 09 Apr 2014, at 13:00, Colin Perkins <csp@csperkins.org> wrote:
>> On 9 Apr 2014, at 11:02, Internet-Drafts@ietf.org wrote:
>>> A New Internet-Draft is available from the on-line Internet-Drafts =
directories.
>>> This draft is a work item of the Real-Time Communication in =
WEB-browsers Working Group of the IETF.
>>>=20
>>>      Title           : WebRTC Data Channels
>>>      Authors         : Randell Jesup
>>>                        Salvatore Loreto
>>>                        Michael Tuexen
>>> 	Filename        : draft-ietf-rtcweb-data-channel-08.txt
>>> 	Pages           : 15
>>> 	Date            : 2014-04-09
>>>=20
>>> Abstract:
>>> The Real-Time Communication in WEB-browsers 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 specifies the non-(S)RTP media data transport aspects of =
the
>>> WebRTC framework.  It provides an architectural overview of how the
>>> Stream Control Transmission Protocol (SCTP) is used in the WebRTC
>>> context as a generic transport service allowing WEB-browsers to
>>> exchange generic data from peer to peer.
>>=20
>> This talks about =93(S)RTP=94 throughout, but the rtp-usage draft =
requires that SRTP be used for WebRTC, and disallows plain RTP. I think =
this draft could be simplified by changing =93(S)RTP=94 to =93SRTP=94 =
throughout.
> Hi Colin,
>=20
> The (S)RTP notion goes back to a comment from Magnus. If I remember it =
correctly he considers SRTP a profile of RTP. Since I don=92t wanted to =
just use RTP, I ended up with (S)RTP based on a discussion with Magnus.
>=20
> However, I=92m fine with changing it to SRTP...

SRTP is an RTP profile. My comment was that if this is for WebRTC only, =
then  only SRTP can be used, and not plain RTP. Using =93(S)RTP=94 =
rather than =93SRTP=94 in this draft suggests that the secure profile is =
optional, which isn=92t the case in WebRTC. If this is for more general =
use than WebRTC, then =93(S)RTP=94 is fine.

Colin



> Magnus: Any comment?
>=20
> Best regards
> Michael
>=20
>>=20
>> --=20
>> Colin Perkins
>> http://csperkins.org/
>>=20
>>=20
>>=20
>> _______________________________________________
>> rtcweb mailing list
>> rtcweb@ietf.org
>> https://www.ietf.org/mailman/listinfo/rtcweb
>>=20
>=20



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




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On 09 Apr 2014, at 14:17, Barry Dingle <btdingle@gmail.com> wrote:

> Hi,
>=20
> The first sentence of the Abstract in the WebRTC Data Channel =
Establishment Protocol incorrectly refers to the "WebRTC working group. =
(The wording in the second line also needs editing.)
>=20
> I suggest copying the first sentence of the Abstract from WebRTC Data =
Channels (-08 version) so it reads -=20
>=20
> The Real-Time Communication in WEB-browsers working group is charged
>    to provide protocol support for direct interactive rich =
communication
>    using audio, video, and data between two peers' web-browsers.
OK. Will be in the next revision.

Best regards
Michael
>=20
>=20
>=20
> Cheers,
> /Barry Dingle
> Australia
>=20
>=20
> On Wed, Apr 9, 2014 at 8:03 PM, <internet-drafts@ietf.org> wrote:
>=20
> A New Internet-Draft is available from the on-line Internet-Drafts =
directories.
>  This draft is a work item of the Real-Time Communication in =
WEB-browsers Working Group of the IETF.
>=20
>         Title           : WebRTC Data Channel Establishment Protocol
>         Authors         : Randell Jesup
>                           Salvatore Loreto
>                           Michael Tuexen
>         Filename        : draft-ietf-rtcweb-data-protocol-04.txt
>         Pages           : 12
>         Date            : 2014-04-09
>=20
> Abstract:
>    The Web Real-Time Communication (WebRTC) working group is charged =
to
>    provide protocols to support for direct interactive rich
>    communication using audio, video, and data between two peers' web-
>    browsers.  This document specifies a simple protocol for =
establishing
>    symmetric data channels between the peers.  It uses a two way
>    handshake and allows sending of user data without waiting for the
>    handshake to complete.
>=20
>=20
> The IETF datatracker status page for this draft is:
> https://datatracker.ietf.org/doc/draft-ietf-rtcweb-data-protocol/
>=20
> There's also a htmlized version available at:
> http://tools.ietf.org/html/draft-ietf-rtcweb-data-protocol-04
>=20
> A diff from the previous version is available at:
> http://www.ietf.org/rfcdiff?url2=3Ddraft-ietf-rtcweb-data-protocol-04
>=20
>=20
> Please note that it may take a couple of minutes from the time of =
submission
> until the htmlized version and diff are available at tools.ietf.org.
>=20
> Internet-Drafts are also available by anonymous FTP at:
> ftp://ftp.ietf.org/internet-drafts/
>=20
> _______________________________________________
> 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 nobody Wed Apr  9 07:32:32 2014
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From: Michael Tuexen <Michael.Tuexen@lurchi.franken.de>
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On 09 Apr 2014, at 16:19, Barry Dingle <btdingle@gmail.com> wrote:

> Data Channel Establishment Protocol Editorials -=20
>=20
> Section 4 - 2nd paragraph -=20
>=20
> Change =20
> "The set of consistent properties includes "   to the following=20
>=20
>=20
>=20
>=20
> "The set of consistent properties includes : "  =20
Done.
>=20
> Section 4 - 3rd para (after the 6 dot points), 2nd sentence reads as =
follows - =20
> "The side wanting to open a data channel selects an SCTP stream =
identifier for which the
>    corresponding incoming and outgoing SCTP stream is unused and sends =
a
>    DATA_CHANNEL_OPEN message on this outgoing SCTP stream. "
>=20
> This is confusing. I suggest it be simplified to -=20
> "The side wanting to open a data channel selects an unused
>  SCTP stream identifier   and sends a
>    DATA_CHANNEL_OPEN message on this outgoing SCTP stream. "
But it is the pair of streams which are unused, not the stream =
identifier.
Is

The side wanting to open a data channel selects an SCTP stream =
identifier for which the
corresponding incoming and outgoing SCTP streams are unused and sends a
DATA_CHANNEL_OPEN message on the outgoing SCTP stream.

better?

Best regards
Michael
>=20
>=20
> Cheers,
> /Barry Dingle
> Australia
>=20
>=20
> On Wed, Apr 9, 2014 at 8:03 PM, <internet-drafts@ietf.org> wrote:
>=20
> A New Internet-Draft is available from the on-line Internet-Drafts =
directories.
>  This draft is a work item of the Real-Time Communication in =
WEB-browsers Working Group of the IETF.
>=20
>         Title           : WebRTC Data Channel Establishment Protocol
>         Authors         : Randell Jesup
>                           Salvatore Loreto
>                           Michael Tuexen
>         Filename        : draft-ietf-rtcweb-data-protocol-04.txt
>         Pages           : 12
>         Date            : 2014-04-09
>=20
> Abstract:
>    The Web Real-Time Communication (WebRTC) working group is charged =
to
>    provide protocols to support for direct interactive rich
>    communication using audio, video, and data between two peers' web-
>    browsers.  This document specifies a simple protocol for =
establishing
>    symmetric data channels between the peers.  It uses a two way
>    handshake and allows sending of user data without waiting for the
>    handshake to complete.
>=20
>=20
> The IETF datatracker status page for this draft is:
> https://datatracker.ietf.org/doc/draft-ietf-rtcweb-data-protocol/
>=20
> There's also a htmlized version available at:
> http://tools.ietf.org/html/draft-ietf-rtcweb-data-protocol-04
>=20
> A diff from the previous version is available at:
> http://www.ietf.org/rfcdiff?url2=3Ddraft-ietf-rtcweb-data-protocol-04
>=20
>=20
> Please note that it may take a couple of minutes from the time of =
submission
> until the htmlized version and diff are available at tools.ietf.org.
>=20
> Internet-Drafts are also available by anonymous FTP at:
> ftp://ftp.ietf.org/internet-drafts/
>=20
> _______________________________________________
> 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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On 09 Apr 2014, at 16:25, Colin Perkins <csp@csperkins.org> wrote:

> On 9 Apr 2014, at 15:20, Michael Tuexen =
<Michael.Tuexen@lurchi.franken.de> wrote:
>> On 09 Apr 2014, at 13:00, Colin Perkins <csp@csperkins.org> wrote:
>>> On 9 Apr 2014, at 11:02, Internet-Drafts@ietf.org wrote:
>>>> A New Internet-Draft is available from the on-line Internet-Drafts =
directories.
>>>> This draft is a work item of the Real-Time Communication in =
WEB-browsers Working Group of the IETF.
>>>>=20
>>>>     Title           : WebRTC Data Channels
>>>>     Authors         : Randell Jesup
>>>>                       Salvatore Loreto
>>>>                       Michael Tuexen
>>>> 	Filename        : draft-ietf-rtcweb-data-channel-08.txt
>>>> 	Pages           : 15
>>>> 	Date            : 2014-04-09
>>>>=20
>>>> Abstract:
>>>> The Real-Time Communication in WEB-browsers 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 specifies the non-(S)RTP media data transport aspects of =
the
>>>> WebRTC framework.  It provides an architectural overview of how the
>>>> Stream Control Transmission Protocol (SCTP) is used in the WebRTC
>>>> context as a generic transport service allowing WEB-browsers to
>>>> exchange generic data from peer to peer.
>>>=20
>>> This talks about =93(S)RTP=94 throughout, but the rtp-usage draft =
requires that SRTP be used for WebRTC, and disallows plain RTP. I think =
this draft could be simplified by changing =93(S)RTP=94 to =93SRTP=94 =
throughout.
>> Hi Colin,
>>=20
>> The (S)RTP notion goes back to a comment from Magnus. If I remember =
it correctly he considers SRTP a profile of RTP. Since I don=92t wanted =
to just use RTP, I ended up with (S)RTP based on a discussion with =
Magnus.
>>=20
>> However, I=92m fine with changing it to SRTP...
>=20
> SRTP is an RTP profile. My comment was that if this is for WebRTC =
only, then  only SRTP can be used, and not plain RTP. Using =93(S)RTP=94 =
rather than =93SRTP=94 in this draft suggests that the secure profile is =
optional, which isn=92t the case in WebRTC. If this is for more general =
use than WebRTC, then =93(S)RTP=94 is fine.
It is clear that in WebRTC only SRTP is used...

Best regards
Michael
>=20
> Colin
>=20
>=20
>=20
>> Magnus: Any comment?
>>=20
>> Best regards
>> Michael
>>=20
>>>=20
>>> --=20
>>> Colin Perkins
>>> http://csperkins.org/
>>>=20
>>>=20
>>>=20
>>> _______________________________________________
>>> rtcweb mailing list
>>> rtcweb@ietf.org
>>> https://www.ietf.org/mailman/listinfo/rtcweb
>>>=20
>>=20
>=20
>=20
>=20
> --=20
> Colin Perkins
> http://csperkins.org/
>=20
>=20
>=20
>=20


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On 9 Apr 2014, at 15:33, Michael Tuexen =
<Michael.Tuexen@lurchi.franken.de> wrote:
> On 09 Apr 2014, at 16:25, Colin Perkins <csp@csperkins.org> wrote:
>> On 9 Apr 2014, at 15:20, Michael Tuexen =
<Michael.Tuexen@lurchi.franken.de> wrote:
>>> On 09 Apr 2014, at 13:00, Colin Perkins <csp@csperkins.org> wrote:
>>>> On 9 Apr 2014, at 11:02, Internet-Drafts@ietf.org wrote:
>>>>> A New Internet-Draft is available from the on-line Internet-Drafts =
directories.
>>>>> This draft is a work item of the Real-Time Communication in =
WEB-browsers Working Group of the IETF.
>>>>>=20
>>>>>    Title           : WebRTC Data Channels
>>>>>    Authors         : Randell Jesup
>>>>>                      Salvatore Loreto
>>>>>                      Michael Tuexen
>>>>> 	Filename        : draft-ietf-rtcweb-data-channel-08.txt
>>>>> 	Pages           : 15
>>>>> 	Date            : 2014-04-09
>>>>>=20
>>>>> Abstract:
>>>>> The Real-Time Communication in WEB-browsers 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 specifies the non-(S)RTP media data transport aspects of =
the
>>>>> WebRTC framework.  It provides an architectural overview of how =
the
>>>>> Stream Control Transmission Protocol (SCTP) is used in the WebRTC
>>>>> context as a generic transport service allowing WEB-browsers to
>>>>> exchange generic data from peer to peer.
>>>>=20
>>>> This talks about =93(S)RTP=94 throughout, but the rtp-usage draft =
requires that SRTP be used for WebRTC, and disallows plain RTP. I think =
this draft could be simplified by changing =93(S)RTP=94 to =93SRTP=94 =
throughout.
>>> Hi Colin,
>>>=20
>>> The (S)RTP notion goes back to a comment from Magnus. If I remember =
it correctly he considers SRTP a profile of RTP. Since I don=92t wanted =
to just use RTP, I ended up with (S)RTP based on a discussion with =
Magnus.
>>>=20
>>> However, I=92m fine with changing it to SRTP...
>>=20
>> SRTP is an RTP profile. My comment was that if this is for WebRTC =
only, then  only SRTP can be used, and not plain RTP. Using =93(S)RTP=94 =
rather than =93SRTP=94 in this draft suggests that the secure profile is =
optional, which isn=92t the case in WebRTC. If this is for more general =
use than WebRTC, then =93(S)RTP=94 is fine.
> It is clear that in WebRTC only SRTP is used...

I don=92t believe your draft is clear on that, due to the use of =
=93(S)RTP=94 terminology. That=92s why I commented.

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




From nobody Wed Apr  9 07:47:26 2014
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On 09 Apr 2014, at 16:39, Colin Perkins <csp@csperkins.org> wrote:

> On 9 Apr 2014, at 15:33, Michael Tuexen =
<Michael.Tuexen@lurchi.franken.de> wrote:
>> On 09 Apr 2014, at 16:25, Colin Perkins <csp@csperkins.org> wrote:
>>> On 9 Apr 2014, at 15:20, Michael Tuexen =
<Michael.Tuexen@lurchi.franken.de> wrote:
>>>> On 09 Apr 2014, at 13:00, Colin Perkins <csp@csperkins.org> wrote:
>>>>> On 9 Apr 2014, at 11:02, Internet-Drafts@ietf.org wrote:
>>>>>> A New Internet-Draft is available from the on-line =
Internet-Drafts directories.
>>>>>> This draft is a work item of the Real-Time Communication in =
WEB-browsers Working Group of the IETF.
>>>>>>=20
>>>>>>   Title           : WebRTC Data Channels
>>>>>>   Authors         : Randell Jesup
>>>>>>                     Salvatore Loreto
>>>>>>                     Michael Tuexen
>>>>>> 	Filename        : draft-ietf-rtcweb-data-channel-08.txt
>>>>>> 	Pages           : 15
>>>>>> 	Date            : 2014-04-09
>>>>>>=20
>>>>>> Abstract:
>>>>>> The Real-Time Communication in WEB-browsers 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 specifies the non-(S)RTP media data transport aspects of =
the
>>>>>> WebRTC framework.  It provides an architectural overview of how =
the
>>>>>> Stream Control Transmission Protocol (SCTP) is used in the WebRTC
>>>>>> context as a generic transport service allowing WEB-browsers to
>>>>>> exchange generic data from peer to peer.
>>>>>=20
>>>>> This talks about =93(S)RTP=94 throughout, but the rtp-usage draft =
requires that SRTP be used for WebRTC, and disallows plain RTP. I think =
this draft could be simplified by changing =93(S)RTP=94 to =93SRTP=94 =
throughout.
>>>> Hi Colin,
>>>>=20
>>>> The (S)RTP notion goes back to a comment from Magnus. If I remember =
it correctly he considers SRTP a profile of RTP. Since I don=92t wanted =
to just use RTP, I ended up with (S)RTP based on a discussion with =
Magnus.
>>>>=20
>>>> However, I=92m fine with changing it to SRTP...
>>>=20
>>> SRTP is an RTP profile. My comment was that if this is for WebRTC =
only, then  only SRTP can be used, and not plain RTP. Using =93(S)RTP=94 =
rather than =93SRTP=94 in this draft suggests that the secure profile is =
optional, which isn=92t the case in WebRTC. If this is for more general =
use than WebRTC, then =93(S)RTP=94 is fine.
>> It is clear that in WebRTC only SRTP is used...
>=20
> I don=92t believe your draft is clear on that, due to the use of =
=93(S)RTP=94 terminology. That=92s why I commented.
I'm happy to change it to SRTP... Let's see what Magnus thinks...

Best regards
Michael
>=20
> --=20
> Colin Perkins
> http://csperkins.org/
>=20
>=20
>=20
>=20


From nobody Wed Apr  9 08:02:14 2014
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To: Michael Tuexen <Michael.Tuexen@lurchi.franken.de>, Colin Perkins <csp@csperkins.org>
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On 2014-04-09 16:47, Michael Tuexen wrote:
> 
> On 09 Apr 2014, at 16:39, Colin Perkins <csp@csperkins.org> wrote:
> 
>> On 9 Apr 2014, at 15:33, Michael Tuexen <Michael.Tuexen@lurchi.franken.de> wrote:
>>> On 09 Apr 2014, at 16:25, Colin Perkins <csp@csperkins.org> wrote:
>>>> On 9 Apr 2014, at 15:20, Michael Tuexen <Michael.Tuexen@lurchi.franken.de> wrote:
>>>>> On 09 Apr 2014, at 13:00, Colin Perkins <csp@csperkins.org> wrote:
>>>>>> On 9 Apr 2014, at 11:02, Internet-Drafts@ietf.org wrote:
>>>>>>> A New Internet-Draft is available from the on-line Internet-Drafts directories.
>>>>>>> This draft is a work item of the Real-Time Communication in WEB-browsers Working Group of the IETF.
>>>>>>>
>>>>>>>   Title           : WebRTC Data Channels
>>>>>>>   Authors         : Randell Jesup
>>>>>>>                     Salvatore Loreto
>>>>>>>                     Michael Tuexen
>>>>>>> 	Filename        : draft-ietf-rtcweb-data-channel-08.txt
>>>>>>> 	Pages           : 15
>>>>>>> 	Date            : 2014-04-09
>>>>>>>
>>>>>>> Abstract:
>>>>>>> The Real-Time Communication in WEB-browsers 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 specifies the non-(S)RTP media data transport aspects of the
>>>>>>> WebRTC framework.  It provides an architectural overview of how the
>>>>>>> Stream Control Transmission Protocol (SCTP) is used in the WebRTC
>>>>>>> context as a generic transport service allowing WEB-browsers to
>>>>>>> exchange generic data from peer to peer.
>>>>>>
>>>>>> This talks about “(S)RTP” throughout, but the rtp-usage draft requires that SRTP be used for WebRTC, and disallows plain RTP. I think this draft could be simplified by changing “(S)RTP” to “SRTP” throughout.
>>>>> Hi Colin,
>>>>>
>>>>> The (S)RTP notion goes back to a comment from Magnus. If I remember it correctly he considers SRTP a profile of RTP. Since I don’t wanted to just use RTP, I ended up with (S)RTP based on a discussion with Magnus.
>>>>>
>>>>> However, I’m fine with changing it to SRTP...
>>>>
>>>> SRTP is an RTP profile. My comment was that if this is for WebRTC only, then  only SRTP can be used, and not plain RTP. Using “(S)RTP” rather than “SRTP” in this draft suggests that the secure profile is optional, which isn’t the case in WebRTC. If this is for more general use than WebRTC, then “(S)RTP” is fine.
>>> It is clear that in WebRTC only SRTP is used...
>>
>> I don’t believe your draft is clear on that, due to the use of “(S)RTP” terminology. That’s why I commented.
> I'm happy to change it to SRTP... Let's see what Magnus thinks...

I am fine with that!

/Magnus

> 
> Best regards
> Michael
>>
>> -- 
>> Colin Perkins
>> http://csperkins.org/
>>
>>
>>
>>
> 
> 
> 


-- 

Magnus Westerlund

----------------------------------------------------------------------
Services, Media and Network features, Ericsson Research EAB/TXM
----------------------------------------------------------------------
Ericsson AB                 | Phone  +46 10 7148287
Färögatan 6                 | Mobile +46 73 0949079
SE-164 80 Stockholm, Sweden | mailto: magnus.westerlund@ericsson.com
----------------------------------------------------------------------


From nobody Wed Apr  9 08:10:43 2014
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Subject: Re: [rtcweb] I-D Action: draft-ietf-rtcweb-data-channel-08.txt
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On 09 Apr 2014, at 17:01, Magnus Westerlund =
<magnus.westerlund@ericsson.com> wrote:

> On 2014-04-09 16:47, Michael Tuexen wrote:
>>=20
>> On 09 Apr 2014, at 16:39, Colin Perkins <csp@csperkins.org> wrote:
>>=20
>>> On 9 Apr 2014, at 15:33, Michael Tuexen =
<Michael.Tuexen@lurchi.franken.de> wrote:
>>>> On 09 Apr 2014, at 16:25, Colin Perkins <csp@csperkins.org> wrote:
>>>>> On 9 Apr 2014, at 15:20, Michael Tuexen =
<Michael.Tuexen@lurchi.franken.de> wrote:
>>>>>> On 09 Apr 2014, at 13:00, Colin Perkins <csp@csperkins.org> =
wrote:
>>>>>>> On 9 Apr 2014, at 11:02, Internet-Drafts@ietf.org wrote:
>>>>>>>> A New Internet-Draft is available from the on-line =
Internet-Drafts directories.
>>>>>>>> This draft is a work item of the Real-Time Communication in =
WEB-browsers Working Group of the IETF.
>>>>>>>>=20
>>>>>>>>  Title           : WebRTC Data Channels
>>>>>>>>  Authors         : Randell Jesup
>>>>>>>>                    Salvatore Loreto
>>>>>>>>                    Michael Tuexen
>>>>>>>> 	Filename        : draft-ietf-rtcweb-data-channel-08.txt
>>>>>>>> 	Pages           : 15
>>>>>>>> 	Date            : 2014-04-09
>>>>>>>>=20
>>>>>>>> Abstract:
>>>>>>>> The Real-Time Communication in WEB-browsers 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 specifies the non-(S)RTP media data transport aspects =
of the
>>>>>>>> WebRTC framework.  It provides an architectural overview of how =
the
>>>>>>>> Stream Control Transmission Protocol (SCTP) is used in the =
WebRTC
>>>>>>>> context as a generic transport service allowing WEB-browsers to
>>>>>>>> exchange generic data from peer to peer.
>>>>>>>=20
>>>>>>> This talks about =93(S)RTP=94 throughout, but the rtp-usage =
draft requires that SRTP be used for WebRTC, and disallows plain RTP. I =
think this draft could be simplified by changing =93(S)RTP=94 to =93SRTP=94=
 throughout.
>>>>>> Hi Colin,
>>>>>>=20
>>>>>> The (S)RTP notion goes back to a comment from Magnus. If I =
remember it correctly he considers SRTP a profile of RTP. Since I don=92t =
wanted to just use RTP, I ended up with (S)RTP based on a discussion =
with Magnus.
>>>>>>=20
>>>>>> However, I=92m fine with changing it to SRTP...
>>>>>=20
>>>>> SRTP is an RTP profile. My comment was that if this is for WebRTC =
only, then  only SRTP can be used, and not plain RTP. Using =93(S)RTP=94 =
rather than =93SRTP=94 in this draft suggests that the secure profile is =
optional, which isn=92t the case in WebRTC. If this is for more general =
use than WebRTC, then =93(S)RTP=94 is fine.
>>>> It is clear that in WebRTC only SRTP is used...
>>>=20
>>> I don=92t believe your draft is clear on that, due to the use of =
=93(S)RTP=94 terminology. That=92s why I commented.
>> I'm happy to change it to SRTP... Let's see what Magnus thinks...
>=20
> I am fine with that!
OK. Next revision used SRTP.

Best regards
Michael
>=20
> /Magnus
>=20
>>=20
>> Best regards
>> Michael
>>>=20
>>> --=20
>>> Colin Perkins
>>> http://csperkins.org/
>>>=20
>>>=20
>>>=20
>>>=20
>>=20
>>=20
>>=20
>=20
>=20
> --=20
>=20
> Magnus Westerlund
>=20
> ----------------------------------------------------------------------
> Services, Media and Network features, Ericsson Research EAB/TXM
> ----------------------------------------------------------------------
> Ericsson AB                 | Phone  +46 10 7148287
> F=E4r=F6gatan 6                 | Mobile +46 73 0949079
> SE-164 80 Stockholm, Sweden | mailto: magnus.westerlund@ericsson.com
> ----------------------------------------------------------------------
>=20
>=20


From nobody Wed Apr  9 11:09:28 2014
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From: Martin Thomson <martin.thomson@gmail.com>
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The intent of this draft is to address the isolation issue.

Unlike a TLS extension, this doesn't require that the TLS WG provide
an official blessing.  Though I'll note that this is the advice that
at least two participants in that working group suggested for
addressing this problem.

In short: negotiate "webrtc" if you are doing the usual, insecure
thing; negotiate "c-webrtc" if you would like media confidentiality.

>>
A new version of I-D, draft-thomson-rtcweb-alpn-00.txt
has been successfully submitted by Martin Thomson and posted to the
IETF repository.

Name:           draft-thomson-rtcweb-alpn
Revision:       00
Title:          Application Layer Protocol Negotiation for Web
Real-Time Communications (WebRTC)
Document date:  2014-04-09
Group:          Individual Submission
Pages:          6
URL:
http://www.ietf.org/internet-drafts/draft-thomson-rtcweb-alpn-00.txt
Status:         https://datatracker.ietf.org/doc/draft-thomson-rtcweb-alpn/
Htmlized:       http://tools.ietf.org/html/draft-thomson-rtcweb-alpn-00


Abstract:
   Application Layer Protocol Negotiation (ALPN) labels are defined for
   use in identifying Web Real-Time Communications (WebRTC) usages of
   Datagram Transport Layer Security (DTLS).  Labels are provided for
   identifying a session that uses a combination of WebRTC compatible
   media and data, and for identifying a session requiring
   confidentiality protection.


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From: Christer Holmberg <christer.holmberg@ericsson.com>
To: Michael Tuexen <Michael.Tuexen@lurchi.franken.de>, Magnus Westerlund <magnus.westerlund@ericsson.com>
Thread-Topic: [rtcweb] I-D Action: draft-ietf-rtcweb-data-channel-08.txt
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References: <20140409100258.9712.74771.idtracker@ietfa.amsl.com> <F09BCD44-1060-4DCB-A796-7A31F1C634DE@csperkins.org> <A05F0177-568C-4B19-AD48-9F415A4C008B@lurchi.franken.de> <02F2BCF4-70B5-47A4-ACE6-C0CCCAB11A50@csperkins.org> <9889BAD9-D9A7-42F2-A0DC-632C26696345@lurchi.franken.de> <73C21E05-E36C-4899-98A4-9D762B75FCE4@csperkins.org> <6C2B5FD8-D9A8-4889-8483-046B4ACC75EC@lurchi.franken.de> <534560B1.2050900@ericsson.com> <2A39482B-5DF0-492E-8CCB-D2E507000816@lurchi.franken.de>
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Hi,

If you want to describe the WebRTC usage, I'm fine using SRTP.

But, again, as there are other usages of the data channel than WebRTC, the =
generic text should not talk about only SRTP.

Regards,

Christer

-----Original Message-----
From: rtcweb [mailto:rtcweb-bounces@ietf.org] On Behalf Of Michael Tuexen
Sent: 09 April 2014 18:11
To: Magnus Westerlund
Cc: rtcweb@ietf.org; Colin Perkins
Subject: Re: [rtcweb] I-D Action: draft-ietf-rtcweb-data-channel-08.txt


On 09 Apr 2014, at 17:01, Magnus Westerlund <magnus.westerlund@ericsson.com=
> wrote:

> On 2014-04-09 16:47, Michael Tuexen wrote:
>>=20
>> On 09 Apr 2014, at 16:39, Colin Perkins <csp@csperkins.org> wrote:
>>=20
>>> On 9 Apr 2014, at 15:33, Michael Tuexen <Michael.Tuexen@lurchi.franken.=
de> wrote:
>>>> On 09 Apr 2014, at 16:25, Colin Perkins <csp@csperkins.org> wrote:
>>>>> On 9 Apr 2014, at 15:20, Michael Tuexen <Michael.Tuexen@lurchi.franke=
n.de> wrote:
>>>>>> On 09 Apr 2014, at 13:00, Colin Perkins <csp@csperkins.org> wrote:
>>>>>>> On 9 Apr 2014, at 11:02, Internet-Drafts@ietf.org wrote:
>>>>>>>> A New Internet-Draft is available from the on-line Internet-Drafts=
 directories.
>>>>>>>> This draft is a work item of the Real-Time Communication in WEB-br=
owsers Working Group of the IETF.
>>>>>>>>=20
>>>>>>>>  Title           : WebRTC Data Channels
>>>>>>>>  Authors         : Randell Jesup
>>>>>>>>                    Salvatore Loreto
>>>>>>>>                    Michael Tuexen
>>>>>>>> 	Filename        : draft-ietf-rtcweb-data-channel-08.txt
>>>>>>>> 	Pages           : 15
>>>>>>>> 	Date            : 2014-04-09
>>>>>>>>=20
>>>>>>>> Abstract:
>>>>>>>> The Real-Time Communication in WEB-browsers working group is=20
>>>>>>>> charged to provide protocol support for direct interactive rich=20
>>>>>>>> communication using audio, video, and data between two peers'=20
>>>>>>>> web-browsers.  This document specifies the non-(S)RTP media=20
>>>>>>>> data transport aspects of the WebRTC framework.  It provides an=20
>>>>>>>> architectural overview of how the Stream Control Transmission=20
>>>>>>>> Protocol (SCTP) is used in the WebRTC context as a generic=20
>>>>>>>> transport service allowing WEB-browsers to exchange generic data f=
rom peer to peer.
>>>>>>>=20
>>>>>>> This talks about "(S)RTP" throughout, but the rtp-usage draft requi=
res that SRTP be used for WebRTC, and disallows plain RTP. I think this dra=
ft could be simplified by changing "(S)RTP" to "SRTP" throughout.
>>>>>> Hi Colin,
>>>>>>=20
>>>>>> The (S)RTP notion goes back to a comment from Magnus. If I remember =
it correctly he considers SRTP a profile of RTP. Since I don't wanted to ju=
st use RTP, I ended up with (S)RTP based on a discussion with Magnus.
>>>>>>=20
>>>>>> However, I'm fine with changing it to SRTP...
>>>>>=20
>>>>> SRTP is an RTP profile. My comment was that if this is for WebRTC onl=
y, then  only SRTP can be used, and not plain RTP. Using "(S)RTP" rather th=
an "SRTP" in this draft suggests that the secure profile is optional, which=
 isn't the case in WebRTC. If this is for more general use than WebRTC, the=
n "(S)RTP" is fine.
>>>> It is clear that in WebRTC only SRTP is used...
>>>=20
>>> I don't believe your draft is clear on that, due to the use of "(S)RTP"=
 terminology. That's why I commented.
>> I'm happy to change it to SRTP... Let's see what Magnus thinks...
>=20
> I am fine with that!
OK. Next revision used SRTP.

Best regards
Michael
>=20
> /Magnus
>=20
>>=20
>> Best regards
>> Michael
>>>=20
>>> --
>>> Colin Perkins
>>> http://csperkins.org/
>>>=20
>>>=20
>>>=20
>>>=20
>>=20
>>=20
>>=20
>=20
>=20
> --
>=20
> Magnus Westerlund
>=20
> ----------------------------------------------------------------------
> Services, Media and Network features, Ericsson Research EAB/TXM
> ----------------------------------------------------------------------
> Ericsson AB                 | Phone  +46 10 7148287
> F=E4r=F6gatan 6                 | Mobile +46 73 0949079
> SE-164 80 Stockholm, Sweden | mailto: magnus.westerlund@ericsson.com
> ----------------------------------------------------------------------
>=20
>=20

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


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If you propose to use ALPN for this, then "webrtc" and "c-webrtc" must 
be Application Level Protocols - presumably *different* protocols.

But what is the *protocol*? I don't see that mentioned in the draft at all.

IIUC it must be the multiplexing of STUN, SRTP, and SCTP over DTLS. 
(Maybe not STUN - maybe that is *below* or *beside* DTLS.) I see how it 
could make sense to use ALPN to verify that this is the intended use of 
the DTLS session. But that isn't even mentioned in the draft.

And, from a protocol perspective, what is the difference between webrtc 
and c-webrtc? AFAICT this is just two different usages of the same 
"protocol", not two different protocols.

	Thanks,
	Paul

On 4/9/14 2:09 PM, Martin Thomson wrote:
> The intent of this draft is to address the isolation issue.
>
> Unlike a TLS extension, this doesn't require that the TLS WG provide
> an official blessing.  Though I'll note that this is the advice that
> at least two participants in that working group suggested for
> addressing this problem.
>
> In short: negotiate "webrtc" if you are doing the usual, insecure
> thing; negotiate "c-webrtc" if you would like media confidentiality.
>
>>>
> A new version of I-D, draft-thomson-rtcweb-alpn-00.txt
> has been successfully submitted by Martin Thomson and posted to the
> IETF repository.
>
> Name:           draft-thomson-rtcweb-alpn
> Revision:       00
> Title:          Application Layer Protocol Negotiation for Web
> Real-Time Communications (WebRTC)
> Document date:  2014-04-09
> Group:          Individual Submission
> Pages:          6
> URL:
> http://www.ietf.org/internet-drafts/draft-thomson-rtcweb-alpn-00.txt
> Status:         https://datatracker.ietf.org/doc/draft-thomson-rtcweb-alpn/
> Htmlized:       http://tools.ietf.org/html/draft-thomson-rtcweb-alpn-00
>
>
> Abstract:
>     Application Layer Protocol Negotiation (ALPN) labels are defined for
>     use in identifying Web Real-Time Communications (WebRTC) usages of
>     Datagram Transport Layer Security (DTLS).  Labels are provided for
>     identifying a session that uses a combination of WebRTC compatible
>     media and data, and for identifying a session requiring
>     confidentiality protection.
>
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb
>


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On 9 April 2014 12:12, Paul Kyzivat <pkyzivat@alum.mit.edu> wrote:
> But what is the *protocol*? I don't see that mentioned in the draft at all.

There are two.  Those two are described (perhaps not explicitly
enough) in the list in Section 2.

> IIUC it must be the multiplexing of STUN, SRTP, and SCTP over DTLS. (Maybe
> not STUN - maybe that is *below* or *beside* DTLS.) I see how it could make
> sense to use ALPN to verify that this is the intended use of the DTLS
> session. But that isn't even mentioned in the draft.

Yeah, STUN isn't inside DTLS, so it doesn't make a lot of sense to
"negotiate" its use.

> And, from a protocol perspective, what is the difference between webrtc and
> c-webrtc? AFAICT this is just two different usages of the same "protocol",
> not two different protocols.

Yep.  But that's what a protocol is.


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On 4/9/14 3:52 PM, Martin Thomson wrote:
> On 9 April 2014 12:12, Paul Kyzivat <pkyzivat@alum.mit.edu> wrote:
>> But what is the *protocol*? I don't see that mentioned in the draft at all.
>
> There are two.  Those two are described (perhaps not explicitly
> enough) in the list in Section 2.

Not explicitly enough, IMO.

>> IIUC it must be the multiplexing of STUN, SRTP, and SCTP over DTLS. (Maybe
>> not STUN - maybe that is *below* or *beside* DTLS.) I see how it could make
>> sense to use ALPN to verify that this is the intended use of the DTLS
>> session. But that isn't even mentioned in the draft.
>
> Yeah, STUN isn't inside DTLS, so it doesn't make a lot of sense to
> "negotiate" its use.
>
>> And, from a protocol perspective, what is the difference between webrtc and
>> c-webrtc? AFAICT this is just two different usages of the same "protocol",
>> not two different protocols.
>
> Yep.  But that's what a protocol is.

Not in my book. I expect two different protocols to look different on 
the wire.

You seem to be saying that SMTP used to talk to ietf mailing lists is a 
different protocol from SMTP used to talk to my lawyer, because I expect 
my lawyer to keep the communications confidential.

	Thanks,
	Paul


From nobody Wed Apr  9 14:26:00 2014
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On 9 April 2014 13:56, Paul Kyzivat <pkyzivat@alum.mit.edu> wrote:
> I expect two different protocols to look different on the wire.
>
> You seem to be saying that SMTP used to talk to ietf mailing lists is a
> different protocol from SMTP used to talk to my lawyer, because I expect my
> lawyer to keep the communications confidential.

They are different on the wire.  They use different identifiers.

That's hair splitting, but there's a real difference between the two
usages.  And I think that it's important enough to do this.

Do you perhaps have an alternative, or is it just that this lack of
solidity is giving you heartburn?  Because I can appreciate that.


From nobody Wed Apr  9 14:41:30 2014
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FWIW, I think the ALPN approach is an improvement, improving the likelihood o=
f deployment.=20


From nobody Wed Apr  9 20:58:37 2014
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Hello,

I've looked into the JSEP draft and WebRTC API (as well as the 
datachannel and DCEP drafts) to get a better understanding of how and 
when the SDP for the DTLS/SCTP Web-RTCDataChannel will be generated and 
what will trigger the DCEP protocol. Unfortunately I find myself more 
confused after the process .

I was trying to find some text regarding what happens in terms of SDP 
and DCEP when an endpoint uses multiple datachannels per peerConnection. 
e.g.

pc = new RTCPeerConnection(configuration);
channel = pc.createDataChannel("chat");
channel = pc.createDataChannel("clue");

Now based on the WebRTC API 
(http://dev.w3.org/2011/webrtc/editor/webrtc.html#peer-to-peer-data-api), I 
understand that a createOffer() is used to generate the required SDP for 
the data channel. This is also mentioned in Clause 5.2.1 "initial 
offers" draft-ietf-rtcweb-jsep-06 indicates: /"Lastly, if a data channel 
has been created, a m= section MUST be generated for data..."/

The WebRTC API cl.5.1.2 "createDataChannel" step 9 says /"Create 
channel's associated underlying data transport and configure it 
according to the relevant properties of channel."/ Now unfortunately it 
doesn't really say want this amounts to. Create could be "a new SCTP 
association" or it could be "using DCEP to do a DATA_CHANNEL_OPEN" or both.

If I look at the RTCPeerConnection Interface for addStream I see that it 
indicates that a negotiationneeded event (WebRTC API cl.4.3.2.3 step 5) 
is fired. This is a trigger for the createOffer(). The example 10.1 
WebRTC API shows this behavior. 10.3 also shows this behavior for 
createDataChannel but the Peer-to-Peer Data API doesn't indicate that 
the negotiationneeded event is triggered when establishing a data channel.

I looked at the RTCDataChannel closure procedure to see if this would 
shed any more light but simply says that the underlying data transport 
may be torn down. The JSEP draft isn't any help here either as it 
doesn't talk about RTCDataChannel closure at all.

So its not clear to me from the drafts/API whether:
1. each createDataChannel results in an offer with new m-line 
"webrtc-datachannel",
2. the first createDataChannel results in an SDP offer with m-line 
"webrtc-datachannel" and subsequent createDataChannels result in DCEP 
DATA_CHANNEL_OPENs on the existing webrtc-datachannel.
3. a mix of the above an endpoint can indicate multiple m-lines 
"webrtc-datachannel" and have multiple DCEP DATA_CHANNEL_OPENs per m-line.

Its clear from draft-ietf-rtcweb-data-channel-07 that "Multiple 
simultaneous data channels MUST be supported in the peer connection" but 
that doesn't clarify the above.

Is there any specification text that clarifies this?

Regards, Christian


From nobody Thu Apr 10 01:52:36 2014
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On 04/09/2014 10:56 PM, Paul Kyzivat wrote:
> On 4/9/14 3:52 PM, Martin Thomson wrote:
>> On 9 April 2014 12:12, Paul Kyzivat <pkyzivat@alum.mit.edu> wrote:
>>> But what is the *protocol*? I don't see that mentioned in the draft 
>>> at all.
>>
>> There are two.  Those two are described (perhaps not explicitly
>> enough) in the list in Section 2.
>
> Not explicitly enough, IMO.
>
>>> IIUC it must be the multiplexing of STUN, SRTP, and SCTP over DTLS. 
>>> (Maybe
>>> not STUN - maybe that is *below* or *beside* DTLS.) I see how it 
>>> could make
>>> sense to use ALPN to verify that this is the intended use of the DTLS
>>> session. But that isn't even mentioned in the draft.
>>
>> Yeah, STUN isn't inside DTLS, so it doesn't make a lot of sense to
>> "negotiate" its use.
>>
>>> And, from a protocol perspective, what is the difference between 
>>> webrtc and
>>> c-webrtc? AFAICT this is just two different usages of the same 
>>> "protocol",
>>> not two different protocols.
>>
>> Yep.  But that's what a protocol is.
>
> Not in my book. I expect two different protocols to look different on 
> the wire.
>
> You seem to be saying that SMTP used to talk to ietf mailing lists is 
> a different protocol from SMTP used to talk to my lawyer, because I 
> expect my lawyer to keep the communications confidential.

(not to be taken as a comment on the proposal, because I haven't 
read/understood it yet)

an adequate parallel would be SMTP (used for MTA-to-MTA relay on port 
25) and SUBMIT (used for message submission on port 587).

They are identical on the wire (apart from an extension in the EHLO 
response), but have different purposes, and are therefore called 
different protocols.



From nobody Thu Apr 10 01:56:08 2014
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Subject: Re: [rtcweb] Asking TLS for help with media isolation
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On 04/08/2014 08:24 PM, Martin Thomson wrote:
> On 8 April 2014 09:50, Bernard Aboba <bernard.aboba@gmail.com> wrote:
>> [BA] I'm not sure that the concept of "isolation" makes sense for those
>> intermediaries (or to voicemail or an audio/video conference, for that
>> matter).   While in a point-to-point call it might be useful, in a
>> conference the whole point is to have audio/video sent to multiple parties,
>> and recording is commonplace.  The problem is that from a protocol point of
>> view the cases are not easily distinguishable -- and so if the browser
>> insists on "isolation" then one wonders what will happen if the conference
>> bridge/video MCU/voicemail system refuses to negotiate it.   Refusing to
>> send media would not be a desirable outcome.
> I think that for this, it's perfectly reasonable to use identity, but
> not stream isolation.  With isolation, if the peer does not agree to
> comply, then the session fails to complete.
Actually I'd say it's "if the peer does not *agree to* comply".
The protocol has no defense against liars, but that's a common issue.
>
> The authenticated party here is an MCU (or bridge, or voicemail,
> etc...).  Rather than sending to "anindividual@example.org", media is
> sent to "mcu@example.com".  Is it reasonable for that MCU to forward
> media to other, unspecified entities?  Clearly it can, but should it?
>
> (Not having thought it through completely, a voicemail box could
> conceivably work.  I think that I'd want to use a different identity
> for it though.)

I can see an use for a recording spec that said "you can record this, 
but only if you do it in such a way that it's only accessible to the 
stated identity".

Would be weird to try to enforce that.... but I agree; MCU and isolation 
have a hard time mixing.
Let's just not.


From nobody Thu Apr 10 04:51:07 2014
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On 04 Mar 2014, at 17:39, Magnus Westerlund =
<magnus.westerlund@ericsson.com> wrote:

> On 2014-02-25 17:07, Michael Tuexen wrote:
>>=20
>> On Feb 24, 2014, at 5:32 PM, Magnus Westerlund =
<magnus.westerlund@ericsson.com> wrote:
>>=20
>>> Hi,
>>> (as individual)
>> Hi Magnus,
>>=20
>> thank you very much for the detailed review. Please see my comments =
in-line.
>>=20
>> Best regards
>> Michael
>>>=20
>>> I have reviewed the WebRTC Data Channel Establishment protocol
>>> (draft-ietf-rtcweb-data-protocol-03) and have some comments:
>>>=20
>>> 1. Section 4:
>>> Shouldn't this section discuss the priority field?
>> I added in the list of consistent properties:
>>=20
>> <t>the priority of the data Channel.</t>
>>=20
>> and in the text below that enumeration:
>>=20
>> ??????
>=20
> Yes, text for this needs to be figured out.
We don't define at this place what priority is. The only point is that =
both
sides use the same priority.
>=20
>>>=20
>>> 2. Section 4:
>>>=20
>>> The method
>>>  used to determine which side uses odd or even is based on the
>>>  underlying DTLS connection role when used in WebRTC, with the side
>>>  acting as the DTLS client using even stream identifiers.
>>>=20
>>> Isn't this unnecessary using the vague word of WebRTC instead of =
simply
>>> pointing to the DTLS roles of the established data channel?
>=20
>> The point is that in the WebRTC you use DCEP/SCTP/DTLS/UDP and =
therefore
>> you can refer to the DTLS role. However, you could use DCEP/SCTP/IP
>> or DCEP/SCTP/UDP/IP or DCEP/SCTP over something not involving DTLS.
>> In that case DTLS is not used and you can not refer to the DTLS role.
>> That is why the restriction is used.
>=20
> Ok, if that concern then you still should be able to write a normative
> specification under the condition that it is SCTP over DTLS. If not =
how
> do you determine that? Are suggesting just to hand way or point to a
> higher signaling layer.
So what about using:

when using <xref target=3D'I-D.ietf-tsvwg-sctp-dtls-encaps'/>, the =
method used to
determine which side uses odd or even is based on the underlying DTLS
connection role: the side acting as the DTLS client MUST use Streams =
with even
SCTP stream identifiers, the side acting as the DTLS server MUST use =
Streams
with odd SCTP stream identifiers.</t>

>>>=20
>>> 3. Section 5.1:
>>>     DATA_CHANNEL_PARTIAL_RELIABLE_TIMED (0x02):  The channel =
provides
>>>        a partial reliable in-order bi-directional Communication
>>>        channel.  User messages might not be transmitted or
>>>        retransmitted after a specified life-time given in milli-
>>>        seconds in the Reliability Parameter.  This life-time starts
>>>        when providing the user message to the Javascript engine.
>>>=20
>>> I think the use of "Javascript engine" in the last sentence is =
unclear.
>>> Can we provide a implementation neutral definition, like "This =
life-time
>>> starts when requesting transmission of the user message."?
>=20
>> This comment was already made on the list and I changed the text to
>>=20
>> This life-time starts when providing the user message to the protocol =
stack.
>>=20
>> OK?
>=20
> Yes.
>=20
>>>=20
>>> 4. Section 5.1:
>>>  Label: Variable Length (sequence of characters)
>>>     The name of the channel.  This may be an empty string.
>>>=20
>>>  Protocol: Variable Length (sequence of characters)
>>>     The protocol for the channel.  If this is an empty string the
>>>     protocol us unspecified.  If it is an non-empty string, it
>>>     specifies an IANA-registered protocol (see Section 8.4).
>>>=20
>>> Both of these fields are strings, shouldn't a particular encoding be
>>> specified here? Like UTF-8. Secondly, what values are allowed, the =
full
>>> set of Unicode?
>=20
>> You are right. Any need for restrictions? We only need to be able to
>> transform it to a DomString.
>> So I changed it to:
>>=20
>> <t hangText=3D'Label: Variable Length (sequence of characters)'>
>> <vspace blankLines=3D'0'/>
>> The name of the channel as a UTF-8 encoded string.
>> This may be an empty string.</t>
>>=20
>> <t hangText=3D'Protocol: Variable Length (sequence of characters)'>
>> <vspace blankLines=3D'0'/>
>> The protocol for the channel as a UTF-8 encoded string.
>> If this is an empty string the protocol us unspecified.
>> If it is an non-empty string, it specifies an IANA-registered =
protocol
>> (see <xref target=3D'iana_protocol'/>).</t>
>=20
> I guess this is slightly overtaken by event. It have to be aligned =
with
> what the websocket sub-protocol identifier.
The text now reads:
<t hangText=3D'Protocol: Variable Length (sequence of characters)'>
<vspace blankLines=3D'0'/>
The sub-protocol for the channel as a UTF-8 encoded string.
If this is an empty string the protocol is unspecified.
If it is a non-empty string, it specifies an protocol registered in the
'WebSocket Subprotocol Name Registry' created in
<xref target=3D'RFC6455'/>.</t>
>=20
>>>=20
>>> 5. Section 6:
>>> All Data Channel Establishment Protocol messages MUST be sent
>>>  requesting ordered delivery and using reliable transmission.
>>>=20
>>> I wonder of the use of requesting ordered delivery and using =
reliable
>>> transmission, from an SCTP stream perspective, wouldn't using in =
both
>>> places be appropriate? Or is how object which has been requested to =
be
>>> transmitted unordered interact in SCTP with the ordered ones?
>=20
>> I'm sorry, I don't understand what you are asking...
>=20
> Sorry, it really is a language issue. The above sentence first states
> "sent 'requesting' ordered" and later "and 'using' reliable".
>=20
> This inconsistency is what I reacted to.
I see. Changed to:
<t>All Data Channel Establishment Protocol messages MUST be sent using
ordered delivery and reliable transmission.
>=20
>>=20
>>> 6. Section 7:
>>>=20
>>> I think this section can be beefed up a bit. First make clear that =
the
>>> Data Channel's required usage of DTLS ensures that the message =
integrity
>>> and possible source authentication as well as confidentiality. Then
>>> going over any security risks with a malicous peer using this =
protocol.
>>> Can a malicous side screw up the peer using this protocol? What are =
the
>>> implications?
>=20
>> Just to double check: Aren't the referenced documents the ones which
>> discuss all security stuff in one place?
>=20
> The security document do discuss system level important aspects. But,
> from my perspective, details that are very specific to this protocol
> should be discussed in this document.
>=20
> I think this is highly relevant in this document as there are others
> that are interested in using it.
OK. Any concrete suggestions what should be covered?
>=20
>>>=20
>>> 7. Section 8.2:
>>>=20
>>> This registry reserves 0xff without any motivation, please add one.
>> I added:
>>=20
>> The value 0xff has been reserved for future extensibility.
>>=20
>=20
> Ok
>=20
>>>=20
>>> 8. Section 8.3:
>>> If I define a channel behavior that allows mixed ordered and =
unordered,
>>> how would I register this?
>=20
>> You can't. For SCTP properties like ordered/unordered delivery are
>> on a per message base. In RTCWeb it was decided to make per channel
>> base. So you can't have a data channel supporting a mixtures of
>> ordered and unordered messages.
>=20
> Ok, fine.
>=20
>>>=20
>>> 9. Section 8.4:
>>>=20
>>> I have a suggestion for this registries name, instead of "Protocol
>>>  Registry" I propose  "DCEP Protocol Identification Registry"
>=20
>> The problem I have with this definition is the following:
>> Your name suggests that the use of this entity is restricted to
>> DCEP. But isn't the protocol a property of a data channel, no matter
>> whether negotiated by used DCEP or not?
>=20
> As we now are going to point at the websocket one this is a mote =
point.
>=20
>>>=20
>>> 10. Section 8.4:
>>>=20
>>> There is missing any specification of what type of strings I may
>>> register. Also, no length limitation, although the protocol allows =
at
>>> maximum of 255 octets of encoded characters.
>=20
>> I don't understand the limitation to 255 octets. The length field
>> is 16 bit, so you can have up to 65535 octets. Therefore I changed:
>> <t>A name for the protocol;</t>
>> to
>> <t>A name for the protocol which length is smaller than 65536 when =
using
>> an UTF-8 encoding;</t>
>=20
> Overtaken by events, limitations will be the ones of the Websocket
> registry.
>=20
>>>=20
>>> 11. This comments concerns the relation to the Data Channel
>>> specification. So my interpretation is that the WebRTC Data channel
>>> draft has become both the base for this specification as well as the =
one
>>> that specifies that the DCEP shall be implemented and supported. For
>>> WebRTC I don't think this matters much, but if someone likes to =
re-use
>>> the basic Data Channel specification, this structure makes it more
>>> difficult and have no need for the bi-directional negotiated =
parameter
>>> settings that the DCEP provides. In that case one have to sub-set =
the
>=20
>> I think the data channel draft discusses the data channels =
independent
>> how they are opened. This covers data parameters, user data =
transmission
>> and closing of data channels. The data channel protocol draft =
discusses
>> an in-band protocol for setting up data channels.
>=20
> I think this was discussed today. And make it clear that DCEP
> requirement shouldn't be from the Data Channel spec.
>=20
>>> data channel specification. I wonder if it would be better to move =
some
>>> normative statements around, so that the chain of implementation
>>> requirement goes draft-ietf-rtcweb-transports ->
>>> draft-ietf-rtcweb-data-protocol -> draft-ietf-rtcweb-data-channel =
thus
>>> providing a cleaner and more modular build up of the protocols.
>> I think you can implement draft-ietf-rtcweb-data-channel without
>> draft-ietf-rtcweb-data-protocol, but not vice versa.
>>=20
>> So you don't like the statement:
>>=20
>>   A simple protocol for internal negotiation is specified in
>>   [I-D.ietf-rtcweb-data-protocol] and MUST be supported.
>>=20
>> from draft-ietf-rtcweb-data-channel. What would you suggest?
>=20
> I think the high level requirement on DCEP might be in =
-rtcweb-transports.
It now reads:
A simple protocol for in-band negotiation is specified in =
[I-D.ietf-rtcweb-data-protocol].
>=20
> Cheers
>=20
> Magnus Westerlund
>=20
> ----------------------------------------------------------------------
> Services, Media and Network features, Ericsson Research EAB/TXM
> ----------------------------------------------------------------------
> Ericsson AB                 | Phone  +46 10 7148287
> F=E4r=F6gatan 6                 | Mobile +46 73 0949079
> SE-164 80 Stockholm, Sweden | mailto: magnus.westerlund@ericsson.com
> ----------------------------------------------------------------------
>=20
>=20


From nobody Thu Apr 10 04:55:33 2014
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Subject: Re: [rtcweb] I-D Action: draft-ietf-rtcweb-data-channel-08.txt
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On 04/09/2014 04:39 PM, Colin Perkins wrote:
> On 9 Apr 2014, at 15:33, Michael Tuexen <Michael.Tuexen@lurchi.franken.de> wrote:
>> On 09 Apr 2014, at 16:25, Colin Perkins <csp@csperkins.org> wrote:
>>> On 9 Apr 2014, at 15:20, Michael Tuexen <Michael.Tuexen@lurchi.franken.de> wrote:
>>>> On 09 Apr 2014, at 13:00, Colin Perkins <csp@csperkins.org> wrote:
>>>>> On 9 Apr 2014, at 11:02, Internet-Drafts@ietf.org wrote:
>>>>>> A New Internet-Draft is available from the on-line Internet-Drafts directories.
>>>>>> This draft is a work item of the Real-Time Communication in WEB-browsers Working Group of the IETF.
>>>>>>
>>>>>>     Title           : WebRTC Data Channels
>>>>>>     Authors         : Randell Jesup
>>>>>>                       Salvatore Loreto
>>>>>>                       Michael Tuexen
>>>>>> 	Filename        : draft-ietf-rtcweb-data-channel-08.txt
>>>>>> 	Pages           : 15
>>>>>> 	Date            : 2014-04-09
>>>>>>
>>>>>> Abstract:
>>>>>> The Real-Time Communication in WEB-browsers 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 specifies the non-(S)RTP media data transport aspects of the
>>>>>> WebRTC framework.  It provides an architectural overview of how the
>>>>>> Stream Control Transmission Protocol (SCTP) is used in the WebRTC
>>>>>> context as a generic transport service allowing WEB-browsers to
>>>>>> exchange generic data from peer to peer.
>>>>> This talks about “(S)RTP” throughout, but the rtp-usage draft requires that SRTP be used for WebRTC, and disallows plain RTP. I think this draft could be simplified by changing “(S)RTP” to “SRTP” throughout.
>>>> Hi Colin,
>>>>
>>>> The (S)RTP notion goes back to a comment from Magnus. If I remember it correctly he considers SRTP a profile of RTP. Since I don’t wanted to just use RTP, I ended up with (S)RTP based on a discussion with Magnus.
>>>>
>>>> However, I’m fine with changing it to SRTP...
>>> SRTP is an RTP profile. My comment was that if this is for WebRTC only, then  only SRTP can be used, and not plain RTP. Using “(S)RTP” rather than “SRTP” in this draft suggests that the secure profile is optional, which isn’t the case in WebRTC. If this is for more general use than WebRTC, then “(S)RTP” is fine.
>> It is clear that in WebRTC only SRTP is used...
> I don’t believe your draft is clear on that, due to the use of “(S)RTP” terminology. That’s why I commented.
>
Enforcing the SRTP-only rule for WebRTC is not the business of this 
draft, so it's not a normative statement in any way, shape or form.

If it's possible to use these channels outside of WebRTC, and some of 
these places claim to be able to use pure RTP with appropriate security, 
then making use of "SRTP" only in this draft is confusing.

I see arguments both ways, and want to leave it at editors' discretion.





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From: Christer Holmberg <christer.holmberg@ericsson.com>
To: Harald Alvestrand <harald@alvestrand.no>, "rtcweb@ietf.org" <rtcweb@ietf.org>
Thread-Topic: [rtcweb] I-D Action: draft-ietf-rtcweb-data-channel-08.txt
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References: <20140409100258.9712.74771.idtracker@ietfa.amsl.com> <F09BCD44-1060-4DCB-A796-7A31F1C634DE@csperkins.org> <A05F0177-568C-4B19-AD48-9F415A4C008B@lurchi.franken.de> <02F2BCF4-70B5-47A4-ACE6-C0CCCAB11A50@csperkins.org> <9889BAD9-D9A7-42F2-A0DC-632C26696345@lurchi.franken.de> <73C21E05-E36C-4899-98A4-9D762B75FCE4@csperkins.org> <534686AA.9040304@alvestrand.no>
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Subject: Re: [rtcweb] I-D Action: draft-ietf-rtcweb-data-channel-08.txt
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Hi,

In BUNDLE we use "RTP-based media" terminology, eventhough BUNDLE will be u=
sed in WebRTC environments where SRTP is mandated.

Why can't we use the same here?

Regards,

Christer

-----Original Message-----
From: rtcweb [mailto:rtcweb-bounces@ietf.org] On Behalf Of Harald Alvestran=
d
Sent: 10. huhtikuuta 2014 14:55
To: rtcweb@ietf.org
Subject: Re: [rtcweb] I-D Action: draft-ietf-rtcweb-data-channel-08.txt

On 04/09/2014 04:39 PM, Colin Perkins wrote:
> On 9 Apr 2014, at 15:33, Michael Tuexen <Michael.Tuexen@lurchi.franken.de=
> wrote:
>> On 09 Apr 2014, at 16:25, Colin Perkins <csp@csperkins.org> wrote:
>>> On 9 Apr 2014, at 15:20, Michael Tuexen <Michael.Tuexen@lurchi.franken.=
de> wrote:
>>>> On 09 Apr 2014, at 13:00, Colin Perkins <csp@csperkins.org> wrote:
>>>>> On 9 Apr 2014, at 11:02, Internet-Drafts@ietf.org wrote:
>>>>>> A New Internet-Draft is available from the on-line Internet-Drafts d=
irectories.
>>>>>> This draft is a work item of the Real-Time Communication in WEB-brow=
sers Working Group of the IETF.
>>>>>>
>>>>>>     Title           : WebRTC Data Channels
>>>>>>     Authors         : Randell Jesup
>>>>>>                       Salvatore Loreto
>>>>>>                       Michael Tuexen
>>>>>> 	Filename        : draft-ietf-rtcweb-data-channel-08.txt
>>>>>> 	Pages           : 15
>>>>>> 	Date            : 2014-04-09
>>>>>>
>>>>>> Abstract:
>>>>>> The Real-Time Communication in WEB-browsers working group is=20
>>>>>> charged to provide protocol support for direct interactive rich=20
>>>>>> communication using audio, video, and data between two peers'=20
>>>>>> web-browsers.  This document specifies the non-(S)RTP media data=20
>>>>>> transport aspects of the WebRTC framework.  It provides an=20
>>>>>> architectural overview of how the Stream Control Transmission=20
>>>>>> Protocol (SCTP) is used in the WebRTC context as a generic=20
>>>>>> transport service allowing WEB-browsers to exchange generic data fro=
m peer to peer.
>>>>> This talks about "(S)RTP" throughout, but the rtp-usage draft require=
s that SRTP be used for WebRTC, and disallows plain RTP. I think this draft=
 could be simplified by changing "(S)RTP" to "SRTP" throughout.
>>>> Hi Colin,
>>>>
>>>> The (S)RTP notion goes back to a comment from Magnus. If I remember it=
 correctly he considers SRTP a profile of RTP. Since I don't wanted to just=
 use RTP, I ended up with (S)RTP based on a discussion with Magnus.
>>>>
>>>> However, I'm fine with changing it to SRTP...
>>> SRTP is an RTP profile. My comment was that if this is for WebRTC only,=
 then  only SRTP can be used, and not plain RTP. Using "(S)RTP" rather than=
 "SRTP" in this draft suggests that the secure profile is optional, which i=
sn't the case in WebRTC. If this is for more general use than WebRTC, then =
"(S)RTP" is fine.
>> It is clear that in WebRTC only SRTP is used...
> I don't believe your draft is clear on that, due to the use of "(S)RTP" t=
erminology. That's why I commented.
>
Enforcing the SRTP-only rule for WebRTC is not the business of this draft, =
so it's not a normative statement in any way, shape or form.

If it's possible to use these channels outside of WebRTC, and some of these=
 places claim to be able to use pure RTP with appropriate security, then ma=
king use of "SRTP" only in this draft is confusing.

I see arguments both ways, and want to leave it at editors' discretion.




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


From nobody Thu Apr 10 06:14:08 2014
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Subject: Re: [rtcweb] Some language on "prioritization"
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Just checking... since there's been no comment to the main body of the 
text here (it's all been on the SCTP aspect), is it OK to include the 
text in my next version of the -transport draft?

(I'm not responding to Karl's comments - it's at a completely different 
level of complexity.)

        Harald

On 03/31/2014 07:08 PM, Harald Alvestrand wrote:
> I see that I have promised to sugggest language for -transport- about
> prioritization.
> Here's an attempt. References and so on to be filled in later.
>
> Comments welcome - this is a first stab!
>
> --------------------------------------------
> The RTCWEB prioritization model is that the application tells the RTCWEB
> implementation about the priority of media and data flows through an API.
>
> The priority associated with a media or data flow is classified as
> "normal", "below normal", "high" or "very high". There are only four
> priority levels at the API.
>
> The RTCWEB implementation is responsible for mapping these to protocol
> mechanics at the protocol interfaces, and to behave appropriately when
> choosing what packets to send when.
>
> [draft-dhesikan] specifies the mapping of the four levels of priority,
> combined with the media type, to DSCP markings. This marking SHOULD be
> done; the implementation MAY turn off use of DSCP markings if it detects
> symptoms of unexpected behaviour like priority inversion or blocking of
> packets with certain DSCP markings. The detection of these conditions is
> implementation dependent. (Question: Does there need to be an API knob
> to turn off DSCP markings?)
>
> When an RTCWEB implementation has packets to send on multiple streams
> that are congestion-controlled under the same congestion controller, the
> RTCWEB implementation SHOULD serve the streams in a weighted round-robin
> fashion, with each level of priority being given twice the transmission
> capacity of the level below; thus, when congestion occurs, a "very high"
> priority flow will have the ability to send 8 times as much data as a
> "below normal" flow if both have data to send. This prioritization is
> independent of the media type, but will lead to packet loss due to full
> send buffers occuring first on the highest volume flows at any given
> priority level. The details of which packet to send first are
> implementation defined.
>
> -- TODO: Specify a relative priority scheme that makes sense with SCTP,
> with an appropriate reference. [draft-ietf-tsvwg-sctp-prpolicies]
> specifies a priority policy, but it's about discarding packets, not
> deciding which packets to send first, and it also makes it impossible to
> specify time-bounded retransmission. --
>


From nobody Thu Apr 10 07:47:45 2014
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On 4/9/14 5:25 PM, Martin Thomson wrote:
> On 9 April 2014 13:56, Paul Kyzivat <pkyzivat@alum.mit.edu> wrote:
>> I expect two different protocols to look different on the wire.
>>
>> You seem to be saying that SMTP used to talk to ietf mailing lists is a
>> different protocol from SMTP used to talk to my lawyer, because I expect my
>> lawyer to keep the communications confidential.
>
> They are different on the wire.  They use different identifiers.
>
> That's hair splitting, but there's a real difference between the two
> usages.  And I think that it's important enough to do this.
>
> Do you perhaps have an alternative, or is it just that this lack of
> solidity is giving you heartburn?  Because I can appreciate that.

Heartburn.

It seems like a hack - tunneling - just looking for *something* that can 
be used to convey one more bit of data.

I don't have another suggestion. I only half understand the problem. But 
to the extent that I do, it seems ill-defined. AFAIK it only really has 
meaning in the context of a browser. Once you start signaling it, it 
gets fuzzier. If you *knew* both ends were controlled by a browser 
following these conventions then it would be a bit clearer. But when 
that ceases to be true I can make no sense of what it all means. And as 
best I understand, the isolation is intended to be per-media-stream, so 
solutions that are at a different level seem problematic.

	Thanks,
	Paul


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On 4/10/14 4:56 AM, Harald Alvestrand wrote:
> On 04/08/2014 08:24 PM, Martin Thomson wrote:
>> On 8 April 2014 09:50, Bernard Aboba <bernard.aboba@gmail.com> wrote:
>>> [BA] I'm not sure that the concept of "isolation" makes sense for those
>>> intermediaries (or to voicemail or an audio/video conference, for that
>>> matter).   While in a point-to-point call it might be useful, in a
>>> conference the whole point is to have audio/video sent to multiple
>>> parties,
>>> and recording is commonplace.  The problem is that from a protocol
>>> point of
>>> view the cases are not easily distinguishable -- and so if the browser
>>> insists on "isolation" then one wonders what will happen if the
>>> conference
>>> bridge/video MCU/voicemail system refuses to negotiate it.   Refusing to
>>> send media would not be a desirable outcome.
>> I think that for this, it's perfectly reasonable to use identity, but
>> not stream isolation.  With isolation, if the peer does not agree to
>> comply, then the session fails to complete.
> Actually I'd say it's "if the peer does not *agree to* comply".
> The protocol has no defense against liars, but that's a common issue.
>>
>> The authenticated party here is an MCU (or bridge, or voicemail,
>> etc...).  Rather than sending to "anindividual@example.org", media is
>> sent to "mcu@example.com".  Is it reasonable for that MCU to forward
>> media to other, unspecified entities?  Clearly it can, but should it?
>>
>> (Not having thought it through completely, a voicemail box could
>> conceivably work.  I think that I'd want to use a different identity
>> for it though.)
>
> I can see an use for a recording spec that said "you can record this,
> but only if you do it in such a way that it's only accessible to the
> stated identity".
>
> Would be weird to try to enforce that.... but I agree; MCU and isolation
> have a hard time mixing.
> Let's just not.

So what does this mean for screen sharing with a conference?

	Thanks,
	Paul


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Subject: Re: [rtcweb] Some language on "prioritization"
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On 4/3/14 12:30 PM, Karl Stahl wrote:

> (B) Each rtcweb file transfer MUST use its own data channel (if
> combined they risk being regulated randomly/unfairly by SCTP itself
> and all files will have just have one single TCP-like flow at the
> network congestion point when sharing the available bandwidth with
> everyone else's flows at that congestion point - That would be
> "negative priority" compared to what we are used to in a browser.).

I presume you mean each *concurrent* file transfer must use its own 
channel. Right? Reusing the same channel to transfer a series of files 
should be fine.

	Thanks,
	Paul


From nobody Thu Apr 10 08:34:32 2014
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To: "'Paul Kyzivat'" <pkyzivat@alum.mit.edu>, <rtcweb@ietf.org>
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Subject: Re: [rtcweb] Some language on "prioritization"
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-----Ursprungligt meddelande-----
Fr=E5n: rtcweb [mailto:rtcweb-bounces@ietf.org] F=F6r Paul Kyzivat
Skickat: den 10 april 2014 17:03
Till: rtcweb@ietf.org
=C4mne: Re: [rtcweb] Some language on "prioritization"

On 4/3/14 12:30 PM, Karl Stahl wrote:

> (B) Each rtcweb file transfer MUST use its own data channel (if=20
> combined they risk being regulated randomly/unfairly by SCTP itself=20
> and all files will have just have one single TCP-like flow at the=20
> network congestion point when sharing the available bandwidth with=20
> everyone else's flows at that congestion point - That would be=20
> "negative priority" compared to what we are used to in a browser.).

I presume you mean each *concurrent* file transfer must use its own =
channel.
Right? Reusing the same channel to transfer a series of files should be
fine.

	Thanks,
	Paul
Yes, That is right. /Karl
_______________________________________________
rtcweb mailing list
rtcweb@ietf.org
https://www.ietf.org/mailman/listinfo/rtcweb


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Subject: Re: [rtcweb] Asking TLS for help with media isolation
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On 10 April 2014 01:56, Harald Alvestrand <harald@alvestrand.no> wrote:
>> I think that for this, it's perfectly reasonable to use identity, but
>> not stream isolation.  With isolation, if the peer does not agree to
>> comply, then the session fails to complete.
>
> Actually I'd say it's "if the peer does not *agree to* comply".
> The protocol has no defense against liars, but that's a common issue.

That's precisely what I said, sans emphasis :)


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On 10 April 2014 07:51, Paul Kyzivat <pkyzivat@alum.mit.edu> wrote:
> So what does this mean for screen sharing with a conference?

It means the same thing, if the stream is isolated, it will not be
possible to share with a conference MCU.

(You can go full mesh to retain good isolation, but that requires
multiple differently isolated streams, and it's likely to annoy the
user with gUM prompts.)


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On 4/10/14 3:33 PM, Martin Thomson wrote:
> On 10 April 2014 07:51, Paul Kyzivat <pkyzivat@alum.mit.edu> wrote:
>> So what does this mean for screen sharing with a conference?
>
> It means the same thing, if the stream is isolated, it will not be
> possible to share with a conference MCU.
>
> (You can go full mesh to retain good isolation, but that requires
> multiple differently isolated streams, and it's likely to annoy the
> user with gUM prompts.)

Well, apparently it will be at the discretion of the conference focus. 
Since it presumably doesn't have a browser, it will only be governed by 
its own policies and whatever normative statements are made about the 
behavior of servers in this case.

There will be a business motivation to allow such sharing if a rationale 
can be made that it is allowed.

So that puts a premium on clear normative statements.

	Thanks,
	Paul


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

        Title           : STUN Usage for Consent Freshness
        Authors         : Muthu Arul Mozhi Perumal
                          Dan Wing
                          Ram Mohan Ravindranath
                          Tirumaleswar Reddy
                          Martin Thomson
	Filename        : draft-ietf-rtcweb-stun-consent-freshness-02.txt
	Pages           : 8
	Date            : 2014-04-10

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

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


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

There's also a htmlized version available at:
http://tools.ietf.org/html/draft-ietf-rtcweb-stun-consent-freshness-02

A diff from the previous version is available at:
http://www.ietf.org/rfcdiff?url2=draft-ietf-rtcweb-stun-consent-freshness-02


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

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


From nobody Thu Apr 10 22:00:50 2014
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Summary of changes in this revision

Addressed the comments received from the WG.
Removed the timers definition from solution overview and made the text
more generic.
Incorporated text from draft-thomson-rtcweb-consent.
Most of the text of solution overview is re-written however the idea is
still kept intact


Comments are welcome.

There is still some dangling reference (text) to SRTP/DTLS mechanism for
consent which we will modify in the next revision


Regards,
Authors

-----Original Message-----
From: "internet-drafts@ietf.org" <internet-drafts@ietf.org>
Date: Friday, 11 April 2014 9:07 am
To: "i-d-announce@ietf.org" <i-d-announce@ietf.org>
Cc: "rtcweb@ietf.org" <rtcweb@ietf.org>
Subject: [rtcweb] I-D Action:
draft-ietf-rtcweb-stun-consent-freshness-02.txt

>
>A New Internet-Draft is available from the on-line Internet-Drafts
>directories.
> This draft is a work item of the Real-Time Communication in WEB-browsers
>Working Group of the IETF.
>
>        Title           : STUN Usage for Consent Freshness
>        Authors         : Muthu Arul Mozhi Perumal
>                          Dan Wing
>                          Ram Mohan Ravindranath
>                          Tirumaleswar Reddy
>                          Martin Thomson
>	Filename        : draft-ietf-rtcweb-stun-consent-freshness-02.txt
>	Pages           : 8
>	Date            : 2014-04-10
>
>Abstract:
>   To prevent sending excessive traffic to an endpoint, periodic consent
>   needs to be obtained from that remote endpoint.
>
>   This document describes a consent mechanism using a new STUN usage.
>   This same mechanism can also determine connection loss ("liveness")
>   with a remote peer.
>
>
>The IETF datatracker status page for this draft is:
>https://datatracker.ietf.org/doc/draft-ietf-rtcweb-stun-consent-freshness/
>
>There's also a htmlized version available at:
>http://tools.ietf.org/html/draft-ietf-rtcweb-stun-consent-freshness-02
>
>A diff from the previous version is available at:
>http://www.ietf.org/rfcdiff?url2=3Ddraft-ietf-rtcweb-stun-consent-freshnes=
s-
>02
>
>
>Please note that it may take a couple of minutes from the time of
>submission
>until the htmlized version and diff are available at tools.ietf.org.
>
>Internet-Drafts are also available by anonymous FTP at:
>ftp://ftp.ietf.org/internet-drafts/
>
>_______________________________________________
>rtcweb mailing list
>rtcweb@ietf.org
>https://www.ietf.org/mailman/listinfo/rtcweb


From nobody Thu Apr 10 23:52:14 2014
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From: Christer Holmberg <christer.holmberg@ericsson.com>
To: "Ram Mohan R (rmohanr)" <rmohanr@cisco.com>, "rtcweb@ietf.org" <rtcweb@ietf.org>
Thread-Topic: [rtcweb] I-D Action: draft-ietf-rtcweb-stun-consent-freshness-02.txt
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Subject: Re: [rtcweb] I-D Action: draft-ietf-rtcweb-stun-consent-freshness-02.txt
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Hi,

I think it would be good to have some text about usage of consent freshness=
 when one entity is ICE lite.

And, I think it would be good to make it more clear that the usage of conse=
nt is always negotiated per direction.

Thanks!

Regards,

Christer


-----Original Message-----
From: rtcweb [mailto:rtcweb-bounces@ietf.org] On Behalf Of Ram Mohan R (rmo=
hanr)
Sent: 11. huhtikuuta 2014 8:01
To: rtcweb@ietf.org
Subject: Re: [rtcweb] I-D Action: draft-ietf-rtcweb-stun-consent-freshness-=
02.txt

Summary of changes in this revision

Addressed the comments received from the WG.
Removed the timers definition from solution overview and made the text more=
 generic.
Incorporated text from draft-thomson-rtcweb-consent.
Most of the text of solution overview is re-written however the idea is sti=
ll kept intact


Comments are welcome.

There is still some dangling reference (text) to SRTP/DTLS mechanism for co=
nsent which we will modify in the next revision


Regards,
Authors

-----Original Message-----
From: "internet-drafts@ietf.org" <internet-drafts@ietf.org>
Date: Friday, 11 April 2014 9:07 am
To: "i-d-announce@ietf.org" <i-d-announce@ietf.org>
Cc: "rtcweb@ietf.org" <rtcweb@ietf.org>
Subject: [rtcweb] I-D Action:
draft-ietf-rtcweb-stun-consent-freshness-02.txt

>
>A New Internet-Draft is available from the on-line Internet-Drafts=20
>directories.
> This draft is a work item of the Real-Time Communication in=20
>WEB-browsers Working Group of the IETF.
>
>        Title           : STUN Usage for Consent Freshness
>        Authors         : Muthu Arul Mozhi Perumal
>                          Dan Wing
>                          Ram Mohan Ravindranath
>                          Tirumaleswar Reddy
>                          Martin Thomson
>	Filename        : draft-ietf-rtcweb-stun-consent-freshness-02.txt
>	Pages           : 8
>	Date            : 2014-04-10
>
>Abstract:
>   To prevent sending excessive traffic to an endpoint, periodic consent
>   needs to be obtained from that remote endpoint.
>
>   This document describes a consent mechanism using a new STUN usage.
>   This same mechanism can also determine connection loss ("liveness")
>   with a remote peer.
>
>
>The IETF datatracker status page for this draft is:
>https://datatracker.ietf.org/doc/draft-ietf-rtcweb-stun-consent-freshne
>ss/
>
>There's also a htmlized version available at:
>http://tools.ietf.org/html/draft-ietf-rtcweb-stun-consent-freshness-02
>
>A diff from the previous version is available at:
>http://www.ietf.org/rfcdiff?url2=3Ddraft-ietf-rtcweb-stun-consent-freshne
>ss-
>02
>
>
>Please note that it may take a couple of minutes from the time of=20
>submission until the htmlized version and diff are available at=20
>tools.ietf.org.
>
>Internet-Drafts are also available by anonymous FTP at:
>ftp://ftp.ietf.org/internet-drafts/
>
>_______________________________________________
>rtcweb mailing list
>rtcweb@ietf.org
>https://www.ietf.org/mailman/listinfo/rtcweb

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


From nobody Thu Apr 10 23:54:09 2014
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From: "Ram Mohan R (rmohanr)" <rmohanr@cisco.com>
To: Christer Holmberg <christer.holmberg@ericsson.com>, "rtcweb@ietf.org" <rtcweb@ietf.org>
Thread-Topic: [rtcweb] I-D Action: draft-ietf-rtcweb-stun-consent-freshness-02.txt
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References: <20140411033753.19230.46577.idtracker@ietfa.amsl.com> <CF6D6F0C.878CF%rmohanr@cisco.com> <7594FB04B1934943A5C02806D1A2204B1D2BD7C3@ESESSMB209.ericsson.se>
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Sure. We will add text for these two in the next revision.

Thanks,
Ram

-----Original Message-----
From: Christer Holmberg <christer.holmberg@ericsson.com>
Date: Friday, 11 April 2014 12:22 pm
To: Ram Mohan Ravindranath <rmohanr@cisco.com>, "rtcweb@ietf.org"
<rtcweb@ietf.org>
Subject: RE: [rtcweb] I-D Action:
draft-ietf-rtcweb-stun-consent-freshness-02.txt

>Hi,
>
>I think it would be good to have some text about usage of consent
>freshness when one entity is ICE lite.
>
>And, I think it would be good to make it more clear that the usage of
>consent is always negotiated per direction.
>
>Thanks!
>
>Regards,
>
>Christer
>
>
>-----Original Message-----
>From: rtcweb [mailto:rtcweb-bounces@ietf.org] On Behalf Of Ram Mohan R
>(rmohanr)
>Sent: 11. huhtikuuta 2014 8:01
>To: rtcweb@ietf.org
>Subject: Re: [rtcweb] I-D Action:
>draft-ietf-rtcweb-stun-consent-freshness-02.txt
>
>Summary of changes in this revision
>
>Addressed the comments received from the WG.
>Removed the timers definition from solution overview and made the text
>more generic.
>Incorporated text from draft-thomson-rtcweb-consent.
>Most of the text of solution overview is re-written however the idea is
>still kept intact
>
>
>Comments are welcome.
>
>There is still some dangling reference (text) to SRTP/DTLS mechanism for
>consent which we will modify in the next revision
>
>
>Regards,
>Authors
>
>-----Original Message-----
>From: "internet-drafts@ietf.org" <internet-drafts@ietf.org>
>Date: Friday, 11 April 2014 9:07 am
>To: "i-d-announce@ietf.org" <i-d-announce@ietf.org>
>Cc: "rtcweb@ietf.org" <rtcweb@ietf.org>
>Subject: [rtcweb] I-D Action:
>draft-ietf-rtcweb-stun-consent-freshness-02.txt
>
>>
>>A New Internet-Draft is available from the on-line Internet-Drafts
>>directories.
>> This draft is a work item of the Real-Time Communication in
>>WEB-browsers Working Group of the IETF.
>>
>>        Title           : STUN Usage for Consent Freshness
>>        Authors         : Muthu Arul Mozhi Perumal
>>                          Dan Wing
>>                          Ram Mohan Ravindranath
>>                          Tirumaleswar Reddy
>>                          Martin Thomson
>>	Filename        : draft-ietf-rtcweb-stun-consent-freshness-02.txt
>>	Pages           : 8
>>	Date            : 2014-04-10
>>
>>Abstract:
>>   To prevent sending excessive traffic to an endpoint, periodic consent
>>   needs to be obtained from that remote endpoint.
>>
>>   This document describes a consent mechanism using a new STUN usage.
>>   This same mechanism can also determine connection loss ("liveness")
>>   with a remote peer.
>>
>>
>>The IETF datatracker status page for this draft is:
>>https://datatracker.ietf.org/doc/draft-ietf-rtcweb-stun-consent-freshne
>>ss/
>>
>>There's also a htmlized version available at:
>>http://tools.ietf.org/html/draft-ietf-rtcweb-stun-consent-freshness-02
>>
>>A diff from the previous version is available at:
>>http://www.ietf.org/rfcdiff?url2=3Ddraft-ietf-rtcweb-stun-consent-freshne
>>ss-
>>02
>>
>>
>>Please note that it may take a couple of minutes from the time of
>>submission until the htmlized version and diff are available at
>>tools.ietf.org.
>>
>>Internet-Drafts are also available by anonymous FTP at:
>>ftp://ftp.ietf.org/internet-drafts/
>>
>>_______________________________________________
>>rtcweb mailing list
>>rtcweb@ietf.org
>>https://www.ietf.org/mailman/listinfo/rtcweb
>
>_______________________________________________
>rtcweb mailing list
>rtcweb@ietf.org
>https://www.ietf.org/mailman/listinfo/rtcweb


From nobody Thu Apr 10 23:58:23 2014
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Subject: Re: [rtcweb] I-D Action: draft-ietf-rtcweb-data-channel-08.txt
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On 10 Apr 2014, at 13:55, Harald Alvestrand <harald@alvestrand.no> =
wrote:

> On 04/09/2014 04:39 PM, Colin Perkins wrote:
>> On 9 Apr 2014, at 15:33, Michael Tuexen =
<Michael.Tuexen@lurchi.franken.de> wrote:
>>> On 09 Apr 2014, at 16:25, Colin Perkins <csp@csperkins.org> wrote:
>>>> On 9 Apr 2014, at 15:20, Michael Tuexen =
<Michael.Tuexen@lurchi.franken.de> wrote:
>>>>> On 09 Apr 2014, at 13:00, Colin Perkins <csp@csperkins.org> wrote:
>>>>>> On 9 Apr 2014, at 11:02, Internet-Drafts@ietf.org wrote:
>>>>>>> A New Internet-Draft is available from the on-line =
Internet-Drafts directories.
>>>>>>> This draft is a work item of the Real-Time Communication in =
WEB-browsers Working Group of the IETF.
>>>>>>>=20
>>>>>>>    Title           : WebRTC Data Channels
>>>>>>>    Authors         : Randell Jesup
>>>>>>>                      Salvatore Loreto
>>>>>>>                      Michael Tuexen
>>>>>>> 	Filename        : draft-ietf-rtcweb-data-channel-08.txt
>>>>>>> 	Pages           : 15
>>>>>>> 	Date            : 2014-04-09
>>>>>>>=20
>>>>>>> Abstract:
>>>>>>> The Real-Time Communication in WEB-browsers 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 specifies the non-(S)RTP media data transport aspects =
of the
>>>>>>> WebRTC framework.  It provides an architectural overview of how =
the
>>>>>>> Stream Control Transmission Protocol (SCTP) is used in the =
WebRTC
>>>>>>> context as a generic transport service allowing WEB-browsers to
>>>>>>> exchange generic data from peer to peer.
>>>>>> This talks about =93(S)RTP=94 throughout, but the rtp-usage draft =
requires that SRTP be used for WebRTC, and disallows plain RTP. I think =
this draft could be simplified by changing =93(S)RTP=94 to =93SRTP=94 =
throughout.
>>>>> Hi Colin,
>>>>>=20
>>>>> The (S)RTP notion goes back to a comment from Magnus. If I =
remember it correctly he considers SRTP a profile of RTP. Since I don=92t =
wanted to just use RTP, I ended up with (S)RTP based on a discussion =
with Magnus.
>>>>>=20
>>>>> However, I=92m fine with changing it to SRTP...
>>>> SRTP is an RTP profile. My comment was that if this is for WebRTC =
only, then  only SRTP can be used, and not plain RTP. Using =93(S)RTP=94 =
rather than =93SRTP=94 in this draft suggests that the secure profile is =
optional, which isn=92t the case in WebRTC. If this is for more general =
use than WebRTC, then =93(S)RTP=94 is fine.
>>> It is clear that in WebRTC only SRTP is used...
>> I don=92t believe your draft is clear on that, due to the use of =
=93(S)RTP=94 terminology. That=92s why I commented.
>>=20
> Enforcing the SRTP-only rule for WebRTC is not the business of this =
draft, so it's not a normative statement in any way, shape or form.
>=20
> If it's possible to use these channels outside of WebRTC, and some of =
these places claim to be able to use pure RTP with appropriate security, =
then making use of "SRTP" only in this draft is confusing.
>=20
> I see arguments both ways, and want to leave it at editors' =
discretion.
At least I don't have a strong position. So will edit the document as it =
is decided by
the WG... (the current version in git uses SRTP, but I'm willing to =
change that back)
Maybe my co-editors do have a strong position?

Best regards
Michael
>=20
>=20
>=20
>=20
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb
>=20


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Subject: Re: [rtcweb] Some language on "prioritization"
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On 10 Apr 2014, at 15:13, Harald Alvestrand <harald@alvestrand.no> =
wrote:

> Just checking... since there's been no comment to the main body of the =
text here (it's all been on the SCTP aspect), is it OK to include the =
text in my next version of the -transport draft?
>=20
> (I'm not responding to Karl's comments - it's at a completely =
different level of complexity.)
>=20
>       Harald
>=20
> On 03/31/2014 07:08 PM, Harald Alvestrand wrote:
>> I see that I have promised to sugggest language for -transport- about
>> prioritization.
>> Here's an attempt. References and so on to be filled in later.
>>=20
>> Comments welcome - this is a first stab!
>>=20
>> --------------------------------------------
>> The RTCWEB prioritization model is that the application tells the =
RTCWEB
>> implementation about the priority of media and data flows through an =
API.
>>=20
>> The priority associated with a media or data flow is classified as
>> "normal", "below normal", "high" or "very high". There are only four
>> priority levels at the API.
>>=20
>> The RTCWEB implementation is responsible for mapping these to =
protocol
>> mechanics at the protocol interfaces, and to behave appropriately =
when
>> choosing what packets to send when.
>>=20
>> [draft-dhesikan] specifies the mapping of the four levels of =
priority,
>> combined with the media type, to DSCP markings. This marking SHOULD =
be
>> done; the implementation MAY turn off use of DSCP markings if it =
detects
>> symptoms of unexpected behaviour like priority inversion or blocking =
of
>> packets with certain DSCP markings. The detection of these conditions =
is
>> implementation dependent. (Question: Does there need to be an API =
knob
>> to turn off DSCP markings?)
>>=20
>> When an RTCWEB implementation has packets to send on multiple streams
>> that are congestion-controlled under the same congestion controller, =
the
>> RTCWEB implementation SHOULD serve the streams in a weighted =
round-robin
>> fashion, with each level of priority being given twice the =
transmission
>> capacity of the level below; thus, when congestion occurs, a "very =
high"
>> priority flow will have the ability to send 8 times as much data as a
>> "below normal" flow if both have data to send. This prioritization is
>> independent of the media type, but will lead to packet loss due to =
full
>> send buffers occuring first on the highest volume flows at any given
>> priority level. The details of which packet to send first are
>> implementation defined.
The text reads good to me.
>>=20
>> -- TODO: Specify a relative priority scheme that makes sense with =
SCTP,
>> with an appropriate reference. [draft-ietf-tsvwg-sctp-prpolicies]
>> specifies a priority policy, but it's about discarding packets, not
>> deciding which packets to send first, and it also makes it impossible =
to
>> specify time-bounded retransmission. --
No sure about this TODO. The priorities in =
draft-ietf-tsvwg-sctp-prpolicies are
not related to the kind of priorities you describe above. Your above =
apply
also to reliable channels.

Best regards
Michael
>>=20
>=20
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb
>=20


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Subject: Re: [rtcweb] Some language on "prioritization"
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On 04/11/2014 09:01 AM, Michael Tuexen wrote:
> On 10 Apr 2014, at 15:13, Harald Alvestrand <harald@alvestrand.no> wrote:
>
>> Just checking... since there's been no comment to the main body of the text here (it's all been on the SCTP aspect), is it OK to include the text in my next version of the -transport draft?
>>
>> (I'm not responding to Karl's comments - it's at a completely different level of complexity.)
>>
>>       Harald
>>
>> On 03/31/2014 07:08 PM, Harald Alvestrand wrote:
>>> I see that I have promised to sugggest language for -transport- about
>>> prioritization.
>>> Here's an attempt. References and so on to be filled in later.
>>>
>>> Comments welcome - this is a first stab!
>>>
>>> --------------------------------------------
>>> The RTCWEB prioritization model is that the application tells the RTCWEB
>>> implementation about the priority of media and data flows through an API.
>>>
>>> The priority associated with a media or data flow is classified as
>>> "normal", "below normal", "high" or "very high". There are only four
>>> priority levels at the API.
>>>
>>> The RTCWEB implementation is responsible for mapping these to protocol
>>> mechanics at the protocol interfaces, and to behave appropriately when
>>> choosing what packets to send when.
>>>
>>> [draft-dhesikan] specifies the mapping of the four levels of priority,
>>> combined with the media type, to DSCP markings. This marking SHOULD be
>>> done; the implementation MAY turn off use of DSCP markings if it detects
>>> symptoms of unexpected behaviour like priority inversion or blocking of
>>> packets with certain DSCP markings. The detection of these conditions is
>>> implementation dependent. (Question: Does there need to be an API knob
>>> to turn off DSCP markings?)
>>>
>>> When an RTCWEB implementation has packets to send on multiple streams
>>> that are congestion-controlled under the same congestion controller, the
>>> RTCWEB implementation SHOULD serve the streams in a weighted round-robin
>>> fashion, with each level of priority being given twice the transmission
>>> capacity of the level below; thus, when congestion occurs, a "very high"
>>> priority flow will have the ability to send 8 times as much data as a
>>> "below normal" flow if both have data to send. This prioritization is
>>> independent of the media type, but will lead to packet loss due to full
>>> send buffers occuring first on the highest volume flows at any given
>>> priority level. The details of which packet to send first are
>>> implementation defined.
> The text reads good to me.
>>> -- TODO: Specify a relative priority scheme that makes sense with SCTP,
>>> with an appropriate reference. [draft-ietf-tsvwg-sctp-prpolicies]
>>> specifies a priority policy, but it's about discarding packets, not
>>> deciding which packets to send first, and it also makes it impossible to
>>> specify time-bounded retransmission. --
> No sure about this TODO. The priorities in draft-ietf-tsvwg-sctp-prpolicies are
> not related to the kind of priorities you describe above. Your above apply
> also to reliable channels.

We've already settled that none of the current policies in -ndata do
what the WG wants (weighted round robin), so I think it's correct to
leave this as a TODO, and punt this to TSVWG.

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


-- 
Surveillance is pervasive. Go Dark.


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From: "Ram Mohan R (rmohanr)" <rmohanr@cisco.com>
To: "rtcweb@ietf.org" <rtcweb@ietf.org>, Oleg Moskalenko <mom040267@gmail.com>
Thread-Topic: [rtcweb] Review of draft-ietf-rtcweb-stun-consent-freshness
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Subject: Re: [rtcweb] Review of draft-ietf-rtcweb-stun-consent-freshness
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Hi Ted,

Thanks for forwarding the feedback from Oleg. Please see inline for answers


>
>
>From:  Ted Hardie <ted.ietf@gmail.com>
>Date:  Monday, 7 April 2014 11:50 pm
>To:  "rtcweb@ietf.org" <rtcweb@ietf.org>, Oleg Moskalenko
><mom040267@gmail.com>
>Subject:  [rtcweb] Review of draft-ietf-rtcweb-stun-consent-freshness
>
>
>Howdy,
>
>
>The chairs recently asked for a review
>draft-ietf-rtcweb-stun-consent-freshness; Oleg was kind enough to do one.
>Below is the review.
>
>regards,
>
>Ted
>
>
>
>On Fri, Apr 4, 2014 at 12:40 AM, Oleg Moskalenko <mom040267@gmail.com>
>wrote:
>Hi Ted
>
>
>I went through the document and I have two things to comment:
>
>
>1) This document defines a "voluntary" pattern of the browser behavior.
>Nothing stops the determined attacker from taking the WebRTC code and
>creating a malicious client application that ignores all proposed
>connectivity checks. May be it is worth mentioning
> in the "Security Considerations" section.

Agree. A malicious browser that does not conform to this spec can do any
thing. I am not sure if we really any text to be added for that in the
draft.

>
>
>2) I have a feeling that the document is written with somewhat optimistic
>idea about the modern IP network qualities. The proposed timeouts are
>probably too small. I am hearing from our TURN server users that in modern
>Wi Fi public networks that's common to
> observe a freeze the IP traffic for several seconds. After that "freeze"
>the connectivity is restored. The users do not want the connection to be
>broken during that time - they want the video screen frozen, temporary. I
>had to make adjustments to the TURN
> server in our recent versions so that it does not disconnects the
>sessions too quickly under those conditions (when TCP is used). I have a
>feeling that you may have the same complains that the browser stops
>transmission in public Wi Fi networks too quickly.
> I'd suggest to review the wording of the proposal (like re-transmission
>after 500 ms and 15 secs timeout) to make it more tolerant for the bad IP
>networks (which are surprisingly are rather common).

Agree. We had revised a lot of text in the solution description section of
the draft (draft-ietf-rtcweb-stun-consent-freshness-02).
The text now is very generic and does not assume any specific timer
values. Please review the latest text and let us know if that looks ok.

Regards,
Ram

>
>
>Overall, I think that this proposal is very useful.
>
>
>Best regards,
>Oleg
>


From nobody Fri Apr 11 03:49:47 2014
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To: "'Harald Alvestrand'" <harald@alvestrand.no>, <rtcweb@ietf.org>
References: <20140411033753.19230.46577.idtracker@ietfa.amsl.com> <CF6D6F0C.878CF%rmohanr@cisco.com> <7594FB04B1934943A5C02806D1A2204B1D2BD7C3@ESESSMB209.ericsson.se>
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Harald,

Just remembered regarding ICE Lite in your transport document; I think =
your
statement regarding ICE, not ICE Lite therein, should be clarified so it =
is
understood that browsers must (also) operate against ICE Lite devices
(typical for a gateway).

(That is the way it already works in Chrome - buggy in Firefox though)
=20
/Karl

-----Ursprungligt meddelande-----
Fr=E5n: rtcweb [mailto:rtcweb-bounces@ietf.org] F=F6r Christer Holmberg
Skickat: den 11 april 2014 08:52
Till: Ram Mohan R (rmohanr); rtcweb@ietf.org
=C4mne: Re: [rtcweb] I-D Action:
draft-ietf-rtcweb-stun-consent-freshness-02.txt

Hi,

I think it would be good to have some text about usage of consent =
freshness
when one entity is ICE lite.

And, I think it would be good to make it more clear that the usage of
consent is always negotiated per direction.

Thanks!

Regards,

Christer


-----Original Message-----
From: rtcweb [mailto:rtcweb-bounces@ietf.org] On Behalf Of Ram Mohan R
(rmohanr)
Sent: 11. huhtikuuta 2014 8:01
To: rtcweb@ietf.org
Subject: Re: [rtcweb] I-D Action:
draft-ietf-rtcweb-stun-consent-freshness-02.txt

Summary of changes in this revision

Addressed the comments received from the WG.
Removed the timers definition from solution overview and made the text =
more
generic.
Incorporated text from draft-thomson-rtcweb-consent.
Most of the text of solution overview is re-written however the idea is
still kept intact


Comments are welcome.

There is still some dangling reference (text) to SRTP/DTLS mechanism for
consent which we will modify in the next revision


Regards,
Authors

-----Original Message-----
From: "internet-drafts@ietf.org" <internet-drafts@ietf.org>
Date: Friday, 11 April 2014 9:07 am
To: "i-d-announce@ietf.org" <i-d-announce@ietf.org>
Cc: "rtcweb@ietf.org" <rtcweb@ietf.org>
Subject: [rtcweb] I-D Action:
draft-ietf-rtcweb-stun-consent-freshness-02.txt

>
>A New Internet-Draft is available from the on-line Internet-Drafts=20
>directories.
> This draft is a work item of the Real-Time Communication in=20
>WEB-browsers Working Group of the IETF.
>
>        Title           : STUN Usage for Consent Freshness
>        Authors         : Muthu Arul Mozhi Perumal
>                          Dan Wing
>                          Ram Mohan Ravindranath
>                          Tirumaleswar Reddy
>                          Martin Thomson
>	Filename        : draft-ietf-rtcweb-stun-consent-freshness-02.txt
>	Pages           : 8
>	Date            : 2014-04-10
>
>Abstract:
>   To prevent sending excessive traffic to an endpoint, periodic =
consent
>   needs to be obtained from that remote endpoint.
>
>   This document describes a consent mechanism using a new STUN usage.
>   This same mechanism can also determine connection loss ("liveness")
>   with a remote peer.
>
>
>The IETF datatracker status page for this draft is:
>https://datatracker.ietf.org/doc/draft-ietf-rtcweb-stun-consent-freshne
>ss/
>
>There's also a htmlized version available at:
>http://tools.ietf.org/html/draft-ietf-rtcweb-stun-consent-freshness-02
>
>A diff from the previous version is available at:
>http://www.ietf.org/rfcdiff?url2=3Ddraft-ietf-rtcweb-stun-consent-freshn=
e
>ss-
>02
>
>
>Please note that it may take a couple of minutes from the time of=20
>submission until the htmlized version and diff are available at=20
>tools.ietf.org.
>
>Internet-Drafts are also available by anonymous FTP at:
>ftp://ftp.ietf.org/internet-drafts/
>
>_______________________________________________
>rtcweb mailing list
>rtcweb@ietf.org
>https://www.ietf.org/mailman/listinfo/rtcweb

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

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


From nobody Fri Apr 11 03:55:04 2014
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On 04/11/2014 12:49 PM, Karl Stahl wrote:
> Harald,
>
> Just remembered regarding ICE Lite in your transport document; I think your
> statement regarding ICE, not ICE Lite therein, should be clarified so it is
> understood that browsers must (also) operate against ICE Lite devices
> (typical for a gateway).

Huh?

As far as I understand ICE-Lite, any full ICE implementation will
interoperate with an ICE-Lite implementation without any special handling.
Thus, any ICE implementation that doesn't do so has a bug.

I don't want to put in the document a statement that says "MUST
implement ICE and MUST NOT have bugs that prevent interoperability with
ICE-Lite". That's just not helpful.

>
> (That is the way it already works in Chrome - buggy in Firefox though)
>  
> /Karl
>
> -----Ursprungligt meddelande-----
> Från: rtcweb [mailto:rtcweb-bounces@ietf.org] För Christer Holmberg
> Skickat: den 11 april 2014 08:52
> Till: Ram Mohan R (rmohanr); rtcweb@ietf.org
> Ämne: Re: [rtcweb] I-D Action:
> draft-ietf-rtcweb-stun-consent-freshness-02.txt
>
> Hi,
>
> I think it would be good to have some text about usage of consent freshness
> when one entity is ICE lite.
>
> And, I think it would be good to make it more clear that the usage of
> consent is always negotiated per direction.
>
> Thanks!
>
> Regards,
>
> Christer
>
>
> -----Original Message-----
> From: rtcweb [mailto:rtcweb-bounces@ietf.org] On Behalf Of Ram Mohan R
> (rmohanr)
> Sent: 11. huhtikuuta 2014 8:01
> To: rtcweb@ietf.org
> Subject: Re: [rtcweb] I-D Action:
> draft-ietf-rtcweb-stun-consent-freshness-02.txt
>
> Summary of changes in this revision
>
> Addressed the comments received from the WG.
> Removed the timers definition from solution overview and made the text more
> generic.
> Incorporated text from draft-thomson-rtcweb-consent.
> Most of the text of solution overview is re-written however the idea is
> still kept intact
>
>
> Comments are welcome.
>
> There is still some dangling reference (text) to SRTP/DTLS mechanism for
> consent which we will modify in the next revision
>
>
> Regards,
> Authors
>
> -----Original Message-----
> From: "internet-drafts@ietf.org" <internet-drafts@ietf.org>
> Date: Friday, 11 April 2014 9:07 am
> To: "i-d-announce@ietf.org" <i-d-announce@ietf.org>
> Cc: "rtcweb@ietf.org" <rtcweb@ietf.org>
> Subject: [rtcweb] I-D Action:
> draft-ietf-rtcweb-stun-consent-freshness-02.txt
>
>> A New Internet-Draft is available from the on-line Internet-Drafts 
>> directories.
>> This draft is a work item of the Real-Time Communication in 
>> WEB-browsers Working Group of the IETF.
>>
>>        Title           : STUN Usage for Consent Freshness
>>        Authors         : Muthu Arul Mozhi Perumal
>>                          Dan Wing
>>                          Ram Mohan Ravindranath
>>                          Tirumaleswar Reddy
>>                          Martin Thomson
>> 	Filename        : draft-ietf-rtcweb-stun-consent-freshness-02.txt
>> 	Pages           : 8
>> 	Date            : 2014-04-10
>>
>> Abstract:
>>   To prevent sending excessive traffic to an endpoint, periodic consent
>>   needs to be obtained from that remote endpoint.
>>
>>   This document describes a consent mechanism using a new STUN usage.
>>   This same mechanism can also determine connection loss ("liveness")
>>   with a remote peer.
>>
>>
>> The IETF datatracker status page for this draft is:
>> https://datatracker.ietf.org/doc/draft-ietf-rtcweb-stun-consent-freshne
>> ss/
>>
>> There's also a htmlized version available at:
>> http://tools.ietf.org/html/draft-ietf-rtcweb-stun-consent-freshness-02
>>
>> A diff from the previous version is available at:
>> http://www.ietf.org/rfcdiff?url2=draft-ietf-rtcweb-stun-consent-freshne
>> ss-
>> 02
>>
>>
>> Please note that it may take a couple of minutes from the time of 
>> submission until the htmlized version and diff are available at 
>> tools.ietf.org.
>>
>> Internet-Drafts are also available by anonymous FTP at:
>> ftp://ftp.ietf.org/internet-drafts/
>>
>> _______________________________________________
>> rtcweb mailing list
>> rtcweb@ietf.org
>> https://www.ietf.org/mailman/listinfo/rtcweb
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb
>
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb
>


-- 
Surveillance is pervasive. Go Dark.


From nobody Fri Apr 11 10:42:50 2014
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--047d7b33c8fa9409f804f6c7dc68
Content-Type: text/plain; charset=ISO-8859-1

I have reviewed this document in preparation for IETF LC.  Overall, it
looks good to me; I have requested LC.

Thanks,
--Richard

--047d7b33c8fa9409f804f6c7dc68
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<div dir=3D"ltr"><div>I have reviewed this document in preparation for IETF=
 LC. =A0Overall, it looks good to me; I have requested LC. =A0</div><div><b=
r></div><div>Thanks,</div><div>--Richard</div><div><br></div></div>

--047d7b33c8fa9409f804f6c7dc68--


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From: "Ram Mohan R (rmohanr)" <rmohanr@cisco.com>
To: Christer Holmberg <christer.holmberg@ericsson.com>, "rtcweb@ietf.org" <rtcweb@ietf.org>
Thread-Topic: [rtcweb] I-D Action: draft-ietf-rtcweb-stun-consent-freshness-02.txt
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Date: Fri, 11 Apr 2014 17:52:00 +0000
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Subject: Re: [rtcweb] I-D Action: draft-ietf-rtcweb-stun-consent-freshness-02.txt
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Found this mail thread in the archives where this issue was discussed but
we have not reached to any conclusion -
https://www.ietf.org/mail-archive/web/rtcweb/current/msg04887.html

With ICE-lite mode since the ICE-lite endpoint does not typically generate
any binding requests, it may not generate STUN consent as well. This would
means a malicious application (running on a ICE-lite endpoint) can use it
for sending unwanted traffic.

Should we mandate an ICE-lite implementation to generate a STUN consent
request to its peer before it can send media ?

Regards,
Ram

-----Original Message-----
From: Ram Mohan Ravindranath <rmohanr@cisco.com>
Date: Friday, 11 April 2014 12:23 pm
To: Christer Holmberg <christer.holmberg@ericsson.com>, "rtcweb@ietf.org"
<rtcweb@ietf.org>
Subject: Re: [rtcweb] I-D Action:
draft-ietf-rtcweb-stun-consent-freshness-02.txt

>Sure. We will add text for these two in the next revision.
>
>Thanks,
>Ram
>
>-----Original Message-----
>From: Christer Holmberg <christer.holmberg@ericsson.com>
>Date: Friday, 11 April 2014 12:22 pm
>To: Ram Mohan Ravindranath <rmohanr@cisco.com>, "rtcweb@ietf.org"
><rtcweb@ietf.org>
>Subject: RE: [rtcweb] I-D Action:
>draft-ietf-rtcweb-stun-consent-freshness-02.txt
>
>>Hi,
>>
>>I think it would be good to have some text about usage of consent
>>freshness when one entity is ICE lite.
>>
>>And, I think it would be good to make it more clear that the usage of
>>consent is always negotiated per direction.
>>
>>Thanks!
>>
>>Regards,
>>
>>Christer
>>
>>
>>-----Original Message-----
>>From: rtcweb [mailto:rtcweb-bounces@ietf.org] On Behalf Of Ram Mohan R
>>(rmohanr)
>>Sent: 11. huhtikuuta 2014 8:01
>>To: rtcweb@ietf.org
>>Subject: Re: [rtcweb] I-D Action:
>>draft-ietf-rtcweb-stun-consent-freshness-02.txt
>>
>>Summary of changes in this revision
>>
>>Addressed the comments received from the WG.
>>Removed the timers definition from solution overview and made the text
>>more generic.
>>Incorporated text from draft-thomson-rtcweb-consent.
>>Most of the text of solution overview is re-written however the idea is
>>still kept intact
>>
>>
>>Comments are welcome.
>>
>>There is still some dangling reference (text) to SRTP/DTLS mechanism for
>>consent which we will modify in the next revision
>>
>>
>>Regards,
>>Authors
>>
>>-----Original Message-----
>>From: "internet-drafts@ietf.org" <internet-drafts@ietf.org>
>>Date: Friday, 11 April 2014 9:07 am
>>To: "i-d-announce@ietf.org" <i-d-announce@ietf.org>
>>Cc: "rtcweb@ietf.org" <rtcweb@ietf.org>
>>Subject: [rtcweb] I-D Action:
>>draft-ietf-rtcweb-stun-consent-freshness-02.txt
>>
>>>
>>>A New Internet-Draft is available from the on-line Internet-Drafts
>>>directories.
>>> This draft is a work item of the Real-Time Communication in
>>>WEB-browsers Working Group of the IETF.
>>>
>>>        Title           : STUN Usage for Consent Freshness
>>>        Authors         : Muthu Arul Mozhi Perumal
>>>                          Dan Wing
>>>                          Ram Mohan Ravindranath
>>>                          Tirumaleswar Reddy
>>>                          Martin Thomson
>>>	Filename        : draft-ietf-rtcweb-stun-consent-freshness-02.txt
>>>	Pages           : 8
>>>	Date            : 2014-04-10
>>>
>>>Abstract:
>>>   To prevent sending excessive traffic to an endpoint, periodic consent
>>>   needs to be obtained from that remote endpoint.
>>>
>>>   This document describes a consent mechanism using a new STUN usage.
>>>   This same mechanism can also determine connection loss ("liveness")
>>>   with a remote peer.
>>>
>>>
>>>The IETF datatracker status page for this draft is:
>>>https://datatracker.ietf.org/doc/draft-ietf-rtcweb-stun-consent-freshne
>>>ss/
>>>
>>>There's also a htmlized version available at:
>>>http://tools.ietf.org/html/draft-ietf-rtcweb-stun-consent-freshness-02
>>>
>>>A diff from the previous version is available at:
>>>http://www.ietf.org/rfcdiff?url2=3Ddraft-ietf-rtcweb-stun-consent-freshn=
e
>>>ss-
>>>02
>>>
>>>
>>>Please note that it may take a couple of minutes from the time of
>>>submission until the htmlized version and diff are available at
>>>tools.ietf.org.
>>>
>>>Internet-Drafts are also available by anonymous FTP at:
>>>ftp://ftp.ietf.org/internet-drafts/
>>>
>>>_______________________________________________
>>>rtcweb mailing list
>>>rtcweb@ietf.org
>>>https://www.ietf.org/mailman/listinfo/rtcweb
>>
>>_______________________________________________
>>rtcweb mailing list
>>rtcweb@ietf.org
>>https://www.ietf.org/mailman/listinfo/rtcweb
>


From nobody Fri Apr 11 11:23:27 2014
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From: Martin Thomson <martin.thomson@gmail.com>
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On 11 April 2014 10:52, Ram Mohan R (rmohanr) <rmohanr@cisco.com> wrote:
> With ICE-lite mode since the ICE-lite endpoint does not typically generate
> any binding requests, it may not generate STUN consent as well.

Wat?

Consent is generated by responding to connectivity checks.  An
ICE-lite endpoint has to do that.


From nobody Fri Apr 11 13:13:44 2014
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The IESG has received a request from the Real-Time Communication in
WEB-browsers WG (rtcweb) to consider the following document:
- 'Web Real-Time Communication Use-cases and Requirements'
  <draft-ietf-rtcweb-use-cases-and-requirements-14.txt> as Informational
RFC

The IESG plans to make a decision in the next few weeks, and solicits
final comments on this action. Please send substantive comments to the
ietf@ietf.org mailing lists by 2014-04-25. Exceptionally, comments may be
sent to iesg@ietf.org instead. In either case, please retain the
beginning of the Subject line to allow automated sorting.

Abstract


   This document describes web based real-time communication use-cases.
   Requirements on the browser functionality are derived from the use-
   cases.




The file can be obtained via
http://datatracker.ietf.org/doc/draft-ietf-rtcweb-use-cases-and-requirements/

IESG discussion can be tracked via
http://datatracker.ietf.org/doc/draft-ietf-rtcweb-use-cases-and-requirements/ballot/


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



From nobody Fri Apr 11 13:38:05 2014
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From nobody Fri Apr 11 16:06:21 2014
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Subject: Re: [rtcweb] I-D Action: draft-ietf-rtcweb-stun-consent-freshness-02.txt
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-----Ursprungligt meddelande-----
Fr=E5n: Harald Alvestrand [mailto:harald@alvestrand.no]=20
Skickat: den 11 april 2014 12:55
Till: Karl Stahl; rtcweb@ietf.org
=C4mne: Re: SV: [rtcweb] I-D Action:
draft-ietf-rtcweb-stun-consent-freshness-02.txt

On 04/11/2014 12:49 PM, Karl Stahl wrote:
> Harald,
>
> Just remembered regarding ICE Lite in your transport document; I think =

> your statement regarding ICE, not ICE Lite therein, should be=20
> clarified so it is understood that browsers must (also) operate=20
> against ICE Lite devices (typical for a gateway).

Huh?

As far as I understand ICE-Lite, any full ICE implementation will
interoperate with an ICE-Lite implementation without any special =
handling.
Thus, any ICE implementation that doesn't do so has a bug.

--- Agree, that is correct.

I don't want to put in the document a statement that says "MUST =
implement
ICE and MUST NOT have bugs that prevent interoperability with ICE-Lite".
That's just not helpful.

--- I am sure you can express it in a way that WOULD be helpful :) (I =
myself
had to ask our developers to get that understanding, so, just a point to
make it clearer and avoid doubts - we are not all experts on the details =
of
all specs.)=20

/Karl


>
> (That is the way it already works in Chrome - buggy in Firefox though)
> =20
> /Karl
>
> -----Ursprungligt meddelande-----
> Fr=E5n: rtcweb [mailto:rtcweb-bounces@ietf.org] F=F6r Christer =
Holmberg
> Skickat: den 11 april 2014 08:52
> Till: Ram Mohan R (rmohanr); rtcweb@ietf.org
> =C4mne: Re: [rtcweb] I-D Action:
> draft-ietf-rtcweb-stun-consent-freshness-02.txt
>
> Hi,
>
> I think it would be good to have some text about usage of consent=20
> freshness when one entity is ICE lite.
>
> And, I think it would be good to make it more clear that the usage of=20
> consent is always negotiated per direction.
>
> Thanks!
>
> Regards,
>
> Christer
>
>
> -----Original Message-----
> From: rtcweb [mailto:rtcweb-bounces@ietf.org] On Behalf Of Ram Mohan R
> (rmohanr)
> Sent: 11. huhtikuuta 2014 8:01
> To: rtcweb@ietf.org
> Subject: Re: [rtcweb] I-D Action:
> draft-ietf-rtcweb-stun-consent-freshness-02.txt
>
> Summary of changes in this revision
>
> Addressed the comments received from the WG.
> Removed the timers definition from solution overview and made the text =

> more generic.
> Incorporated text from draft-thomson-rtcweb-consent.
> Most of the text of solution overview is re-written however the idea=20
> is still kept intact
>
>
> Comments are welcome.
>
> There is still some dangling reference (text) to SRTP/DTLS mechanism=20
> for consent which we will modify in the next revision
>
>
> Regards,
> Authors
>
> -----Original Message-----
> From: "internet-drafts@ietf.org" <internet-drafts@ietf.org>
> Date: Friday, 11 April 2014 9:07 am
> To: "i-d-announce@ietf.org" <i-d-announce@ietf.org>
> Cc: "rtcweb@ietf.org" <rtcweb@ietf.org>
> Subject: [rtcweb] I-D Action:
> draft-ietf-rtcweb-stun-consent-freshness-02.txt
>
>> A New Internet-Draft is available from the on-line Internet-Drafts=20
>> directories.
>> This draft is a work item of the Real-Time Communication in=20
>> WEB-browsers Working Group of the IETF.
>>
>>        Title           : STUN Usage for Consent Freshness
>>        Authors         : Muthu Arul Mozhi Perumal
>>                          Dan Wing
>>                          Ram Mohan Ravindranath
>>                          Tirumaleswar Reddy
>>                          Martin Thomson
>> 	Filename        : draft-ietf-rtcweb-stun-consent-freshness-02.txt
>> 	Pages           : 8
>> 	Date            : 2014-04-10
>>
>> Abstract:
>>   To prevent sending excessive traffic to an endpoint, periodic =
consent
>>   needs to be obtained from that remote endpoint.
>>
>>   This document describes a consent mechanism using a new STUN usage.
>>   This same mechanism can also determine connection loss ("liveness")
>>   with a remote peer.
>>
>>
>> The IETF datatracker status page for this draft is:
>> https://datatracker.ietf.org/doc/draft-ietf-rtcweb-stun-consent-fresh
>> ne
>> ss/
>>
>> There's also a htmlized version available at:
>> http://tools.ietf.org/html/draft-ietf-rtcweb-stun-consent-freshness-0
>> 2
>>
>> A diff from the previous version is available at:
>> =
http://www.ietf.org/rfcdiff?url2=3Ddraft-ietf-rtcweb-stun-consent-fresh
>> ne
>> ss-
>> 02
>>
>>
>> Please note that it may take a couple of minutes from the time of=20
>> submission until the htmlized version and diff are available at=20
>> tools.ietf.org.
>>
>> Internet-Drafts are also available by anonymous FTP at:
>> ftp://ftp.ietf.org/internet-drafts/
>>
>> _______________________________________________
>> rtcweb mailing list
>> rtcweb@ietf.org
>> https://www.ietf.org/mailman/listinfo/rtcweb
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb
>
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb
>


--
Surveillance is pervasive. Go Dark.


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Seems we have another reason why running a browser on ice-lite is a bad idea. Good that we do not allow that.

On 11. april 2014 22:37:55 CEST, Christer Holmberg <christer.holmberg@ericsson.com> wrote:
>Hi,
>
>>> With ICE-lite mode since the ICE-lite endpoint does not typically 
>>> generate any binding requests, it may not generate STUN consent as
>well.
>>
>>Wat?
>>
>>Consent is generated by responding to connectivity checks.  An
>ICE-lite endpoint has to do that.
>
>Sure, but I guess what is meant is that the ICE-lite endpoint does not
>generate STUN consent REQUESTS.
>
>Regards,
>
>Christer
>_______________________________________________
>rtcweb mailing list
>rtcweb@ietf.org
>https://www.ietf.org/mailman/listinfo/rtcweb

-- 
Sent from my Android device with K-9 Mail. Please excuse my brevity.
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<html><head></head><body>Seems we have another reason why running a browser on ice-lite is a bad idea. Good that we do not allow that.<br><br><div class="gmail_quote">On 11. april 2014 22:37:55 CEST, Christer Holmberg &lt;christer.holmberg@ericsson.com&gt; wrote:<blockquote class="gmail_quote" style="margin: 0pt 0pt 0pt 0.8ex; border-left: 1px solid rgb(204, 204, 204); padding-left: 1ex;">
<pre class="k9mail">Hi,<br /><br /><blockquote class="gmail_quote" style="margin: 0pt 0pt 1ex 0.8ex; border-left: 1px solid #729fcf; padding-left: 1ex;"><blockquote class="gmail_quote" style="margin: 0pt 0pt 1ex 0.8ex; border-left: 1px solid #ad7fa8; padding-left: 1ex;"> With ICE-lite mode since the ICE-lite endpoint does not typically <br /> generate any binding requests, it may not generate STUN consent as well.<br /></blockquote><br />Wat?<br /><br />Consent is generated by responding to connectivity checks.  An ICE-lite endpoint has to do that.<br /></blockquote><br />Sure, but I guess what is meant is that the ICE-lite endpoint does not generate STUN consent REQUESTS.<br /><br />Regards,<br /><br />Christer<br /><hr /><br />rtcweb mailing list<br />rtcweb@ietf.org<br /><a href="https://www.ietf.org/mailman/listinfo/rtcweb">https://www.ietf.org/mailman/listinfo/rtcweb</a><br /></pre></blockquote></div><br>
-- <br>
Sent from my Android device with K-9 Mail. Please excuse my brevity.</body></html>
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From: "Karl Stahl" <karl.stahl@intertex.se>
To: "'Christer Holmberg'" <christer.holmberg@ericsson.com>, "'Harald Alvestrand'" <harald@alvestrand.no>, "'Martin Thomson'" <martin.thomson@gmail.com>, "'Ram Mohan R \(rmohanr\)'" <rmohanr@cisco.com>
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Date: Sat, 12 Apr 2014 11:57:10 +0200
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Subject: Re: [rtcweb] ICE-Lite, Was: I-D Action: draft-ietf-rtcweb-stun-consent-freshness-02.txt
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To avoid confusion=E2=80=A6

=20

Gateways are not mentioned in draft-ietf-rtcweb-transports-03

=20

Under 3.5.  Middle box related functions=20

It too simply says:

ICE [RFC5245] MUST be supported.  The implementation MUST be a full

   ICE implementation, not ICE-Lite.

=20

It needs to be said that gateways, i.e. devices having a media port on a =
public IP address =E2=80=93 never behind a NAT/firewall =E2=80=93 SHOULD =
implement ICE-Lite. (which browsers implementing full ICE correctly are =
compatible with =E2=80=93 and has to be =E2=80=93 which we all have the =
same understanding of, I believe)

=20

=20

/Karl

=20

=20

Fr=C3=A5n: rtcweb [mailto:rtcweb-bounces@ietf.org] F=C3=B6r Christer =
Holmberg
Skickat: den 12 april 2014 10:26
Till: Harald Alvestrand; Martin Thomson; Ram Mohan R (rmohanr)
Kopia: rtcweb@ietf.org
=C3=84mne: Re: [rtcweb] I-D Action: =
draft-ietf-rtcweb-stun-consent-freshness-02.txt

=20

Hi,

=20

It=E2=80=99s not about running a browser on ice-lite =E2=80=93 it is =
about a browser being prepared to run with a remote endpoint that is =
running ice-lite (e.g. a gateway). Certain browser implementations have =
had some problems with that, afaik=E2=80=A6

=20

Regards,

=20

Christer

=20

From: Harald Alvestrand [mailto:harald@alvestrand.no]=20
Sent: 12 April 2014 08:59
To: Christer Holmberg; Martin Thomson; Ram Mohan R (rmohanr)
Cc: rtcweb@ietf.org
Subject: Re: [rtcweb] I-D Action: =
draft-ietf-rtcweb-stun-consent-freshness-02.txt

=20

Seems we have another reason why running a browser on ice-lite is a bad =
idea. Good that we do not allow that.

On 11. april 2014 22:37:55 CEST, Christer Holmberg =
<christer.holmberg@ericsson.com> wrote:

Hi,

 With ICE-lite mode since the ICE-lite endpoint does not typically=20
 generate any binding requests, it may not generate STUN consent as =
well.


Wat?

Consent is generated by responding to connectivity checks.  An ICE-lite =
endpoint has to do that.


Sure, but I guess what is meant is that the ICE-lite endpoint does not =
generate STUN consent REQUESTS.

Regards,

Christer


  _____ =20


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


--=20
Sent from my Android device with K-9 Mail. Please excuse my brevity.


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</o:shapelayout></xml><![endif]--></head><body lang=3DSV link=3Dblue =
vlink=3Dpurple><div class=3DWordSection1><p class=3DMsoPlainText><span =
lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'>To=
 avoid confusion=E2=80=A6<o:p></o:p></span></p><p =
class=3DMsoPlainText><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'><o=
:p>&nbsp;</o:p></span></p><p class=3DMsoPlainText><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'>Ga=
teways are not mentioned</span><span lang=3DEN-US =
style=3D'font-family:"Courier New"'> in =
draft-ietf-rtcweb-transports-03<o:p></o:p></span></p><p =
class=3DMsoPlainText><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'><o=
:p>&nbsp;</o:p></span></p><p class=3DMsoPlainText><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'>Un=
der </span><span lang=3DEN-US style=3D'font-family:"Courier =
New"'>3.5.=C2=A0 Middle box related functions <o:p></o:p></span></p><p =
class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'>It=
 too simply says:<o:p></o:p></span></p><p class=3DMsoPlainText><span =
lang=3DEN-US style=3D'font-family:"Courier New"'>ICE [RFC5245] MUST be =
supported.=C2=A0 The implementation MUST be a =
full<o:p></o:p></span></p><p class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-family:"Courier New"'>=C2=A0=C2=A0 ICE implementation, not =
ICE-Lite.</span><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'><o=
:p></o:p></span></p><p class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'><o=
:p>&nbsp;</o:p></span></p><p class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'>It=
 needs to be said that gateways, i.e. devices having a media port on a =
public IP address =E2=80=93 never behind a NAT/firewall =E2=80=93 SHOULD =
implement ICE-Lite. (which browsers implementing full ICE correctly are =
compatible with =E2=80=93 and has to be =E2=80=93 which we all have the =
same understanding of, I believe)<o:p></o:p></span></p><p =
class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'><o=
:p>&nbsp;</o:p></span></p><p class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'><o=
:p>&nbsp;</o:p></span></p><p class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'>/K=
arl<o:p></o:p></span></p><p class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'><o=
:p>&nbsp;</o:p></span></p><p class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'><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"'>Fr=C3=A5n:</=
span></b><span =
style=3D'font-size:10.0pt;font-family:"Tahoma","sans-serif"'> rtcweb =
[mailto:rtcweb-bounces@ietf.org] <b>F=C3=B6r </b>Christer =
Holmberg<br><b>Skickat:</b> den 12 april 2014 10:26<br><b>Till:</b> =
Harald Alvestrand; Martin Thomson; Ram Mohan R =
(rmohanr)<br><b>Kopia:</b> rtcweb@ietf.org<br><b>=C3=84mne:</b> Re: =
[rtcweb] I-D Action: =
draft-ietf-rtcweb-stun-consent-freshness-02.txt<o:p></o:p></span></p></di=
v></div><p class=3DMsoNormal><o:p>&nbsp;</o:p></p><p =
class=3DMsoNormal><span lang=3DEN-GB =
style=3D'font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497=
D'>Hi,<o:p></o:p></span></p><p class=3DMsoNormal><span lang=3DEN-GB =
style=3D'font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497=
D'><o:p>&nbsp;</o:p></span></p><p class=3DMsoNormal><span lang=3DEN-GB =
style=3D'font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497=
D'>It=E2=80=99s not about running a browser on ice-lite =E2=80=93 it is =
about a browser being prepared to run with a remote endpoint that is =
running ice-lite (e.g. a gateway). Certain browser implementations have =
had some problems with that, afaik=E2=80=A6<o:p></o:p></span></p><p =
class=3DMsoNormal><span lang=3DEN-GB =
style=3D'font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497=
D'><o:p>&nbsp;</o:p></span></p><p class=3DMsoNormal><span lang=3DEN-GB =
style=3D'font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497=
D'>Regards,<o:p></o:p></span></p><p class=3DMsoNormal><span lang=3DEN-GB =
style=3D'font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497=
D'><o:p>&nbsp;</o:p></span></p><p class=3DMsoNormal><span lang=3DEN-GB =
style=3D'font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497=
D'>Christer<o:p></o:p></span></p><p class=3DMsoNormal><a =
name=3D"_MailEndCompose"></a><span lang=3DEN-GB =
style=3D'font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497=
D'><o:p>&nbsp;</o:p></span></p><div><div =
style=3D'border:none;border-top:solid #E1E1E1 1.0pt;padding:3.0pt 0cm =
0cm 0cm'><p class=3DMsoNormal><b><span lang=3DEN-US =
style=3D'font-size:11.0pt;font-family:"Calibri","sans-serif"'>From:</span=
></b><span lang=3DEN-US =
style=3D'font-size:11.0pt;font-family:"Calibri","sans-serif"'> Harald =
Alvestrand [<a =
href=3D"mailto:harald@alvestrand.no">mailto:harald@alvestrand.no</a>] =
<br><b>Sent:</b> 12 April 2014 08:59<br><b>To:</b> Christer Holmberg; =
Martin Thomson; Ram Mohan R (rmohanr)<br><b>Cc:</b> <a =
href=3D"mailto:rtcweb@ietf.org">rtcweb@ietf.org</a><br><b>Subject:</b> =
Re: [rtcweb] I-D Action: =
draft-ietf-rtcweb-stun-consent-freshness-02.txt<o:p></o:p></span></p></di=
v></div><p class=3DMsoNormal><span =
lang=3DEN-GB><o:p>&nbsp;</o:p></span></p><p class=3DMsoNormal =
style=3D'margin-bottom:12.0pt'><span lang=3DEN-GB>Seems we have another =
reason why running a browser on ice-lite is a bad idea. Good that we do =
not allow that.<o:p></o:p></span></p><div><p class=3DMsoNormal><span =
lang=3DEN-GB>On 11. april 2014 22:37:55 CEST, Christer Holmberg &lt;<a =
href=3D"mailto:christer.holmberg@ericsson.com">christer.holmberg@ericsson=
.com</a>&gt; wrote:<o:p></o:p></span></p><blockquote =
style=3D'border:none;border-left:solid #CCCCCC 1.0pt;padding:0cm 0cm 0cm =
6.0pt;margin-left:4.8pt;margin-top:5.0pt;margin-right:0cm;margin-bottom:5=
.0pt'><pre style=3D'margin-bottom:12.0pt'><span =
lang=3DEN-GB>Hi,<o:p></o:p></span></pre><blockquote =
style=3D'border:none;border-left:solid #729FCF 1.0pt;padding:0cm 0cm 0cm =
6.0pt;margin-left:4.8pt;margin-top:5.0pt;margin-right:0cm;margin-bottom:6=
.0pt'><blockquote style=3D'border:none;border-left:solid #AD7FA8 =
1.0pt;padding:0cm 0cm 0cm =
6.0pt;margin-left:4.8pt;margin-top:5.0pt;margin-right:0cm;margin-bottom:6=
.0pt'><pre><span lang=3DEN-GB> With ICE-lite mode since the ICE-lite =
endpoint does not typically <br>&nbsp;generate any binding requests, it =
may not generate STUN consent as =
well.<o:p></o:p></span></pre></blockquote><pre><span =
lang=3DEN-GB><br>Wat?<br><br>Consent is generated by responding to =
connectivity checks.&nbsp; An ICE-lite endpoint has to do =
that.<o:p></o:p></span></pre></blockquote><pre><span =
lang=3DEN-GB><br>Sure, but I guess what is meant is that the ICE-lite =
endpoint does not generate STUN consent =
REQUESTS.<br><br>Regards,<br><br>Christer<o:p></o:p></span></pre><pre =
style=3D'text-align:center'><span lang=3DEN-GB><hr size=3D3 =
width=3D"100%" align=3Dcenter></span></pre><pre><span =
lang=3DEN-GB><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">https://www.ietf.or=
g/mailman/listinfo/rtcweb</a><o:p></o:p></span></pre></blockquote></div><=
p class=3DMsoNormal><span lang=3DEN-GB><br>-- <br>Sent from my Android =
device with K-9 Mail. Please excuse my =
brevity.<o:p></o:p></span></p></div></body></html>
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Date: Sat, 12 Apr 2014 14:55:34 +0200
From: Harald Alvestrand <harald@alvestrand.no>
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To: Karl Stahl <karl.stahl@intertex.se>,  'Christer Holmberg' <christer.holmberg@ericsson.com>, 'Martin Thomson' <martin.thomson@gmail.com>,  "'Ram Mohan R (rmohanr)'" <rmohanr@cisco.com>
References: <20140411033753.19230.46577.idtracker@ietfa.amsl.com> <CF6D6F0C.878CF%rmohanr@cisco.com> <7594FB04B1934943A5C02806D1A2204B1D2BD7C3@ESESSMB209.ericsson.se> <CF6D8F50.87A2E%rmohanr@cisco.com> <CF6E24CE.87C6C%rmohanr@cisco.com> <CABkgnnUHXXEyNrRetFMUvpqF5mreHWL4LijvhG+QSQAQxkzHZQ@mail.gmail.com> <7594FB04B1934943A5C02806D1A2204B1D2BF6E5@ESESSMB209.ericsson.se> <eb4ad62a-d30d-4a03-8c28-061cd0105d5f@email.android.com> <7594FB04B1934943A5C02806D1A2204B1D2BFE90@ESESSMB209.ericsson.se> <024c01cf5635$92835650$b78a02f0$@stahl@intertex.se>
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Subject: Re: [rtcweb] ICE-Lite, Was: I-D Action: draft-ietf-rtcweb-stun-consent-freshness-02.txt
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On 04/12/2014 11:57 AM, Karl Stahl wrote:
>
> To avoid confusionâ€¦
>
> Gateways are not mentionedin draft-ietf-rtcweb-transports-03
>
> Under 3.5. Middle box related functions
>
> It too simply says:
>
> ICE [RFC5245] MUST be supported. The implementation MUST be a full
>
>    ICE implementation, not ICE-Lite.
>
> It needs to be said that gateways, i.e. devices having a media port on 
> a public IP address â€“ never behind a NAT/firewall â€“ SHOULD implement 
> ICE-Lite. (which browsers implementing full ICE correctly are 
> compatible with â€“ and has to be â€“ which we all have the same 
> understanding of, I believe)
>

I think we've been over this ground before.

At the moment, the RTCWEB protocol suite documentation deals with 
requirements for browsers, not requirements for gateways.

If someone wishes to start a requirements-for-gateways documents, I 
think they should be free to do that - but I hope we can finish this 
document set before that.


> /Karl
>
> *FrÃ¥n:*rtcweb [mailto:rtcweb-bounces@ietf.org] *FÃ¶r *Christer Holmberg
> *Skickat:* den 12 april 2014 10:26
> *Till:* Harald Alvestrand; Martin Thomson; Ram Mohan R (rmohanr)
> *Kopia:* rtcweb@ietf.org
> *Ã„mne:* Re: [rtcweb] I-D Action: 
> draft-ietf-rtcweb-stun-consent-freshness-02.txt
>
> Hi,
>
> Itâ€™s not about running a browser on ice-lite â€“ it is about a browser 
> being prepared to run with a remote endpoint that is running ice-lite 
> (e.g. a gateway). Certain browser implementations have had some 
> problems with that, afaikâ€¦
>
> Regards,
>
> Christer
>
> *From:*Harald Alvestrand [mailto:harald@alvestrand.no]
> *Sent:* 12 April 2014 08:59
> *To:* Christer Holmberg; Martin Thomson; Ram Mohan R (rmohanr)
> *Cc:* rtcweb@ietf.org <mailto:rtcweb@ietf.org>
> *Subject:* Re: [rtcweb] I-D Action: 
> draft-ietf-rtcweb-stun-consent-freshness-02.txt
>
> Seems we have another reason why running a browser on ice-lite is a 
> bad idea. Good that we do not allow that.
>
> On 11. april 2014 22:37:55 CEST, Christer Holmberg 
> <christer.holmberg@ericsson.com 
> <mailto:christer.holmberg@ericsson.com>> wrote:
>
>     Hi,
>
>               With ICE-lite mode since the ICE-lite endpoint does not typically
>               generate any binding requests, it may not generate STUN consent as well.
>
>
>         Wat?
>
>         Consent is generated by responding to connectivity checks.  An ICE-lite endpoint has to do that.
>
>
>     Sure, but I guess what is meant is that the ICE-lite endpoint does not generate STUN consent REQUESTS.
>
>     Regards,
>
>     Christer
>
>     ------------------------------------------------------------------------
>
>
>     rtcweb mailing list
>     rtcweb@ietf.org  <mailto:rtcweb@ietf.org>
>     https://www.ietf.org/mailman/listinfo/rtcweb
>
>
> -- 
> Sent from my Android device with K-9 Mail. Please excuse my brevity.
>


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    <div class="moz-cite-prefix">On 04/12/2014 11:57 AM, Karl Stahl
      wrote:<br>
    </div>
    <blockquote
      cite="mid:024c01cf5635$92835650$b78a02f0$@stahl@intertex.se"
      type="cite">
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        <p class="MsoPlainText"><span
style="font-size:10.0pt;font-family:&quot;Arial&quot;,&quot;sans-serif&quot;;color:blue"
            lang="EN-US">To avoid confusionâ€¦<o:p></o:p></span></p>
        <p class="MsoPlainText"><span
style="font-size:10.0pt;font-family:&quot;Arial&quot;,&quot;sans-serif&quot;;color:blue"
            lang="EN-US"><o:p>Â </o:p></span></p>
        <p class="MsoPlainText"><span
style="font-size:10.0pt;font-family:&quot;Arial&quot;,&quot;sans-serif&quot;;color:blue"
            lang="EN-US">Gateways are not mentioned</span><span
            style="font-family:&quot;Courier New&quot;" lang="EN-US"> in
            draft-ietf-rtcweb-transports-03<o:p></o:p></span></p>
        <p class="MsoPlainText"><span
style="font-size:10.0pt;font-family:&quot;Arial&quot;,&quot;sans-serif&quot;;color:blue"
            lang="EN-US"><o:p>Â </o:p></span></p>
        <p class="MsoPlainText"><span
style="font-size:10.0pt;font-family:&quot;Arial&quot;,&quot;sans-serif&quot;;color:blue"
            lang="EN-US">Under </span><span
            style="font-family:&quot;Courier New&quot;" lang="EN-US">3.5.Â 
            Middle box related functions <o:p></o:p></span></p>
        <p class="MsoNormal"><span
style="font-size:10.0pt;font-family:&quot;Arial&quot;,&quot;sans-serif&quot;;color:blue"
            lang="EN-US">It too simply says:<o:p></o:p></span></p>
        <p class="MsoPlainText"><span style="font-family:&quot;Courier
            New&quot;" lang="EN-US">ICE [RFC5245] MUST be supported.Â 
            The implementation MUST be a full<o:p></o:p></span></p>
        <p class="MsoNormal"><span style="font-family:&quot;Courier
            New&quot;" lang="EN-US">Â Â  ICE implementation, not ICE-Lite.</span><span
style="font-size:10.0pt;font-family:&quot;Arial&quot;,&quot;sans-serif&quot;;color:blue"
            lang="EN-US"><o:p></o:p></span></p>
        <p class="MsoNormal"><span
style="font-size:10.0pt;font-family:&quot;Arial&quot;,&quot;sans-serif&quot;;color:blue"
            lang="EN-US"><o:p>Â </o:p></span></p>
        <p class="MsoNormal"><span
style="font-size:10.0pt;font-family:&quot;Arial&quot;,&quot;sans-serif&quot;;color:blue"
            lang="EN-US">It needs to be said that gateways, i.e. devices
            having a media port on a public IP address â€“ never behind a
            NAT/firewall â€“ SHOULD implement ICE-Lite. (which browsers
            implementing full ICE correctly are compatible with â€“ and
            has to be â€“ which we all have the same understanding of, I
            believe)</span></p>
      </div>
    </blockquote>
    <br>
    I think we've been over this ground before.<br>
    <br>
    At the moment, the RTCWEB protocol suite documentation deals with
    requirements for browsers, not requirements for gateways.<br>
    <br>
    If someone wishes to start a requirements-for-gateways documents, I
    think they should be free to do that - but I hope we can finish this
    document set before that.<br>
    <br>
    <br>
    <blockquote
      cite="mid:024c01cf5635$92835650$b78a02f0$@stahl@intertex.se"
      type="cite">
      <div class="WordSection1">
        <p class="MsoNormal"><span
style="font-size:10.0pt;font-family:&quot;Arial&quot;,&quot;sans-serif&quot;;color:blue"
            lang="EN-US"><o:p></o:p></span></p>
        <p class="MsoNormal"><span
style="font-size:10.0pt;font-family:&quot;Arial&quot;,&quot;sans-serif&quot;;color:blue"
            lang="EN-US"><o:p>Â </o:p></span></p>
        <p class="MsoNormal"><span
style="font-size:10.0pt;font-family:&quot;Arial&quot;,&quot;sans-serif&quot;;color:blue"
            lang="EN-US"><o:p>Â </o:p></span></p>
        <p class="MsoNormal"><span
style="font-size:10.0pt;font-family:&quot;Arial&quot;,&quot;sans-serif&quot;;color:blue"
            lang="EN-US">/Karl<o:p></o:p></span></p>
        <p class="MsoNormal"><span
style="font-size:10.0pt;font-family:&quot;Arial&quot;,&quot;sans-serif&quot;;color:blue"
            lang="EN-US"><o:p>Â </o:p></span></p>
        <p class="MsoNormal"><span
style="font-size:10.0pt;font-family:&quot;Arial&quot;,&quot;sans-serif&quot;;color:blue"
            lang="EN-US"><o:p>Â </o:p></span></p>
        <div>
          <div style="border:none;border-top:solid #B5C4DF
            1.0pt;padding:3.0pt 0cm 0cm 0cm">
            <p class="MsoNormal"><b><span
style="font-size:10.0pt;font-family:&quot;Tahoma&quot;,&quot;sans-serif&quot;">FrÃ¥n:</span></b><span
style="font-size:10.0pt;font-family:&quot;Tahoma&quot;,&quot;sans-serif&quot;">
                rtcweb [<a class="moz-txt-link-freetext" href="mailto:rtcweb-bounces@ietf.org">mailto:rtcweb-bounces@ietf.org</a>] <b>FÃ¶r </b>Christer
                Holmberg<br>
                <b>Skickat:</b> den 12 april 2014 10:26<br>
                <b>Till:</b> Harald Alvestrand; Martin Thomson; Ram
                Mohan R (rmohanr)<br>
                <b>Kopia:</b> <a class="moz-txt-link-abbreviated" href="mailto:rtcweb@ietf.org">rtcweb@ietf.org</a><br>
                <b>Ã„mne:</b> Re: [rtcweb] I-D Action:
                draft-ietf-rtcweb-stun-consent-freshness-02.txt<o:p></o:p></span></p>
          </div>
        </div>
        <p class="MsoNormal"><o:p>Â </o:p></p>
        <p class="MsoNormal"><span
style="font-size:11.0pt;font-family:&quot;Calibri&quot;,&quot;sans-serif&quot;;color:#1F497D"
            lang="EN-GB">Hi,<o:p></o:p></span></p>
        <p class="MsoNormal"><span
style="font-size:11.0pt;font-family:&quot;Calibri&quot;,&quot;sans-serif&quot;;color:#1F497D"
            lang="EN-GB"><o:p>Â </o:p></span></p>
        <p class="MsoNormal"><span
style="font-size:11.0pt;font-family:&quot;Calibri&quot;,&quot;sans-serif&quot;;color:#1F497D"
            lang="EN-GB">Itâ€™s not about running a browser on ice-lite â€“
            it is about a browser being prepared to run with a remote
            endpoint that is running ice-lite (e.g. a gateway). Certain
            browser implementations have had some problems with that,
            afaikâ€¦<o:p></o:p></span></p>
        <p class="MsoNormal"><span
style="font-size:11.0pt;font-family:&quot;Calibri&quot;,&quot;sans-serif&quot;;color:#1F497D"
            lang="EN-GB"><o:p>Â </o:p></span></p>
        <p class="MsoNormal"><span
style="font-size:11.0pt;font-family:&quot;Calibri&quot;,&quot;sans-serif&quot;;color:#1F497D"
            lang="EN-GB">Regards,<o:p></o:p></span></p>
        <p class="MsoNormal"><span
style="font-size:11.0pt;font-family:&quot;Calibri&quot;,&quot;sans-serif&quot;;color:#1F497D"
            lang="EN-GB"><o:p>Â </o:p></span></p>
        <p class="MsoNormal"><span
style="font-size:11.0pt;font-family:&quot;Calibri&quot;,&quot;sans-serif&quot;;color:#1F497D"
            lang="EN-GB">Christer<o:p></o:p></span></p>
        <p class="MsoNormal"><a moz-do-not-send="true"
            name="_MailEndCompose"></a><span
style="font-size:11.0pt;font-family:&quot;Calibri&quot;,&quot;sans-serif&quot;;color:#1F497D"
            lang="EN-GB"><o:p>Â </o:p></span></p>
        <div>
          <div style="border:none;border-top:solid #E1E1E1
            1.0pt;padding:3.0pt 0cm 0cm 0cm">
            <p class="MsoNormal"><b><span
style="font-size:11.0pt;font-family:&quot;Calibri&quot;,&quot;sans-serif&quot;"
                  lang="EN-US">From:</span></b><span
style="font-size:11.0pt;font-family:&quot;Calibri&quot;,&quot;sans-serif&quot;"
                lang="EN-US"> Harald Alvestrand [<a
                  moz-do-not-send="true"
                  href="mailto:harald@alvestrand.no">mailto:harald@alvestrand.no</a>]
                <br>
                <b>Sent:</b> 12 April 2014 08:59<br>
                <b>To:</b> Christer Holmberg; Martin Thomson; Ram Mohan
                R (rmohanr)<br>
                <b>Cc:</b> <a moz-do-not-send="true"
                  href="mailto:rtcweb@ietf.org">rtcweb@ietf.org</a><br>
                <b>Subject:</b> Re: [rtcweb] I-D Action:
                draft-ietf-rtcweb-stun-consent-freshness-02.txt<o:p></o:p></span></p>
          </div>
        </div>
        <p class="MsoNormal"><span lang="EN-GB"><o:p>Â </o:p></span></p>
        <p class="MsoNormal" style="margin-bottom:12.0pt"><span
            lang="EN-GB">Seems we have another reason why running a
            browser on ice-lite is a bad idea. Good that we do not allow
            that.<o:p></o:p></span></p>
        <div>
          <p class="MsoNormal"><span lang="EN-GB">On 11. april 2014
              22:37:55 CEST, Christer Holmberg &lt;<a
                moz-do-not-send="true"
                href="mailto:christer.holmberg@ericsson.com">christer.holmberg@ericsson.com</a>&gt;
              wrote:<o:p></o:p></span></p>
          <blockquote style="border:none;border-left:solid #CCCCCC
            1.0pt;padding:0cm 0cm 0cm
6.0pt;margin-left:4.8pt;margin-top:5.0pt;margin-right:0cm;margin-bottom:5.0pt">
            <pre style="margin-bottom:12.0pt"><span lang="EN-GB">Hi,<o:p></o:p></span></pre>
            <blockquote style="border:none;border-left:solid #729FCF
              1.0pt;padding:0cm 0cm 0cm
6.0pt;margin-left:4.8pt;margin-top:5.0pt;margin-right:0cm;margin-bottom:6.0pt">
              <blockquote style="border:none;border-left:solid #AD7FA8
                1.0pt;padding:0cm 0cm 0cm
6.0pt;margin-left:4.8pt;margin-top:5.0pt;margin-right:0cm;margin-bottom:6.0pt">
                <pre><span lang="EN-GB"> With ICE-lite mode since the ICE-lite endpoint does not typically 
Â generate any binding requests, it may not generate STUN consent as well.<o:p></o:p></span></pre>
              </blockquote>
              <pre><span lang="EN-GB">
Wat?

Consent is generated by responding to connectivity checks.Â  An ICE-lite endpoint has to do that.<o:p></o:p></span></pre>
            </blockquote>
            <pre><span lang="EN-GB">
Sure, but I guess what is meant is that the ICE-lite endpoint does not generate STUN consent REQUESTS.

Regards,

Christer<o:p></o:p></span></pre>
            <pre style="text-align:center"><span lang="EN-GB"><hr size="3" width="100%" align="center"></span></pre>
            <pre><span lang="EN-GB">
rtcweb mailing list
<a moz-do-not-send="true" href="mailto:rtcweb@ietf.org">rtcweb@ietf.org</a>
<a moz-do-not-send="true" href="https://www.ietf.org/mailman/listinfo/rtcweb">https://www.ietf.org/mailman/listinfo/rtcweb</a><o:p></o:p></span></pre>
          </blockquote>
        </div>
        <p class="MsoNormal"><span lang="EN-GB"><br>
            -- <br>
            Sent from my Android device with K-9 Mail. Please excuse my
            brevity.<o:p></o:p></span></p>
      </div>
    </blockquote>
    <br>
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Archived-At: http://mailarchive.ietf.org/arch/msg/rtcweb/chf5rSfcBqvHEx-ogCmzTy-Mlkk
Subject: Re: [rtcweb] JSEP/WebRTC API Datachannel question
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I'll take a punt that people were thinking about 2 below when data 
channel was added. In order to clarify that I propose to add the 
following text to the JSEP draft to clarify this.

http://tools.ietf.org/html/draft-ietf-rtcweb-jsep-06#section-5.2.1
Replace:
"Lastly, if a data channel has been created, a m= section MUST be
    generated for data. The <media> field MUST be set to "application" 
and the <proto> field MUST be set to "DTLS/SCTP", as specified in 
[I-D.ietf-mmusic-sctp-sdp], Section 3; the "fmt" value MUST be set to 
the SCTP port number, as specified in Section 4.1."

With:
/  "Lastly, if createDataChannel has been called a m= section MUST be 
generated for data. In the case of multiple createDataChannel calls only 
one m= section is generated. The <media> field MUST be set to 
"application" and the <proto> field MUST be set to "DTLS/SCTP", as 
specified in [I-D.ietf-mmusic-sctp-sdp], Section 3; the "fmt" value MUST 
be set to the SCTP port number, as specified in Section 4.1."/

New text in 
http://tools.ietf.org/html/draft-ietf-rtcweb-jsep-06#section-5.2.2:
(above last paragraph)
/ "If createDataChannel has been called and a m= section has not 
previously been generated, a m= section MUST be generated for data. In 
the case of multiple createDataChannel calls only one m= section is 
generated. The <media> field MUST be set to "application" and the 
<proto> field MUST be set to "DTLS/SCTP", as specified in 
[I-D.ietf-mmusic-sctp-sdp], Section 3; the "fmt" value MUST be set to 
the SCTP port number, as specified in Section 4.1.//
//
//If all data channels have been closed, the m= section related to the 
data channels MUST be marked as recvonly by changing the value of the 
[RFC3264] directional attribute to "a=recvonly".//"/

It would also be good to add some text regarding the relationship 
between a createDataChannel and DCEP DATA_CHANNEL_OPEN I'm not sure if 
the JSEP draft is the place?

Also the W3C draft could be updated to indicate the interaction of a 
createDataChannel with the negotiationneeded event in section 5.2 
something along the lines of:
/7. For the first creation of a RTCDataChannel object fire a 
negotiationneeded event at connection./


Regards,
Christian



On 10/04/2014 1:58 PM, Christian Groves wrote:
> Hello,
>
> I've looked into the JSEP draft and WebRTC API (as well as the 
> datachannel and DCEP drafts) to get a better understanding of how and 
> when the SDP for the DTLS/SCTP Web-RTCDataChannel will be generated 
> and what will trigger the DCEP protocol. Unfortunately I find myself 
> more confused after the process .
>
> I was trying to find some text regarding what happens in terms of SDP 
> and DCEP when an endpoint uses multiple datachannels per 
> peerConnection. e.g.
>
> pc = new RTCPeerConnection(configuration);
> channel = pc.createDataChannel("chat");
> channel = pc.createDataChannel("clue");
>
> Now based on the WebRTC API 
> (http://dev.w3.org/2011/webrtc/editor/webrtc.html#peer-to-peer-data-api), 
> I understand that a createOffer() is used to generate the required SDP 
> for the data channel. This is also mentioned in Clause 5.2.1 "initial 
> offers" draft-ietf-rtcweb-jsep-06 indicates: /"Lastly, if a data 
> channel has been created, a m= section MUST be generated for data..."/
>
> The WebRTC API cl.5.1.2 "createDataChannel" step 9 says /"Create 
> channel's associated underlying data transport and configure it 
> according to the relevant properties of channel."/ Now unfortunately 
> it doesn't really say want this amounts to. Create could be "a new 
> SCTP association" or it could be "using DCEP to do a 
> DATA_CHANNEL_OPEN" or both.
>
> If I look at the RTCPeerConnection Interface for addStream I see that 
> it indicates that a negotiationneeded event (WebRTC API cl.4.3.2.3 
> step 5) is fired. This is a trigger for the createOffer(). The example 
> 10.1 WebRTC API shows this behavior. 10.3 also shows this behavior for 
> createDataChannel but the Peer-to-Peer Data API doesn't indicate that 
> the negotiationneeded event is triggered when establishing a data 
> channel.
>
> I looked at the RTCDataChannel closure procedure to see if this would 
> shed any more light but simply says that the underlying data transport 
> may be torn down. The JSEP draft isn't any help here either as it 
> doesn't talk about RTCDataChannel closure at all.
>
> So its not clear to me from the drafts/API whether:
> 1. each createDataChannel results in an offer with new m-line 
> "webrtc-datachannel",
> 2. the first createDataChannel results in an SDP offer with m-line 
> "webrtc-datachannel" and subsequent createDataChannels result in DCEP 
> DATA_CHANNEL_OPENs on the existing webrtc-datachannel.
> 3. a mix of the above an endpoint can indicate multiple m-lines 
> "webrtc-datachannel" and have multiple DCEP DATA_CHANNEL_OPENs per 
> m-line.
>
> Its clear from draft-ietf-rtcweb-data-channel-07 that "Multiple 
> simultaneous data channels MUST be supported in the peer connection" 
> but that doesn't clarify the above.
>
> Is there any specification text that clarifies this?
>
> Regards, Christian
>
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb
>


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Hi Martin and WG,

Colin has helped clarify the text. Thus, I wanted to provide an updated
version of the text to see if this resolves your concerns properly.


Section 4.1 (Part)

   o  Support for multiple synchronisation contexts.  Participants that
      send multiple simultaneous RTP packet streams SHOULD do so as part
      of a single synchronisation context, using a single RTCP CNAME for
      all streams and allowing receivers to play the streams out in a
      synchronised manner.  For compatibility with potential future
      versions of this specification, or for interoperability with non-
      WebRTC devices through a gateway, receivers MUST support multiple
      synchronisation contexts, indicated by the use of multiple RTCP
      CNAMEs in an RTP session.  This specification requires the usage
      of a single CNAME when sending RTP Packet Streams in some
      circumstances, see Section 4.9.

4.9.  Generation of the RTCP Canonical Name (CNAME)

   The RTCP Canonical Name (CNAME) provides a persistent transport-level
   identifier for an RTP end-point.  While the Synchronisation Source
   (SSRC) identifier for an RTP end-point can change if a collision is
   detected, or when the RTP application is restarted, its RTCP CNAME is
   meant to stay unchanged for the duration of a RTCPeerConnection
   [W3C.WD-webrtc-20130910], so that RTP end-points can be uniquely
   identified and associated with their RTP packet streams within a set
   of related RTP sessions.

   Each RTP end-point MUST have at least one RTCP CNAME, and that RTCP
   CNAME MUST be unique within the RTCPeerConnection.  RTCP CNAMEs
   identify a particular synchronisation context, i.e., all SSRCs
   associated with a single RTCP CNAME share a common reference clock.
   If an end-point has SSRCs that are associated with several
   unsynchronised reference clocks, and hence different synchronisation
   contexts, it will need to use multiple RTCP CNAMEs, one for each
   synchronisation context.

   Taking the discussion in Section 11 into account, a WebRTC end-point
   MUST NOT use more than one RTCP CNAME in the RTP sessions belonging
   to single RTCPeerConnection (that is, an RTCPeerConnection forms a
   synchronisation context).  RTP middleboxes MAY generate RTP packet
   streams associated with more than one RTCP CNAME, to allow them to
   avoid having to resynchronize media from multiple different end-
   points part of a multi-party RTP session.

   The RTP specification [RFC3550] includes guidelines for choosing a
   unique RTP CNAME, but these are not sufficient in the presence of NAT
   devices.  In addition, long-term persistent identifiers can be
   problematic from a privacy viewpoint (Section 13).  Accordingly, a
   WebRTC endpoint MUST generate a new, unique, short-term persistent
   RTCP CNAME for each RTCPeerConnection, following [RFC7022].

   An WebRTC end-point MUST support reception of any CNAME that matches
   the syntax limitations specified by the RTP specification [RFC3550]
   and cannot assume that any CNAME will be chosen according to the form
   suggested above.


Section 11 (Part)

   The same MediaStreamTrack can also be included in multiple
   MediaStreams, thus multiple sets of MediaStreams can implicitly need
   to use the same synchronisation base.  To ensure that this works in
   all cases, and don't forces a end-point to change synchronisation
   base and CNAME in the middle of a ongoing delivery of any packet
   streams, which would cause media disruption; all MediaStreamTracks
   and their associated SSRCs originating from the same end-point needs
   to be sent using the same CNAME within one RTCPeerConnection.  This
   is motivating the strong recommendation in Section 4.9 to only use a
   single CNAME.

      Note: It is important that the same CNAME is not used in different
      communication session contexts or origins, as that could enable
      tracking of a user and its device usage of different services.
      See Section 4.4.1 of Security Considerations for WebRTC
      [I-D.ietf-rtcweb-security] for further discussion.

      The requirement on using the same CNAME for all SSRCs that
      originates from the same end-point, does not require middleboxes
      that forwards traffic from multiple end-points to only use a
      single CNAME.

   The above will currently force a WebRTC end-point that receives an
   MediaStreamTrack on one RTCPeerConnection and adds it as an outgoing
   on any RTCPeerConnection to perform resynchronisation of the stream.
   This, as the sending party needs to change the CNAME, which implies
   that it has to use a locally available system clock as timebase for
   the synchronisation.  Thus, the relative relation between the
   timebase of the incoming stream and the system sending out needs to
   defined.  This relation also needs monitoring for clock drift and
   likely adjustments of the synchronisation.  The sending entity is
   also responsible for congestion control for its the sent streams.  In
   cases of packet loss the loss of incoming data also needs to be
   handled.  This leads to the observation that the method that is least
   likely to cause issues or interruptions in the outgoing source packet
   stream is a model of full decoding, including repair etc followed by
   encoding of the media again into the outgoing packet stream.
   Optimisations of this method is clearly possible and implementation
   specific.

   A WebRTC end-point MUST support receiving multiple MediaStreamTracks,
   where each of different MediaStreamTracks (and their sets of
   associated packet streams) uses different CNAMEs.  However,
   MediaStreamTracks that are received with different CNAMEs have no
   defined synchronisation.

      Note: The motivation for supporting reception of multiple CNAMEs
      are to allow for forward compatibility with any future changes
      that enables more efficient stream handling when end-points relay/
      forward streams.  It also ensures that end-points can interoperate
      with certain types of multi-stream middleboxes or end-points that
      are not WebRTC.


Section 13 (Part)

   RTCP packets convey a Canonical Name (CNAME) identifier that is used
   to associate RTP packet streams that need to be synchronised across
   related RTP sessions.  Inappropriate choice of CNAME values can be a
   privacy concern, since long-term persistent CNAME identifiers can be
   used to track users across multiple WebRTC calls.  Section 4.9 of
   this memo provides guidelines for generation of untraceable CNAME
   values that alleviate this risk.

-- 

Magnus Westerlund

----------------------------------------------------------------------
Services, Media and Network features, Ericsson Research EAB/TXM
----------------------------------------------------------------------
Ericsson AB                 | Phone  +46 10 7148287
FÃ¤rÃ¶gatan 6                 | Mobile +46 73 0949079
SE-164 80 Stockholm, Sweden | mailto: magnus.westerlund@ericsson.com
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On 2014-04-08 17:13, Roni Even wrote:
> Hi,
> Comment on video switching in-line

>>
>>    o  Video switching MCUs (Topo-Video-switch-MCU) SHOULD NOT be used,
>>       since they make the use of RTCP for congestion control and quality
>>       of service reports problematic (see Section 3.6.2 of
>>       [I-D.ietf-avtcore-rtp-topologies-update]).
> [Roni Even] I think that SHOULD NOT is too strong here better use MAY be
> used, this is a simple common mode of centralized video conferencing being
> used. If you are talking about the RTP layer the MCU gets all the RTCP
> reports. The problem of loop detection may occur when cascading MCUs and the
> avtcore topology should point it out. I do not see it as a major problem
> since the video switching is managed by the application layer who should
> prevent such situations.
> 

Roni,

I think you need to go read the definition of the topology in the draft.
As defined in the topologies update draft, that behaviour is not
acceptable towards an WebRTC end-point that uses RTCP for congestion
control.

I quote the most important parts (Section 3.8) from -01:

  "A video switching MCU forwards to a participant a single media
   stream, selected from the available streams.  The criteria for
   selection are often based on voice activity in the audio-visual
   conference, but other conference management mechanisms (like
   presentation mode or explicit floor control) are known to exist as
   well.

   The video switching MCU may also perform media translation to modify
   the content in bit-rate, encoding, or resolution.  However, it still
   may indicate the original sender of the content through the SSRC.  In
   this case, the values of the CC and CSRC fields are retained.

   If not terminating RTP, the RTCP Sender Reports are forwarded for the
   currently selected sender.  All RTCP Receiver Reports are freely
   forwarded between the participants.  In addition, the MCU may also
   originate RTCP control traffic in order to control the session and/or
   report on status from its viewpoint.

   The video switching MCU has most of the attributes of a Translator.
   However, its stream selection is a mixing behavior.  This behavior
   has some RTP and RTCP issues associated with it.  The suppression of
   all but one media stream results in most participants seeing only a
   subset of the sent media streams at any given time, often a single
   stream per conference.  Therefore, RTCP Receiver Reports only report
   on these streams.  Consequently, the media senders that are not
   currently forwarded receive a view of the session that indicates
   their media streams disappear somewhere en route.  This makes the use
   of RTCP for congestion control, or any type of quality reporting,
   very problematic."

I think the behaviour you are interested in falls under the RTP mixer
behaviour either the Mixing (Section 3.6.1) or the Switching (Section
3.6.2) where RTCP is properly handled.


Cheers


Magnus Westerlund

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Hi,

I have considered these comments and are okay with modifying the
recommendation. I have modified the proposed text from Harald somewhat.
My proposal is the following:


   WebRTC implementations of RTP endpoints implemented according to this
   memo are expected to support all the topologies described in
   [I-D.ietf-avtcore-rtp-topologies-update] where the RTP endpoints send
   and receive unicast RTP packet streams to and from some peer device,
   provided that peer can participate in performing congestion control
   on the RTP packet streams.  The peer device could be another RTP
   endpoint, or it could be an RTP middlebox that redistributes the RTP
   packet streams to other RTP endpoints.  This limitation means that
   some of the RTP middlebox-based topologies are not suitable for use
   in the WebRTC environment.  Specifically:

   o  Video switching MCUs (Topo-Video-switch-MCU) SHOULD NOT be used,
      since they make the use of RTCP for congestion control and quality
      of service reports problematic (see Section 3.8 of
      [I-D.ietf-avtcore-rtp-topologies-update]).

   o  The Relay-Transport Translator (Topo-PtM-Trn-Translator) topology
      SHOULD NOT be used because its safe use requires a point to
      multipoint congestion control algorithm or RTP circuit breaker,
      which has not yet been standardised.

   The following topology may be used, however it has considerations
   worth noting:

   o  Content modifying MCUs with RTCP termination (Topo-RTCP-
      terminating-MCU) MAY be used.  Note that in this RTP Topology, RTP
      loop detection and identification of active senders is the
      responsibility of the WebRTC application; since the clients are
      isolated from each other at the RTP layer, RTP cannot assist with
      these functions (see section 3.9 of
      [I-D.ietf-avtcore-rtp-topologies-update]).


Please provide feedback on this text!

Please see further response to questions and comments inline below.


On 2014-04-07 15:44, Harald Alvestrand wrote:
> On 04/07/2014 01:15 PM, Magnus Westerlund wrote:
>> On 2014-04-07 10:12, Harald Alvestrand wrote:

>>> In my (strong) opinion: This recommendation is wrong.
>>>
>>> All central conferencing units that use individual RTP sessions with
>>> clients fall into this category. This includes, I believe, every single
>>> multiuser video conferencing system based on WebRTC in existence (and
>>> there are many of them).
>> Do they? The point of this topology is that it terminates the source
>> identification and hides when it is performing any type of mixing or
>> switching operation. You can perform this with a back to back RTP
>> session style without breaking this property. Just by correctly linking
>> information between the sessions.
> 
> Is there a definition for "correctly linking information"?
> In particular, are you thinking of mappings that preserve SSRC?
> Because that's not the ones I'm thinking of.

The correctly linking information is mostly a question of preserving
CNAMEs across the middlebox to ensure that an end-point can determine
synchronization contexts, and origin if CSRC are used. As we add
additional meta data about streams they may also needs to preserved
depending on scope. Here it becomes a question of how one deals with
APPID and MSID.

Preserving SSRC is a choice, if one does it then loop detection works,
if one doesn't preserve it it doesn't. Preserving it avoids having to
remap any information that have global scope across the legs.

It becomes a choice on what complexities one needs to handle, common
uniqueness requirements versus remapping I think is the major trade-off.

> 
>>   Which means that the individual RTP
>> session is from the identification part inseparable from an SFM or
>> classic RTP mixer. Only the SSRC level loop detection gets impacted.
>>
>> I do understand that people have implemented it this way. Implementation
>> are not necessarily equivalent to recommended practice.
> 
> But when there's one common practice, and no recommended practice, that
> might be a gentle hint that there's a reason why people are doing it -
> indeed, it might be worth making it a recommended practice.

You are welcome to propose changes regarding this to the
topologies-update document in AVTCORE WG.

>>
>>> I believe the recommendation should be:
>>>
>>> * Content modifying MCUs with RTCP termination
>>> (Topo-RTCP-terminating-MCU) MAY be used. Note that in this scenario, RTP
>>> loop detection and identification of active senders is the resposibility
>>> of the application; since the clients are isolated from each other at
>>> the RTP layer, RTP cannot assist with these functions.
>>>
>> If this topology was without issues, then one would simply remove the
>> bullet, and let it fall under the general clause. However, it clearly
>> needs that warning. Which results in this topology being both positively
>> and negatively treated in regards to the other "you may encounter this
>> topology".
> 
> It might need that warning - I'm not sure it's a real issue; loop
> detection is only an issue when you have potential loops, and in a
> single-mixer architecture, that potential is minimal (each endpoint
> knows perfectly well the difference between what it's sourcing and what
> it's receiving). Identification of active senders can be an
> application-layer thing.

Yes, I agree that in a basic conference application with only a single
server instance this is unlikely to form a loop. If one starts doing
more complex things, including multi-server conferences or
PeerConnection Meshes things become more complex. The later we anyway
have removed the RTP loop detection due to the CNAME being unique on
PeerConnection level, but I do think that is the right trade-off from
privacy point of view.

> 
> Anyway, I stand by my statement: SHOULD NOT is the wrong thing to say.

I do accept this, and are willing to change the recommendation.

Cheers

Magnus Westerlund

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Hi,

Please see inline for replies.

On 2014-04-10 13:50, Michael Tuexen wrote:
> On 04 Mar 2014, at 17:39, Magnus Westerlund <magnus.westerlund@ericsson.com> wrote:
> 
>> On 2014-02-25 17:07, Michael Tuexen wrote:
>>>
>>> On Feb 24, 2014, at 5:32 PM, Magnus Westerlund <magnus.westerlund@ericsson.com> wrote:
>>>> 1. Section 4:
>>>> Shouldn't this section discuss the priority field?
>>> I added in the list of consistent properties:
>>>
>>> <t>the priority of the data Channel.</t>
>>>
>>> and in the text below that enumeration:
>>>
>>> ??????
>>
>> Yes, text for this needs to be figured out.
> We don't define at this place what priority is. The only point is that both
> sides use the same priority.

Okay, I agree, the Priority is a property of the established Data
Channel, and the values and implication of these needs to be defined in
the Data Channel draft and not here.

>>
>>>>
>>>> 2. Section 4:
>>>>
>>>> The method
>>>>  used to determine which side uses odd or even is based on the
>>>>  underlying DTLS connection role when used in WebRTC, with the side
>>>>  acting as the DTLS client using even stream identifiers.
>>>>
>>>> Isn't this unnecessary using the vague word of WebRTC instead of simply
>>>> pointing to the DTLS roles of the established data channel?
>>
>>> The point is that in the WebRTC you use DCEP/SCTP/DTLS/UDP and therefore
>>> you can refer to the DTLS role. However, you could use DCEP/SCTP/IP
>>> or DCEP/SCTP/UDP/IP or DCEP/SCTP over something not involving DTLS.
>>> In that case DTLS is not used and you can not refer to the DTLS role.
>>> That is why the restriction is used.
>>
>> Ok, if that concern then you still should be able to write a normative
>> specification under the condition that it is SCTP over DTLS. If not how
>> do you determine that? Are suggesting just to hand way or point to a
>> higher signaling layer.
> So what about using:
> 
> when using <xref target='I-D.ietf-tsvwg-sctp-dtls-encaps'/>, the method used to
> determine which side uses odd or even is based on the underlying DTLS
> connection role: the side acting as the DTLS client MUST use Streams with even
> SCTP stream identifiers, the side acting as the DTLS server MUST use Streams
> with odd SCTP stream identifiers.</t>

I think that is fine for when using over DTLS. And to my understanding
this do require DTLS? If not we need alternative text.


>>>>
>>>> 4. Section 5.1:
>>>>  Label: Variable Length (sequence of characters)
>>>>     The name of the channel.  This may be an empty string.
>>>>
>>>>  Protocol: Variable Length (sequence of characters)
>>>>     The protocol for the channel.  If this is an empty string the
>>>>     protocol us unspecified.  If it is an non-empty string, it
>>>>     specifies an IANA-registered protocol (see Section 8.4).
>>>>
>>>> Both of these fields are strings, shouldn't a particular encoding be
>>>> specified here? Like UTF-8. Secondly, what values are allowed, the full
>>>> set of Unicode?
>>
>>> You are right. Any need for restrictions? We only need to be able to
>>> transform it to a DomString.
>>> So I changed it to:
>>>
>>> <t hangText='Label: Variable Length (sequence of characters)'>
>>> <vspace blankLines='0'/>
>>> The name of the channel as a UTF-8 encoded string.
>>> This may be an empty string.</t>
>>>
>>> <t hangText='Protocol: Variable Length (sequence of characters)'>
>>> <vspace blankLines='0'/>
>>> The protocol for the channel as a UTF-8 encoded string.
>>> If this is an empty string the protocol us unspecified.
>>> If it is an non-empty string, it specifies an IANA-registered protocol
>>> (see <xref target='iana_protocol'/>).</t>
>>
>> I guess this is slightly overtaken by event. It have to be aligned with
>> what the websocket sub-protocol identifier.
> The text now reads:
> <t hangText='Protocol: Variable Length (sequence of characters)'>
> <vspace blankLines='0'/>
> The sub-protocol for the channel as a UTF-8 encoded string.
> If this is an empty string the protocol is unspecified.
> If it is a non-empty string, it specifies an protocol registered in the
> 'WebSocket Subprotocol Name Registry' created in
> <xref target='RFC6455'/>.</t>

Ok.

>>
>>>>
>>>> 5. Section 6:
>>>> All Data Channel Establishment Protocol messages MUST be sent
>>>>  requesting ordered delivery and using reliable transmission.
>>>>
>>>> I wonder of the use of requesting ordered delivery and using reliable
>>>> transmission, from an SCTP stream perspective, wouldn't using in both
>>>> places be appropriate? Or is how object which has been requested to be
>>>> transmitted unordered interact in SCTP with the ordered ones?
>>
>>> I'm sorry, I don't understand what you are asking...
>>
>> Sorry, it really is a language issue. The above sentence first states
>> "sent 'requesting' ordered" and later "and 'using' reliable".
>>
>> This inconsistency is what I reacted to.
> I see. Changed to:
> <t>All Data Channel Establishment Protocol messages MUST be sent using
> ordered delivery and reliable transmission.

Ok

>>
>>>
>>>> 6. Section 7:
>>>>
>>>> I think this section can be beefed up a bit. First make clear that the
>>>> Data Channel's required usage of DTLS ensures that the message integrity
>>>> and possible source authentication as well as confidentiality. Then
>>>> going over any security risks with a malicous peer using this protocol.
>>>> Can a malicous side screw up the peer using this protocol? What are the
>>>> implications?
>>
>>> Just to double check: Aren't the referenced documents the ones which
>>> discuss all security stuff in one place?
>>
>> The security document do discuss system level important aspects. But,
>> from my perspective, details that are very specific to this protocol
>> should be discussed in this document.
>>
>> I think this is highly relevant in this document as there are others
>> that are interested in using it.
> OK. Any concrete suggestions what should be covered?

I can at least think of potentially relevant issues:

- Exhaustion attacks by opening all 64k Data channels
- Attempting to overflow buffers by including 64k of Strings in Label
and Protocol fields in the OPEN message.
- Attempt to get undocumented behaviour by sending Reliability
Parameters that aren't defined. Or use out of range Priorities or
message types.

But that I guess is most of the things I can think of to attempt to
break the protocol.

Then it is the question of requiring DTLS to ensure it is safe from
Privacy, Confidentiality, Integrity and possibly source authentication
concerns.


>>>>
>>>> 11. This comments concerns the relation to the Data Channel
>>>> specification. So my interpretation is that the WebRTC Data channel
>>>> draft has become both the base for this specification as well as the one
>>>> that specifies that the DCEP shall be implemented and supported. For
>>>> WebRTC I don't think this matters much, but if someone likes to re-use
>>>> the basic Data Channel specification, this structure makes it more
>>>> difficult and have no need for the bi-directional negotiated parameter
>>>> settings that the DCEP provides. In that case one have to sub-set the
>>
>>> I think the data channel draft discusses the data channels independent
>>> how they are opened. This covers data parameters, user data transmission
>>> and closing of data channels. The data channel protocol draft discusses
>>> an in-band protocol for setting up data channels.
>>
>> I think this was discussed today. And make it clear that DCEP
>> requirement shouldn't be from the Data Channel spec.
>>
>>>> data channel specification. I wonder if it would be better to move some
>>>> normative statements around, so that the chain of implementation
>>>> requirement goes draft-ietf-rtcweb-transports ->
>>>> draft-ietf-rtcweb-data-protocol -> draft-ietf-rtcweb-data-channel thus
>>>> providing a cleaner and more modular build up of the protocols.
>>> I think you can implement draft-ietf-rtcweb-data-channel without
>>> draft-ietf-rtcweb-data-protocol, but not vice versa.
>>>
>>> So you don't like the statement:
>>>
>>>   A simple protocol for internal negotiation is specified in
>>>   [I-D.ietf-rtcweb-data-protocol] and MUST be supported.
>>>
>>> from draft-ietf-rtcweb-data-channel. What would you suggest?
>>
>> I think the high level requirement on DCEP might be in -rtcweb-transports.
> It now reads:
> A simple protocol for in-band negotiation is specified in [I-D.ietf-rtcweb-data-protocol].


Ok


Cheers


Magnus Westerlund

----------------------------------------------------------------------
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Subject: Re: [rtcweb] Text proposal for CNAME in draft-ietf-rtcweb-rtp-usage
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The text looks fine, with one comment.

On 15 April 2014 05:10, Magnus Westerlund
<magnus.westerlund@ericsson.com> wrote:
>       Note: It is important that the same CNAME is not used in different
>       communication session contexts or origins, as that could enable
>       tracking of a user and its device usage of different services.
>       See Section 4.4.1 of Security Considerations for WebRTC
>       [I-D.ietf-rtcweb-security] for further discussion.

I'd rather not have this hidden in a "Note", and I'd prefer if it were
more concrete.  I think that we need to say something like:

A different CNAME MUST be used for different RTCPeerConnection
instances.  Having two communication sessions with the same CNAME
could enable tracking of a user or device across different services
(see Section 4.4.1 of [security] for details).  A web application MAY
override the CNAME that is selected using the process in [RFC7022] to
allow for synchronization of disjoint sessions; [[this doesn't result
in a tracking issue, since the creation of matching CNAMEs depends on
existing tracking]].


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On 04/15/2014 05:14 PM, Magnus Westerlund wrote:
> Hi,
>
> I have considered these comments and are okay with modifying the
> recommendation. I have modified the proposed text from Harald somewhat.
> My proposal is the following:
>
>
>     WebRTC implementations of RTP endpoints implemented according to this
>     memo are expected to support all the topologies described in
>     [I-D.ietf-avtcore-rtp-topologies-update] where the RTP endpoints send
>     and receive unicast RTP packet streams to and from some peer device,
>     provided that peer can participate in performing congestion control
>     on the RTP packet streams.  The peer device could be another RTP
>     endpoint, or it could be an RTP middlebox that redistributes the RTP
>     packet streams to other RTP endpoints.  This limitation means that
>     some of the RTP middlebox-based topologies are not suitable for use
>     in the WebRTC environment.  Specifically:
>
>     o  Video switching MCUs (Topo-Video-switch-MCU) SHOULD NOT be used,
>        since they make the use of RTCP for congestion control and quality
>        of service reports problematic (see Section 3.8 of
>        [I-D.ietf-avtcore-rtp-topologies-update]).
>
>     o  The Relay-Transport Translator (Topo-PtM-Trn-Translator) topology
>        SHOULD NOT be used because its safe use requires a point to
>        multipoint congestion control algorithm or RTP circuit breaker,
>        which has not yet been standardised.

Grammar warning: The above can be parsed as either

(multipoint congestion control algorithm) or (RTP circuit breaker)

or

multipoint ((congestion control algorithm) or (RTP circuit breaker))

Only the second parse will be a true statement once -circuit-breaker
(a normative dependency) is published. Perhaps this needs rephrasing?

>
>     The following topology may be used, however it has considerations
>     worth noting:
>
>     o  Content modifying MCUs with RTCP termination (Topo-RTCP-
>        terminating-MCU) MAY be used.  Note that in this RTP Topology, RTP
>        loop detection and identification of active senders is the
>        responsibility of the WebRTC application; since the clients are
>        isolated from each other at the RTP layer, RTP cannot assist with
>        these functions (see section 3.9 of
>        [I-D.ietf-avtcore-rtp-topologies-update]).
>
>
> Please provide feedback on this text!

This text looks reasonable to me.

>
> Please see further response to questions and comments inline below.
>
>
> On 2014-04-07 15:44, Harald Alvestrand wrote:
>> On 04/07/2014 01:15 PM, Magnus Westerlund wrote:
>>> On 2014-04-07 10:12, Harald Alvestrand wrote:
>>>> In my (strong) opinion: This recommendation is wrong.
>>>>
>>>> All central conferencing units that use individual RTP sessions with
>>>> clients fall into this category. This includes, I believe, every single
>>>> multiuser video conferencing system based on WebRTC in existence (and
>>>> there are many of them).
>>> Do they? The point of this topology is that it terminates the source
>>> identification and hides when it is performing any type of mixing or
>>> switching operation. You can perform this with a back to back RTP
>>> session style without breaking this property. Just by correctly linking
>>> information between the sessions.
>> Is there a definition for "correctly linking information"?
>> In particular, are you thinking of mappings that preserve SSRC?
>> Because that's not the ones I'm thinking of.
> The correctly linking information is mostly a question of preserving
> CNAMEs across the middlebox to ensure that an end-point can determine
> synchronization contexts, and origin if CSRC are used. As we add
> additional meta data about streams they may also needs to preserved
> depending on scope. Here it becomes a question of how one deals with
> APPID and MSID.
>
> Preserving SSRC is a choice, if one does it then loop detection works,
> if one doesn't preserve it it doesn't. Preserving it avoids having to
> remap any information that have global scope across the legs.
>
> It becomes a choice on what complexities one needs to handle, common
> uniqueness requirements versus remapping I think is the major trade-off.
>
>>>    Which means that the individual RTP
>>> session is from the identification part inseparable from an SFM or
>>> classic RTP mixer. Only the SSRC level loop detection gets impacted.
>>>
>>> I do understand that people have implemented it this way. Implementation
>>> are not necessarily equivalent to recommended practice.
>> But when there's one common practice, and no recommended practice, that
>> might be a gentle hint that there's a reason why people are doing it -
>> indeed, it might be worth making it a recommended practice.
> You are welcome to propose changes regarding this to the
> topologies-update document in AVTCORE WG.
>
>>>> I believe the recommendation should be:
>>>>
>>>> * Content modifying MCUs with RTCP termination
>>>> (Topo-RTCP-terminating-MCU) MAY be used. Note that in this scenario, RTP
>>>> loop detection and identification of active senders is the resposibility
>>>> of the application; since the clients are isolated from each other at
>>>> the RTP layer, RTP cannot assist with these functions.
>>>>
>>> If this topology was without issues, then one would simply remove the
>>> bullet, and let it fall under the general clause. However, it clearly
>>> needs that warning. Which results in this topology being both positively
>>> and negatively treated in regards to the other "you may encounter this
>>> topology".
>> It might need that warning - I'm not sure it's a real issue; loop
>> detection is only an issue when you have potential loops, and in a
>> single-mixer architecture, that potential is minimal (each endpoint
>> knows perfectly well the difference between what it's sourcing and what
>> it's receiving). Identification of active senders can be an
>> application-layer thing.
> Yes, I agree that in a basic conference application with only a single
> server instance this is unlikely to form a loop. If one starts doing
> more complex things, including multi-server conferences or
> PeerConnection Meshes things become more complex. The later we anyway
> have removed the RTP loop detection due to the CNAME being unique on
> PeerConnection level, but I do think that is the right trade-off from
> privacy point of view.
>
>> Anyway, I stand by my statement: SHOULD NOT is the wrong thing to say.
> I do accept this, and are willing to change the recommendation.
>
> Cheers
>
> Magnus Westerlund
>
> ----------------------------------------------------------------------
> Services, Media and Network features, Ericsson Research EAB/TXM
> ----------------------------------------------------------------------
> Ericsson AB                 | Phone  +46 10 7148287
> Färögatan 6                 | Mobile +46 73 0949079
> SE-164 80 Stockholm, Sweden | mailto: magnus.westerlund@ericsson.com
> ----------------------------------------------------------------------
>


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Subject: Re: [rtcweb] Draft proposal for updating Multiparty topologies in draft-ietf-rtcweb-rtp-usage
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On 2014-04-15 23:57, Harald Alvestrand wrote:
> On 04/15/2014 05:14 PM, Magnus Westerlund wrote:
>> Hi,
>>
>> I have considered these comments and are okay with modifying the
>> recommendation. I have modified the proposed text from Harald somewhat.
>> My proposal is the following:
>>
>>
>>     WebRTC implementations of RTP endpoints implemented according to this
>>     memo are expected to support all the topologies described in
>>     [I-D.ietf-avtcore-rtp-topologies-update] where the RTP endpoints send
>>     and receive unicast RTP packet streams to and from some peer device,
>>     provided that peer can participate in performing congestion control
>>     on the RTP packet streams.  The peer device could be another RTP
>>     endpoint, or it could be an RTP middlebox that redistributes the RTP
>>     packet streams to other RTP endpoints.  This limitation means that
>>     some of the RTP middlebox-based topologies are not suitable for use
>>     in the WebRTC environment.  Specifically:
>>
>>     o  Video switching MCUs (Topo-Video-switch-MCU) SHOULD NOT be used,
>>        since they make the use of RTCP for congestion control and quality
>>        of service reports problematic (see Section 3.8 of
>>        [I-D.ietf-avtcore-rtp-topologies-update]).
>>
>>     o  The Relay-Transport Translator (Topo-PtM-Trn-Translator) topology
>>        SHOULD NOT be used because its safe use requires a point to
>>        multipoint congestion control algorithm or RTP circuit breaker,
>>        which has not yet been standardised.
> 
> Grammar warning: The above can be parsed as either
> 
> (multipoint congestion control algorithm) or (RTP circuit breaker)
> 
> or
> 
> multipoint ((congestion control algorithm) or (RTP circuit breaker))
> 

It is the later that was intended. Because the topology requires
congestion control for multi endpoint.

Will reword.

Thanks

Magnus Westerlund

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Services, Media and Network features, Ericsson Research EAB/TXM
----------------------------------------------------------------------
Ericsson AB                 | Phone  +46 10 7148287
Färögatan 6                 | Mobile +46 73 0949079
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Subject: Re: [rtcweb] Text proposal for CNAME in draft-ietf-rtcweb-rtp-usage
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On 2014-04-15 19:33, Martin Thomson wrote:
> The text looks fine, with one comment.
> 
> On 15 April 2014 05:10, Magnus Westerlund
> <magnus.westerlund@ericsson.com> wrote:
>>       Note: It is important that the same CNAME is not used in different
>>       communication session contexts or origins, as that could enable
>>       tracking of a user and its device usage of different services.
>>       See Section 4.4.1 of Security Considerations for WebRTC
>>       [I-D.ietf-rtcweb-security] for further discussion.
> 
> I'd rather not have this hidden in a "Note", and I'd prefer if it were
> more concrete.  I think that we need to say something like:
> 
> A different CNAME MUST be used for different RTCPeerConnection
> instances.  Having two communication sessions with the same CNAME
> could enable tracking of a user or device across different services
> (see Section 4.4.1 of [security] for details).  A web application MAY
> override the CNAME that is selected using the process in [RFC7022] to
> allow for synchronization of disjoint sessions; [[this doesn't result
> in a tracking issue, since the creation of matching CNAMEs depends on
> existing tracking]].
> 
> 

The first part is already said in Section 4.9 text:

   The RTP specification [RFC3550] includes guidelines for choosing a
   unique RTP CNAME, but these are not sufficient in the presence of NAT
   devices.  In addition, long-term persistent identifiers can be
   problematic from a privacy viewpoint (Section 13).  Accordingly, a
   WebRTC endpoint MUST generate a new, unique, short-term persistent
   RTCP CNAME for each RTCPeerConnection, following [RFC7022].

Thus, I don't want in two places to use Normative RFC 2119 text for the
same purpose.

For the second half, I have to ask what the scope of this overriding is
intended to be. Is it that within the same origin, a web application
creating multiple PeerConnections can request them to use the same (by
browser) selected random CNAME value? Or is it that the web application
can set the CNAME to be used at PeerConnection creation time. The first
I am fine with, as it appear to have very little downsides, and can
really help application that tries to do advanced things where multiple
PCs are used across multiple endpoints and middleboxes.

Thus, I would propose to modify your proposal to result in a change in
Section 4.9:

Old:
   The RTP specification [RFC3550] includes guidelines for choosing a
   unique RTP CNAME, but these are not sufficient in the presence of NAT
   devices.  In addition, long-term persistent identifiers can be
   problematic from a privacy viewpoint (Section 13).  Accordingly, a
   WebRTC endpoint MUST generate a new, unique, short-term persistent
   RTCP CNAME for each RTCPeerConnection, following [RFC7022].

NEW:

   The RTP specification [RFC3550] includes guidelines for choosing a
   unique RTP CNAME, but these are not sufficient in the presence of NAT
   devices.  In addition, long-term persistent identifiers can be
   problematic from a privacy viewpoint (Section 13).  Accordingly, a
   WebRTC endpoint MUST generate a new, unique, short-term persistent
   RTCP CNAME for each RTCPeerConnection, following [RFC7022], with a
   single exception; if explicitly requested at creation an
   RTCPeerConnection MAY use the same CNAME as as an existing
   RTCPeerConnection within their common same-origin context.


Section 11:

OLD:

   The same MediaStreamTrack can also be included in multiple
   MediaStreams, thus multiple sets of MediaStreams can implicitly need
   to use the same synchronisation base.  To ensure that this works in
   all cases, and don't forces a end-point to change synchronisation
   base and CNAME in the middle of a ongoing delivery of any packet
   streams, which would cause media disruption; all MediaStreamTracks
   and their associated SSRCs originating from the same end-point needs
   to be sent using the same CNAME within one RTCPeerConnection.  This
   is motivating the strong recommendation in Section 4.9 to only use a
   single CNAME.

      Note: It is important that the same CNAME is not used in different
      communication session contexts or origins, as that could enable
      tracking of a user and its device usage of different services.
      See Section 4.4.1 of Security Considerations for WebRTC
      [I-D.ietf-rtcweb-security] for further discussion.

      The requirement on using the same CNAME for all SSRCs that
      originates from the same end-point, does not require middleboxes
      that forwards traffic from multiple end-points to only use a
      single CNAME.

NEW:

   The same MediaStreamTrack can also be included in multiple
   MediaStreams, thus multiple sets of MediaStreams can implicitly need
   to use the same synchronisation base.  To ensure that this works in
   all cases, and don't forces a end-point to change synchronisation
   base and CNAME in the middle of a ongoing delivery of any packet
   streams, which would cause media disruption; all MediaStreamTracks
   and their associated SSRCs originating from the same end-point needs
   to be sent using the same CNAME within one RTCPeerConnection.  This
   is motivating the strong recommendation in Section 4.9 to only use a
   single CNAME.

      The requirement on using the same CNAME for all SSRCs that
      originates from the same end-point, does not require middleboxes
      that forwards traffic from multiple end-points to only use a
      single CNAME.

   Different CNAMEs are normally required to be used for different
   RTCPeerConnection instances, as specified in Section 4.9.  Having two
   communication sessions with the same CNAME could enable tracking of a
   user or device across different services (see Section 4.4.1 of
   [I-D.ietf-rtcweb-security] for details).  A web application can
   request that the CNAMEs used in different RTCPeerConnection within a
   same-orign context to be the same, this allow for synchronization of
   the endpoint's RTP packet streams across the different
   RTCPeerConnections.

      Note: this doesn't result in a tracking issue, since the creation
      of matching CNAMEs depends on existing tracking.


Is this better/acceptable?

Cheers

Magnus Westerlund

----------------------------------------------------------------------
Services, Media and Network features, Ericsson Research EAB/TXM
----------------------------------------------------------------------
Ericsson AB                 | Phone  +46 10 7148287
FÃ¤rÃ¶gatan 6                 | Mobile +46 73 0949079
SE-164 80 Stockholm, Sweden | mailto: magnus.westerlund@ericsson.com
----------------------------------------------------------------------


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Dear RTCWEB,

We're having a minor change of management in this WG.  Magnus is stepping
down in preparation for his upcoming parental leave.  Sean Turner has
kindly agreed to take his place.  For those of you who don't know Sean, he
was recently SEC AD, he chaired the S/MIME working group before that, and
he's currently providing security expertise in the STIR working group.

Many thanks to Magnus for his leadership in this working group!

--Richard

--089e01184bae698ea304f72965db
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<div dir=3D"ltr">Dear RTCWEB,<div><br></div><div>We&#39;re having a minor c=
hange of management in this WG. =A0Magnus is stepping down in preparation f=
or his upcoming parental leave. =A0Sean Turner has kindly agreed to take hi=
s place. =A0For those of you who don&#39;t know Sean, he was recently SEC A=
D, he chaired the S/MIME working group before that, and he&#39;s currently =
providing security expertise in the STIR working group.</div>
<div><br></div><div>Many thanks to Magnus for his leadership in this workin=
g group!</div><div><br></div><div>--Richard</div><div><br></div></div>

--089e01184bae698ea304f72965db--


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On Wed, Apr 16, 2014 at 7:04 AM, Richard Barnes <rlb@ipv.sx> wrote:

> Dear RTCWEB,
>
> We're having a minor change of management in this WG.  Magnus is stepping
> down in preparation for his upcoming parental leave.  Sean Turner has
> kindly agreed to take his place.  For those of you who don't know Sean, h=
e
> was recently SEC AD, he chaired the S/MIME working group before that, and
> he's currently providing security expertise in the STIR working group.
>
> Many thanks to Magnus for his leadership in this working group!
>
>
=E2=80=8BI'd like to add my thanks to Magnus; as a chair, contributor (and =
host for
meetings), he's been a tremendous and positive influence on the development
of WebRTC.

Sean, welcome, you have big boots to fill!

Ted





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

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<div dir=3D"ltr"><div class=3D"gmail_extra">On Wed, Apr 16, 2014 at 7:04 AM=
, Richard Barnes <span dir=3D"ltr">&lt;<a href=3D"mailto:rlb@ipv.sx" target=
=3D"_blank">rlb@ipv.sx</a>&gt;</span> wrote:<br><div class=3D"gmail_quote">=
<blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1p=
x #ccc solid;padding-left:1ex">
<div dir=3D"ltr">Dear RTCWEB,<div><br></div><div>We&#39;re having a minor c=
hange of management in this WG. =C2=A0Magnus is stepping down in preparatio=
n for his upcoming parental leave. =C2=A0Sean Turner has kindly agreed to t=
ake his place. =C2=A0For those of you who don&#39;t know Sean, he was recen=
tly SEC AD, he chaired the S/MIME working group before that, and he&#39;s c=
urrently providing security expertise in the STIR working group.</div>

<div><br></div><div>Many thanks to Magnus for his leadership in this workin=
g group!</div><span class=3D"HOEnZb"><font color=3D"#888888"><div><br></div=
></font></span></div></blockquote><div><br><div class=3D"gmail_default" sty=
le=3D"font-family:georgia,serif">
=E2=80=8BI&#39;d like to add my thanks to Magnus; as a chair, contributor (=
and host for meetings), he&#39;s been a tremendous and positive influence o=
n the development of WebRTC. <br><br></div><div class=3D"gmail_default" sty=
le=3D"font-family:georgia,serif">
Sean, welcome, you have big boots to fill!<br><br>Ted<br></div><br><br><br>=
=C2=A0</div><blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;bo=
rder-left:1px #ccc solid;padding-left:1ex"><div dir=3D"ltr"><span class=3D"=
HOEnZb"><font color=3D"#888888"><div>
</div><div>--Richard</div><div><br></div></font></span></div>
<br>_______________________________________________<br>
rtcweb mailing list<br>
<a href=3D"mailto:rtcweb@ietf.org">rtcweb@ietf.org</a><br>
<a href=3D"https://www.ietf.org/mailman/listinfo/rtcweb" target=3D"_blank">=
https://www.ietf.org/mailman/listinfo/rtcweb</a><br>
<br></blockquote></div><br></div></div>

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From nobody Wed Apr 16 08:07:34 2014
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From: Eric Rescorla <ekr@rtfm.com>
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Yes, thanks to Magnus! Welcome to Sean.

-Ekr



On Wed, Apr 16, 2014 at 7:17 AM, Ted Hardie <ted.ietf@gmail.com> wrote:

> On Wed, Apr 16, 2014 at 7:04 AM, Richard Barnes <rlb@ipv.sx> wrote:
>
>> Dear RTCWEB,
>>
>> We're having a minor change of management in this WG.  Magnus is stepping
>> down in preparation for his upcoming parental leave.  Sean Turner has
>> kindly agreed to take his place.  For those of you who don't know Sean, he
>> was recently SEC AD, he chaired the S/MIME working group before that, and
>> he's currently providing security expertise in the STIR working group.
>>
>> Many thanks to Magnus for his leadership in this working group!
>>
>>
> I'd like to add my thanks to Magnus; as a chair, contributor (and host for
> meetings), he's been a tremendous and positive influence on the development
> of WebRTC.
>
> Sean, welcome, you have big boots to fill!
>
> Ted
>
>
>
>
>
>> --Richard
>>
>>
>> _______________________________________________
>> rtcweb mailing list
>> rtcweb@ietf.org
>> https://www.ietf.org/mailman/listinfo/rtcweb
>>
>>
>
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb
>
>

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<div dir=3D"ltr">Yes, thanks to Magnus! Welcome to Sean.<div><br></div><div=
>-Ekr</div><div><br></div></div><div class=3D"gmail_extra"><br><br><div cla=
ss=3D"gmail_quote">On Wed, Apr 16, 2014 at 7:17 AM, Ted Hardie <span dir=3D=
"ltr">&lt;<a href=3D"mailto:ted.ietf@gmail.com" target=3D"_blank">ted.ietf@=
gmail.com</a>&gt;</span> wrote:<br>

<blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1p=
x #ccc solid;padding-left:1ex"><div dir=3D"ltr"><div class=3D"gmail_extra">=
<div><div class=3D"h5">On Wed, Apr 16, 2014 at 7:04 AM, Richard Barnes <spa=
n dir=3D"ltr">&lt;<a href=3D"mailto:rlb@ipv.sx" target=3D"_blank">rlb@ipv.s=
x</a>&gt;</span> wrote:<br>

</div></div><div class=3D"gmail_quote"><div><div class=3D"h5"><blockquote c=
lass=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1px #ccc solid;=
padding-left:1ex">
<div dir=3D"ltr">Dear RTCWEB,<div><br></div><div>We&#39;re having a minor c=
hange of management in this WG. =A0Magnus is stepping down in preparation f=
or his upcoming parental leave. =A0Sean Turner has kindly agreed to take hi=
s place. =A0For those of you who don&#39;t know Sean, he was recently SEC A=
D, he chaired the S/MIME working group before that, and he&#39;s currently =
providing security expertise in the STIR working group.</div>



<div><br></div><div>Many thanks to Magnus for his leadership in this workin=
g group!</div><span><font color=3D"#888888"><div><br></div></font></span></=
div></blockquote></div></div><div><br><div class=3D"gmail_default" style=3D=
"font-family:georgia,serif">


I&#39;d like to add my thanks to Magnus; as a chair, contributor (and host =
for meetings), he&#39;s been a tremendous and positive influence on the dev=
elopment of WebRTC. <br><br></div><div class=3D"gmail_default" style=3D"fon=
t-family:georgia,serif">


Sean, welcome, you have big boots to fill!<br><br>Ted<br></div><br><br><br>=
=A0</div><blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;borde=
r-left:1px #ccc solid;padding-left:1ex"><div dir=3D"ltr"><span><font color=
=3D"#888888"><div>


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

--047d7b87389620368d04f72a4321--


From nobody Wed Apr 16 14:09:43 2014
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From: Sean Turner <TurnerS@ieca.com>
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Thanks.  I=92ll try my best!  Obviously, going to take a wee-bit of time =
to get up to speed.

spt

On Apr 16, 2014, at 11:05, Eric Rescorla <ekr@rtfm.com> wrote:

> Yes, thanks to Magnus! Welcome to Sean.
>=20
> -Ekr
>=20
>=20
>=20
> On Wed, Apr 16, 2014 at 7:17 AM, Ted Hardie <ted.ietf@gmail.com> =
wrote:
> On Wed, Apr 16, 2014 at 7:04 AM, Richard Barnes <rlb@ipv.sx> wrote:
> Dear RTCWEB,
>=20
> We're having a minor change of management in this WG.  Magnus is =
stepping down in preparation for his upcoming parental leave.  Sean =
Turner has kindly agreed to take his place.  For those of you who don't =
know Sean, he was recently SEC AD, he chaired the S/MIME working group =
before that, and he's currently providing security expertise in the STIR =
working group.
>=20
> Many thanks to Magnus for his leadership in this working group!
>=20
>=20
> I'd like to add my thanks to Magnus; as a chair, contributor (and host =
for meetings), he's been a tremendous and positive influence on the =
development of WebRTC.=20
>=20
> Sean, welcome, you have big boots to fill!
>=20
> Ted
>=20
>=20
>=20
> =20
> --Richard
>=20
>=20
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb
>=20
>=20
>=20
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb
>=20
>=20
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb


From nobody Wed Apr 16 15:41:48 2014
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From: Andrei Popov <Andrei.Popov@microsoft.com>
To: "rtcweb@ietf.org" <rtcweb@ietf.org>
Thread-Topic: draft-ietf-rtcweb-security-arch-09: DTLS 1.2 only?
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Subject: [rtcweb] draft-ietf-rtcweb-security-arch-09: DTLS 1.2 only?
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draft-ietf-rtcweb-security-arch-09. says:

" [[OPEN ISSUE:  Are these the right cipher suites?]]  All
   implementations MUST implement the following two cipher suites:
   TLS_DHE_RSA_WITH_AES_128_GCM_SHA256 and
   TLS_ECDHE_RSA_WITH_AES_128_GCM_SHA256 ..."

These are (D)TLS 1.2 cipher suites; I think this requirement would exclude =
DTLS 1.0 implementations. For easier/broader adoption, would it make sense =
to allow DTLS 1.0?

Cheers,

Andrei

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</head>
<body lang=3D"EN-US" link=3D"#0563C1" vlink=3D"#954F72">
<div class=3D"WordSection1">
<p class=3D"MsoNormal">draft-ietf-rtcweb-security-arch-09. says:<o:p></o:p>=
</p>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
<p class=3D"MsoNormal">&#8220; [[OPEN ISSUE:&nbsp; Are these the right ciph=
er suites?]]&nbsp; All<o:p></o:p></p>
<p class=3D"MsoNormal">&nbsp;&nbsp; implementations MUST implement the foll=
owing two cipher suites:<o:p></o:p></p>
<p class=3D"MsoNormal">&nbsp;&nbsp; TLS_DHE_RSA_WITH_AES_128_GCM_SHA256 and=
<o:p></o:p></p>
<p class=3D"MsoNormal">&nbsp;&nbsp; TLS_ECDHE_RSA_WITH_AES_128_GCM_SHA256 &=
#8230;&#8221;<o:p></o:p></p>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
<p class=3D"MsoNormal">These are (D)TLS 1.2 cipher suites; I think this req=
uirement would exclude DTLS 1.0 implementations. For easier/broader adoptio=
n, would it make sense to allow DTLS 1.0?<o:p></o:p></p>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
<p class=3D"MsoNormal">Cheers,<o:p></o:p></p>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
<p class=3D"MsoNormal">Andrei<o:p></o:p></p>
</div>
</body>
</html>

--_000_99c75200c5e742b5946ec8e0a850e13dBY2PR03MB427namprd03pro_--


From nobody Wed Apr 16 16:27:12 2014
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On 16 April 2014 05:54, Magnus Westerlund
<magnus.westerlund@ericsson.com> wrote:
> Is this better/acceptable?

Better, but I don't think that the proposed solution is sufficient for
some use cases.  There is no downside to enabling correlation of
sessions when the signaling entity is able to force correlation.  Let
the app set CNAME.


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On 2014-04-17 01:27, Martin Thomson wrote:
> On 16 April 2014 05:54, Magnus Westerlund
> <magnus.westerlund@ericsson.com> wrote:
>> Is this better/acceptable?
> 
> Better, but I don't think that the proposed solution is sufficient for
> some use cases.  There is no downside to enabling correlation of
> sessions when the signaling entity is able to force correlation.  Let
> the app set CNAME.
> 

Sure from privacy point of view, but there are a potential to really
screw up RTP/RTCP by setting the same CNAME in multiple endpoints part
of the same communication session. That is why I wrote like I wrote.
Even using the same CNAME between different instances of the media
framework, which I assume is the likely result of establishing
PeerConnection in different tabs/windows in a browser instance could get
into issues using the same CNAME, due to them really not being based on
a common context.

Cheers

Magnus Westerlund

----------------------------------------------------------------------
Services, Media and Network features, Ericsson Research EAB/TXM
----------------------------------------------------------------------
Ericsson AB                 | Phone  +46 10 7148287
FÃ¤rÃ¶gatan 6                 | Mobile +46 73 0949079
SE-164 80 Stockholm, Sweden | mailto: magnus.westerlund@ericsson.com
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From nobody Thu Apr 17 09:16:13 2014
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From: Martin Thomson <martin.thomson@gmail.com>
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On 17 April 2014 03:50, Magnus Westerlund
<magnus.westerlund@ericsson.com> wrote:
> On 2014-04-17 01:27, Martin Thomson wrote:
>> On 16 April 2014 05:54, Magnus Westerlund
>> <magnus.westerlund@ericsson.com> wrote:
>>> Is this better/acceptable?
>>
>> Better, but I don't think that the proposed solution is sufficient for
>> some use cases.  There is no downside to enabling correlation of
>> sessions when the signaling entity is able to force correlation.  Let
>> the app set CNAME.
>>
>
> Sure from privacy point of view, but there are a potential to really
> screw up RTP/RTCP by setting the same CNAME in multiple endpoints part
> of the same communication session. That is why I wrote like I wrote.
> Even using the same CNAME between different instances of the media
> framework, which I assume is the likely result of establishing
> PeerConnection in different tabs/windows in a browser instance could get
> into issues using the same CNAME, due to them really not being based on
> a common context.


Sure, WFM.


From nobody Fri Apr 18 07:53:15 2014
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Hi,

  I am new to rtcweb so pl excuse my lack of knowledge. 
  
1)  Refer 
   F33     The browser must be able to initiate and
           accept a media session where the data needed
           for establishment can be carried in SIP.

   In other places in other documents it is stated that browser is 
given freedom to do signaling it likes. Then this requirement will 
be met with signaling gateway. And there is no work in progress for 
that. Whether IETF mandates for browser to understand SIP or not?

2) It will be better if 1-800-Go-Fedex (Fedex call) is referred as 
IVR call and Fedex should be taken out as SKYPE was taken out.

Thanks
Samir


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To: "srivastava_samir@hush.com" <srivastava_samir@hush.com>, "rtcweb@ietf.org" <rtcweb@ietf.org>
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Hi,
 =20
>1)  Refer=20
>   F33     The browser must be able to initiate and
>           accept a media session where the data needed
>           for establishment can be carried in SIP.
>
>   In other places in other documents it is stated that browser is given f=
reedom to do signaling it=20
> likes. Then this requirement will be met with signaling gateway. And ther=
e is no work in progress=20
> for that. Whether IETF mandates for browser to understand SIP or not?

Not.

IETF does mandate the browser to understand SDP, though (as part of JSEP).

>2) It will be better if 1-800-Go-Fedex (Fedex call) is referred as IVR cal=
l and Fedex should be taken >out as SKYPE was taken out.

Which version did you look at? Skype has been removed from the latest versi=
on (-14). Fedex is still there, though.

Regards,

Christer


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From: "Roni Even" <ron.even.tlv@gmail.com>
To: "'Magnus Westerlund'" <magnus.westerlund@ericsson.com>, "'Harald Alvestrand'" <harald@alvestrand.no>, <rtcweb@ietf.org>
References: <533E7A50.5040909@ericsson.com> <53425DDE.2030005@alvestrand.no> <534288C2.6010906@ericsson.com> <5342ABBB.9050300@alvestrand.no> <534D4CC4.9040107@ericsson.com>
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Subject: Re: [rtcweb] Draft proposal for updating Multiparty topologies in draft-ietf-rtcweb-rtp-usage
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Hi Magnus,
I am OK with the text.
Do you think that for Multiparty we should state that a mixed conference
(WebRTC and non WebRTC participants) is possible and should also support
these rules.
=20
Roni=20


> -----Original Message-----
> From: rtcweb [mailto:rtcweb-bounces@ietf.org] On Behalf Of Magnus
> Westerlund
> Sent: 15 April, 2014 6:14 PM
> To: Harald Alvestrand; rtcweb@ietf.org
> Subject: Re: [rtcweb] Draft proposal for updating Multiparty =
topologies in
draft-
> ietf-rtcweb-rtp-usage
>=20
> Hi,
>=20
> I have considered these comments and are okay with modifying the
> recommendation. I have modified the proposed text from Harald =
somewhat.
> My proposal is the following:
>=20
>=20
>    WebRTC implementations of RTP endpoints implemented according to =
this
>    memo are expected to support all the topologies described in
>    [I-D.ietf-avtcore-rtp-topologies-update] where the RTP endpoints =
send
>    and receive unicast RTP packet streams to and from some peer =
device,
>    provided that peer can participate in performing congestion control
>    on the RTP packet streams.  The peer device could be another RTP
>    endpoint, or it could be an RTP middlebox that redistributes the =
RTP
>    packet streams to other RTP endpoints.  This limitation means that
>    some of the RTP middlebox-based topologies are not suitable for use
>    in the WebRTC environment.  Specifically:
>=20
>    o  Video switching MCUs (Topo-Video-switch-MCU) SHOULD NOT be used,
>       since they make the use of RTCP for congestion control and =
quality
>       of service reports problematic (see Section 3.8 of
>       [I-D.ietf-avtcore-rtp-topologies-update]).
>=20
>    o  The Relay-Transport Translator (Topo-PtM-Trn-Translator) =
topology
>       SHOULD NOT be used because its safe use requires a point to
>       multipoint congestion control algorithm or RTP circuit breaker,
>       which has not yet been standardised.
>=20
>    The following topology may be used, however it has considerations
>    worth noting:
>=20
>    o  Content modifying MCUs with RTCP termination (Topo-RTCP-
>       terminating-MCU) MAY be used.  Note that in this RTP Topology, =
RTP
>       loop detection and identification of active senders is the
>       responsibility of the WebRTC application; since the clients are
>       isolated from each other at the RTP layer, RTP cannot assist =
with
>       these functions (see section 3.9 of
>       [I-D.ietf-avtcore-rtp-topologies-update]).
>=20
>=20
> Please provide feedback on this text!
>=20
> Please see further response to questions and comments inline below.
>=20
>=20
> On 2014-04-07 15:44, Harald Alvestrand wrote:
> > On 04/07/2014 01:15 PM, Magnus Westerlund wrote:
> >> On 2014-04-07 10:12, Harald Alvestrand wrote:
>=20
> >>> In my (strong) opinion: This recommendation is wrong.
> >>>
> >>> All central conferencing units that use individual RTP sessions =
with
> >>> clients fall into this category. This includes, I believe, every
> >>> single multiuser video conferencing system based on WebRTC in
> >>> existence (and there are many of them).
> >> Do they? The point of this topology is that it terminates the =
source
> >> identification and hides when it is performing any type of mixing =
or
> >> switching operation. You can perform this with a back to back RTP
> >> session style without breaking this property. Just by correctly
> >> linking information between the sessions.
> >
> > Is there a definition for "correctly linking information"?
> > In particular, are you thinking of mappings that preserve SSRC?
> > Because that's not the ones I'm thinking of.
>=20
> The correctly linking information is mostly a question of preserving
CNAMEs
> across the middlebox to ensure that an end-point can determine
> synchronization contexts, and origin if CSRC are used. As we add
additional
> meta data about streams they may also needs to preserved depending on
> scope. Here it becomes a question of how one deals with APPID and =
MSID.
>=20
> Preserving SSRC is a choice, if one does it then loop detection works, =
if
one
> doesn't preserve it it doesn't. Preserving it avoids having to remap =
any
> information that have global scope across the legs.
>=20
> It becomes a choice on what complexities one needs to handle, common
> uniqueness requirements versus remapping I think is the major =
trade-off.
>=20
> >
> >>   Which means that the individual RTP session is from the
> >> identification part inseparable from an SFM or classic RTP mixer.
> >> Only the SSRC level loop detection gets impacted.
> >>
> >> I do understand that people have implemented it this way.
> >> Implementation are not necessarily equivalent to recommended =
practice.
> >
> > But when there's one common practice, and no recommended practice,
> > that might be a gentle hint that there's a reason why people are =
doing
> > it - indeed, it might be worth making it a recommended practice.
>=20
> You are welcome to propose changes regarding this to the =
topologies-update
> document in AVTCORE WG.
>=20
> >>
> >>> I believe the recommendation should be:
> >>>
> >>> * Content modifying MCUs with RTCP termination
> >>> (Topo-RTCP-terminating-MCU) MAY be used. Note that in this =
scenario,
> >>> RTP loop detection and identification of active senders is the
> >>> resposibility of the application; since the clients are isolated
> >>> from each other at the RTP layer, RTP cannot assist with these
functions.
> >>>
> >> If this topology was without issues, then one would simply remove =
the
> >> bullet, and let it fall under the general clause. However, it =
clearly
> >> needs that warning. Which results in this topology being both
> >> positively and negatively treated in regards to the other "you may
> >> encounter this topology".
> >
> > It might need that warning - I'm not sure it's a real issue; loop
> > detection is only an issue when you have potential loops, and in a
> > single-mixer architecture, that potential is minimal (each endpoint
> > knows perfectly well the difference between what it's sourcing and
> > what it's receiving). Identification of active senders can be an
> > application-layer thing.
>=20
> Yes, I agree that in a basic conference application with only a single
server
> instance this is unlikely to form a loop. If one starts doing more =
complex
things,
> including multi-server conferences or PeerConnection Meshes things =
become
> more complex. The later we anyway have removed the RTP loop detection =
due
> to the CNAME being unique on PeerConnection level, but I do think that =
is
the
> right trade-off from privacy point of view.
>=20
> >
> > Anyway, I stand by my statement: SHOULD NOT is the wrong thing to =
say.
>=20
> I do accept this, and are willing to change the recommendation.
>=20
> Cheers
>=20
> Magnus Westerlund
>=20
> ----------------------------------------------------------------------
> Services, Media and Network features, Ericsson Research EAB/TXM
> ----------------------------------------------------------------------
> Ericsson AB                 | Phone  +46 10 7148287
> F=E4r=F6gatan 6                 | Mobile +46 73 0949079
> SE-164 80 Stockholm, Sweden | mailto: magnus.westerlund@ericsson.com
> ----------------------------------------------------------------------
>=20
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb


From nobody Mon Apr 21 05:13:28 2014
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To: "'Simon Perreault'" <simon.perreault@viagenie.ca>, "'Pal Martinsen \(palmarti\)'" <palmarti@cisco.com>, "'Harald Alvestrand'" <harald@alvestrand.no>, "'Magnus Westerlund'" <magnus.westerlund@ericsson.com>
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Subject: Re: [rtcweb] Some language on "prioritization"
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1) AVOIDING SELF DESTRUCTING PRIORITAZATION FROM FILE TRANSFER IN DATA
CHANNEL

Simon,

I saw you are an author of TURN-TCP RFC6062. Doesn't that RFC allow a=20
webrtc file transfer channel (separate from the rtcweb data channel)=20
to be created, using TLS file transfer P2P? ICE/STUN/TURN would set=20
it up, NOT BUNDLED with chunk data and RTP media.

Freeing the current rtcweb data channel from file transfer would remedy=20
the self destruction of prioritization/QoS for data chunks and RTP media =

as explained in (A), (B) and (C) below.

I also think it would be much better for a web programmer to have a=20
separate p2p file transfer channel (that is not and cannot be =
prioritized)=20
than pouring files into the data channel.


2) MARKING PRIORITY DATA AND MEDIA FOR NETWORK LEVEL PRIORITIZATION/QoS

P=E5l, Harald, Magnus, Collins, Tiru, Dan,

We have previously discussed how to convey traffic info (bandwidth=20
requirement, and traffic type/prio) from the browser/application to=20
the network layer to prioritize or take other QoS measure by some=20
network element.

I have been pushing for inserting such traffic info into every RTP=20
media packet in the RTP header extension, see [1] below.

After looking how to mark data channel priority data chunks similarly,=20
I suggested [2] below, i.e. a new DTLS ContentType for the rtcweb=20
data channel, where the content would be the unencrypted traffic=20
info (preferable the same as in the RTP header extension) followed by=20
the SCTP data.=20

Are there other ways? - Maybe more similar, consistent and/or better?

We need the traffic info (contrary to e.g. DSCP bits) to be unchanged=20
end-to-end and available in both STUN and TURN flows for network=20
elements to see and act upon (e.g. to set DSCP bits or reserve=20
bandwidth or otherwise regulate) for various network types. The=20
traffic info must also be possible for network elements to find.

No way b!) Since webrtc now mandates DTLS-SRTP for media, cannot=20
that DTLS also hold the traffic info, just as for the data chunks=20
- instead of putting the traffic info into the RTP header extension?=20
NO, here the DTLS is only in the key exchange packets - not in every=20
packet so that would NOT work, unfortunately.

A way c?) Why not send data chunks in SRTP (instead of SCTP) and=20
reuse the RTP header extension for the traffic info for the network=20
layer. (File transfer is done in 1) above - not here!)

What other things from SCTP do we actually want? (Not everything=20
from that complex protocol for carrying telephony signaling can be=20
of interest for webrtc.)=20

- Priority (local): The browser could have a prioritizing mechanism=20
including both media and data chunks before putting it in the UDP=20
socket (simpler than SCTP).

- Reliable data: A small transaction layer with retransmission of=20
reliable data chunks could be implemented by the browser (simpler=20
than SCTP).

So with this way c, SCTP would go away totally and only the RTP=20
header extension would be used for traffic info to the network layer.


Are there more ways? We need this traffic info to allow WebRTC with=20
acceptable QoS over all network types (especially mobile Internet/OTT=20
LTE and Cable Networks that are of reservation type).


Below are also a few comments to other feedback.
=20
/Karl


On 10 Apr 2014, at 17:35, Paul Kyzivat [pkyzivat@alum.mit.edu] wrote:
I presume you mean each *concurrent* file transfer must use its own =
channel.
Right? Reusing the same channel to transfer a series of files should be
fine.

	Thanks,
	Paul
[Karl] Yes, That is right. /Karl

On 10 Apr 2014, at 15:14, Harald Alvestrand [harald@alvestrand.no] =
wrote:
Just checking... since there's been no comment to the main body of the =
text
here (it's all been on the SCTP aspect), is it OK to include the text in =
my
next version of the -transport draft?

(I'm not responding to Karl's comments - it's at a completely different
level of complexity.)

[Karl] Yes, it is but needs to be addressed. I think file transfer=20
just be taken out of the data channel, and a file transfer channel=20
(on a separate flow, as suggested above) be defined for Web=20
programmers to use. That would remedy (A), (B) and (C) below (p2p=20
file transfer not destroying priority for data chunks and media).=20
The file transfer channel would not/cannot be prioritized.=20

On 03 Apr 2014, at 21:12, Michael Tuexen
[Michael.Tuexen@lurchi.franken.de]wrote:
>Is there an
> overall design/thought somewhere that I missed/am unaware of?)
The basis idea is to use a stream scheduler to handle streams different.
Some data channels can get a higher bandwidth than others. Since the
priority stuff is still in flux in W3C, it hasn't been nailed down in =
the
SCTP spec. I guess, the place will be in
http://tools.ietf.org/html/draft-ietf-tsvwg-sctp-ndata-00
This document is not finalized yet.
>
[Karl] There are severe limitations for doing prioritization in a=20
protocol (SCTP or other) executed in the endpoints. In most cases=20
there are no packets in the endpoint to put first among queued up=20
traffic, because the queue is somewhere else between the endpoints.=20
Just consider the most common case where the endpoints are connected=20
with a fast Ethernet. The data has then already left the endpoint and=20
is queued up at some congestion point in the network where the SCTP=20
protocol cannot rearrange the queue. First when that unreachable=20
queue is full, there is something in the endpoint to act upon...


-----Ursprungligt meddelande-----
Fr=E5n: rtcweb [mailto:rtcweb-bounces@ietf.org] F=F6r Karl Stahl
Skickat: den 3 april 2014 18:30
Till: 'Matthew Kaufman (SKYPE)'; 'Harald Alvestrand'; 'Magnus =
Westerlund'
Kopia: rtcweb@ietf.org
=C4mne: Re: [rtcweb] Some language on "prioritization"

The questions raised here don't have simple "local or endpoint=20
answers". There are overall and basic things to be made clear first.=20
Combining P2P file transfer over the rtcweb data channels with=20
prioritized traffic chunks, brings further complications to the=20
prioritization/QoS issue. This cannot be handled/done by endpoint=20
or applications without considering the transport network (The=20
congestion to handle by prioritization is in the network - NOT in=20
the endpoints)!=20

There are below (A), (B) and (C) - to get us back to normal -=20
then there is (D) to improve - to allow the network level to=20
give us the prioritization required for WebRTC critical traffic=20
(real-time and data chunks). Without (D) - an interface to=20
network QoS equipment - the network does not even know what to=20
prioritize.

I am afraid that this will be considered "back to the drawing board"=20
rather than "some language" but I cannot see the way forward if not=20
considered.=20

Has the SCTP data channel been constructed with hopes of=20
prioritization/quality functionality that simply are not there=20
(cannot be there)? (I have browsed the (complex) SCTP spec and=20
some rtcweb drafts to get an understanding of the=20
prioritization/QoS thoughts and means in this rtcweb context=20
and the intended usage. Is there an overall design/thought=20
somewhere that I missed/am unaware of?)

File transfer - getting a larger amount of data through between=20
endpoints as quickly as possible - CANNOT be prioritized over=20
IP networks! For file transfer we normally use TCP, meaning we=20
fill the pipe, until THE ENDPOINTS see packet drop and regulate=20
down their traffic flow. File transfer over IP networks works=20
this way and cannot be improved upon ("Internet would stop" if=20
we prioritized such traffic! The available bandwidth at congestion=20
points are shared between everyone's TCP-flows hitting such=20
congestion points in the network.) File transfer's nature is=20
to have "infinite bandwidth demand" - the mechanism to handle=20
this is to allow file transfer to "take the rest of the bandwidth"=20
after prioritized media and chunks have taken their needed share.=20

The improvement to make - prioritization of media traffic and=20
data chunks - is to allow level 3 QoS methods - not only relying=20
on the level 4 TCP backing off process. Without level 3 QoS=20
(diffserve/DSCP, reservation or separate pipes) media and data=20
chunks packets are also lost in the level 4 TCP backing off=20
process (=3Dtoday's situation over best effort Internet). That is=20
where the prioritization/QoS measures that we need for rtcweb=20
have to be.

SCTP in itself (instead of TLS/TCP) does not/cannot/will not=20
improve things. It can destroy though if wrongly/badly=20
implemented or used!

>From the SCTP RFC4960: 7.  Congestion Control
   ...adequate resources will  be allocated to SCTP traffic=20
   to ensure prompt delivery of time-critical data ... unlikely,=20
   during normal operations, that transmissions encounter severe=20
   congestion conditions. =20

   However, SCTP must operate under adverse operational
   conditions, which can develop upon partial network failures=20
   or unexpected traffic surges.  In such situations, SCTP must=20
   follow correct congestion control steps to recover from=20
   congestion quickly in order to get data delivered as soon as=20
   possible. =20

   ... SCTP congestion control is always applied to the entire=20
   association, and not to individual streams.

Doing file transfer with the data channel will use SCTP in the=20
"adverse operational conditions, which can develop upon partial=20
network failures or unexpected traffic surges"-mode! Has that been=20
considered when constructing the data channel? Hope so...

Assuming SCTP is implemented correctly and as good as TCP for=20
file transfer (even though file transfer =3D filling the pipe=20
mode is an "adverse condition..." for SCTP). Then we must specify:


(A) Rtcweb file transfer MUST NOT use the same data channel as=20
data chunks expected to have prioritized delivery (otherwise,=20
as per above, SCTP will regulate the "the entire association"=20
to share the available bandwidth at the congestion point -=20
self-destroying the expected prioritized delivery of data chunks=20
in the same channel).

(B) Each rtcweb file transfer MUST use its own data channel (if=20
combined they risk being regulated randomly/unfairly by SCTP=20
itself and all files will have just have one single TCP-like=20
flow at the network congestion point when sharing the available=20
bandwidth with everyone else's flows at that congestion point=20
- That would be "negative priority" compared to what we are used=20
to in a browser.).

FURTHER, to allow the underlying network the ability to handle=20
the real-time and data chunks as least as good as "before rtcweb"=20
(Unfortunately this will make NAT-firewall less good - not sending=20
everything P2P in a single bundle/flow - I don't like it either...=20
Sorry...):

(C) File transfer using the data channel MUST NOT be bundled=20
with media and data chunk traffic in the same flow - file=20
transfer must use a separate socket creating another flow/5-tuple=20
over the network.=20
(In reservation type of networks (e.g. mobile OTT, cable), file=20
transfer - having no bandwidth limitation - would otherwise=20
overflow the reserved bandwidth for media and chunck traffic -=20
destroying prioritized delivery of media and chunck traffic=20
over that flow.) Such file transfer data channels MUST NOT be=20
prioritized by the underlying network (applies to all IP network=20
types, network prioritization of file transfer will STOP traffic=20
between endpoints
- Endpoint's TCP-like regulation has to function!). Several file=20
transfers could share such a separate flow (without specifying in=20
SDP I guess) but the underlying network must be able to separate=20
such flows (not to be prioritized) from media and chunk flows.=20

In the API, such file transfer data channel could be mapped to=20
the "below normal" classification (but would it not be better=20
with a "file transfer channel" API instead?). (And notice that=20
without (D) below, the network is not informed how to separate a=20
file transfer data channel flow from a prioritized data channel=20
flow.)


Having (A), (B) and (C) in place we are BACK TO THE CURRENT=20
Internet prioritization/QoS level - before the rtcweb p2p data=20
channel file transfer was introduced...

Now remains the prioritization improvement required for quality=20

hungry rtcweb! Both for the real-time media and the data chuncks.


(D) The underlying transport network needs to be informed about=20
the traffic requirements.=20

- In RTP media this traffic info could be inserted by the rtcweb=20
browser in the RTP header extension [1] depending on codec usage=20
etc. The API classification could be mapped to the=20
"Prioritize X(2 bits)" suggested in [1] below.

- for the data channel, the traffic info could be inserted as=20
suggested in previous email ([2] below)=20
http://www.ietf.org/mail-archive/web/rtcweb/current/msg11973.html =20

Note that the required bandwidth (only known by the=20
browser/application) has to be passed to reservation types of=20
network. Requested prioritization from the API or DSCP bits are=20
not sufficient. The browser knows the bandwidth requirements for=20
codecs, but for data chunks, the application has the knowledge.=20
In lack of a bandwidth parameter in the data channel API, a simple=20
convention - e.g. reserve 200 kbps for data chunks could serve as=20
a compromise (or 4 bandwidths mapped to the API prioritize=20
classifications...). A modified API with the bandwidth requirement=20
spelled out would of course be better for the chuck data channel.=20
For a file transfer data channel - prioritization classification=20
does not even apply. =20


Having prioritization over the network cleared, the application=20
should use that transport properly - having media and data chunks=20
transported with prioritization (as produced/inserted), NOT queuing=20
them up, rearranged by protocols or applications and NOT involving=20
file transfer (that has to be unprioritized).=20

Using DSCP bits if available through the OS in some situations does=20
not help either, as already pointed out in the transport document.=20
For (D) above, DSCP bits could in a few situations function locally,=20
but not end-to-end, so of no use generally.=20
=20
/Karl


[1] Traffic type information to encode (recreating the=20
idea/intention of the RTP payload type (PT) header that is not=20
available for the network level anymore, by instead having the=20
browser filling the RTP header extension with relevant traffic=20
info that all IP network types can use.), as suggested already=20
October 22. Two parameters to encode=20
http://www.ietf.org/mail-archive/web/rtcweb/current/msg09129.html=20

A) The maximum bandwidth requirement: Two bytes could contain=20
everything from some bps for real-time text to Gbps for future 3D=20
supersize telepresence on a logarithmic scale.
=20
B) The quality characteristics for the stream, with the highest=20
bit set to 1, we could allocate a bit each for quality type e.g:=20
Best Effort, Audio, Video, Supplemental Video, Gaming, Data,=20
Delay Insensitive (e.g. video streaming), Minimum Delay,=20
Reliable Delivery, Prioritize X(2 bits), Variation Y(2 bits),=20
that could be combined as required to describe the stream.

And with the highest bit set to 0, there could instead be a number=20
for special usage that does not fit the general description of=20
the individual bits.

Without this information, how could e.g. a reservation type=20
network (e.g. mobile OTT and cable) reserve the bandwidth=20
required for incoming rtcweb media or data chunks from a diffserve=20
network (carrying no bandwidth requirement)?=20

=20
[2] A new DTLS ContentType (e.g. 24 instead of current 23) for the=20
rtcweb data channel, where the content would be the unencrypted=20
traffic info (preferable the same as in the RTP header extension)=20
followed by the SCTP data as it is today (where it today comes=20
directly in the DTLS as the only content).:

TODAY:
Content Type: 23 (1 byte) content type 23 means =93application data=94
DTLS version:  0xfeff (2 byte)
Epoch: 0001 (2 byte)
Sequence number (e.g. 00 00 00 00 00 01) 6 byte
Length (e.g. 01 c0) 2 byte
Encrypted application data: ...... =3D the SCTP
=20
SUGGESTION WITH ADDED TRAFFIC INFO FOR LOWER LEVELS:=20
Content Type: 24 (1 byte) content type 24 meaning =93application data =
preceded
by traffic info=94
DTLS version:  0xfeff (2 byte)
Epoch: 0001 (2 byte)
Sequence number (e.g. 00 00 00 00 00 01) 6 byte
Length (e.g. 01 c0) 2 byte: length of the encrypted application data =
after
the new header extension
TRAFFIC INFO, Same as suggested for the RTP Header extension:=20
Encrypted application data: ...... ...... =3D the SCTP unchanged


-----Ursprungligt meddelande-----
Fr=E5n: rtcweb [mailto:rtcweb-bounces@ietf.org] F=F6r Matthew Kaufman =
(SKYPE)
Skickat: den 31 mars 2014 19:50
Till: Harald Alvestrand
Kopia: rtcweb@ietf.org
=C4mne: Re: [rtcweb] Some language on "prioritization"

And if SCTP and RTP have packets to send, which goes first, why, and =
under
what circumstances?

Matthew Kaufman

(Sent from my iPhone)

> On Mar 31, 2014, at 10:48 AM, "Harald Alvestrand" =
<harald@alvestrand.no>
wrote:
>=20
>> On 03/31/2014 07:42 PM, Michael Tuexen wrote:
>>> On 31 Mar 2014, at 19:38, Martin Thomson <martin.thomson@gmail.com>
wrote:
>>>=20
>>>> On 31 March 2014 10:08, Harald Alvestrand <harald@alvestrand.no> =
wrote:
>>>> -- TODO: Specify a relative priority scheme that makes sense with=20
>>>> SCTP, with an appropriate reference.=20
>>>> [draft-ietf-tsvwg-sctp-prpolicies]
>>>> specifies a priority policy, but it's about discarding packets, not =

>>>> deciding which packets to send first, and it also makes it=20
>>>> impossible to specify time-bounded retransmission. --
>>>=20
>>> Why would SCTP need special treatment?  I can understand if there=20
>>> are particular time-sensitive control messages that need to be given =

>>> higher priority, but this is all time-sensitive.
>> A single transport connection (in this case an SCTP association) can=20
>> only use a single DSCP. So it would be OK to use the same priority=20
>> for all data channels, but things get complicated when when some data =

>> channels would have different priorities requiring different DSCP
markings.
>=20
> The SCTP protocol machine will, at some given moment in time, have=20
> packets in its send buffers that belong to multiple data channels.=20
> Only one of them can go on the wire first.
>=20
> Which packet does it choose to send first, and why?
>=20
> That's the question I'm looking for an answer to.
>=20
> If the answer has an RFC number, chapter and section, I'd be very =
happy.
> If it doesn't - perhaps we have to invent one.
> Or say that it's "implementation dependent".
>=20



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Subject: Re: [rtcweb] Some language on "prioritization"
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It bothers me to see file transfer singled out for special treatment.

File transfer is one (common) example of a potentially high volume 
stream that has low requirements on latency. There are certainly other 
use cases with similar characteristics that are not file transfers.

If it isn't feasible to manage data channels with differing priority 
needs within the same SCTP association, then why not simply make 
provision for multiple SCTP associations, with a priority for each?

	Thanks,
	Paul

On 4/21/14 8:13 AM, Karl Stahl wrote:
> 1) AVOIDING SELF DESTRUCTING PRIORITAZATION FROM FILE TRANSFER IN DATA
> CHANNEL
>
> Simon,
>
> I saw you are an author of TURN-TCP RFC6062. Doesn't that RFC allow a
> webrtc file transfer channel (separate from the rtcweb data channel)
> to be created, using TLS file transfer P2P? ICE/STUN/TURN would set
> it up, NOT BUNDLED with chunk data and RTP media.
>
> Freeing the current rtcweb data channel from file transfer would remedy
> the self destruction of prioritization/QoS for data chunks and RTP media
> as explained in (A), (B) and (C) below.
>
> I also think it would be much better for a web programmer to have a
> separate p2p file transfer channel (that is not and cannot be prioritized)
> than pouring files into the data channel.
>
>
> 2) MARKING PRIORITY DATA AND MEDIA FOR NETWORK LEVEL PRIORITIZATION/QoS
>
> Pål, Harald, Magnus, Collins, Tiru, Dan,
>
> We have previously discussed how to convey traffic info (bandwidth
> requirement, and traffic type/prio) from the browser/application to
> the network layer to prioritize or take other QoS measure by some
> network element.
>
> I have been pushing for inserting such traffic info into every RTP
> media packet in the RTP header extension, see [1] below.
>
> After looking how to mark data channel priority data chunks similarly,
> I suggested [2] below, i.e. a new DTLS ContentType for the rtcweb
> data channel, where the content would be the unencrypted traffic
> info (preferable the same as in the RTP header extension) followed by
> the SCTP data.
>
> Are there other ways? - Maybe more similar, consistent and/or better?
>
> We need the traffic info (contrary to e.g. DSCP bits) to be unchanged
> end-to-end and available in both STUN and TURN flows for network
> elements to see and act upon (e.g. to set DSCP bits or reserve
> bandwidth or otherwise regulate) for various network types. The
> traffic info must also be possible for network elements to find.
>
> No way b!) Since webrtc now mandates DTLS-SRTP for media, cannot
> that DTLS also hold the traffic info, just as for the data chunks
> - instead of putting the traffic info into the RTP header extension?
> NO, here the DTLS is only in the key exchange packets - not in every
> packet so that would NOT work, unfortunately.
>
> A way c?) Why not send data chunks in SRTP (instead of SCTP) and
> reuse the RTP header extension for the traffic info for the network
> layer. (File transfer is done in 1) above - not here!)
>
> What other things from SCTP do we actually want? (Not everything
> from that complex protocol for carrying telephony signaling can be
> of interest for webrtc.)
>
> - Priority (local): The browser could have a prioritizing mechanism
> including both media and data chunks before putting it in the UDP
> socket (simpler than SCTP).
>
> - Reliable data: A small transaction layer with retransmission of
> reliable data chunks could be implemented by the browser (simpler
> than SCTP).
>
> So with this way c, SCTP would go away totally and only the RTP
> header extension would be used for traffic info to the network layer.
>
>
> Are there more ways? We need this traffic info to allow WebRTC with
> acceptable QoS over all network types (especially mobile Internet/OTT
> LTE and Cable Networks that are of reservation type).
>
>
> Below are also a few comments to other feedback.
>
> /Karl
>
>
> On 10 Apr 2014, at 17:35, Paul Kyzivat [pkyzivat@alum.mit.edu] wrote:
> I presume you mean each *concurrent* file transfer must use its own channel.
> Right? Reusing the same channel to transfer a series of files should be
> fine.
>
> 	Thanks,
> 	Paul
> [Karl] Yes, That is right. /Karl
>
> On 10 Apr 2014, at 15:14, Harald Alvestrand [harald@alvestrand.no] wrote:
> Just checking... since there's been no comment to the main body of the text
> here (it's all been on the SCTP aspect), is it OK to include the text in my
> next version of the -transport draft?
>
> (I'm not responding to Karl's comments - it's at a completely different
> level of complexity.)
>
> [Karl] Yes, it is but needs to be addressed. I think file transfer
> just be taken out of the data channel, and a file transfer channel
> (on a separate flow, as suggested above) be defined for Web
> programmers to use. That would remedy (A), (B) and (C) below (p2p
> file transfer not destroying priority for data chunks and media).
> The file transfer channel would not/cannot be prioritized.
>
> On 03 Apr 2014, at 21:12, Michael Tuexen
> [Michael.Tuexen@lurchi.franken.de]wrote:
>> Is there an
>> overall design/thought somewhere that I missed/am unaware of?)
> The basis idea is to use a stream scheduler to handle streams different.
> Some data channels can get a higher bandwidth than others. Since the
> priority stuff is still in flux in W3C, it hasn't been nailed down in the
> SCTP spec. I guess, the place will be in
> http://tools.ietf.org/html/draft-ietf-tsvwg-sctp-ndata-00
> This document is not finalized yet.
>>
> [Karl] There are severe limitations for doing prioritization in a
> protocol (SCTP or other) executed in the endpoints. In most cases
> there are no packets in the endpoint to put first among queued up
> traffic, because the queue is somewhere else between the endpoints.
> Just consider the most common case where the endpoints are connected
> with a fast Ethernet. The data has then already left the endpoint and
> is queued up at some congestion point in the network where the SCTP
> protocol cannot rearrange the queue. First when that unreachable
> queue is full, there is something in the endpoint to act upon...
>
>
> -----Ursprungligt meddelande-----
> Från: rtcweb [mailto:rtcweb-bounces@ietf.org] För Karl Stahl
> Skickat: den 3 april 2014 18:30
> Till: 'Matthew Kaufman (SKYPE)'; 'Harald Alvestrand'; 'Magnus Westerlund'
> Kopia: rtcweb@ietf.org
> Ämne: Re: [rtcweb] Some language on "prioritization"
>
> The questions raised here don't have simple "local or endpoint
> answers". There are overall and basic things to be made clear first.
> Combining P2P file transfer over the rtcweb data channels with
> prioritized traffic chunks, brings further complications to the
> prioritization/QoS issue. This cannot be handled/done by endpoint
> or applications without considering the transport network (The
> congestion to handle by prioritization is in the network - NOT in
> the endpoints)!
>
> There are below (A), (B) and (C) - to get us back to normal -
> then there is (D) to improve - to allow the network level to
> give us the prioritization required for WebRTC critical traffic
> (real-time and data chunks). Without (D) - an interface to
> network QoS equipment - the network does not even know what to
> prioritize.
>
> I am afraid that this will be considered "back to the drawing board"
> rather than "some language" but I cannot see the way forward if not
> considered.
>
> Has the SCTP data channel been constructed with hopes of
> prioritization/quality functionality that simply are not there
> (cannot be there)? (I have browsed the (complex) SCTP spec and
> some rtcweb drafts to get an understanding of the
> prioritization/QoS thoughts and means in this rtcweb context
> and the intended usage. Is there an overall design/thought
> somewhere that I missed/am unaware of?)
>
> File transfer - getting a larger amount of data through between
> endpoints as quickly as possible - CANNOT be prioritized over
> IP networks! For file transfer we normally use TCP, meaning we
> fill the pipe, until THE ENDPOINTS see packet drop and regulate
> down their traffic flow. File transfer over IP networks works
> this way and cannot be improved upon ("Internet would stop" if
> we prioritized such traffic! The available bandwidth at congestion
> points are shared between everyone's TCP-flows hitting such
> congestion points in the network.) File transfer's nature is
> to have "infinite bandwidth demand" - the mechanism to handle
> this is to allow file transfer to "take the rest of the bandwidth"
> after prioritized media and chunks have taken their needed share.
>
> The improvement to make - prioritization of media traffic and
> data chunks - is to allow level 3 QoS methods - not only relying
> on the level 4 TCP backing off process. Without level 3 QoS
> (diffserve/DSCP, reservation or separate pipes) media and data
> chunks packets are also lost in the level 4 TCP backing off
> process (=today's situation over best effort Internet). That is
> where the prioritization/QoS measures that we need for rtcweb
> have to be.
>
> SCTP in itself (instead of TLS/TCP) does not/cannot/will not
> improve things. It can destroy though if wrongly/badly
> implemented or used!
>
>>From the SCTP RFC4960: 7.  Congestion Control
>     ...adequate resources will  be allocated to SCTP traffic
>     to ensure prompt delivery of time-critical data ... unlikely,
>     during normal operations, that transmissions encounter severe
>     congestion conditions.
>
>     However, SCTP must operate under adverse operational
>     conditions, which can develop upon partial network failures
>     or unexpected traffic surges.  In such situations, SCTP must
>     follow correct congestion control steps to recover from
>     congestion quickly in order to get data delivered as soon as
>     possible.
>
>     ... SCTP congestion control is always applied to the entire
>     association, and not to individual streams.
>
> Doing file transfer with the data channel will use SCTP in the
> "adverse operational conditions, which can develop upon partial
> network failures or unexpected traffic surges"-mode! Has that been
> considered when constructing the data channel? Hope so...
>
> Assuming SCTP is implemented correctly and as good as TCP for
> file transfer (even though file transfer = filling the pipe
> mode is an "adverse condition..." for SCTP). Then we must specify:
>
>
> (A) Rtcweb file transfer MUST NOT use the same data channel as
> data chunks expected to have prioritized delivery (otherwise,
> as per above, SCTP will regulate the "the entire association"
> to share the available bandwidth at the congestion point -
> self-destroying the expected prioritized delivery of data chunks
> in the same channel).
>
> (B) Each rtcweb file transfer MUST use its own data channel (if
> combined they risk being regulated randomly/unfairly by SCTP
> itself and all files will have just have one single TCP-like
> flow at the network congestion point when sharing the available
> bandwidth with everyone else's flows at that congestion point
> - That would be "negative priority" compared to what we are used
> to in a browser.).
>
> FURTHER, to allow the underlying network the ability to handle
> the real-time and data chunks as least as good as "before rtcweb"
> (Unfortunately this will make NAT-firewall less good - not sending
> everything P2P in a single bundle/flow - I don't like it either...
> Sorry...):
>
> (C) File transfer using the data channel MUST NOT be bundled
> with media and data chunk traffic in the same flow - file
> transfer must use a separate socket creating another flow/5-tuple
> over the network.
> (In reservation type of networks (e.g. mobile OTT, cable), file
> transfer - having no bandwidth limitation - would otherwise
> overflow the reserved bandwidth for media and chunck traffic -
> destroying prioritized delivery of media and chunck traffic
> over that flow.) Such file transfer data channels MUST NOT be
> prioritized by the underlying network (applies to all IP network
> types, network prioritization of file transfer will STOP traffic
> between endpoints
> - Endpoint's TCP-like regulation has to function!). Several file
> transfers could share such a separate flow (without specifying in
> SDP I guess) but the underlying network must be able to separate
> such flows (not to be prioritized) from media and chunk flows.
>
> In the API, such file transfer data channel could be mapped to
> the "below normal" classification (but would it not be better
> with a "file transfer channel" API instead?). (And notice that
> without (D) below, the network is not informed how to separate a
> file transfer data channel flow from a prioritized data channel
> flow.)
>
>
> Having (A), (B) and (C) in place we are BACK TO THE CURRENT
> Internet prioritization/QoS level - before the rtcweb p2p data
> channel file transfer was introduced...
>
> Now remains the prioritization improvement required for quality
>
> hungry rtcweb! Both for the real-time media and the data chuncks.
>
>
> (D) The underlying transport network needs to be informed about
> the traffic requirements.
>
> - In RTP media this traffic info could be inserted by the rtcweb
> browser in the RTP header extension [1] depending on codec usage
> etc. The API classification could be mapped to the
> "Prioritize X(2 bits)" suggested in [1] below.
>
> - for the data channel, the traffic info could be inserted as
> suggested in previous email ([2] below)
> http://www.ietf.org/mail-archive/web/rtcweb/current/msg11973.html
>
> Note that the required bandwidth (only known by the
> browser/application) has to be passed to reservation types of
> network. Requested prioritization from the API or DSCP bits are
> not sufficient. The browser knows the bandwidth requirements for
> codecs, but for data chunks, the application has the knowledge.
> In lack of a bandwidth parameter in the data channel API, a simple
> convention - e.g. reserve 200 kbps for data chunks could serve as
> a compromise (or 4 bandwidths mapped to the API prioritize
> classifications...). A modified API with the bandwidth requirement
> spelled out would of course be better for the chuck data channel.
> For a file transfer data channel - prioritization classification
> does not even apply.
>
>
> Having prioritization over the network cleared, the application
> should use that transport properly - having media and data chunks
> transported with prioritization (as produced/inserted), NOT queuing
> them up, rearranged by protocols or applications and NOT involving
> file transfer (that has to be unprioritized).
>
> Using DSCP bits if available through the OS in some situations does
> not help either, as already pointed out in the transport document.
> For (D) above, DSCP bits could in a few situations function locally,
> but not end-to-end, so of no use generally.
>
> /Karl
>
>
> [1] Traffic type information to encode (recreating the
> idea/intention of the RTP payload type (PT) header that is not
> available for the network level anymore, by instead having the
> browser filling the RTP header extension with relevant traffic
> info that all IP network types can use.), as suggested already
> October 22. Two parameters to encode
> http://www.ietf.org/mail-archive/web/rtcweb/current/msg09129.html
>
> A) The maximum bandwidth requirement: Two bytes could contain
> everything from some bps for real-time text to Gbps for future 3D
> supersize telepresence on a logarithmic scale.
>
> B) The quality characteristics for the stream, with the highest
> bit set to 1, we could allocate a bit each for quality type e.g:
> Best Effort, Audio, Video, Supplemental Video, Gaming, Data,
> Delay Insensitive (e.g. video streaming), Minimum Delay,
> Reliable Delivery, Prioritize X(2 bits), Variation Y(2 bits),
> that could be combined as required to describe the stream.
>
> And with the highest bit set to 0, there could instead be a number
> for special usage that does not fit the general description of
> the individual bits.
>
> Without this information, how could e.g. a reservation type
> network (e.g. mobile OTT and cable) reserve the bandwidth
> required for incoming rtcweb media or data chunks from a diffserve
> network (carrying no bandwidth requirement)?
>
>
> [2] A new DTLS ContentType (e.g. 24 instead of current 23) for the
> rtcweb data channel, where the content would be the unencrypted
> traffic info (preferable the same as in the RTP header extension)
> followed by the SCTP data as it is today (where it today comes
> directly in the DTLS as the only content).:
>
> TODAY:
> Content Type: 23 (1 byte) content type 23 means “application data”
> DTLS version:  0xfeff (2 byte)
> Epoch: 0001 (2 byte)
> Sequence number (e.g. 00 00 00 00 00 01) 6 byte
> Length (e.g. 01 c0) 2 byte
> Encrypted application data: ...... = the SCTP
>
> SUGGESTION WITH ADDED TRAFFIC INFO FOR LOWER LEVELS:
> Content Type: 24 (1 byte) content type 24 meaning “application data preceded
> by traffic info”
> DTLS version:  0xfeff (2 byte)
> Epoch: 0001 (2 byte)
> Sequence number (e.g. 00 00 00 00 00 01) 6 byte
> Length (e.g. 01 c0) 2 byte: length of the encrypted application data after
> the new header extension
> TRAFFIC INFO, Same as suggested for the RTP Header extension:
> Encrypted application data: ...... ...... = the SCTP unchanged
>
>
> -----Ursprungligt meddelande-----
> Från: rtcweb [mailto:rtcweb-bounces@ietf.org] För Matthew Kaufman (SKYPE)
> Skickat: den 31 mars 2014 19:50
> Till: Harald Alvestrand
> Kopia: rtcweb@ietf.org
> Ämne: Re: [rtcweb] Some language on "prioritization"
>
> And if SCTP and RTP have packets to send, which goes first, why, and under
> what circumstances?
>
> Matthew Kaufman
>
> (Sent from my iPhone)
>
>> On Mar 31, 2014, at 10:48 AM, "Harald Alvestrand" <harald@alvestrand.no>
> wrote:
>>
>>> On 03/31/2014 07:42 PM, Michael Tuexen wrote:
>>>> On 31 Mar 2014, at 19:38, Martin Thomson <martin.thomson@gmail.com>
> wrote:
>>>>
>>>>> On 31 March 2014 10:08, Harald Alvestrand <harald@alvestrand.no> wrote:
>>>>> -- TODO: Specify a relative priority scheme that makes sense with
>>>>> SCTP, with an appropriate reference.
>>>>> [draft-ietf-tsvwg-sctp-prpolicies]
>>>>> specifies a priority policy, but it's about discarding packets, not
>>>>> deciding which packets to send first, and it also makes it
>>>>> impossible to specify time-bounded retransmission. --
>>>>
>>>> Why would SCTP need special treatment?  I can understand if there
>>>> are particular time-sensitive control messages that need to be given
>>>> higher priority, but this is all time-sensitive.
>>> A single transport connection (in this case an SCTP association) can
>>> only use a single DSCP. So it would be OK to use the same priority
>>> for all data channels, but things get complicated when when some data
>>> channels would have different priorities requiring different DSCP
> markings.
>>
>> The SCTP protocol machine will, at some given moment in time, have
>> packets in its send buffers that belong to multiple data channels.
>> Only one of them can go on the wire first.
>>
>> Which packet does it choose to send first, and why?
>>
>> That's the question I'm looking for an answer to.
>>
>> If the answer has an RFC number, chapter and section, I'd be very happy.
>> If it doesn't - perhaps we have to invent one.
>> Or say that it's "implementation dependent".
>>
>
>
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb
>


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Inline comments/explanations below. /Karl

-----Ursprungligt meddelande-----
Fr=E5n: rtcweb [mailto:rtcweb-bounces@ietf.org] F=F6r Paul Kyzivat
Skickat: den 21 april 2014 16:45
Till: rtcweb@ietf.org
=C4mne: Re: [rtcweb] Some language on "prioritization"

It bothers me to see file transfer singled out for special treatment.

File transfer is one (common) example of a potentially high volume =
stream
that has low requirements on latency. There are certainly other use =
cases
with similar characteristics that are not file transfers.


--- When I say "file transfer" I mean ALL traffic of the character =
"transfer
this high amount of data as fast as possible", which usually is done by =
TCP,
allowing all "bandwidth left for me at the congestion point" to be used =
(by
the TCP endpoints regulating their flow to share the available bandwidth =
at
the congestion point). That cannot be transferred better or with higher
priority, due to its "maximum bandwidth for minimum time" nature. =
Congestion
always occur - it is the way the available bandwidth is shared (best). =
That
is what we use TCP for.

--- "Stream" I would reserve for traffic ticking along with time, NOT
filling the pipe. That could be real-time data and be prioritized or
reserved bandwidth for.=20

--- If you know your bandwidth at the congestion point (which you don't =
-
TCP tries it out) you could convert a file transfer into a stream by
spreading it out over time, NOT filling the pipe. But then it would just
take longer time to get the file transferred. File transfer =
cannot/should
not be prioritized.  =20

--- These traffic types are different and need to be so.

If it isn't feasible to manage data channels with differing priority =
needs
within the same SCTP association, then why not simply make provision for
multiple SCTP associations, with a priority for each?

--- Yes multiple SCTP associations are possible as said in (B) below =
when
file transfer is involved. However, for the sharing of bandwidth at
congestion in the network, such SCTP associations should NOT be bundled =
over
the same flow/5-tuple as the real-time traffic (=3DC below). You then =
need to
set up another ICE connection - That is where I now suggested to do it =
with
TLS instead (which already is perfect for file transfer) instead of SCTP
(which could be used if working equally well, although much more =
complex).
Neither can/should prioritize file transfer though. =20



	Thanks,
	Paul

On 4/21/14 8:13 AM, Karl Stahl wrote:
> 1) AVOIDING SELF DESTRUCTING PRIORITAZATION FROM FILE TRANSFER IN DATA =

> CHANNEL
>
> Simon,
>
> I saw you are an author of TURN-TCP RFC6062. Doesn't that RFC allow a=20
> webrtc file transfer channel (separate from the rtcweb data channel)=20
> to be created, using TLS file transfer P2P? ICE/STUN/TURN would set it =

> up, NOT BUNDLED with chunk data and RTP media.
>
> Freeing the current rtcweb data channel from file transfer would=20
> remedy the self destruction of prioritization/QoS for data chunks and=20
> RTP media as explained in (A), (B) and (C) below.
>
> I also think it would be much better for a web programmer to have a=20
> separate p2p file transfer channel (that is not and cannot be=20
> prioritized) than pouring files into the data channel.
>
>
> 2) MARKING PRIORITY DATA AND MEDIA FOR NETWORK LEVEL=20
> PRIORITIZATION/QoS
>
> P=E5l, Harald, Magnus, Collins, Tiru, Dan,
>
> We have previously discussed how to convey traffic info (bandwidth=20
> requirement, and traffic type/prio) from the browser/application to=20
> the network layer to prioritize or take other QoS measure by some=20
> network element.
>
> I have been pushing for inserting such traffic info into every RTP=20
> media packet in the RTP header extension, see [1] below.
>
> After looking how to mark data channel priority data chunks similarly, =

> I suggested [2] below, i.e. a new DTLS ContentType for the rtcweb data =

> channel, where the content would be the unencrypted traffic info=20
> (preferable the same as in the RTP header extension) followed by the=20
> SCTP data.
>
> Are there other ways? - Maybe more similar, consistent and/or better?
>
> We need the traffic info (contrary to e.g. DSCP bits) to be unchanged=20
> end-to-end and available in both STUN and TURN flows for network=20
> elements to see and act upon (e.g. to set DSCP bits or reserve=20
> bandwidth or otherwise regulate) for various network types. The=20
> traffic info must also be possible for network elements to find.
>
> No way b!) Since webrtc now mandates DTLS-SRTP for media, cannot that=20
> DTLS also hold the traffic info, just as for the data chunks
> - instead of putting the traffic info into the RTP header extension?
> NO, here the DTLS is only in the key exchange packets - not in every=20
> packet so that would NOT work, unfortunately.
>
> A way c?) Why not send data chunks in SRTP (instead of SCTP) and reuse =

> the RTP header extension for the traffic info for the network layer.=20
> (File transfer is done in 1) above - not here!)
>
> What other things from SCTP do we actually want? (Not everything from=20
> that complex protocol for carrying telephony signaling can be of=20
> interest for webrtc.)
>
> - Priority (local): The browser could have a prioritizing mechanism=20
> including both media and data chunks before putting it in the UDP=20
> socket (simpler than SCTP).
>
> - Reliable data: A small transaction layer with retransmission of=20
> reliable data chunks could be implemented by the browser (simpler than =

> SCTP).
>
> So with this way c, SCTP would go away totally and only the RTP header =

> extension would be used for traffic info to the network layer.
>
>
> Are there more ways? We need this traffic info to allow WebRTC with=20
> acceptable QoS over all network types (especially mobile Internet/OTT=20
> LTE and Cable Networks that are of reservation type).
>
>
> Below are also a few comments to other feedback.
>
> /Karl
>
>
> On 10 Apr 2014, at 17:35, Paul Kyzivat [pkyzivat@alum.mit.edu] wrote:
> I presume you mean each *concurrent* file transfer must use its own
channel.
> Right? Reusing the same channel to transfer a series of files should=20
> be fine.
>
> 	Thanks,
> 	Paul
> [Karl] Yes, That is right. /Karl
>
> On 10 Apr 2014, at 15:14, Harald Alvestrand [harald@alvestrand.no] =
wrote:
> Just checking... since there's been no comment to the main body of the =

> text here (it's all been on the SCTP aspect), is it OK to include the=20
> text in my next version of the -transport draft?
>
> (I'm not responding to Karl's comments - it's at a completely=20
> different level of complexity.)
>
> [Karl] Yes, it is but needs to be addressed. I think file transfer=20
> just be taken out of the data channel, and a file transfer channel (on =

> a separate flow, as suggested above) be defined for Web programmers to =

> use. That would remedy (A), (B) and (C) below (p2p file transfer not=20
> destroying priority for data chunks and media).
> The file transfer channel would not/cannot be prioritized.
>
> On 03 Apr 2014, at 21:12, Michael Tuexen
> [Michael.Tuexen@lurchi.franken.de]wrote:
>> Is there an
>> overall design/thought somewhere that I missed/am unaware of?)
> The basis idea is to use a stream scheduler to handle streams =
different.
> Some data channels can get a higher bandwidth than others. Since the=20
> priority stuff is still in flux in W3C, it hasn't been nailed down in=20
> the SCTP spec. I guess, the place will be in
> http://tools.ietf.org/html/draft-ietf-tsvwg-sctp-ndata-00
> This document is not finalized yet.
>>
> [Karl] There are severe limitations for doing prioritization in a=20
> protocol (SCTP or other) executed in the endpoints. In most cases=20
> there are no packets in the endpoint to put first among queued up=20
> traffic, because the queue is somewhere else between the endpoints.
> Just consider the most common case where the endpoints are connected=20
> with a fast Ethernet. The data has then already left the endpoint and=20
> is queued up at some congestion point in the network where the SCTP=20
> protocol cannot rearrange the queue. First when that unreachable queue =

> is full, there is something in the endpoint to act upon...
>
>
> -----Ursprungligt meddelande-----
> Fr=E5n: rtcweb [mailto:rtcweb-bounces@ietf.org] F=F6r Karl Stahl
> Skickat: den 3 april 2014 18:30
> Till: 'Matthew Kaufman (SKYPE)'; 'Harald Alvestrand'; 'Magnus =
Westerlund'
> Kopia: rtcweb@ietf.org
> =C4mne: Re: [rtcweb] Some language on "prioritization"
>
> The questions raised here don't have simple "local or endpoint=20
> answers". There are overall and basic things to be made clear first.
> Combining P2P file transfer over the rtcweb data channels with=20
> prioritized traffic chunks, brings further complications to the=20
> prioritization/QoS issue. This cannot be handled/done by endpoint or=20
> applications without considering the transport network (The congestion =

> to handle by prioritization is in the network - NOT in the endpoints)!
>
> There are below (A), (B) and (C) - to get us back to normal - then=20
> there is (D) to improve - to allow the network level to give us the=20
> prioritization required for WebRTC critical traffic (real-time and=20
> data chunks). Without (D) - an interface to network QoS equipment -=20
> the network does not even know what to prioritize.
>
> I am afraid that this will be considered "back to the drawing board"
> rather than "some language" but I cannot see the way forward if not=20
> considered.
>
> Has the SCTP data channel been constructed with hopes of=20
> prioritization/quality functionality that simply are not there (cannot =

> be there)? (I have browsed the (complex) SCTP spec and some rtcweb=20
> drafts to get an understanding of the prioritization/QoS thoughts and=20
> means in this rtcweb context and the intended usage. Is there an=20
> overall design/thought somewhere that I missed/am unaware of?)
>
> File transfer - getting a larger amount of data through between=20
> endpoints as quickly as possible - CANNOT be prioritized over IP=20
> networks! For file transfer we normally use TCP, meaning we fill the=20
> pipe, until THE ENDPOINTS see packet drop and regulate down their=20
> traffic flow. File transfer over IP networks works this way and cannot =

> be improved upon ("Internet would stop" if we prioritized such=20
> traffic! The available bandwidth at congestion points are shared=20
> between everyone's TCP-flows hitting such congestion points in the=20
> network.) File transfer's nature is to have "infinite bandwidth=20
> demand" - the mechanism to handle this is to allow file transfer to=20
> "take the rest of the bandwidth"
> after prioritized media and chunks have taken their needed share.
>
> The improvement to make - prioritization of media traffic and data=20
> chunks - is to allow level 3 QoS methods - not only relying on the=20
> level 4 TCP backing off process. Without level 3 QoS (diffserve/DSCP,=20
> reservation or separate pipes) media and data chunks packets are also=20
> lost in the level 4 TCP backing off process (=3Dtoday's situation over =

> best effort Internet). That is where the prioritization/QoS measures=20
> that we need for rtcweb have to be.
>
> SCTP in itself (instead of TLS/TCP) does not/cannot/will not improve=20
> things. It can destroy though if wrongly/badly implemented or used!
>
>>From the SCTP RFC4960: 7.  Congestion Control
>     ...adequate resources will  be allocated to SCTP traffic
>     to ensure prompt delivery of time-critical data ... unlikely,
>     during normal operations, that transmissions encounter severe
>     congestion conditions.
>
>     However, SCTP must operate under adverse operational
>     conditions, which can develop upon partial network failures
>     or unexpected traffic surges.  In such situations, SCTP must
>     follow correct congestion control steps to recover from
>     congestion quickly in order to get data delivered as soon as
>     possible.
>
>     ... SCTP congestion control is always applied to the entire
>     association, and not to individual streams.
>
> Doing file transfer with the data channel will use SCTP in the=20
> "adverse operational conditions, which can develop upon partial=20
> network failures or unexpected traffic surges"-mode! Has that been=20
> considered when constructing the data channel? Hope so...
>
> Assuming SCTP is implemented correctly and as good as TCP for file=20
> transfer (even though file transfer =3D filling the pipe mode is an=20
> "adverse condition..." for SCTP). Then we must specify:
>
>
> (A) Rtcweb file transfer MUST NOT use the same data channel as data=20
> chunks expected to have prioritized delivery (otherwise, as per above, =

> SCTP will regulate the "the entire association"
> to share the available bandwidth at the congestion point -=20
> self-destroying the expected prioritized delivery of data chunks in=20
> the same channel).
>
> (B) Each rtcweb file transfer MUST use its own data channel (if=20
> combined they risk being regulated randomly/unfairly by SCTP itself=20
> and all files will have just have one single TCP-like flow at the=20
> network congestion point when sharing the available bandwidth with=20
> everyone else's flows at that congestion point
> - That would be "negative priority" compared to what we are used to in =

> a browser.).
>
> FURTHER, to allow the underlying network the ability to handle the=20
> real-time and data chunks as least as good as "before rtcweb"
> (Unfortunately this will make NAT-firewall less good - not sending=20
> everything P2P in a single bundle/flow - I don't like it either...
> Sorry...):
>
> (C) File transfer using the data channel MUST NOT be bundled with=20
> media and data chunk traffic in the same flow - file transfer must use =

> a separate socket creating another flow/5-tuple over the network.
> (In reservation type of networks (e.g. mobile OTT, cable), file=20
> transfer - having no bandwidth limitation - would otherwise overflow=20
> the reserved bandwidth for media and chunck traffic - destroying=20
> prioritized delivery of media and chunck traffic over that flow.) Such =

> file transfer data channels MUST NOT be prioritized by the underlying=20
> network (applies to all IP network types, network prioritization of=20
> file transfer will STOP traffic between endpoints
> - Endpoint's TCP-like regulation has to function!). Several file=20
> transfers could share such a separate flow (without specifying in SDP=20
> I guess) but the underlying network must be able to separate such=20
> flows (not to be prioritized) from media and chunk flows.
>
> In the API, such file transfer data channel could be mapped to the=20
> "below normal" classification (but would it not be better with a "file =

> transfer channel" API instead?). (And notice that without (D) below,=20
> the network is not informed how to separate a file transfer data=20
> channel flow from a prioritized data channel
> flow.)
>
>
> Having (A), (B) and (C) in place we are BACK TO THE CURRENT Internet=20
> prioritization/QoS level - before the rtcweb p2p data channel file=20
> transfer was introduced...
>
> Now remains the prioritization improvement required for quality
>
> hungry rtcweb! Both for the real-time media and the data chuncks.
>
>
> (D) The underlying transport network needs to be informed about the=20
> traffic requirements.
>
> - In RTP media this traffic info could be inserted by the rtcweb=20
> browser in the RTP header extension [1] depending on codec usage etc.=20
> The API classification could be mapped to the "Prioritize X(2 bits)"=20
> suggested in [1] below.
>
> - for the data channel, the traffic info could be inserted as=20
> suggested in previous email ([2] below)=20
> http://www.ietf.org/mail-archive/web/rtcweb/current/msg11973.html
>
> Note that the required bandwidth (only known by the
> browser/application) has to be passed to reservation types of network. =

> Requested prioritization from the API or DSCP bits are not sufficient. =

> The browser knows the bandwidth requirements for codecs, but for data=20
> chunks, the application has the knowledge.
> In lack of a bandwidth parameter in the data channel API, a simple=20
> convention - e.g. reserve 200 kbps for data chunks could serve as a=20
> compromise (or 4 bandwidths mapped to the API prioritize=20
> classifications...). A modified API with the bandwidth requirement=20
> spelled out would of course be better for the chuck data channel.
> For a file transfer data channel - prioritization classification does=20
> not even apply.
>
>
> Having prioritization over the network cleared, the application should =

> use that transport properly - having media and data chunks transported =

> with prioritization (as produced/inserted), NOT queuing them up,=20
> rearranged by protocols or applications and NOT involving file=20
> transfer (that has to be unprioritized).
>
> Using DSCP bits if available through the OS in some situations does=20
> not help either, as already pointed out in the transport document.
> For (D) above, DSCP bits could in a few situations function locally,=20
> but not end-to-end, so of no use generally.
>
> /Karl
>
>
> [1] Traffic type information to encode (recreating the idea/intention=20
> of the RTP payload type (PT) header that is not available for the=20
> network level anymore, by instead having the browser filling the RTP=20
> header extension with relevant traffic info that all IP network types=20
> can use.), as suggested already October 22. Two parameters to encode=20
> http://www.ietf.org/mail-archive/web/rtcweb/current/msg09129.html
>
> A) The maximum bandwidth requirement: Two bytes could contain=20
> everything from some bps for real-time text to Gbps for future 3D=20
> supersize telepresence on a logarithmic scale.
>
> B) The quality characteristics for the stream, with the highest bit=20
> set to 1, we could allocate a bit each for quality type e.g:
> Best Effort, Audio, Video, Supplemental Video, Gaming, Data, Delay=20
> Insensitive (e.g. video streaming), Minimum Delay, Reliable Delivery,=20
> Prioritize X(2 bits), Variation Y(2 bits), that could be combined as=20
> required to describe the stream.
>
> And with the highest bit set to 0, there could instead be a number for =

> special usage that does not fit the general description of the=20
> individual bits.
>
> Without this information, how could e.g. a reservation type network=20
> (e.g. mobile OTT and cable) reserve the bandwidth required for=20
> incoming rtcweb media or data chunks from a diffserve network=20
> (carrying no bandwidth requirement)?
>
>
> [2] A new DTLS ContentType (e.g. 24 instead of current 23) for the=20
> rtcweb data channel, where the content would be the unencrypted=20
> traffic info (preferable the same as in the RTP header extension)=20
> followed by the SCTP data as it is today (where it today comes=20
> directly in the DTLS as the only content).:
>
> TODAY:
> Content Type: 23 (1 byte) content type 23 means =93application data=94
> DTLS version:  0xfeff (2 byte)
> Epoch: 0001 (2 byte)
> Sequence number (e.g. 00 00 00 00 00 01) 6 byte Length (e.g. 01 c0) 2=20
> byte Encrypted application data: ...... =3D the SCTP
>
> SUGGESTION WITH ADDED TRAFFIC INFO FOR LOWER LEVELS:
> Content Type: 24 (1 byte) content type 24 meaning =93application data=20
> preceded by traffic info=94
> DTLS version:  0xfeff (2 byte)
> Epoch: 0001 (2 byte)
> Sequence number (e.g. 00 00 00 00 00 01) 6 byte Length (e.g. 01 c0) 2=20
> byte: length of the encrypted application data after the new header=20
> extension TRAFFIC INFO, Same as suggested for the RTP Header=20
> extension:
> Encrypted application data: ...... ...... =3D the SCTP unchanged
>
>
> -----Ursprungligt meddelande-----
> Fr=E5n: rtcweb [mailto:rtcweb-bounces@ietf.org] F=F6r Matthew Kaufman=20
> (SKYPE)
> Skickat: den 31 mars 2014 19:50
> Till: Harald Alvestrand
> Kopia: rtcweb@ietf.org
> =C4mne: Re: [rtcweb] Some language on "prioritization"
>
> And if SCTP and RTP have packets to send, which goes first, why, and=20
> under what circumstances?
>
> Matthew Kaufman
>
> (Sent from my iPhone)
>
>> On Mar 31, 2014, at 10:48 AM, "Harald Alvestrand"=20
>> <harald@alvestrand.no>
> wrote:
>>
>>> On 03/31/2014 07:42 PM, Michael Tuexen wrote:
>>>> On 31 Mar 2014, at 19:38, Martin Thomson <martin.thomson@gmail.com>
> wrote:
>>>>
>>>>> On 31 March 2014 10:08, Harald Alvestrand <harald@alvestrand.no>
wrote:
>>>>> -- TODO: Specify a relative priority scheme that makes sense with=20
>>>>> SCTP, with an appropriate reference.
>>>>> [draft-ietf-tsvwg-sctp-prpolicies]
>>>>> specifies a priority policy, but it's about discarding packets,=20
>>>>> not deciding which packets to send first, and it also makes it=20
>>>>> impossible to specify time-bounded retransmission. --
>>>>
>>>> Why would SCTP need special treatment?  I can understand if there=20
>>>> are particular time-sensitive control messages that need to be=20
>>>> given higher priority, but this is all time-sensitive.
>>> A single transport connection (in this case an SCTP association) can =

>>> only use a single DSCP. So it would be OK to use the same priority=20
>>> for all data channels, but things get complicated when when some=20
>>> data channels would have different priorities requiring different=20
>>> DSCP
> markings.
>>
>> The SCTP protocol machine will, at some given moment in time, have=20
>> packets in its send buffers that belong to multiple data channels.
>> Only one of them can go on the wire first.
>>
>> Which packet does it choose to send first, and why?
>>
>> That's the question I'm looking for an answer to.
>>
>> If the answer has an RFC number, chapter and section, I'd be very =
happy.
>> If it doesn't - perhaps we have to invent one.
>> Or say that it's "implementation dependent".
>>
>
>
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb
>

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


From nobody Tue Apr 22 06:46:42 2014
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Le 2014-04-21 08:13, Karl Stahl a écrit :
> 1) AVOIDING SELF DESTRUCTING PRIORITAZATION FROM FILE TRANSFER IN DATA
> CHANNEL
> 
> Simon,
> 
> I saw you are an author of TURN-TCP RFC6062. Doesn't that RFC allow a 
> webrtc file transfer channel (separate from the rtcweb data channel) 
> to be created, using TLS file transfer P2P? ICE/STUN/TURN would set 
> it up, NOT BUNDLED with chunk data and RTP media.

RFC6062 knows nothing about WebRTC or TLS inside the peer connection
payload. So it doesn't explicitly "allow" what you suggest. But I guess
it doesn't prevent it either. It doesn't care.

> Freeing the current rtcweb data channel from file transfer would remedy 
> the self destruction of prioritization/QoS for data chunks and RTP media 
> as explained in (A), (B) and (C) below.
> 
> I also think it would be much better for a web programmer to have a 
> separate p2p file transfer channel (that is not and cannot be prioritized) 
> than pouring files into the data channel.

Why do you think using a separate channel would eliminate the issue of
prioritization? We would end up with multiple channels contending for
the same bandwidth. You have just moved the problem one level up. And
made things needlessly more complex, IMHO.

Simon
-- 
DTN made easy, lean, and smart --> http://postellation.viagenie.ca
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From nobody Tue Apr 22 11:36:54 2014
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Subject: Re: [rtcweb] Some language on "prioritization"
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On 4/21/14 6:38 PM, Karl Stahl wrote:
> Inline comments/explanations below. /Karl
>
> -----Ursprungligt meddelande-----
> Från: rtcweb [mailto:rtcweb-bounces@ietf.org] För Paul Kyzivat
> Skickat: den 21 april 2014 16:45
> Till: rtcweb@ietf.org
> Ämne: Re: [rtcweb] Some language on "prioritization"
>
> It bothers me to see file transfer singled out for special treatment.
>
> File transfer is one (common) example of a potentially high volume stream
> that has low requirements on latency. There are certainly other use cases
> with similar characteristics that are not file transfers.
>
>
> --- When I say "file transfer" I mean ALL traffic of the character "transfer
> this high amount of data as fast as possible",

OK. So you are really talking about a usage with a particular traffic class.

> which usually is done by TCP,
> allowing all "bandwidth left for me at the congestion point" to be used (by
> the TCP endpoints regulating their flow to share the available bandwidth at
> the congestion point). That cannot be transferred better or with higher
> priority, due to its "maximum bandwidth for minimum time" nature. Congestion
> always occur - it is the way the available bandwidth is shared (best). That
> is what we use TCP for.
>
> --- "Stream" I would reserve for traffic ticking along with time, NOT
> filling the pipe. That could be real-time data and be prioritized or
> reserved bandwidth for.
>
> --- If you know your bandwidth at the congestion point (which you don't -
> TCP tries it out) you could convert a file transfer into a stream by
> spreading it out over time, NOT filling the pipe. But then it would just
> take longer time to get the file transferred. File transfer cannot/should
> not be prioritized.
>
> --- These traffic types are different and need to be so.
>
> If it isn't feasible to manage data channels with differing priority needs
> within the same SCTP association, then why not simply make provision for
> multiple SCTP associations, with a priority for each?
>
> --- Yes multiple SCTP associations are possible as said in (B) below when
> file transfer is involved. However, for the sharing of bandwidth at
> congestion in the network, such SCTP associations should NOT be bundled over
> the same flow/5-tuple as the real-time traffic (=C below). You then need to
> set up another ICE connection - That is where I now suggested to do it with
> TLS instead (which already is perfect for file transfer) instead of SCTP
> (which could be used if working equally well, although much more complex).
> Neither can/should prioritize file transfer though.

ISTM that avoiding setting up another 5-tuple, with the trouble of ICE, 
etc. is one of the goals. The data channels give you a cheap and easy 
way to set up these things.

IIUC there is already a desire/intent to mark the different streams 
within a bundle as different priorities/traffic classes. (I thought that 
is what DART is about.) Isn't that good enough to achieve the same end 
as running the data over a different 5-tuple?

Apparently there is a problem with doing that at the data channel level. 
But I wouldn't think it would be a problem for multiple SCTP 
associations in the same bundle. (Different SCTP port numbers.) That 
would allow you to have one association for "file transfers" and one for 
more interactive data.

	Thanks,
	Paul

> 	Thanks,
> 	Paul
>
> On 4/21/14 8:13 AM, Karl Stahl wrote:
>> 1) AVOIDING SELF DESTRUCTING PRIORITAZATION FROM FILE TRANSFER IN DATA
>> CHANNEL
>>
>> Simon,
>>
>> I saw you are an author of TURN-TCP RFC6062. Doesn't that RFC allow a
>> webrtc file transfer channel (separate from the rtcweb data channel)
>> to be created, using TLS file transfer P2P? ICE/STUN/TURN would set it
>> up, NOT BUNDLED with chunk data and RTP media.
>>
>> Freeing the current rtcweb data channel from file transfer would
>> remedy the self destruction of prioritization/QoS for data chunks and
>> RTP media as explained in (A), (B) and (C) below.
>>
>> I also think it would be much better for a web programmer to have a
>> separate p2p file transfer channel (that is not and cannot be
>> prioritized) than pouring files into the data channel.
>>
>>
>> 2) MARKING PRIORITY DATA AND MEDIA FOR NETWORK LEVEL
>> PRIORITIZATION/QoS
>>
>> Pål, Harald, Magnus, Collins, Tiru, Dan,
>>
>> We have previously discussed how to convey traffic info (bandwidth
>> requirement, and traffic type/prio) from the browser/application to
>> the network layer to prioritize or take other QoS measure by some
>> network element.
>>
>> I have been pushing for inserting such traffic info into every RTP
>> media packet in the RTP header extension, see [1] below.
>>
>> After looking how to mark data channel priority data chunks similarly,
>> I suggested [2] below, i.e. a new DTLS ContentType for the rtcweb data
>> channel, where the content would be the unencrypted traffic info
>> (preferable the same as in the RTP header extension) followed by the
>> SCTP data.
>>
>> Are there other ways? - Maybe more similar, consistent and/or better?
>>
>> We need the traffic info (contrary to e.g. DSCP bits) to be unchanged
>> end-to-end and available in both STUN and TURN flows for network
>> elements to see and act upon (e.g. to set DSCP bits or reserve
>> bandwidth or otherwise regulate) for various network types. The
>> traffic info must also be possible for network elements to find.
>>
>> No way b!) Since webrtc now mandates DTLS-SRTP for media, cannot that
>> DTLS also hold the traffic info, just as for the data chunks
>> - instead of putting the traffic info into the RTP header extension?
>> NO, here the DTLS is only in the key exchange packets - not in every
>> packet so that would NOT work, unfortunately.
>>
>> A way c?) Why not send data chunks in SRTP (instead of SCTP) and reuse
>> the RTP header extension for the traffic info for the network layer.
>> (File transfer is done in 1) above - not here!)
>>
>> What other things from SCTP do we actually want? (Not everything from
>> that complex protocol for carrying telephony signaling can be of
>> interest for webrtc.)
>>
>> - Priority (local): The browser could have a prioritizing mechanism
>> including both media and data chunks before putting it in the UDP
>> socket (simpler than SCTP).
>>
>> - Reliable data: A small transaction layer with retransmission of
>> reliable data chunks could be implemented by the browser (simpler than
>> SCTP).
>>
>> So with this way c, SCTP would go away totally and only the RTP header
>> extension would be used for traffic info to the network layer.
>>
>>
>> Are there more ways? We need this traffic info to allow WebRTC with
>> acceptable QoS over all network types (especially mobile Internet/OTT
>> LTE and Cable Networks that are of reservation type).
>>
>>
>> Below are also a few comments to other feedback.
>>
>> /Karl
>>
>>
>> On 10 Apr 2014, at 17:35, Paul Kyzivat [pkyzivat@alum.mit.edu] wrote:
>> I presume you mean each *concurrent* file transfer must use its own
> channel.
>> Right? Reusing the same channel to transfer a series of files should
>> be fine.
>>
>> 	Thanks,
>> 	Paul
>> [Karl] Yes, That is right. /Karl
>>
>> On 10 Apr 2014, at 15:14, Harald Alvestrand [harald@alvestrand.no] wrote:
>> Just checking... since there's been no comment to the main body of the
>> text here (it's all been on the SCTP aspect), is it OK to include the
>> text in my next version of the -transport draft?
>>
>> (I'm not responding to Karl's comments - it's at a completely
>> different level of complexity.)
>>
>> [Karl] Yes, it is but needs to be addressed. I think file transfer
>> just be taken out of the data channel, and a file transfer channel (on
>> a separate flow, as suggested above) be defined for Web programmers to
>> use. That would remedy (A), (B) and (C) below (p2p file transfer not
>> destroying priority for data chunks and media).
>> The file transfer channel would not/cannot be prioritized.
>>
>> On 03 Apr 2014, at 21:12, Michael Tuexen
>> [Michael.Tuexen@lurchi.franken.de]wrote:
>>> Is there an
>>> overall design/thought somewhere that I missed/am unaware of?)
>> The basis idea is to use a stream scheduler to handle streams different.
>> Some data channels can get a higher bandwidth than others. Since the
>> priority stuff is still in flux in W3C, it hasn't been nailed down in
>> the SCTP spec. I guess, the place will be in
>> http://tools.ietf.org/html/draft-ietf-tsvwg-sctp-ndata-00
>> This document is not finalized yet.
>>>
>> [Karl] There are severe limitations for doing prioritization in a
>> protocol (SCTP or other) executed in the endpoints. In most cases
>> there are no packets in the endpoint to put first among queued up
>> traffic, because the queue is somewhere else between the endpoints.
>> Just consider the most common case where the endpoints are connected
>> with a fast Ethernet. The data has then already left the endpoint and
>> is queued up at some congestion point in the network where the SCTP
>> protocol cannot rearrange the queue. First when that unreachable queue
>> is full, there is something in the endpoint to act upon...
>>
>>
>> -----Ursprungligt meddelande-----
>> Från: rtcweb [mailto:rtcweb-bounces@ietf.org] För Karl Stahl
>> Skickat: den 3 april 2014 18:30
>> Till: 'Matthew Kaufman (SKYPE)'; 'Harald Alvestrand'; 'Magnus Westerlund'
>> Kopia: rtcweb@ietf.org
>> Ämne: Re: [rtcweb] Some language on "prioritization"
>>
>> The questions raised here don't have simple "local or endpoint
>> answers". There are overall and basic things to be made clear first.
>> Combining P2P file transfer over the rtcweb data channels with
>> prioritized traffic chunks, brings further complications to the
>> prioritization/QoS issue. This cannot be handled/done by endpoint or
>> applications without considering the transport network (The congestion
>> to handle by prioritization is in the network - NOT in the endpoints)!
>>
>> There are below (A), (B) and (C) - to get us back to normal - then
>> there is (D) to improve - to allow the network level to give us the
>> prioritization required for WebRTC critical traffic (real-time and
>> data chunks). Without (D) - an interface to network QoS equipment -
>> the network does not even know what to prioritize.
>>
>> I am afraid that this will be considered "back to the drawing board"
>> rather than "some language" but I cannot see the way forward if not
>> considered.
>>
>> Has the SCTP data channel been constructed with hopes of
>> prioritization/quality functionality that simply are not there (cannot
>> be there)? (I have browsed the (complex) SCTP spec and some rtcweb
>> drafts to get an understanding of the prioritization/QoS thoughts and
>> means in this rtcweb context and the intended usage. Is there an
>> overall design/thought somewhere that I missed/am unaware of?)
>>
>> File transfer - getting a larger amount of data through between
>> endpoints as quickly as possible - CANNOT be prioritized over IP
>> networks! For file transfer we normally use TCP, meaning we fill the
>> pipe, until THE ENDPOINTS see packet drop and regulate down their
>> traffic flow. File transfer over IP networks works this way and cannot
>> be improved upon ("Internet would stop" if we prioritized such
>> traffic! The available bandwidth at congestion points are shared
>> between everyone's TCP-flows hitting such congestion points in the
>> network.) File transfer's nature is to have "infinite bandwidth
>> demand" - the mechanism to handle this is to allow file transfer to
>> "take the rest of the bandwidth"
>> after prioritized media and chunks have taken their needed share.
>>
>> The improvement to make - prioritization of media traffic and data
>> chunks - is to allow level 3 QoS methods - not only relying on the
>> level 4 TCP backing off process. Without level 3 QoS (diffserve/DSCP,
>> reservation or separate pipes) media and data chunks packets are also
>> lost in the level 4 TCP backing off process (=today's situation over
>> best effort Internet). That is where the prioritization/QoS measures
>> that we need for rtcweb have to be.
>>
>> SCTP in itself (instead of TLS/TCP) does not/cannot/will not improve
>> things. It can destroy though if wrongly/badly implemented or used!
>>
>> >From the SCTP RFC4960: 7.  Congestion Control
>>      ...adequate resources will  be allocated to SCTP traffic
>>      to ensure prompt delivery of time-critical data ... unlikely,
>>      during normal operations, that transmissions encounter severe
>>      congestion conditions.
>>
>>      However, SCTP must operate under adverse operational
>>      conditions, which can develop upon partial network failures
>>      or unexpected traffic surges.  In such situations, SCTP must
>>      follow correct congestion control steps to recover from
>>      congestion quickly in order to get data delivered as soon as
>>      possible.
>>
>>      ... SCTP congestion control is always applied to the entire
>>      association, and not to individual streams.
>>
>> Doing file transfer with the data channel will use SCTP in the
>> "adverse operational conditions, which can develop upon partial
>> network failures or unexpected traffic surges"-mode! Has that been
>> considered when constructing the data channel? Hope so...
>>
>> Assuming SCTP is implemented correctly and as good as TCP for file
>> transfer (even though file transfer = filling the pipe mode is an
>> "adverse condition..." for SCTP). Then we must specify:
>>
>>
>> (A) Rtcweb file transfer MUST NOT use the same data channel as data
>> chunks expected to have prioritized delivery (otherwise, as per above,
>> SCTP will regulate the "the entire association"
>> to share the available bandwidth at the congestion point -
>> self-destroying the expected prioritized delivery of data chunks in
>> the same channel).
>>
>> (B) Each rtcweb file transfer MUST use its own data channel (if
>> combined they risk being regulated randomly/unfairly by SCTP itself
>> and all files will have just have one single TCP-like flow at the
>> network congestion point when sharing the available bandwidth with
>> everyone else's flows at that congestion point
>> - That would be "negative priority" compared to what we are used to in
>> a browser.).
>>
>> FURTHER, to allow the underlying network the ability to handle the
>> real-time and data chunks as least as good as "before rtcweb"
>> (Unfortunately this will make NAT-firewall less good - not sending
>> everything P2P in a single bundle/flow - I don't like it either...
>> Sorry...):
>>
>> (C) File transfer using the data channel MUST NOT be bundled with
>> media and data chunk traffic in the same flow - file transfer must use
>> a separate socket creating another flow/5-tuple over the network.
>> (In reservation type of networks (e.g. mobile OTT, cable), file
>> transfer - having no bandwidth limitation - would otherwise overflow
>> the reserved bandwidth for media and chunck traffic - destroying
>> prioritized delivery of media and chunck traffic over that flow.) Such
>> file transfer data channels MUST NOT be prioritized by the underlying
>> network (applies to all IP network types, network prioritization of
>> file transfer will STOP traffic between endpoints
>> - Endpoint's TCP-like regulation has to function!). Several file
>> transfers could share such a separate flow (without specifying in SDP
>> I guess) but the underlying network must be able to separate such
>> flows (not to be prioritized) from media and chunk flows.
>>
>> In the API, such file transfer data channel could be mapped to the
>> "below normal" classification (but would it not be better with a "file
>> transfer channel" API instead?). (And notice that without (D) below,
>> the network is not informed how to separate a file transfer data
>> channel flow from a prioritized data channel
>> flow.)
>>
>>
>> Having (A), (B) and (C) in place we are BACK TO THE CURRENT Internet
>> prioritization/QoS level - before the rtcweb p2p data channel file
>> transfer was introduced...
>>
>> Now remains the prioritization improvement required for quality
>>
>> hungry rtcweb! Both for the real-time media and the data chuncks.
>>
>>
>> (D) The underlying transport network needs to be informed about the
>> traffic requirements.
>>
>> - In RTP media this traffic info could be inserted by the rtcweb
>> browser in the RTP header extension [1] depending on codec usage etc.
>> The API classification could be mapped to the "Prioritize X(2 bits)"
>> suggested in [1] below.
>>
>> - for the data channel, the traffic info could be inserted as
>> suggested in previous email ([2] below)
>> http://www.ietf.org/mail-archive/web/rtcweb/current/msg11973.html
>>
>> Note that the required bandwidth (only known by the
>> browser/application) has to be passed to reservation types of network.
>> Requested prioritization from the API or DSCP bits are not sufficient.
>> The browser knows the bandwidth requirements for codecs, but for data
>> chunks, the application has the knowledge.
>> In lack of a bandwidth parameter in the data channel API, a simple
>> convention - e.g. reserve 200 kbps for data chunks could serve as a
>> compromise (or 4 bandwidths mapped to the API prioritize
>> classifications...). A modified API with the bandwidth requirement
>> spelled out would of course be better for the chuck data channel.
>> For a file transfer data channel - prioritization classification does
>> not even apply.
>>
>>
>> Having prioritization over the network cleared, the application should
>> use that transport properly - having media and data chunks transported
>> with prioritization (as produced/inserted), NOT queuing them up,
>> rearranged by protocols or applications and NOT involving file
>> transfer (that has to be unprioritized).
>>
>> Using DSCP bits if available through the OS in some situations does
>> not help either, as already pointed out in the transport document.
>> For (D) above, DSCP bits could in a few situations function locally,
>> but not end-to-end, so of no use generally.
>>
>> /Karl
>>
>>
>> [1] Traffic type information to encode (recreating the idea/intention
>> of the RTP payload type (PT) header that is not available for the
>> network level anymore, by instead having the browser filling the RTP
>> header extension with relevant traffic info that all IP network types
>> can use.), as suggested already October 22. Two parameters to encode
>> http://www.ietf.org/mail-archive/web/rtcweb/current/msg09129.html
>>
>> A) The maximum bandwidth requirement: Two bytes could contain
>> everything from some bps for real-time text to Gbps for future 3D
>> supersize telepresence on a logarithmic scale.
>>
>> B) The quality characteristics for the stream, with the highest bit
>> set to 1, we could allocate a bit each for quality type e.g:
>> Best Effort, Audio, Video, Supplemental Video, Gaming, Data, Delay
>> Insensitive (e.g. video streaming), Minimum Delay, Reliable Delivery,
>> Prioritize X(2 bits), Variation Y(2 bits), that could be combined as
>> required to describe the stream.
>>
>> And with the highest bit set to 0, there could instead be a number for
>> special usage that does not fit the general description of the
>> individual bits.
>>
>> Without this information, how could e.g. a reservation type network
>> (e.g. mobile OTT and cable) reserve the bandwidth required for
>> incoming rtcweb media or data chunks from a diffserve network
>> (carrying no bandwidth requirement)?
>>
>>
>> [2] A new DTLS ContentType (e.g. 24 instead of current 23) for the
>> rtcweb data channel, where the content would be the unencrypted
>> traffic info (preferable the same as in the RTP header extension)
>> followed by the SCTP data as it is today (where it today comes
>> directly in the DTLS as the only content).:
>>
>> TODAY:
>> Content Type: 23 (1 byte) content type 23 means “application data”
>> DTLS version:  0xfeff (2 byte)
>> Epoch: 0001 (2 byte)
>> Sequence number (e.g. 00 00 00 00 00 01) 6 byte Length (e.g. 01 c0) 2
>> byte Encrypted application data: ...... = the SCTP
>>
>> SUGGESTION WITH ADDED TRAFFIC INFO FOR LOWER LEVELS:
>> Content Type: 24 (1 byte) content type 24 meaning “application data
>> preceded by traffic info”
>> DTLS version:  0xfeff (2 byte)
>> Epoch: 0001 (2 byte)
>> Sequence number (e.g. 00 00 00 00 00 01) 6 byte Length (e.g. 01 c0) 2
>> byte: length of the encrypted application data after the new header
>> extension TRAFFIC INFO, Same as suggested for the RTP Header
>> extension:
>> Encrypted application data: ...... ...... = the SCTP unchanged
>>
>>
>> -----Ursprungligt meddelande-----
>> Från: rtcweb [mailto:rtcweb-bounces@ietf.org] För Matthew Kaufman
>> (SKYPE)
>> Skickat: den 31 mars 2014 19:50
>> Till: Harald Alvestrand
>> Kopia: rtcweb@ietf.org
>> Ämne: Re: [rtcweb] Some language on "prioritization"
>>
>> And if SCTP and RTP have packets to send, which goes first, why, and
>> under what circumstances?
>>
>> Matthew Kaufman
>>
>> (Sent from my iPhone)
>>
>>> On Mar 31, 2014, at 10:48 AM, "Harald Alvestrand"
>>> <harald@alvestrand.no>
>> wrote:
>>>
>>>> On 03/31/2014 07:42 PM, Michael Tuexen wrote:
>>>>> On 31 Mar 2014, at 19:38, Martin Thomson <martin.thomson@gmail.com>
>> wrote:
>>>>>
>>>>>> On 31 March 2014 10:08, Harald Alvestrand <harald@alvestrand.no>
> wrote:
>>>>>> -- TODO: Specify a relative priority scheme that makes sense with
>>>>>> SCTP, with an appropriate reference.
>>>>>> [draft-ietf-tsvwg-sctp-prpolicies]
>>>>>> specifies a priority policy, but it's about discarding packets,
>>>>>> not deciding which packets to send first, and it also makes it
>>>>>> impossible to specify time-bounded retransmission. --
>>>>>
>>>>> Why would SCTP need special treatment?  I can understand if there
>>>>> are particular time-sensitive control messages that need to be
>>>>> given higher priority, but this is all time-sensitive.
>>>> A single transport connection (in this case an SCTP association) can
>>>> only use a single DSCP. So it would be OK to use the same priority
>>>> for all data channels, but things get complicated when when some
>>>> data channels would have different priorities requiring different
>>>> DSCP
>> markings.
>>>
>>> The SCTP protocol machine will, at some given moment in time, have
>>> packets in its send buffers that belong to multiple data channels.
>>> Only one of them can go on the wire first.
>>>
>>> Which packet does it choose to send first, and why?
>>>
>>> That's the question I'm looking for an answer to.
>>>
>>> If the answer has an RFC number, chapter and section, I'd be very happy.
>>> If it doesn't - perhaps we have to invent one.
>>> Or say that it's "implementation dependent".
>>>
>>
>>
>> _______________________________________________
>> 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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-----Ursprungligt meddelande-----
Fr=E5n: Simon Perreault [mailto:simon.perreault@viagenie.ca]=20
Skickat: den 22 april 2014 15:46
Till: Karl Stahl; 'Pal Martinsen (palmarti)'; 'Harald Alvestrand'; =
'Magnus
Westerlund'
Kopia: rtcweb@ietf.org
=C4mne: Re: SV: [rtcweb] Some language on "prioritization"

Le 2014-04-21 08:13, Karl Stahl a =E9crit :
> 1) AVOIDING SELF DESTRUCTING PRIORITAZATION FROM FILE TRANSFER IN DATA =

> CHANNEL
>=20
> Simon,
>=20
> I saw you are an author of TURN-TCP RFC6062. Doesn't that RFC allow a=20
> webrtc file transfer channel (separate from the rtcweb data channel)=20
> to be created, using TLS file transfer P2P? ICE/STUN/TURN would set it =

> up, NOT BUNDLED with chunk data and RTP media.

RFC6062 knows nothing about WebRTC or TLS inside the peer connection
payload. So it doesn't explicitly "allow" what you suggest. But I guess =
it
doesn't prevent it either. It doesn't care.

[Karl] OK (just realized TURN was not limited to setting up UDP with
TURN-TCP RFC6062 now available), so meant "allow" in the sense: Fine, we =
can
use TLS for file-transfer, rather than new/complex UDP with DTLS with =
SCTP.=20

> Freeing the current rtcweb data channel from file transfer would=20
> remedy the self destruction of prioritization/QoS for data chunks and=20
> RTP media as explained in (A), (B) and (C) below.
>=20
> I also think it would be much better for a web programmer to have a=20
> separate p2p file transfer channel (that is not and cannot be=20
> prioritized) than pouring files into the data channel.

Why do you think using a separate channel would eliminate the issue of
prioritization? We would end up with multiple channels contending for =
the
same bandwidth. You have just moved the problem one level up. And made
things needlessly more complex, IMHO.

[Karl] The (A) and (B) and (C) at the bottom in the email you answered
DESTROYS the QoS/Prioritization we are used over the Internet (By SCTP =
data
channel mixing real-time stream data with filling the pipe (congesting
creating) file transfer data in the same bundle/flow/5-tuple).

SCTP's own prioritization mechanism can only act on what is in the
endpoint's buffers - SCTP cannot help/remedy congestion happening IN THE
NETWORK where its data transfer has caused congestion when competing =
with
other parties file transfer TCP flows, and there filled a network buffer
with the mixture of our own file transfer AND REAL-TIME data.=20

Yes, pouring a file into current data channel will even cause QoS damage =
to
our own bundled Video/Voice that SCTP's prio-mechanism cannot fix, =
because
it cannot rearrange what is queued up in a Network buffer.

First after getting back to having file transfer separated out of the =
bundle
in another flow/5-tuple can we start thinking about improving
prioritization/QoS for real-time data.


[Karl] I wrote this sometime before, regarding how QoS over the internet
works in general - may be useful for understanding

For better understanding of these QoS/QoE problems, and methods for
improving (not specifically discussing the policies of sharing real-time
traffic space), I would like to explain:

Over an IP pipe only used for real-traffic (no TCP data traffic), it is
sufficient that the pipe is wide enough for good QoS. That is often used =
and
implemented by separating IP pipes at a lower level using e.g. Ethernet
VLAN, MPLS, Ethernet over ATM (std for ADSL modems). TURN servers can
enforce real-traffic into such pipes and QoS is achieved. Let=92s call =
this
Level 2 QoS (Network level)

Over an IP pipe shared between quality requiring real-time traffic and =
less
demanding data or streaming (usually TCP) traffic, we have the sharing
between these two traffic classes to consider. The main method (and the
mechanism making today=92s real-time QoE as good as it often is) is that =
TCP
endpoints back-off and share their bandwidth usage. Real-time traffic =
using
UDP transport do not back-off, the endpoints using UDP occupy the =
bandwidth
needed. When an IP pipe gets filled, it is all endpoint=92s TCP =
bandwidth
usage that is back-off and shared between them, leaving room for the UDP
traffic. This is mechanism we experience everyday over the Internet, =
using
our =93Surf IP pipes=94. Let=92s call this Level 4 QoS (Transport level)

However, this Level 4 QoS is based on that at congestion times (which =
happen
every time we click =96 setting up a TCP flow and transferring some =
amount of
data as quick as possible) the router handling the most narrow part of =
the
pipe (the congestion point) drop packets. It is this packet dropping =
that
(i) signals to TCP endpoints to reduce their bandwidth (via TCP=92s =
error
correction/retransmission mechanism) and (ii) destroys the QoE of =
real-time
traffic. (Both TCP and UDP packets are dropped in this process that is
triggered by flow intensive TCP traffic.)

We need (i) but don=92t want (ii) and to improve on this we can e.g. use
diffserve, DSCP bits in IP packets to instruct routers to always forward =
the
real-time traffic before any unmarked TCP traffic (which usually fills =
most
of the pipe). Then QoS is then achieved for real-time traffic. This is =
Level
3 QoS (IP level). The method used is =93traffic shaping=94: backing off =
data
traffic, leaving real-time traffic free passage without packet loss.

Here, in TRAM we want to go beyond Level 4 QoS (already available and
working as good as it can on the Internet), to give quality demanding =
WebRTC
real-time traffic better QoE by:
a. Forcing real-time traffic into IP-pipes having Level 2 QoS (using
auto-discovered TURN servers)=20
or
b. Forcing real-time traffic into IP-pipes having Level 3 QoS (using
auto-discovered TURN servers). Here we must have traffic shaping =
mechanisms
working, and with correct and sufficient information to do the job. This =
is
why we in TRAM discuss DISCUSS/MALICE,
draft-thomson-tram-turn-bandwidth-00.txt attributes and recreating the
payload type (PT) idea/intent in RTP packets (by now conveying bandwidth =
and
traffic type in the RTP extension header
http://www.ietf.org/mail-archive/web/rtcweb/current/msg09129.html.

How these QoS/QoE things are done and used in practice is also
illustrated/exemplified in this email discussion:
http://www.ietf.org/mail-archive/web/rtcweb/current/msg09031.html=20

Prioritization of individual real-time packets (e.g. using diffserve =
DSCP
bits) have today little impact on the QoE, UNLESS a pipe full with only
real-time traffic (which is unusual), because in today=92s IP pipes with =
high
bandwidth, packets are not stored for later delivery, but rather dropped =
by
routers implementing diffserve (after only a short time period =3D small
buffer size). The only good remedy is higher bandwidth, that can handle =
ALL
real-time traffic.=20

Prioritization of real-time packets (any diffserve DSCP bits marking) =
=93makes
QoS perfect=94, when there are best effort (=3Dno DSCP bits set) TCP =
(=3Dnon
real-time) traffic going over the IP pipe that can be pushed off. =
Routers
implementing diffserve do that.



Simon
--
DTN made easy, lean, and smart --> http://postellation.viagenie.ca
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-----Ursprungligt meddelande-----
Fr=E5n: Paul Kyzivat [mailto:pkyzivat@alum.mit.edu]=20
Skickat: den 22 april 2014 20:37
Till: Karl Stahl; rtcweb@ietf.org
=C4mne: Re: SV: [rtcweb] Some language on "prioritization"

On 4/21/14 6:38 PM, Karl Stahl wrote:
> Inline comments/explanations below. /Karl
>
> -----Ursprungligt meddelande-----
> Fr=E5n: rtcweb [mailto:rtcweb-bounces@ietf.org] F=F6r Paul Kyzivat
> Skickat: den 21 april 2014 16:45
> Till: rtcweb@ietf.org
> =C4mne: Re: [rtcweb] Some language on "prioritization"
>
> It bothers me to see file transfer singled out for special treatment.
>
> File transfer is one (common) example of a potentially high volume=20
> stream that has low requirements on latency. There are certainly other =

> use cases with similar characteristics that are not file transfers.
>
>
> --- When I say "file transfer" I mean ALL traffic of the character=20
> "transfer this high amount of data as fast as possible",

OK. So you are really talking about a usage with a particular traffic =
class.
[Karl] Yes

> which usually is done by TCP,
> allowing all "bandwidth left for me at the congestion point" to be=20
> used (by the TCP endpoints regulating their flow to share the=20
> available bandwidth at the congestion point). That cannot be=20
> transferred better or with higher priority, due to its "maximum=20
> bandwidth for minimum time" nature. Congestion always occur - it is=20
> the way the available bandwidth is shared (best). That is what we use =
TCP
for.
>
> --- "Stream" I would reserve for traffic ticking along with time, NOT=20
> filling the pipe. That could be real-time data and be prioritized or=20
> reserved bandwidth for.
>
> --- If you know your bandwidth at the congestion point (which you=20
> don't - TCP tries it out) you could convert a file transfer into a=20
> stream by spreading it out over time, NOT filling the pipe. But then=20
> it would just take longer time to get the file transferred. File=20
> transfer cannot/should not be prioritized.
>
> --- These traffic types are different and need to be so.
>
> If it isn't feasible to manage data channels with differing priority=20
> needs within the same SCTP association, then why not simply make=20
> provision for multiple SCTP associations, with a priority for each?
>
> --- Yes multiple SCTP associations are possible as said in (B) below=20
> when file transfer is involved. However, for the sharing of bandwidth=20
> at congestion in the network, such SCTP associations should NOT be=20
> bundled over the same flow/5-tuple as the real-time traffic (=3DC=20
> below). You then need to set up another ICE connection - That is where =

> I now suggested to do it with TLS instead (which already is perfect=20
> for file transfer) instead of SCTP (which could be used if working =
equally
well, although much more complex).
> Neither can/should prioritize file transfer though.

ISTM that avoiding setting up another 5-tuple, with the trouble of ICE, =
etc.
is one of the goals.=20
[Karl] Yes, that goal will unfortunately have to be sacrificed=20

The data channels give you a cheap and easy way to set up these things.

IIUC there is already a desire/intent to mark the different streams =
within a
bundle as different priorities/traffic classes. (I thought that is what =
DART
is about.)=20
[Karl] For whom to act upon? At the network level? Do you have a =
pointer?

Isn't that good enough to achieve the same end as running the data over =
a
different 5-tuple?
[Karl] Nope, I don't think reservation type of networks could do that =
even
with very complex network equipment doing reservations (which I believe =
is
per flow).=20

Apparently there is a problem with doing that at the data channel level. =

But I wouldn't think it would be a problem for multiple SCTP =
associations in
the same bundle. (Different SCTP port numbers.) That would allow you to =
have
one association for "file transfers" and one for more interactive data.
[Karl] Still one flow/5-tuple for one bundle - I also wish there was a =
way
around that

	Thanks,
	Paul

> 	Thanks,
> 	Paul
>
> On 4/21/14 8:13 AM, Karl Stahl wrote:
>> 1) AVOIDING SELF DESTRUCTING PRIORITAZATION FROM FILE TRANSFER IN=20
>> DATA CHANNEL
>>
>> Simon,
>>
>> I saw you are an author of TURN-TCP RFC6062. Doesn't that RFC allow a =

>> webrtc file transfer channel (separate from the rtcweb data channel)=20
>> to be created, using TLS file transfer P2P? ICE/STUN/TURN would set=20
>> it up, NOT BUNDLED with chunk data and RTP media.
>>
>> Freeing the current rtcweb data channel from file transfer would=20
>> remedy the self destruction of prioritization/QoS for data chunks and =

>> RTP media as explained in (A), (B) and (C) below.
>>
>> I also think it would be much better for a web programmer to have a=20
>> separate p2p file transfer channel (that is not and cannot be
>> prioritized) than pouring files into the data channel.
>>
>>
>> 2) MARKING PRIORITY DATA AND MEDIA FOR NETWORK LEVEL=20
>> PRIORITIZATION/QoS
>>
>> P=E5l, Harald, Magnus, Collins, Tiru, Dan,
>>
>> We have previously discussed how to convey traffic info (bandwidth=20
>> requirement, and traffic type/prio) from the browser/application to=20
>> the network layer to prioritize or take other QoS measure by some=20
>> network element.
>>
>> I have been pushing for inserting such traffic info into every RTP=20
>> media packet in the RTP header extension, see [1] below.
>>
>> After looking how to mark data channel priority data chunks=20
>> similarly, I suggested [2] below, i.e. a new DTLS ContentType for the =

>> rtcweb data channel, where the content would be the unencrypted=20
>> traffic info (preferable the same as in the RTP header extension)=20
>> followed by the SCTP data.
>>
>> Are there other ways? - Maybe more similar, consistent and/or better?
>>
>> We need the traffic info (contrary to e.g. DSCP bits) to be unchanged =

>> end-to-end and available in both STUN and TURN flows for network=20
>> elements to see and act upon (e.g. to set DSCP bits or reserve=20
>> bandwidth or otherwise regulate) for various network types. The=20
>> traffic info must also be possible for network elements to find.
>>
>> No way b!) Since webrtc now mandates DTLS-SRTP for media, cannot that =

>> DTLS also hold the traffic info, just as for the data chunks
>> - instead of putting the traffic info into the RTP header extension?
>> NO, here the DTLS is only in the key exchange packets - not in every=20
>> packet so that would NOT work, unfortunately.
>>
>> A way c?) Why not send data chunks in SRTP (instead of SCTP) and=20
>> reuse the RTP header extension for the traffic info for the network
layer.
>> (File transfer is done in 1) above - not here!)
>>
>> What other things from SCTP do we actually want? (Not everything from =

>> that complex protocol for carrying telephony signaling can be of=20
>> interest for webrtc.)
>>
>> - Priority (local): The browser could have a prioritizing mechanism=20
>> including both media and data chunks before putting it in the UDP=20
>> socket (simpler than SCTP).
>>
>> - Reliable data: A small transaction layer with retransmission of=20
>> reliable data chunks could be implemented by the browser (simpler=20
>> than SCTP).
>>
>> So with this way c, SCTP would go away totally and only the RTP=20
>> header extension would be used for traffic info to the network layer.
>>
>>
>> Are there more ways? We need this traffic info to allow WebRTC with=20
>> acceptable QoS over all network types (especially mobile Internet/OTT =

>> LTE and Cable Networks that are of reservation type).
>>
>>
>> Below are also a few comments to other feedback.
>>
>> /Karl
>>
>>
>> On 10 Apr 2014, at 17:35, Paul Kyzivat [pkyzivat@alum.mit.edu] wrote:
>> I presume you mean each *concurrent* file transfer must use its own
> channel.
>> Right? Reusing the same channel to transfer a series of files should=20
>> be fine.
>>
>> 	Thanks,
>> 	Paul
>> [Karl] Yes, That is right. /Karl
>>
>> On 10 Apr 2014, at 15:14, Harald Alvestrand [harald@alvestrand.no] =
wrote:
>> Just checking... since there's been no comment to the main body of=20
>> the text here (it's all been on the SCTP aspect), is it OK to include =

>> the text in my next version of the -transport draft?
>>
>> (I'm not responding to Karl's comments - it's at a completely=20
>> different level of complexity.)
>>
>> [Karl] Yes, it is but needs to be addressed. I think file transfer=20
>> just be taken out of the data channel, and a file transfer channel=20
>> (on a separate flow, as suggested above) be defined for Web=20
>> programmers to use. That would remedy (A), (B) and (C) below (p2p=20
>> file transfer not destroying priority for data chunks and media).
>> The file transfer channel would not/cannot be prioritized.
>>
>> On 03 Apr 2014, at 21:12, Michael Tuexen
>> [Michael.Tuexen@lurchi.franken.de]wrote:
>>> Is there an
>>> overall design/thought somewhere that I missed/am unaware of?)
>> The basis idea is to use a stream scheduler to handle streams =
different.
>> Some data channels can get a higher bandwidth than others. Since the=20
>> priority stuff is still in flux in W3C, it hasn't been nailed down in =

>> the SCTP spec. I guess, the place will be in
>> http://tools.ietf.org/html/draft-ietf-tsvwg-sctp-ndata-00
>> This document is not finalized yet.
>>>
>> [Karl] There are severe limitations for doing prioritization in a=20
>> protocol (SCTP or other) executed in the endpoints. In most cases=20
>> there are no packets in the endpoint to put first among queued up=20
>> traffic, because the queue is somewhere else between the endpoints.
>> Just consider the most common case where the endpoints are connected=20
>> with a fast Ethernet. The data has then already left the endpoint and =

>> is queued up at some congestion point in the network where the SCTP=20
>> protocol cannot rearrange the queue. First when that unreachable=20
>> queue is full, there is something in the endpoint to act upon...
>>
>>
>> -----Ursprungligt meddelande-----
>> Fr=E5n: rtcweb [mailto:rtcweb-bounces@ietf.org] F=F6r Karl Stahl
>> Skickat: den 3 april 2014 18:30
>> Till: 'Matthew Kaufman (SKYPE)'; 'Harald Alvestrand'; 'Magnus =
Westerlund'
>> Kopia: rtcweb@ietf.org
>> =C4mne: Re: [rtcweb] Some language on "prioritization"
>>
>> The questions raised here don't have simple "local or endpoint=20
>> answers". There are overall and basic things to be made clear first.
>> Combining P2P file transfer over the rtcweb data channels with=20
>> prioritized traffic chunks, brings further complications to the=20
>> prioritization/QoS issue. This cannot be handled/done by endpoint or=20
>> applications without considering the transport network (The=20
>> congestion to handle by prioritization is in the network - NOT in the
endpoints)!
>>
>> There are below (A), (B) and (C) - to get us back to normal - then=20
>> there is (D) to improve - to allow the network level to give us the=20
>> prioritization required for WebRTC critical traffic (real-time and=20
>> data chunks). Without (D) - an interface to network QoS equipment -=20
>> the network does not even know what to prioritize.
>>
>> I am afraid that this will be considered "back to the drawing board"
>> rather than "some language" but I cannot see the way forward if not=20
>> considered.
>>
>> Has the SCTP data channel been constructed with hopes of=20
>> prioritization/quality functionality that simply are not there=20
>> (cannot be there)? (I have browsed the (complex) SCTP spec and some=20
>> rtcweb drafts to get an understanding of the prioritization/QoS=20
>> thoughts and means in this rtcweb context and the intended usage. Is=20
>> there an overall design/thought somewhere that I missed/am unaware=20
>> of?)
>>
>> File transfer - getting a larger amount of data through between=20
>> endpoints as quickly as possible - CANNOT be prioritized over IP=20
>> networks! For file transfer we normally use TCP, meaning we fill the=20
>> pipe, until THE ENDPOINTS see packet drop and regulate down their=20
>> traffic flow. File transfer over IP networks works this way and=20
>> cannot be improved upon ("Internet would stop" if we prioritized such =

>> traffic! The available bandwidth at congestion points are shared=20
>> between everyone's TCP-flows hitting such congestion points in the
>> network.) File transfer's nature is to have "infinite bandwidth=20
>> demand" - the mechanism to handle this is to allow file transfer to=20
>> "take the rest of the bandwidth"
>> after prioritized media and chunks have taken their needed share.
>>
>> The improvement to make - prioritization of media traffic and data=20
>> chunks - is to allow level 3 QoS methods - not only relying on the=20
>> level 4 TCP backing off process. Without level 3 QoS (diffserve/DSCP, =

>> reservation or separate pipes) media and data chunks packets are also =

>> lost in the level 4 TCP backing off process (=3Dtoday's situation =
over=20
>> best effort Internet). That is where the prioritization/QoS measures=20
>> that we need for rtcweb have to be.
>>
>> SCTP in itself (instead of TLS/TCP) does not/cannot/will not improve=20
>> things. It can destroy though if wrongly/badly implemented or used!
>>
>> >From the SCTP RFC4960: 7.  Congestion Control
>>      ...adequate resources will  be allocated to SCTP traffic
>>      to ensure prompt delivery of time-critical data ... unlikely,
>>      during normal operations, that transmissions encounter severe
>>      congestion conditions.
>>
>>      However, SCTP must operate under adverse operational
>>      conditions, which can develop upon partial network failures
>>      or unexpected traffic surges.  In such situations, SCTP must
>>      follow correct congestion control steps to recover from
>>      congestion quickly in order to get data delivered as soon as
>>      possible.
>>
>>      ... SCTP congestion control is always applied to the entire
>>      association, and not to individual streams.
>>
>> Doing file transfer with the data channel will use SCTP in the=20
>> "adverse operational conditions, which can develop upon partial=20
>> network failures or unexpected traffic surges"-mode! Has that been=20
>> considered when constructing the data channel? Hope so...
>>
>> Assuming SCTP is implemented correctly and as good as TCP for file=20
>> transfer (even though file transfer =3D filling the pipe mode is an=20
>> "adverse condition..." for SCTP). Then we must specify:
>>
>>
>> (A) Rtcweb file transfer MUST NOT use the same data channel as data=20
>> chunks expected to have prioritized delivery (otherwise, as per=20
>> above, SCTP will regulate the "the entire association"
>> to share the available bandwidth at the congestion point -=20
>> self-destroying the expected prioritized delivery of data chunks in=20
>> the same channel).
>>
>> (B) Each rtcweb file transfer MUST use its own data channel (if=20
>> combined they risk being regulated randomly/unfairly by SCTP itself=20
>> and all files will have just have one single TCP-like flow at the=20
>> network congestion point when sharing the available bandwidth with=20
>> everyone else's flows at that congestion point
>> - That would be "negative priority" compared to what we are used to=20
>> in a browser.).
>>
>> FURTHER, to allow the underlying network the ability to handle the=20
>> real-time and data chunks as least as good as "before rtcweb"
>> (Unfortunately this will make NAT-firewall less good - not sending=20
>> everything P2P in a single bundle/flow - I don't like it either...
>> Sorry...):
>>
>> (C) File transfer using the data channel MUST NOT be bundled with=20
>> media and data chunk traffic in the same flow - file transfer must=20
>> use a separate socket creating another flow/5-tuple over the network.
>> (In reservation type of networks (e.g. mobile OTT, cable), file=20
>> transfer - having no bandwidth limitation - would otherwise overflow=20
>> the reserved bandwidth for media and chunck traffic - destroying=20
>> prioritized delivery of media and chunck traffic over that flow.)=20
>> Such file transfer data channels MUST NOT be prioritized by the=20
>> underlying network (applies to all IP network types, network=20
>> prioritization of file transfer will STOP traffic between endpoints
>> - Endpoint's TCP-like regulation has to function!). Several file=20
>> transfers could share such a separate flow (without specifying in SDP =

>> I guess) but the underlying network must be able to separate such=20
>> flows (not to be prioritized) from media and chunk flows.
>>
>> In the API, such file transfer data channel could be mapped to the=20
>> "below normal" classification (but would it not be better with a=20
>> "file transfer channel" API instead?). (And notice that without (D)=20
>> below, the network is not informed how to separate a file transfer=20
>> data channel flow from a prioritized data channel
>> flow.)
>>
>>
>> Having (A), (B) and (C) in place we are BACK TO THE CURRENT Internet=20
>> prioritization/QoS level - before the rtcweb p2p data channel file=20
>> transfer was introduced...
>>
>> Now remains the prioritization improvement required for quality
>>
>> hungry rtcweb! Both for the real-time media and the data chuncks.
>>
>>
>> (D) The underlying transport network needs to be informed about the=20
>> traffic requirements.
>>
>> - In RTP media this traffic info could be inserted by the rtcweb=20
>> browser in the RTP header extension [1] depending on codec usage etc.
>> The API classification could be mapped to the "Prioritize X(2 bits)"
>> suggested in [1] below.
>>
>> - for the data channel, the traffic info could be inserted as=20
>> suggested in previous email ([2] below)=20
>> http://www.ietf.org/mail-archive/web/rtcweb/current/msg11973.html
>>
>> Note that the required bandwidth (only known by the
>> browser/application) has to be passed to reservation types of =
network.
>> Requested prioritization from the API or DSCP bits are not =
sufficient.
>> The browser knows the bandwidth requirements for codecs, but for data =

>> chunks, the application has the knowledge.
>> In lack of a bandwidth parameter in the data channel API, a simple=20
>> convention - e.g. reserve 200 kbps for data chunks could serve as a=20
>> compromise (or 4 bandwidths mapped to the API prioritize=20
>> classifications...). A modified API with the bandwidth requirement=20
>> spelled out would of course be better for the chuck data channel.
>> For a file transfer data channel - prioritization classification does =

>> not even apply.
>>
>>
>> Having prioritization over the network cleared, the application=20
>> should use that transport properly - having media and data chunks=20
>> transported with prioritization (as produced/inserted), NOT queuing=20
>> them up, rearranged by protocols or applications and NOT involving=20
>> file transfer (that has to be unprioritized).
>>
>> Using DSCP bits if available through the OS in some situations does=20
>> not help either, as already pointed out in the transport document.
>> For (D) above, DSCP bits could in a few situations function locally,=20
>> but not end-to-end, so of no use generally.
>>
>> /Karl
>>
>>
>> [1] Traffic type information to encode (recreating the idea/intention =

>> of the RTP payload type (PT) header that is not available for the=20
>> network level anymore, by instead having the browser filling the RTP=20
>> header extension with relevant traffic info that all IP network types =

>> can use.), as suggested already October 22. Two parameters to encode=20
>> http://www.ietf.org/mail-archive/web/rtcweb/current/msg09129.html
>>
>> A) The maximum bandwidth requirement: Two bytes could contain=20
>> everything from some bps for real-time text to Gbps for future 3D=20
>> supersize telepresence on a logarithmic scale.
>>
>> B) The quality characteristics for the stream, with the highest bit=20
>> set to 1, we could allocate a bit each for quality type e.g:
>> Best Effort, Audio, Video, Supplemental Video, Gaming, Data, Delay=20
>> Insensitive (e.g. video streaming), Minimum Delay, Reliable Delivery, =

>> Prioritize X(2 bits), Variation Y(2 bits), that could be combined as=20
>> required to describe the stream.
>>
>> And with the highest bit set to 0, there could instead be a number=20
>> for special usage that does not fit the general description of the=20
>> individual bits.
>>
>> Without this information, how could e.g. a reservation type network=20
>> (e.g. mobile OTT and cable) reserve the bandwidth required for=20
>> incoming rtcweb media or data chunks from a diffserve network=20
>> (carrying no bandwidth requirement)?
>>
>>
>> [2] A new DTLS ContentType (e.g. 24 instead of current 23) for the=20
>> rtcweb data channel, where the content would be the unencrypted=20
>> traffic info (preferable the same as in the RTP header extension)=20
>> followed by the SCTP data as it is today (where it today comes=20
>> directly in the DTLS as the only content).:
>>
>> TODAY:
>> Content Type: 23 (1 byte) content type 23 means =93application =
data=94
>> DTLS version:  0xfeff (2 byte)
>> Epoch: 0001 (2 byte)
>> Sequence number (e.g. 00 00 00 00 00 01) 6 byte Length (e.g. 01 c0) 2 =

>> byte Encrypted application data: ...... =3D the SCTP
>>
>> SUGGESTION WITH ADDED TRAFFIC INFO FOR LOWER LEVELS:
>> Content Type: 24 (1 byte) content type 24 meaning =93application data =

>> preceded by traffic info=94
>> DTLS version:  0xfeff (2 byte)
>> Epoch: 0001 (2 byte)
>> Sequence number (e.g. 00 00 00 00 00 01) 6 byte Length (e.g. 01 c0) 2
>> byte: length of the encrypted application data after the new header=20
>> extension TRAFFIC INFO, Same as suggested for the RTP Header
>> extension:
>> Encrypted application data: ...... ...... =3D the SCTP unchanged
>>
>>
>> -----Ursprungligt meddelande-----
>> Fr=E5n: rtcweb [mailto:rtcweb-bounces@ietf.org] F=F6r Matthew Kaufman
>> (SKYPE)
>> Skickat: den 31 mars 2014 19:50
>> Till: Harald Alvestrand
>> Kopia: rtcweb@ietf.org
>> =C4mne: Re: [rtcweb] Some language on "prioritization"
>>
>> And if SCTP and RTP have packets to send, which goes first, why, and=20
>> under what circumstances?
>>
>> Matthew Kaufman
>>
>> (Sent from my iPhone)
>>
>>> On Mar 31, 2014, at 10:48 AM, "Harald Alvestrand"
>>> <harald@alvestrand.no>
>> wrote:
>>>
>>>> On 03/31/2014 07:42 PM, Michael Tuexen wrote:
>>>>> On 31 Mar 2014, at 19:38, Martin Thomson=20
>>>>> <martin.thomson@gmail.com>
>> wrote:
>>>>>
>>>>>> On 31 March 2014 10:08, Harald Alvestrand <harald@alvestrand.no>
> wrote:
>>>>>> -- TODO: Specify a relative priority scheme that makes sense with =

>>>>>> SCTP, with an appropriate reference.
>>>>>> [draft-ietf-tsvwg-sctp-prpolicies]
>>>>>> specifies a priority policy, but it's about discarding packets,=20
>>>>>> not deciding which packets to send first, and it also makes it=20
>>>>>> impossible to specify time-bounded retransmission. --
>>>>>
>>>>> Why would SCTP need special treatment?  I can understand if there=20
>>>>> are particular time-sensitive control messages that need to be=20
>>>>> given higher priority, but this is all time-sensitive.
>>>> A single transport connection (in this case an SCTP association)=20
>>>> can only use a single DSCP. So it would be OK to use the same=20
>>>> priority for all data channels, but things get complicated when=20
>>>> when some data channels would have different priorities requiring=20
>>>> different DSCP
>> markings.
>>>
>>> The SCTP protocol machine will, at some given moment in time, have=20
>>> packets in its send buffers that belong to multiple data channels.
>>> Only one of them can go on the wire first.
>>>
>>> Which packet does it choose to send first, and why?
>>>
>>> That's the question I'm looking for an answer to.
>>>
>>> If the answer has an RFC number, chapter and section, I'd be very =
happy.
>>> If it doesn't - perhaps we have to invent one.
>>> Or say that it's "implementation dependent".
>>>
>>
>>
>> _______________________________________________
>> rtcweb mailing list
>> rtcweb@ietf.org
>> https://www.ietf.org/mailman/listinfo/rtcweb
>>
>
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb
>
>


From nobody Wed Apr 23 05:31:09 2014
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Subject: [rtcweb] Pictures of congestion control on the Internet - which is more realistic?
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I quote Karl's piece below, because I want other's opinion of whether it's a realistic picture.

My impression is that it is not - in particular that:

- Lots of interactive traffic goes over the Internet today without layer 2 separation
- All UDP endpoints that use the open Internet have some form of backoff on congestion - using delay, packet drop or (most likely) both to detect congestion

I'm writing the transport document based on those assumptions - that we should specify something that will work as well as it can over an Internet that does not care, and try to allow for better things to come along later.

Am I wrong to write it like this?

Harald



------------------------------------
[Karl] I wrote this sometime before, regarding how QoS over the internet
works in general - may be useful for understanding

For better understanding of these QoS/QoE problems, and methods for
improving (not specifically discussing the policies of sharing real-time
traffic space), I would like to explain:

Over an IP pipe only used for real-traffic (no TCP data traffic), it is
sufficient that the pipe is wide enough for good QoS. That is often used and
implemented by separating IP pipes at a lower level using e.g. Ethernet
VLAN, MPLS, Ethernet over ATM (std for ADSL modems). TURN servers can
enforce real-traffic into such pipes and QoS is achieved. Let’s call this
Level 2 QoS (Network level)

Over an IP pipe shared between quality requiring real-time traffic and less
demanding data or streaming (usually TCP) traffic, we have the sharing
between these two traffic classes to consider. The main method (and the
mechanism making today’s real-time QoE as good as it often is) is that TCP
endpoints back-off and share their bandwidth usage. Real-time traffic using
UDP transport do not back-off, the endpoints using UDP occupy the bandwidth
needed. When an IP pipe gets filled, it is all endpoint’s TCP bandwidth
usage that is back-off and shared between them, leaving room for the UDP
traffic. This is mechanism we experience everyday over the Internet, using
our “Surf IP pipes”. Let’s call this Level 4 QoS (Transport level)

However, this Level 4 QoS is based on that at congestion times (which happen
every time we click – setting up a TCP flow and transferring some amount of
data as quick as possible) the router handling the most narrow part of the
pipe (the congestion point) drop packets. It is this packet dropping that
(i) signals to TCP endpoints to reduce their bandwidth (via TCP’s error
correction/retransmission mechanism) and (ii) destroys the QoE of real-time
traffic. (Both TCP and UDP packets are dropped in this process that is
triggered by flow intensive TCP traffic.)

We need (i) but don’t want (ii) and to improve on this we can e.g. use
diffserve, DSCP bits in IP packets to instruct routers to always forward the
real-time traffic before any unmarked TCP traffic (which usually fills most
of the pipe). Then QoS is then achieved for real-time traffic. This is Level
3 QoS (IP level). The method used is “traffic shaping”: backing off data
traffic, leaving real-time traffic free passage without packet loss.


From nobody Wed Apr 23 06:27:49 2014
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A New Internet-Draft is available from the on-line Internet-Drafts directories.
 This draft is a work item of the Real-Time Communication in WEB-browsers Working Group of the IETF.

        Title           : Web Real-Time Communication (WebRTC): Media Transport and Use of RTP
        Authors         : Colin Perkins
                          Magnus Westerlund
                          Joerg Ott
	Filename        : draft-ietf-rtcweb-rtp-usage-13.txt
	Pages           : 45
	Date            : 2014-04-23

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


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

There's also a htmlized version available at:
http://tools.ietf.org/html/draft-ietf-rtcweb-rtp-usage-13

A diff from the previous version is available at:
http://www.ietf.org/rfcdiff?url2=draft-ietf-rtcweb-rtp-usage-13


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

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


From nobody Wed Apr 23 06:33:01 2014
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WG,

We authors have been doing some editing on this version to take care of
the last known issues. This includes the text that has been discussed on
the list lately. It also contains an expanded example about RTCP
bandwidth (see Section 7.2). Further we have reviewed the terminology
usage in the document, and applied the AVTEXT grouping taxonomy.

So, we authors now believe this to be ready for WG last call.

Cheers

Magnus

On 2014-04-23 15:27, internet-drafts@ietf.org wrote:
> 
> A New Internet-Draft is available from the on-line Internet-Drafts directories.
>  This draft is a work item of the Real-Time Communication in WEB-browsers Working Group of the IETF.
> 
>         Title           : Web Real-Time Communication (WebRTC): Media Transport and Use of RTP
>         Authors         : Colin Perkins
>                           Magnus Westerlund
>                           Joerg Ott
> 	Filename        : draft-ietf-rtcweb-rtp-usage-13.txt
> 	Pages           : 45
> 	Date            : 2014-04-23
> 
> Abstract:
>    The Web Real-Time Communication (WebRTC) framework provides support
>    for direct interactive rich communication using audio, video, text,
>    collaboration, games, etc. between two peers' web-browsers.  This
>    memo describes the media transport aspects of the WebRTC framework.
>    It specifies how the Real-time Transport Protocol (RTP) is used in
>    the WebRTC context, and gives requirements for which RTP features,
>    profiles, and extensions need to be supported.
> 
> 
> The IETF datatracker status page for this draft is:
> https://datatracker.ietf.org/doc/draft-ietf-rtcweb-rtp-usage/
> 
> There's also a htmlized version available at:
> http://tools.ietf.org/html/draft-ietf-rtcweb-rtp-usage-13
> 
> A diff from the previous version is available at:
> http://www.ietf.org/rfcdiff?url2=draft-ietf-rtcweb-rtp-usage-13
> 
> 
> Please note that it may take a couple of minutes from the time of submission
> until the htmlized version and diff are available at tools.ietf.org.
> 
> Internet-Drafts are also available by anonymous FTP at:
> ftp://ftp.ietf.org/internet-drafts/
> 
> _______________________________________________
> 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
> 
> 


-- 

Magnus Westerlund

----------------------------------------------------------------------
Services, Media and Network features, Ericsson Research EAB/TXM
----------------------------------------------------------------------
Ericsson AB                 | Phone  +46 10 7148287
Färögatan 6                 | Mobile +46 73 0949079
SE-164 80 Stockholm, Sweden | mailto: magnus.westerlund@ericsson.com
----------------------------------------------------------------------


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Subject: Re: [rtcweb] Pictures of congestion control on the Internet - which is more realistic?
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--001a11c38cc2f75ddf04f7b70699
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I would say that some video end points have congestion back off.  Almost
all audio end points have none. Based on this, most UDP endpoints do not
deal well with congestion. Ideally this should change to support better
bandwidth management. Futhermore, reliably transmitting real time media
over public internet would also require changes to biffer management
algorithms in routers, which will, in turn, affect congestion handling
strategies in the end points.

On Apr 23, 2014 8:31 AM, "Harald Alvestrand" <harald@alvestrand.no> wrote:
>
> I quote Karl's piece below, because I want other's opinion of whether
it's a realistic picture.
>
> My impression is that it is not - in particular that:
>
> - Lots of interactive traffic goes over the Internet today without layer
2 separation
> - All UDP endpoints that use the open Internet have some form of backoff
on congestion - using delay, packet drop or (most likely) both to detect
congestion
>
> I'm writing the transport document based on those assumptions - that we
should specify something that will work as well as it can over an Internet
that does not care, and try to allow for better things to come along later.
>
> Am I wrong to write it like this?
>
> Harald
>
>
>
> ------------------------------------
> [Karl] I wrote this sometime before, regarding how QoS over the internet
> works in general - may be useful for understanding
>
> For better understanding of these QoS/QoE problems, and methods for
> improving (not specifically discussing the policies of sharing real-time
> traffic space), I would like to explain:
>
> Over an IP pipe only used for real-traffic (no TCP data traffic), it is
> sufficient that the pipe is wide enough for good QoS. That is often used
and
> implemented by separating IP pipes at a lower level using e.g. Ethernet
> VLAN, MPLS, Ethernet over ATM (std for ADSL modems). TURN servers can
> enforce real-traffic into such pipes and QoS is achieved. Let=E2=80=99s c=
all this
> Level 2 QoS (Network level)
>
> Over an IP pipe shared between quality requiring real-time traffic and
less
> demanding data or streaming (usually TCP) traffic, we have the sharing
> between these two traffic classes to consider. The main method (and the
> mechanism making today=E2=80=99s real-time QoE as good as it often is) is=
 that TCP
> endpoints back-off and share their bandwidth usage. Real-time traffic
using
> UDP transport do not back-off, the endpoints using UDP occupy the
bandwidth
> needed. When an IP pipe gets filled, it is all endpoint=E2=80=99s TCP ban=
dwidth
> usage that is back-off and shared between them, leaving room for the UDP
> traffic. This is mechanism we experience everyday over the Internet, usin=
g
> our =E2=80=9CSurf IP pipes=E2=80=9D. Let=E2=80=99s call this Level 4 QoS =
(Transport level)
>
> However, this Level 4 QoS is based on that at congestion times (which
happen
> every time we click =E2=80=93 setting up a TCP flow and transferring some=
 amount
of
> data as quick as possible) the router handling the most narrow part of th=
e
> pipe (the congestion point) drop packets. It is this packet dropping that
> (i) signals to TCP endpoints to reduce their bandwidth (via TCP=E2=80=99s=
 error
> correction/retransmission mechanism) and (ii) destroys the QoE of
real-time
> traffic. (Both TCP and UDP packets are dropped in this process that is
> triggered by flow intensive TCP traffic.)
>
> We need (i) but don=E2=80=99t want (ii) and to improve on this we can e.g=
. use
> diffserve, DSCP bits in IP packets to instruct routers to always forward
the
> real-time traffic before any unmarked TCP traffic (which usually fills
most
> of the pipe). Then QoS is then achieved for real-time traffic. This is
Level
> 3 QoS (IP level). The method used is =E2=80=9Ctraffic shaping=E2=80=9D: b=
acking off data
> traffic, leaving real-time traffic free passage without packet loss.
>
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb

--001a11c38cc2f75ddf04f7b70699
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<p dir=3D"ltr">I would say that some video end points have congestion back =
off.=C2=A0 Almost all audio end points have none. Based on this, most UDP e=
ndpoints do not deal well with congestion. Ideally this should change to su=
pport better bandwidth management. Futhermore, reliably transmitting real t=
ime media over public internet would also require changes to biffer managem=
ent algorithms in routers, which will, in turn, affect congestion handling =
strategies in the end points.</p>

<p dir=3D"ltr">On Apr 23, 2014 8:31 AM, &quot;Harald Alvestrand&quot; &lt;<=
a href=3D"mailto:harald@alvestrand.no">harald@alvestrand.no</a>&gt; wrote:<=
br>
&gt;<br>
&gt; I quote Karl&#39;s piece below, because I want other&#39;s opinion of =
whether it&#39;s a realistic picture.<br>
&gt;<br>
&gt; My impression is that it is not - in particular that:<br>
&gt;<br>
&gt; - Lots of interactive traffic goes over the Internet today without lay=
er 2 separation<br>
&gt; - All UDP endpoints that use the open Internet have some form of backo=
ff on congestion - using delay, packet drop or (most likely) both to detect=
 congestion<br>
&gt;<br>
&gt; I&#39;m writing the transport document based on those assumptions - th=
at we should specify something that will work as well as it can over an Int=
ernet that does not care, and try to allow for better things to come along =
later.<br>

&gt;<br>
&gt; Am I wrong to write it like this?<br>
&gt;<br>
&gt; Harald<br>
&gt;<br>
&gt;<br>
&gt;<br>
&gt; ------------------------------------<br>
&gt; [Karl] I wrote this sometime before, regarding how QoS over the intern=
et<br>
&gt; works in general - may be useful for understanding<br>
&gt;<br>
&gt; For better understanding of these QoS/QoE problems, and methods for<br=
>
&gt; improving (not specifically discussing the policies of sharing real-ti=
me<br>
&gt; traffic space), I would like to explain:<br>
&gt;<br>
&gt; Over an IP pipe only used for real-traffic (no TCP data traffic), it i=
s<br>
&gt; sufficient that the pipe is wide enough for good QoS. That is often us=
ed and<br>
&gt; implemented by separating IP pipes at a lower level using e.g. Etherne=
t<br>
&gt; VLAN, MPLS, Ethernet over ATM (std for ADSL modems). TURN servers can<=
br>
&gt; enforce real-traffic into such pipes and QoS is achieved. Let=E2=80=99=
s call this<br>
&gt; Level 2 QoS (Network level)<br>
&gt;<br>
&gt; Over an IP pipe shared between quality requiring real-time traffic and=
 less<br>
&gt; demanding data or streaming (usually TCP) traffic, we have the sharing=
<br>
&gt; between these two traffic classes to consider. The main method (and th=
e<br>
&gt; mechanism making today=E2=80=99s real-time QoE as good as it often is)=
 is that TCP<br>
&gt; endpoints back-off and share their bandwidth usage. Real-time traffic =
using<br>
&gt; UDP transport do not back-off, the endpoints using UDP occupy the band=
width<br>
&gt; needed. When an IP pipe gets filled, it is all endpoint=E2=80=99s TCP =
bandwidth<br>
&gt; usage that is back-off and shared between them, leaving room for the U=
DP<br>
&gt; traffic. This is mechanism we experience everyday over the Internet, u=
sing<br>
&gt; our =E2=80=9CSurf IP pipes=E2=80=9D. Let=E2=80=99s call this Level 4 Q=
oS (Transport level)<br>
&gt;<br>
&gt; However, this Level 4 QoS is based on that at congestion times (which =
happen<br>
&gt; every time we click =E2=80=93 setting up a TCP flow and transferring s=
ome amount of<br>
&gt; data as quick as possible) the router handling the most narrow part of=
 the<br>
&gt; pipe (the congestion point) drop packets. It is this packet dropping t=
hat<br>
&gt; (i) signals to TCP endpoints to reduce their bandwidth (via TCP=E2=80=
=99s error<br>
&gt; correction/retransmission mechanism) and (ii) destroys the QoE of real=
-time<br>
&gt; traffic. (Both TCP and UDP packets are dropped in this process that is=
<br>
&gt; triggered by flow intensive TCP traffic.)<br>
&gt;<br>
&gt; We need (i) but don=E2=80=99t want (ii) and to improve on this we can =
e.g. use<br>
&gt; diffserve, DSCP bits in IP packets to instruct routers to always forwa=
rd the<br>
&gt; real-time traffic before any unmarked TCP traffic (which usually fills=
 most<br>
&gt; of the pipe). Then QoS is then achieved for real-time traffic. This is=
 Level<br>
&gt; 3 QoS (IP level). The method used is =E2=80=9Ctraffic shaping=E2=80=9D=
: backing off data<br>
&gt; traffic, leaving real-time traffic free passage without packet loss.<b=
r>
&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">https://www.i=
etf.org/mailman/listinfo/rtcweb</a><br>
</p>

--001a11c38cc2f75ddf04f7b70699--


From nobody Wed Apr 23 08:19:09 2014
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On Wed, Apr 23, 2014 at 5:30 AM, Harald Alvestrand <harald@alvestrand.no> w=
rote:
> I quote Karl's piece below, because I want other's opinion of whether it'=
s a
> realistic picture.
>
> My impression is that it is not - in particular that:
>
> - Lots of interactive traffic goes over the Internet today without layer =
2
> separation
> - All UDP endpoints that use the open Internet have some form of backoff =
on
> congestion - using delay, packet drop or (most likely) both to detect
> congestion

I'm not sure if we have a shared definition of "congestion".  Mine is
network delays exceeding that of what human factors research notes
as "perceptible" with about 20ms as the outer bound.

http://gettys.wordpress.com/2013/07/10/low-latency-requires-smart-queuing-t=
raditional-aqm-is-not-enough/

Kathie/Van something like a sustained  delay > 1.1 * RTT =3D bad queue

Bittorrent folk  put it at > 100ms

Others use even bigger numbers.

> I'm writing the transport document based on those assumptions - that we
> should specify something that will work as well as it can over an Interne=
t
> that does not care, and try to allow for better things to come along late=
r.
>
> Am I wrong to write it like this?
>
> Harald
>
>
>
> ------------------------------------
> [Karl] I wrote this sometime before, regarding how QoS over the internet
> works in general - may be useful for understanding
>
> For better understanding of these QoS/QoE problems, and methods for
> improving (not specifically discussing the policies of sharing real-time
> traffic space), I would like to explain:
>
> Over an IP pipe only used for real-traffic (no TCP data traffic), it is
> sufficient that the pipe is wide enough for good QoS. That is often used =
and
> implemented by separating IP pipes at a lower level using e.g. Ethernet
> VLAN, MPLS, Ethernet over ATM (std for ADSL modems). TURN servers can
> enforce real-traffic into such pipes and QoS is achieved. Let=E2=80=99s c=
all this
> Level 2 QoS (Network level)
>
> Over an IP pipe shared between quality requiring real-time traffic and le=
ss
> demanding data or streaming (usually TCP) traffic, we have the sharing
> between these two traffic classes to consider. The main method (and the
> mechanism making today=E2=80=99s real-time QoE as good as it often is) is=
 that TCP
> endpoints back-off and share their bandwidth usage. Real-time traffic usi=
ng
> UDP transport do not back-off, the endpoints using UDP occupy the bandwid=
th
> needed. When an IP pipe gets filled, it is all endpoint=E2=80=99s TCP ban=
dwidth
> usage that is back-off and shared between them, leaving room for the UDP
> traffic. This is mechanism we experience everyday over the Internet, usin=
g
> our =E2=80=9CSurf IP pipes=E2=80=9D. Let=E2=80=99s call this Level 4 QoS =
(Transport level)
>
> However, this Level 4 QoS is based on that at congestion times (which hap=
pen
> every time we click =E2=80=93 setting up a TCP flow and transferring some=
 amount of
> data as quick as possible) the router handling the most narrow part of th=
e
> pipe (the congestion point) drop packets. It is this packet dropping that
> (i) signals to TCP endpoints to reduce their bandwidth (via TCP=E2=80=99s=
 error
> correction/retransmission mechanism) and (ii) destroys the QoE of real-ti=
me
> traffic. (Both TCP and UDP packets are dropped in this process that is
> triggered by flow intensive TCP traffic.)

Um, er, the situation today is one of dramatic overbuffering on everything,
triggering packet drop too infrequently, which messes with QoE far worse
for much interactive traffic  (notably  games  and VOIP) than  drop  does.

"once you have bad latency you're stuck with it"  -

http://rescomp.stanford.edu/~cheshire/rants/Latency.html

>
> We need (i) but don=E2=80=99t want (ii) and to improve on this we can e.g=
. use
> diffserve, DSCP bits in IP packets to instruct routers to always forward =
the
> real-time traffic before any unmarked TCP traffic (which usually fills mo=
st
> of the pipe). Then QoS is then achieved for real-time traffic. This is Le=
vel
> 3 QoS (IP level). The method used is =E2=80=9Ctraffic shaping=E2=80=9D: b=
acking off data
> traffic, leaving real-time traffic free passage without packet loss.

Well,  there  are several techniques involved in traffic  shaping, and in
the end  you  have  to use most of  them comprehensively to  get
good  QoE.

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

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



--=20
Dave T=C3=A4ht

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


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From: Stephan Wenger <stewe@stewe.org>
To: Roman Shpount <roman@telurix.com>, Harald Alvestrand <harald@alvestrand.no>
Thread-Topic: [rtcweb] Pictures of congestion control on the Internet - which is more realistic?
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Subject: Re: [rtcweb] Pictures of congestion control on the Internet - which is more realistic?
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--_000_CF7D2A6D464A8stewesteweorg_
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Hi,
Please see inline.
Stephan

From: Roman Shpount <roman@telurix.com<mailto:roman@telurix.com>>
Date: Wednesday, April 23, 2014 at 8:02
To: Harald Alvestrand <harald@alvestrand.no<mailto:harald@alvestrand.no>>
Cc: "rtcweb@ietf.org<mailto:rtcweb@ietf.org>" <rtcweb@ietf.org<mailto:rtcwe=
b@ietf.org>>
Subject: Re: [rtcweb] Pictures of congestion control on the Internet - whic=
h is more realistic?


I would say that some video end points have congestion back off.

Concur.  In fact, I would replace "some" with "very few", and I think the s=
tatement would still be correct.  At least when it comes to the hardware ba=
sed solutions, and your typical SIP client.  The proprietary soft-clients (=
Skype, hangouts, etc.) may be a better in that regard, especially the moder=
n ones that use scalability (like ours, brag brag).  (One can, of course, a=
rgue that at any given moment there are more sessions active using modern s=
oft-clients than the combined lifetime sales of all traditional hardware vi=
deo conference vendors-and that's why we haven't had a meltdown yet.  I att=
ribute the lack of a meltdown to other reasons, though.)

Almost all audio end points have none.

Concur as well.  And, I believe the number of bits spent on interactive aud=
io over the open Internet is still considerably higher than those for inter=
active video.

In both cases, the typical reaction of an endpoint is to expose the user to=
 crappy quality, until the user hangs up.  Sometimes, slightly smarter impl=
ementations hang up themselves if the packet loss rate becomes excessive.I =
would suggest not to paint a rosier picture in the draft than what is warra=
nted based on real-world observations.

Based on this, most UDP endpoints do not deal well with congestion. Ideally=
 this should change to support better bandwidth management. Futhermore, rel=
iably transmitting real time media over public internet would also require =
changes to biffer management algorithms in routers, which will, in turn, af=
fect congestion handling strategies in the end points.

On Apr 23, 2014 8:31 AM, "Harald Alvestrand" <harald@alvestrand.no<mailto:h=
arald@alvestrand.no>> wrote:
>
> I quote Karl's piece below, because I want other's opinion of whether it'=
s a realistic picture.
>
> My impression is that it is not - in particular that:
>
> - Lots of interactive traffic goes over the Internet today without layer =
2 separation
> - All UDP endpoints that use the open Internet have some form of backoff =
on congestion - using delay, packet drop or (most likely) both to detect co=
ngestion
>
> I'm writing the transport document based on those assumptions - that we s=
hould specify something that will work as well as it can over an Internet t=
hat does not care, and try to allow for better things to come along later.
>
> Am I wrong to write it like this?
>
> Harald
>
>
>
> ------------------------------------
> [Karl] I wrote this sometime before, regarding how QoS over the internet
> works in general - may be useful for understanding
>
> For better understanding of these QoS/QoE problems, and methods for
> improving (not specifically discussing the policies of sharing real-time
> traffic space), I would like to explain:
>
> Over an IP pipe only used for real-traffic (no TCP data traffic), it is
> sufficient that the pipe is wide enough for good QoS. That is often used =
and
> implemented by separating IP pipes at a lower level using e.g. Ethernet
> VLAN, MPLS, Ethernet over ATM (std for ADSL modems). TURN servers can
> enforce real-traffic into such pipes and QoS is achieved. Let's call this
> Level 2 QoS (Network level)
>
> Over an IP pipe shared between quality requiring real-time traffic and le=
ss
> demanding data or streaming (usually TCP) traffic, we have the sharing
> between these two traffic classes to consider. The main method (and the
> mechanism making today's real-time QoE as good as it often is) is that TC=
P
> endpoints back-off and share their bandwidth usage. Real-time traffic usi=
ng
> UDP transport do not back-off, the endpoints using UDP occupy the bandwid=
th
> needed. When an IP pipe gets filled, it is all endpoint's TCP bandwidth
> usage that is back-off and shared between them, leaving room for the UDP
> traffic. This is mechanism we experience everyday over the Internet, usin=
g
> our "Surf IP pipes". Let's call this Level 4 QoS (Transport level)
>
> However, this Level 4 QoS is based on that at congestion times (which hap=
pen
> every time we click - setting up a TCP flow and transferring some amount =
of
> data as quick as possible) the router handling the most narrow part of th=
e
> pipe (the congestion point) drop packets. It is this packet dropping that
> (i) signals to TCP endpoints to reduce their bandwidth (via TCP's error
> correction/retransmission mechanism) and (ii) destroys the QoE of real-ti=
me
> traffic. (Both TCP and UDP packets are dropped in this process that is
> triggered by flow intensive TCP traffic.)
>
> We need (i) but don't want (ii) and to improve on this we can e.g. use
> diffserve, DSCP bits in IP packets to instruct routers to always forward =
the
> real-time traffic before any unmarked TCP traffic (which usually fills mo=
st
> of the pipe). Then QoS is then achieved for real-time traffic. This is Le=
vel
> 3 QoS (IP level). The method used is "traffic shaping": backing off data
> traffic, leaving real-time traffic free passage without packet loss.
>
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org<mailto:rtcweb@ietf.org>
> https://www.ietf.org/mailman/listinfo/rtcweb

--_000_CF7D2A6D464A8stewesteweorg_
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<html>
<head>
<meta http-equiv=3D"Content-Type" content=3D"text/html; charset=3Diso-8859-=
1">
</head>
<body style=3D"word-wrap: break-word; -webkit-nbsp-mode: space; -webkit-lin=
e-break: after-white-space; color: rgb(0, 0, 0); font-size: 14px; font-fami=
ly: Calibri, sans-serif;">
<div>Hi,</div>
<div>Please see inline.</div>
<div>Stephan</div>
<blockquote style=3D"margin:0 0 0 40px; border:none; padding:0px;">
<div><br>
</div>
</blockquote>
<span id=3D"OLK_SRC_BODY_SECTION">
<blockquote style=3D"margin:0 0 0 40px; border:none; padding:0px;">
<div style=3D"font-family:Calibri; font-size:11pt; text-align:left; color:b=
lack; BORDER-BOTTOM: medium none; BORDER-LEFT: medium none; PADDING-BOTTOM:=
 0in; PADDING-LEFT: 0in; PADDING-RIGHT: 0in; BORDER-TOP: #b5c4df 1pt solid;=
 BORDER-RIGHT: medium none; PADDING-TOP: 3pt">
<span style=3D"font-weight:bold">From: </span>Roman Shpount &lt;<a href=3D"=
mailto:roman@telurix.com">roman@telurix.com</a>&gt;<br>
<span style=3D"font-weight:bold">Date: </span>Wednesday, April 23, 2014 at =
8:02<br>
<span style=3D"font-weight:bold">To: </span>Harald Alvestrand &lt;<a href=
=3D"mailto:harald@alvestrand.no">harald@alvestrand.no</a>&gt;<br>
<span style=3D"font-weight:bold">Cc: </span>&quot;<a href=3D"mailto:rtcweb@=
ietf.org">rtcweb@ietf.org</a>&quot; &lt;<a href=3D"mailto:rtcweb@ietf.org">=
rtcweb@ietf.org</a>&gt;<br>
<span style=3D"font-weight:bold">Subject: </span>Re: [rtcweb] Pictures of c=
ongestion control on the Internet - which is more realistic?<br>
</div>
<div><br>
</div>
</blockquote>
<div>
<blockquote style=3D"margin:0 0 0 40px; border:none; padding:0px;"></blockq=
uote>
<div>
<blockquote style=3D"margin:0 0 0 40px; border:none; padding:0px;">
<p dir=3D"ltr">I would say that some video end points have congestion back =
off.&nbsp; </p>
</blockquote>
</div>
</div>
</span>
<div>Concur. &nbsp;In fact, I would replace &#8220;some&#8221; with &#8220;=
very few&#8221;, and I think the statement would still be correct. &nbsp;At=
 least when it comes to the hardware based solutions, and your typical SIP =
client. &nbsp;The proprietary soft-clients (Skype, hangouts, etc.) may
 be a better in that regard, especially the modern ones that use scalabilit=
y (like ours, brag brag). &nbsp;(One can, of course, argue that at any give=
n moment there are more sessions active using modern soft-clients than the =
combined lifetime sales of all traditional
 hardware video conference vendors&#8212;and that&#8217;s why we haven&#821=
7;t had a meltdown yet. &nbsp;I attribute the lack of a meltdown to other r=
easons, though.)&nbsp;</div>
<span id=3D"OLK_SRC_BODY_SECTION">
<div>
<div>
<blockquote style=3D"margin:0 0 0 40px; border:none; padding:0px;">
<p dir=3D"ltr">Almost all audio end points have none. </p>
</blockquote>
</div>
</div>
</span>
<div>Concur as well. &nbsp;And, I believe the number of bits spent on inter=
active audio over the open Internet is still considerably higher than those=
 for interactive video.</div>
<div><br>
</div>
<div>In both cases, the typical reaction of an endpoint is to expose the us=
er to crappy quality, until the user hangs up. &nbsp;Sometimes, slightly sm=
arter implementations hang up themselves if the packet loss rate becomes ex=
cessive.I would suggest not to paint
 a rosier picture in the draft than what is warranted based on real-world o=
bservations. &nbsp;</div>
<span id=3D"OLK_SRC_BODY_SECTION">
<div>
<div>
<blockquote style=3D"margin:0 0 0 40px; border:none; padding:0px;">
<p dir=3D"ltr">Based on this, most UDP endpoints do not deal well with cong=
estion. Ideally this should change to support better bandwidth management. =
Futhermore, reliably transmitting real time media over public internet woul=
d also require changes to biffer management
 algorithms in routers, which will, in turn, affect congestion handling str=
ategies in the end points.</p>
<p dir=3D"ltr">On Apr 23, 2014 8:31 AM, &quot;Harald Alvestrand&quot; &lt;<=
a href=3D"mailto:harald@alvestrand.no">harald@alvestrand.no</a>&gt; wrote:<=
br>
&gt;<br>
&gt; I quote Karl's piece below, because I want other's opinion of whether =
it's a realistic picture.<br>
&gt;<br>
&gt; My impression is that it is not - in particular that:<br>
&gt;<br>
&gt; - Lots of interactive traffic goes over the Internet today without lay=
er 2 separation<br>
&gt; - All UDP endpoints that use the open Internet have some form of backo=
ff on congestion - using delay, packet drop or (most likely) both to detect=
 congestion<br>
&gt;<br>
&gt; I'm writing the transport document based on those assumptions - that w=
e should specify something that will work as well as it can over an Interne=
t that does not care, and try to allow for better things to come along late=
r.<br>
&gt;<br>
&gt; Am I wrong to write it like this?<br>
&gt;<br>
&gt; Harald<br>
&gt;<br>
&gt;<br>
&gt;<br>
&gt; ------------------------------------<br>
&gt; [Karl] I wrote this sometime before, regarding how QoS over the intern=
et<br>
&gt; works in general - may be useful for understanding<br>
&gt;<br>
&gt; For better understanding of these QoS/QoE problems, and methods for<br=
>
&gt; improving (not specifically discussing the policies of sharing real-ti=
me<br>
&gt; traffic space), I would like to explain:<br>
&gt;<br>
&gt; Over an IP pipe only used for real-traffic (no TCP data traffic), it i=
s<br>
&gt; sufficient that the pipe is wide enough for good QoS. That is often us=
ed and<br>
&gt; implemented by separating IP pipes at a lower level using e.g. Etherne=
t<br>
&gt; VLAN, MPLS, Ethernet over ATM (std for ADSL modems). TURN servers can<=
br>
&gt; enforce real-traffic into such pipes and QoS is achieved. Let&#8217;s =
call this<br>
&gt; Level 2 QoS (Network level)<br>
&gt;<br>
&gt; Over an IP pipe shared between quality requiring real-time traffic and=
 less<br>
&gt; demanding data or streaming (usually TCP) traffic, we have the sharing=
<br>
&gt; between these two traffic classes to consider. The main method (and th=
e<br>
&gt; mechanism making today&#8217;s real-time QoE as good as it often is) i=
s that TCP<br>
&gt; endpoints back-off and share their bandwidth usage. Real-time traffic =
using<br>
&gt; UDP transport do not back-off, the endpoints using UDP occupy the band=
width<br>
&gt; needed. When an IP pipe gets filled, it is all endpoint&#8217;s TCP ba=
ndwidth<br>
&gt; usage that is back-off and shared between them, leaving room for the U=
DP<br>
&gt; traffic. This is mechanism we experience everyday over the Internet, u=
sing<br>
&gt; our &#8220;Surf IP pipes&#8221;. Let&#8217;s call this Level 4 QoS (Tr=
ansport level)<br>
&gt;<br>
&gt; However, this Level 4 QoS is based on that at congestion times (which =
happen<br>
&gt; every time we click &#8211; setting up a TCP flow and transferring som=
e amount of<br>
&gt; data as quick as possible) the router handling the most narrow part of=
 the<br>
&gt; pipe (the congestion point) drop packets. It is this packet dropping t=
hat<br>
&gt; (i) signals to TCP endpoints to reduce their bandwidth (via TCP&#8217;=
s error<br>
&gt; correction/retransmission mechanism) and (ii) destroys the QoE of real=
-time<br>
&gt; traffic. (Both TCP and UDP packets are dropped in this process that is=
<br>
&gt; triggered by flow intensive TCP traffic.)<br>
&gt;<br>
&gt; We need (i) but don&#8217;t want (ii) and to improve on this we can e.=
g. use<br>
&gt; diffserve, DSCP bits in IP packets to instruct routers to always forwa=
rd the<br>
&gt; real-time traffic before any unmarked TCP traffic (which usually fills=
 most<br>
&gt; of the pipe). Then QoS is then achieved for real-time traffic. This is=
 Level<br>
&gt; 3 QoS (IP level). The method used is &#8220;traffic shaping&#8221;: ba=
cking off data<br>
&gt; traffic, leaving real-time traffic free passage without packet loss.<b=
r>
&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">https://www.i=
etf.org/mailman/listinfo/rtcweb</a><br>
</p>
</blockquote>
</div>
</div>
</span>
</body>
</html>

--_000_CF7D2A6D464A8stewesteweorg_--


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On 23 April 2014 05:30, Harald Alvestrand <harald@alvestrand.no> wrote:
> - All UDP endpoints that use the open Internet have some form of backoff on
> congestion - using delay, packet drop or (most likely) both to detect
> congestion

There's some evidence that this is true, but we wouldn't have RMCAT if
it weren't for the fact that there was pretty wide consensus that it's
not good enough.  Maybe the vendors of those soft video clients
Stephan mentions are confident that they have the answer, I don't
know.


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Dave Taht <dave.taht@gmail.com> wrote:
> 
> I'm not sure if we have a shared definition of "congestion".  Mine is
> network delays exceeding that of what human factors research notes
> as "perceptible" with about 20ms as the outer bound.

   I, OTOH, am quite sure we don't have a shared definition of congestion.

   My preferred definition is rate-of-arrival exceeding rate-of-forwarding.

   But, of course, you don't have to agree with mine -- "shared" merely
implies _some_ overlap of the sets of definitions we use...

   Dave's, I fear, isn't terribly useful. There is a base latency (in the
absence of any traffic) which exceeds 20 milliseconds for many paths
of interest.

   To tell truth, though, most users probably only notice the total delay,
not the components of it; so I think Dave's metric is useful -- it's just
not helpful as a measure of "congestion".

   I'd like to follow up a bit on Dave's metric, regardless.

   There's a lot of talk about "buffer-bloat" where I believe we all
agree the situation would be better with less buffering.

   But can we talk about the other end of that spectrum?

   Surely there is some minimal amount of buffering, for any particular
traffic, which is needed in order to maintain a reasonable flow rate.
Most likely, that minimal buffering varies with factors such as RTT. Do
we even have a shared concept of what those factors are?

   I'd like to see a thread on what is needed as a minimum buffering
for various kinds of traffic. I suspect that minimum can be fairly small
except for TCP transport of "large" files...

> http://gettys.wordpress.com/2013/07/10/low-latency-requires-smart-queuing-traditional-aqm-is-not-enough/

   Excellent as food for thought...

--
John Leslie <john@jlc.net>


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Would that be a good discussion to have on the rmcat list?

I hear they're close to finishing their requirements and test 
methodology documents.....


On 04/23/2014 06:37 PM, John Leslie wrote:
> Dave Taht <dave.taht@gmail.com> wrote:
>> I'm not sure if we have a shared definition of "congestion".  Mine is
>> network delays exceeding that of what human factors research notes
>> as "perceptible" with about 20ms as the outer bound.
>     I, OTOH, am quite sure we don't have a shared definition of congestion.
>
>     My preferred definition is rate-of-arrival exceeding rate-of-forwarding.
>
>     But, of course, you don't have to agree with mine -- "shared" merely
> implies _some_ overlap of the sets of definitions we use...
>
>     Dave's, I fear, isn't terribly useful. There is a base latency (in the
> absence of any traffic) which exceeds 20 milliseconds for many paths
> of interest.
>
>     To tell truth, though, most users probably only notice the total delay,
> not the components of it; so I think Dave's metric is useful -- it's just
> not helpful as a measure of "congestion".
>
>     I'd like to follow up a bit on Dave's metric, regardless.
>
>     There's a lot of talk about "buffer-bloat" where I believe we all
> agree the situation would be better with less buffering.
>
>     But can we talk about the other end of that spectrum?
>
>     Surely there is some minimal amount of buffering, for any particular
> traffic, which is needed in order to maintain a reasonable flow rate.
> Most likely, that minimal buffering varies with factors such as RTT. Do
> we even have a shared concept of what those factors are?
>
>     I'd like to see a thread on what is needed as a minimum buffering
> for various kinds of traffic. I suspect that minimum can be fairly small
> except for TCP transport of "large" files...
>
>> http://gettys.wordpress.com/2013/07/10/low-latency-requires-smart-queuing-traditional-aqm-is-not-enough/
>     Excellent as food for thought...
>
> --
> John Leslie <john@jlc.net>


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From: "Karl Stahl" <karl.stahl@intertex.se>
To: "'Harald Alvestrand'" <harald@alvestrand.no>, <rtcweb@ietf.org>
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-----Ursprungligt meddelande-----
Fr=E5n: rtcweb [mailto:rtcweb-bounces@ietf.org] F=F6r Harald Alvestrand
Skickat: den 23 april 2014 14:31
Till: rtcweb@ietf.org
=C4mne: [rtcweb] Pictures of congestion control on the Internet - which =
is
more realistic?

I quote Karl's piece below, because I want other's opinion of whether =
it's a
realistic picture.

My impression is that it is not - in particular that:

- Lots of interactive traffic goes over the Internet today without layer =
2
separation

[Karl] I am NOT for more level 2 separation (was describing it though) - =
we
are on the level 3 Internet! We can do equally well with level 3 QoS
(diffserve and reservation, but for that we need the traffic info in =
each
packet, unchanged end-to-end, so network elements can act upon it.)

[Karl] Today we mostly only have level 4 QoS over the Internet. We need
level 3 QoS for WebRTC.=20

- All UDP endpoints that use the open Internet have some form of backoff =
on
congestion - using delay, packet drop or (most likely) both to detect
congestion

[Karl] Unfortunately that is not very effective. With level 4 QoS, UDP
backing off means that TCP flows take that bandwidth instead (today's
Internet traffic is very TCP-flow intensive), and the congestion is back
again. (Had we level 3 QoS, and starting to fill the whole pipe with
priority data, then priority data backing off would be effective. =
Otherwise,
the improvement is small, if any.)=20


I'm writing the transport document based on those assumptions - that we
should specify something that will work as well as it can over an =
Internet
that does not care, and try to allow for better things to come along =
later.

[Karl] Two things to address for achieving that:

1. Doing file transfer in the data channel bundled with real-time =
traffic in
the same flow/5-tuple, destroys the level 4 QoS we already have over the
Internet. Getting file transfer out of the bundle restores that. SCTP's =
prio
mechanism cannot restore that when the congestion is in the network =
(only
when in the endpoints)=20

2). To "try to allow for better things to come along" we need the =
traffic
info (bandwidth requirement and traffic type) in every priority packet,
readable at the network level, unchanged end-to-end. Then network =
elements
can be programmed to act upon that traffic info to give us level 3 QoS.
(When routing, the priority packets are simply put first in the out =
queue,
i.e. never dropped, but TCP file transfer flows will have packets =
discarded
and be backed off. Reservation type of networks will have to do some
calculations also.) It is easy to put in that traffic info now - very
difficult to do it later.


Am I wrong to write it like this?

Harald



------------------------------------
[Karl] I wrote this sometime before, regarding how QoS over the internet
works in general - may be useful for understanding

For better understanding of these QoS/QoE problems, and methods for
improving (not specifically discussing the policies of sharing real-time
traffic space), I would like to explain:

Over an IP pipe only used for real-traffic (no TCP data traffic), it is
sufficient that the pipe is wide enough for good QoS. That is often used =
and
implemented by separating IP pipes at a lower level using e.g. Ethernet
VLAN, MPLS, Ethernet over ATM (std for ADSL modems). TURN servers can
enforce real-traffic into such pipes and QoS is achieved. Let=92s call =
this
Level 2 QoS (Network level)

Over an IP pipe shared between quality requiring real-time traffic and =
less
demanding data or streaming (usually TCP) traffic, we have the sharing
between these two traffic classes to consider. The main method (and the
mechanism making today=92s real-time QoE as good as it often is) is that =
TCP
endpoints back-off and share their bandwidth usage. Real-time traffic =
using
UDP transport do not back-off, the endpoints using UDP occupy the =
bandwidth
needed. When an IP pipe gets filled, it is all endpoint=92s TCP =
bandwidth
usage that is back-off and shared between them, leaving room for the UDP
traffic. This is mechanism we experience everyday over the Internet, =
using
our =93Surf IP pipes=94. Let=92s call this Level 4 QoS (Transport level)

However, this Level 4 QoS is based on that at congestion times (which =
happen
every time we click =96 setting up a TCP flow and transferring some =
amount of
data as quick as possible) the router handling the most narrow part of =
the
pipe (the congestion point) drop packets. It is this packet dropping =
that
(i) signals to TCP endpoints to reduce their bandwidth (via TCP=92s =
error
correction/retransmission mechanism) and (ii) destroys the QoE of =
real-time
traffic. (Both TCP and UDP packets are dropped in this process that is
triggered by flow intensive TCP traffic.)

We need (i) but don=92t want (ii) and to improve on this we can e.g. use
diffserve, DSCP bits in IP packets to instruct routers to always forward =
the
real-time traffic before any unmarked TCP traffic (which usually fills =
most
of the pipe). Then QoS is then achieved for real-time traffic. This is =
Level
3 QoS (IP level). The method used is =93traffic shaping=94: backing off =
data
traffic, leaving real-time traffic free passage without packet loss.

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


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To: "'Stephan Wenger'" <stewe@stewe.org>, "'Roman Shpount'" <roman@telurix.com>, "'Harald Alvestrand'" <harald@alvestrand.no>
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I think real-time media backing-off is rarely used, because it is seldom
efficient. As long as we rely on level 4 QoS, the freed bandwidth is =
simply
taken by flow intensive TCP file transfer traffic and congestion is back
again.

> Futhermore, reliably transmitting real time media over public internet
would also require changes to biffer management algorithms in routers=85

--- Yes, need to be done in the routers =96 endpoints cannot simply do =
it
alone.

And routers need to know what to prioritize or reserve for: We need the
traffic info (bandwidth requirement and traffic type) in each RTP or =
data
channel priority packet, readable by the network and unchanged =
end-to-end
over the Internet [1] [2].

=20

/Karl

=20

[1] Traffic type information to encode (recreating the=20

idea/intention of the RTP payload type (PT) byte that is not=20

available for the network layer anymore, by instead having the=20

browser filling the RTP header extension with relevant traffic=20

info that all IP network types can use.).Two parameters to encode=20

(Bandwidth and traffic type)

 <http://www.ietf.org/mail-archive/web/rtcweb/current/msg09129.html>
http://www.ietf.org/mail-archive/web/rtcweb/current/msg09129.html=20

this draft should be filled for use with WebRTC:=20

http://tools.ietf.org/html/draft-carlberg-avtext-classifier-01

=20

[2] For the WebRTC data channel, the traffic info could be inserted as=20

suggested (in the same format as in the RTP header extension):=20

 <http://www.ietf.org/mail-archive/web/rtcweb/current/msg11973.html>
http://www.ietf.org/mail-archive/web/rtcweb/current/msg11973.html =20

=20

=20

Fr=E5n: rtcweb [mailto:rtcweb-bounces@ietf.org] F=F6r Stephan Wenger
Skickat: den 23 april 2014 17:47
Till: Roman Shpount; Harald Alvestrand
Kopia: rtcweb@ietf.org
=C4mne: Re: [rtcweb] Pictures of congestion control on the Internet - =
which is
more realistic?

=20

Hi,

Please see inline.

Stephan

=20

From: Roman Shpount <roman@telurix.com>
Date: Wednesday, April 23, 2014 at 8:02
To: Harald Alvestrand <harald@alvestrand.no>
Cc: "rtcweb@ietf.org" <rtcweb@ietf.org>
Subject: Re: [rtcweb] Pictures of congestion control on the Internet - =
which
is more realistic?

=20

I would say that some video end points have congestion back off. =20

Concur.  In fact, I would replace =93some=94 with =93very few=94, and I =
think the
statement would still be correct.  At least when it comes to the =
hardware
based solutions, and your typical SIP client.  The proprietary =
soft-clients
(Skype, hangouts, etc.) may be a better in that regard, especially the
modern ones that use scalability (like ours, brag brag).  (One can, of
course, argue that at any given moment there are more sessions active =
using
modern soft-clients than the combined lifetime sales of all traditional
hardware video conference vendors=97and that=92s why we haven=92t had a =
meltdown
yet.  I attribute the lack of a meltdown to other reasons, though.)=20

Almost all audio end points have none.=20

Concur as well.  And, I believe the number of bits spent on interactive
audio over the open Internet is still considerably higher than those for
interactive video.

=20

In both cases, the typical reaction of an endpoint is to expose the user =
to
crappy quality, until the user hangs up.  Sometimes, slightly smarter
implementations hang up themselves if the packet loss rate becomes
excessive.I would suggest not to paint a rosier picture in the draft =
than
what is warranted based on real-world observations. =20

Based on this, most UDP endpoints do not deal well with congestion. =
Ideally
this should change to support better bandwidth management. Futhermore,
reliably transmitting real time media over public internet would also
require changes to biffer management algorithms in routers, which will, =
in
turn, affect congestion handling strategies in the end points.

On Apr 23, 2014 8:31 AM, "Harald Alvestrand" <harald@alvestrand.no> =
wrote:
>
> I quote Karl's piece below, because I want other's opinion of whether =
it's
a realistic picture.
>
> My impression is that it is not - in particular that:
>
> - Lots of interactive traffic goes over the Internet today without =
layer 2
separation
> - All UDP endpoints that use the open Internet have some form of =
backoff
on congestion - using delay, packet drop or (most likely) both to detect
congestion
>
> I'm writing the transport document based on those assumptions - that =
we
should specify something that will work as well as it can over an =
Internet
that does not care, and try to allow for better things to come along =
later.
>
> Am I wrong to write it like this?
>
> Harald
>
>
>
> ------------------------------------
> [Karl] I wrote this sometime before, regarding how QoS over the =
internet
> works in general - may be useful for understanding
>
> For better understanding of these QoS/QoE problems, and methods for
> improving (not specifically discussing the policies of sharing =
real-time
> traffic space), I would like to explain:
>
> Over an IP pipe only used for real-traffic (no TCP data traffic), it =
is
> sufficient that the pipe is wide enough for good QoS. That is often =
used
and
> implemented by separating IP pipes at a lower level using e.g. =
Ethernet
> VLAN, MPLS, Ethernet over ATM (std for ADSL modems). TURN servers can
> enforce real-traffic into such pipes and QoS is achieved. Let=92s call =
this
> Level 2 QoS (Network level)
>
> Over an IP pipe shared between quality requiring real-time traffic and
less
> demanding data or streaming (usually TCP) traffic, we have the sharing
> between these two traffic classes to consider. The main method (and =
the
> mechanism making today=92s real-time QoE as good as it often is) is =
that TCP
> endpoints back-off and share their bandwidth usage. Real-time traffic
using
> UDP transport do not back-off, the endpoints using UDP occupy the
bandwidth
> needed. When an IP pipe gets filled, it is all endpoint=92s TCP =
bandwidth
> usage that is back-off and shared between them, leaving room for the =
UDP
> traffic. This is mechanism we experience everyday over the Internet, =
using
> our =93Surf IP pipes=94. Let=92s call this Level 4 QoS (Transport =
level)
>
> However, this Level 4 QoS is based on that at congestion times (which
happen
> every time we click =96 setting up a TCP flow and transferring some =
amount
of
> data as quick as possible) the router handling the most narrow part of =
the
> pipe (the congestion point) drop packets. It is this packet dropping =
that
> (i) signals to TCP endpoints to reduce their bandwidth (via TCP=92s =
error
> correction/retransmission mechanism) and (ii) destroys the QoE of
real-time
> traffic. (Both TCP and UDP packets are dropped in this process that is
> triggered by flow intensive TCP traffic.)
>
> We need (i) but don=92t want (ii) and to improve on this we can e.g. =
use
> diffserve, DSCP bits in IP packets to instruct routers to always =
forward
the
> real-time traffic before any unmarked TCP traffic (which usually fills
most
> of the pipe). Then QoS is then achieved for real-time traffic. This is
Level
> 3 QoS (IP level). The method used is =93traffic shaping=94: backing =
off data
> traffic, leaving real-time traffic free passage without packet loss.
>
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb


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vlink=3Dpurple><div class=3DWordSection1><p class=3DMsoPlainText><span =
lang=3DEN-US style=3D'font-family:"Calibri","sans-serif";color:black'>I =
think real-time media backing-off is rarely used, because it is seldom =
efficient. As long as we rely on level 4 QoS, the freed bandwidth is =
simply taken by flow intensive TCP file transfer traffic and congestion =
is back again.<o:p></o:p></span></p><p><span lang=3DEN-US =
style=3D'font-family:"Calibri","sans-serif";color:black'>&gt; =
</span><span lang=3DEN-US =
style=3D'font-size:10.5pt;font-family:"Calibri","sans-serif";color:black'=
>Futhermore, reliably transmitting real time media over public internet =
would also require changes to biffer management algorithms in =
routers&#8230;<o:p></o:p></span></p><p class=3DMsoPlainText><span =
lang=3DEN-US =
style=3D'font-family:"Calibri","sans-serif";color:black'>--- Yes, need =
to be done in the routers &#8211; endpoints cannot simply do it =
alone.<o:p></o:p></span></p><p class=3DMsoPlainText><span lang=3DEN-US =
style=3D'font-family:"Calibri","sans-serif";color:black'>And routers =
need to know what to prioritize or reserve for: We need the traffic info =
(bandwidth requirement and traffic type) in each RTP or data channel =
priority packet, readable by the network and unchanged end-to-end over =
the Internet [1] [2].<o:p></o:p></span></p><p class=3DMsoPlainText><span =
lang=3DEN-US =
style=3D'font-family:"Calibri","sans-serif";color:black'><o:p>&nbsp;</o:p=
></span></p><p class=3DMsoPlainText><span lang=3DEN-US =
style=3D'font-family:"Calibri","sans-serif";color:black'>/Karl<o:p></o:p>=
</span></p><p class=3DMsoPlainText><span lang=3DEN-US =
style=3D'color:black'><o:p>&nbsp;</o:p></span></p><p =
class=3DMsoPlainText><span lang=3DEN-US>[1] Traffic type information to =
encode (recreating the <o:p></o:p></span></p><p =
class=3DMsoPlainText><span lang=3DEN-US>idea/intention of the RTP =
payload type (PT) byte that is not <o:p></o:p></span></p><p =
class=3DMsoPlainText><span lang=3DEN-US>available for the network layer =
anymore, by instead having the <o:p></o:p></span></p><p =
class=3DMsoPlainText><span lang=3DEN-US>browser filling the RTP header =
extension with relevant traffic <o:p></o:p></span></p><p =
class=3DMsoPlainText><span lang=3DEN-US>info that all IP network types =
can use.).Two parameters to encode <o:p></o:p></span></p><p =
class=3DMsoPlainText><span lang=3DEN-US>(Bandwidth and traffic =
type)<o:p></o:p></span></p><p class=3DMsoPlainText><a =
href=3D"http://www.ietf.org/mail-archive/web/rtcweb/current/msg09129.html=
"><span =
lang=3DEN-US>http://www.ietf.org/mail-archive/web/rtcweb/current/msg09129=
.html</span></a> <span lang=3DEN-US><o:p></o:p></span></p><p =
class=3DMsoPlainText><span lang=3DEN-US>this draft should be filled for =
use with WebRTC: <o:p></o:p></span></p><p class=3DMsoPlainText><span =
lang=3DEN-US><a =
href=3D"http://tools.ietf.org/html/draft-carlberg-avtext-classifier-01">h=
ttp://tools.ietf.org/html/draft-carlberg-avtext-classifier-01</a><o:p></o=
:p></span></p><p class=3DMsoPlainText><span =
lang=3DEN-US><o:p>&nbsp;</o:p></span></p><p class=3DMsoPlainText><span =
lang=3DEN-US>[2] For the WebRTC data channel, the traffic info could be =
inserted as <o:p></o:p></span></p><p class=3DMsoPlainText><span =
lang=3DEN-US>suggested (in the same format as in the RTP header =
extension): <o:p></o:p></span></p><p class=3DMsoPlainText><a =
href=3D"http://www.ietf.org/mail-archive/web/rtcweb/current/msg11973.html=
"><span =
lang=3DEN-US>http://www.ietf.org/mail-archive/web/rtcweb/current/msg11973=
.html</span></a><span lang=3DEN-US>&nbsp; <o:p></o:p></span></p><p =
class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'><o=
:p>&nbsp;</o:p></span></p><p class=3DMsoNormal><span lang=3DEN-US =
style=3D'font-size:10.0pt;font-family:"Arial","sans-serif";color:blue'><o=
:p>&nbsp;</o:p></span></p><div><div =
style=3D'border:none;border-top:solid #B5C4DF 1.0pt;padding:3.0pt 0cm =
0cm 0cm'><p class=3DMsoNormal><b><span =
style=3D'font-size:10.0pt;font-family:"Tahoma","sans-serif"'>Fr=E5n:</spa=
n></b><span =
style=3D'font-size:10.0pt;font-family:"Tahoma","sans-serif"'> rtcweb =
[mailto:rtcweb-bounces@ietf.org] <b>F=F6r </b>Stephan =
Wenger<br><b>Skickat:</b> den 23 april 2014 17:47<br><b>Till:</b> Roman =
Shpount; Harald Alvestrand<br><b>Kopia:</b> =
rtcweb@ietf.org<br><b>=C4mne:</b> Re: [rtcweb] Pictures of congestion =
control on the Internet - which is more =
realistic?<o:p></o:p></span></p></div></div><p =
class=3DMsoNormal><o:p>&nbsp;</o:p></p><div><p class=3DMsoNormal><span =
style=3D'font-size:10.5pt;font-family:"Calibri","sans-serif";color:black'=
>Hi,<o:p></o:p></span></p></div><div><p class=3DMsoNormal><span =
style=3D'font-size:10.5pt;font-family:"Calibri","sans-serif";color:black'=
>Please see inline.<o:p></o:p></span></p></div><div><p =
class=3DMsoNormal><span =
style=3D'font-size:10.5pt;font-family:"Calibri","sans-serif";color:black'=
>Stephan<o:p></o:p></span></p></div><blockquote =
style=3D'margin-left:30.0pt;margin-right:0cm'><div><p =
class=3DMsoNormal><span =
style=3D'font-size:10.5pt;font-family:"Calibri","sans-serif";color:black'=
><o:p>&nbsp;</o:p></span></p></div></blockquote><blockquote =
style=3D'margin-left:30.0pt;margin-right:0cm'><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:11.0pt;font-family:"Calibri","sans-serif";color:black'=
>From: </span></b><span =
style=3D'font-size:11.0pt;font-family:"Calibri","sans-serif";color:black'=
>Roman Shpount &lt;<a =
href=3D"mailto:roman@telurix.com">roman@telurix.com</a>&gt;<br><b>Date: =
</b>Wednesday, April 23, 2014 at 8:02<br><b>To: </b>Harald Alvestrand =
&lt;<a =
href=3D"mailto:harald@alvestrand.no">harald@alvestrand.no</a>&gt;<br><b>C=
c: </b>&quot;<a =
href=3D"mailto:rtcweb@ietf.org">rtcweb@ietf.org</a>&quot; &lt;<a =
href=3D"mailto:rtcweb@ietf.org">rtcweb@ietf.org</a>&gt;<br><b>Subject: =
</b>Re: [rtcweb] Pictures of congestion control on the Internet - which =
is more realistic?<o:p></o:p></span></p></div><div><p =
class=3DMsoNormal><span =
style=3D'font-size:10.5pt;font-family:"Calibri","sans-serif";color:black'=
><o:p>&nbsp;</o:p></span></p></div></blockquote><div><div><blockquote =
style=3D'margin-left:30.0pt;margin-right:0cm'><p><span =
style=3D'font-size:10.5pt;font-family:"Calibri","sans-serif";color:black'=
>I would say that some video end points have congestion back off.&nbsp; =
<o:p></o:p></span></p></blockquote></div></div><div><p =
class=3DMsoNormal><span =
style=3D'font-size:10.5pt;font-family:"Calibri","sans-serif";color:black'=
>Concur. &nbsp;In fact, I would replace &#8220;some&#8221; with =
&#8220;very few&#8221;, and I think the statement would still be =
correct. &nbsp;At least when it comes to the hardware based solutions, =
and your typical SIP client. &nbsp;The proprietary soft-clients (Skype, =
hangouts, etc.) may be a better in that regard, especially the modern =
ones that use scalability (like ours, brag brag). &nbsp;(One can, of =
course, argue that at any given moment there are more sessions active =
using modern soft-clients than the combined lifetime sales of all =
traditional hardware video conference vendors&#8212;and that&#8217;s why =
we haven&#8217;t had a meltdown yet. &nbsp;I attribute the lack of a =
meltdown to other reasons, =
though.)&nbsp;<o:p></o:p></span></p></div><div><div><blockquote =
style=3D'margin-left:30.0pt;margin-right:0cm'><p><span =
style=3D'font-size:10.5pt;font-family:"Calibri","sans-serif";color:black'=
>Almost all audio end points have none. =
<o:p></o:p></span></p></blockquote></div></div><div><p =
class=3DMsoNormal><span =
style=3D'font-size:10.5pt;font-family:"Calibri","sans-serif";color:black'=
>Concur as well. &nbsp;And, I believe the number of bits spent on =
interactive audio over the open Internet is still considerably higher =
than those for interactive video.<o:p></o:p></span></p></div><div><p =
class=3DMsoNormal><span =
style=3D'font-size:10.5pt;font-family:"Calibri","sans-serif";color:black'=
><o:p>&nbsp;</o:p></span></p></div><div><p class=3DMsoNormal><span =
style=3D'font-size:10.5pt;font-family:"Calibri","sans-serif";color:black'=
>In both cases, the typical reaction of an endpoint is to expose the =
user to crappy quality, until the user hangs up. &nbsp;Sometimes, =
slightly smarter implementations hang up themselves if the packet loss =
rate becomes excessive.I would suggest not to paint a rosier picture in =
the draft than what is warranted based on real-world observations. =
&nbsp;<o:p></o:p></span></p></div><div><div><blockquote =
style=3D'margin-left:30.0pt;margin-right:0cm'><p><span =
style=3D'font-size:10.5pt;font-family:"Calibri","sans-serif";color:black'=
>Based on this, most UDP endpoints do not deal well with congestion. =
Ideally this should change to support better bandwidth management. =
Futhermore, reliably transmitting real time media over public internet =
would also require changes to biffer management algorithms in routers, =
which will, in turn, affect congestion handling strategies in the end =
points.<o:p></o:p></span></p><p><span =
style=3D'font-size:10.5pt;font-family:"Calibri","sans-serif";color:black'=
>On Apr 23, 2014 8:31 AM, &quot;Harald Alvestrand&quot; &lt;<a =
href=3D"mailto:harald@alvestrand.no">harald@alvestrand.no</a>&gt; =
wrote:<br>&gt;<br>&gt; I quote Karl's piece below, because I want =
other's opinion of whether it's a realistic picture.<br>&gt;<br>&gt; My =
impression is that it is not - in particular that:<br>&gt;<br>&gt; - =
Lots of interactive traffic goes over the Internet today without layer 2 =
separation<br>&gt; - All UDP endpoints that use the open Internet have =
some form of backoff on congestion - using delay, packet drop or (most =
likely) both to detect congestion<br>&gt;<br>&gt; I'm writing the =
transport document based on those assumptions - that we should specify =
something that will work as well as it can over an Internet that does =
not care, and try to allow for better things to come along =
later.<br>&gt;<br>&gt; Am I wrong to write it like this?<br>&gt;<br>&gt; =
Harald<br>&gt;<br>&gt;<br>&gt;<br>&gt; =
------------------------------------<br>&gt; [Karl] I wrote this =
sometime before, regarding how QoS over the internet<br>&gt; works in =
general - may be useful for understanding<br>&gt;<br>&gt; For better =
understanding of these QoS/QoE problems, and methods for<br>&gt; =
improving (not specifically discussing the policies of sharing =
real-time<br>&gt; traffic space), I would like to =
explain:<br>&gt;<br>&gt; Over an IP pipe only used for real-traffic (no =
TCP data traffic), it is<br>&gt; sufficient that the pipe is wide enough =
for good QoS. That is often used and<br>&gt; implemented by separating =
IP pipes at a lower level using e.g. Ethernet<br>&gt; VLAN, MPLS, =
Ethernet over ATM (std for ADSL modems). TURN servers can<br>&gt; =
enforce real-traffic into such pipes and QoS is achieved. Let&#8217;s =
call this<br>&gt; Level 2 QoS (Network level)<br>&gt;<br>&gt; Over an IP =
pipe shared between quality requiring real-time traffic and less<br>&gt; =
demanding data or streaming (usually TCP) traffic, we have the =
sharing<br>&gt; between these two traffic classes to consider. The main =
method (and the<br>&gt; mechanism making today&#8217;s real-time QoE as =
good as it often is) is that TCP<br>&gt; endpoints back-off and share =
their bandwidth usage. Real-time traffic using<br>&gt; UDP transport do =
not back-off, the endpoints using UDP occupy the bandwidth<br>&gt; =
needed. When an IP pipe gets filled, it is all endpoint&#8217;s TCP =
bandwidth<br>&gt; usage that is back-off and shared between them, =
leaving room for the UDP<br>&gt; traffic. This is mechanism we =
experience everyday over the Internet, using<br>&gt; our &#8220;Surf IP =
pipes&#8221;. Let&#8217;s call this Level 4 QoS (Transport =
level)<br>&gt;<br>&gt; However, this Level 4 QoS is based on that at =
congestion times (which happen<br>&gt; every time we click &#8211; =
setting up a TCP flow and transferring some amount of<br>&gt; data as =
quick as possible) the router handling the most narrow part of =
the<br>&gt; pipe (the congestion point) drop packets. It is this packet =
dropping that<br>&gt; (i) signals to TCP endpoints to reduce their =
bandwidth (via TCP&#8217;s error<br>&gt; correction/retransmission =
mechanism) and (ii) destroys the QoE of real-time<br>&gt; traffic. (Both =
TCP and UDP packets are dropped in this process that is<br>&gt; =
triggered by flow intensive TCP traffic.)<br>&gt;<br>&gt; We need (i) =
but don&#8217;t want (ii) and to improve on this we can e.g. use<br>&gt; =
diffserve, DSCP bits in IP packets to instruct routers to always forward =
the<br>&gt; real-time traffic before any unmarked TCP traffic (which =
usually fills most<br>&gt; of the pipe). Then QoS is then achieved for =
real-time traffic. This is Level<br>&gt; 3 QoS (IP level). The method =
used is &#8220;traffic shaping&#8221;: backing off data<br>&gt; traffic, =
leaving real-time traffic free passage without packet =
loss.<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">https://www.ietf.or=
g/mailman/listinfo/rtcweb</a><o:p></o:p></span></p></blockquote></div></d=
iv></div></body></html>
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To: Harald Alvestrand <harald@alvestrand.no>, "rtcweb@ietf.org" <rtcweb@ietf.org>
Thread-Topic: [rtcweb] Pictures of congestion control on the Internet - which is more realistic?
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On 2014-04-23 14:31, Harald Alvestrand wrote:=0A=
> I quote Karl's piece below, because I want other's opinion of whether=0A=
> it's a realistic picture.=0A=
>=0A=
> My impression is that it is not - in particular that:=0A=
>=0A=
> - Lots of interactive traffic goes over the Internet today without=0A=
> layer 2 separation - All UDP endpoints that use the open Internet=0A=
> have some form of backoff on congestion - using delay, packet drop or=0A=
> (most likely) both to detect congestion=0A=
>=0A=
> I'm writing the transport document based on those assumptions - that=0A=
> we should specify something that will work as well as it can over an=0A=
> Internet that does not care, and try to allow for better things to=0A=
> come along later.=0A=
>=0A=
> Am I wrong to write it like this?=0A=
=0A=
My impression (I have no facts, so I can be completely wrong; it would =0A=
be nice if someone could show figures) is that most UDP traffic on the =0A=
internet comes from modern (SW implementations) audiovisual =0A=
communication, from torrent traffic and from other special transport =0A=
built on top of UDP (quic is one example, but there are other used by =0A=
e.g. games). I think all of them back off, so I think you are right.=0A=
=0A=
There is some older stuff that do not back off, but does it really =0A=
generate enough traffic to be relevant?=0A=
=0A=
>=0A=
> Harald=0A=
>=0A=
>=0A=
>=0A=
> ------------------------------------ [Karl] I wrote this sometime=0A=
> before, regarding how QoS over the internet works in general - may be=0A=
> useful for understanding=0A=
>=0A=
> For better understanding of these QoS/QoE problems, and methods for=0A=
> improving (not specifically discussing the policies of sharing=0A=
> real-time traffic space), I would like to explain:=0A=
>=0A=
> Over an IP pipe only used for real-traffic (no TCP data traffic), it=0A=
> is sufficient that the pipe is wide enough for good QoS. That is=0A=
> often used and implemented by separating IP pipes at a lower level=0A=
> using e.g. Ethernet VLAN, MPLS, Ethernet over ATM (std for ADSL=0A=
> modems). TURN servers can enforce real-traffic into such pipes and=0A=
> QoS is achieved. Let=92s call this Level 2 QoS (Network level)=0A=
>=0A=
> Over an IP pipe shared between quality requiring real-time traffic=0A=
> and less demanding data or streaming (usually TCP) traffic, we have=0A=
> the sharing between these two traffic classes to consider. The main=0A=
> method (and the mechanism making today=92s real-time QoE as good as it=0A=
> often is) is that TCP endpoints back-off and share their bandwidth=0A=
> usage. Real-time traffic using UDP transport do not back-off, the=0A=
> endpoints using UDP occupy the bandwidth needed. When an IP pipe gets=0A=
> filled, it is all endpoint=92s TCP bandwidth usage that is back-off and=
=0A=
> shared between them, leaving room for the UDP traffic. This is=0A=
> mechanism we experience everyday over the Internet, using our =93Surf=0A=
> IP pipes=94. Let=92s call this Level 4 QoS (Transport level)=0A=
>=0A=
> However, this Level 4 QoS is based on that at congestion times (which=0A=
> happen every time we click =96 setting up a TCP flow and transferring=0A=
> some amount of data as quick as possible) the router handling the=0A=
> most narrow part of the pipe (the congestion point) drop packets. It=0A=
> is this packet dropping that (i) signals to TCP endpoints to reduce=0A=
> their bandwidth (via TCP=92s error correction/retransmission mechanism)=
=0A=
> and (ii) destroys the QoE of real-time traffic. (Both TCP and UDP=0A=
> packets are dropped in this process that is triggered by flow=0A=
> intensive TCP traffic.)=0A=
>=0A=
> We need (i) but don=92t want (ii) and to improve on this we can e.g.=0A=
> use diffserve, DSCP bits in IP packets to instruct routers to always=0A=
> forward the real-time traffic before any unmarked TCP traffic (which=0A=
> usually fills most of the pipe). Then QoS is then achieved for=0A=
> real-time traffic. This is Level 3 QoS (IP level). The method used is=0A=
> =93traffic shaping=94: backing off data traffic, leaving real-time=0A=
> traffic free passage without packet loss.=0A=
>=0A=
> _______________________________________________ rtcweb mailing list=0A=
> rtcweb@ietf.org https://www.ietf.org/mailman/listinfo/rtcweb=0A=
>=0A=
=0A=


From nobody Thu Apr 24 04:04:47 2014
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Subject: Re: [rtcweb] Pictures of congestion control on the Internet - which is more realistic?
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On 23. apr. 2014, at 18:37, John Leslie wrote:

> Dave Taht <dave.taht@gmail.com> wrote:
>>=20
>> I'm not sure if we have a shared definition of "congestion".  Mine is
>> network delays exceeding that of what human factors research notes
>> as "perceptible" with about 20ms as the outer bound.
>=20
>   I, OTOH, am quite sure we don't have a shared definition of =
congestion.
>=20
>   My preferred definition is rate-of-arrival exceeding =
rate-of-forwarding.

FWIW, there's RFC 6077, which discusses definitions of congestion and =
congestion control in its introduction. It says:

" Congestion can be defined as a state or condition that occurs when
   network resources are overloaded, resulting in impairments for
   network users as objectively measured by the probability of loss
   and/or delay.  The overload results in the reduction of utility in
   networks that support both spatial and temporal multiplexing, but no
   reservation"

which is based on a presentation by Keshav in 2007, which, in turn, is =
based on the long and boring discussion about this definition that =
happened when ICCRG was born. Been there, done that, let's not repeat =
please.

Buuuuuut this seems off topic for the context here anyway... if you =
really want to discuss this definition, I suggest to take it to ICCRG. =
There, we will tell you to read the mailing list archives  :-)

Cheers,
Michael


From nobody Thu Apr 24 04:24:27 2014
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From: "Karl Stahl" <karl.stahl@intertex.se>
To: =?iso-8859-1?Q?'Stefan_H=E5kansson_LK'?= <stefan.lk.hakansson@ericsson.com>,  "'Harald Alvestrand'" <harald@alvestrand.no>, <rtcweb@ietf.org>
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-----Ursprungligt meddelande-----
Fr=E5n: rtcweb [mailto:rtcweb-bounces@ietf.org] F=F6r Stefan H=E5kansson =
LK
Skickat: den 24 april 2014 11:57
Till: Harald Alvestrand; rtcweb@ietf.org
=C4mne: Re: [rtcweb] Pictures of congestion control on the Internet - =
which is
more realistic?

On 2014-04-23 14:31, Harald Alvestrand wrote:
> I quote Karl's piece below, because I want other's opinion of whether=20
> it's a realistic picture.
>
> My impression is that it is not - in particular that:
>
> - Lots of interactive traffic goes over the Internet today without=20
> layer 2 separation - All UDP endpoints that use the open Internet have =

> some form of backoff on congestion - using delay, packet drop or (most =

> likely) both to detect congestion
>
> I'm writing the transport document based on those assumptions - that=20
> we should specify something that will work as well as it can over an=20
> Internet that does not care, and try to allow for better things to=20
> come along later.
>
> Am I wrong to write it like this?

My impression (I have no facts, so I can be completely wrong; it would =
be
nice if someone could show figures) is that most UDP traffic on the =
internet
comes from modern (SW implementations) audiovisual communication, from
torrent traffic and from other special transport built on top of UDP =
(quic
is one example, but there are other used by e.g. games). I think all of =
them
back off, so I think you are right.

There is some older stuff that do not back off, but does it really =
generate
enough traffic to be relevant?

[Karl] If file transfer type of traffic (that wants max bandwidth during
shortest time) run over UDP (like e.g. torrent) their endpoints need to
back-off when seeing congestion - just like TCP has built-in. That is =
also
one of the mechanisms build into SCTP (to "operate under adverse =
operational
conditions" Quoting SCTP RFC4960: 7.  Congestion Control).

To add level 3 QoS to the level 4 QoS we already had for WebRTC =
real-time
traffic (NOT file transfer traffic type), the network need to see the
traffic info in those packets and assure that those are prioritized =
instead
of dropped at congestion in routers when fighting with file transfer =
traffic
about the available bandwidth.=20

If I remember correctly bit-torrent has also a setting of max bandwidth
usage (maybe that is why torrent don't just use TCP's "take all you get" =
),
so if you have a 2 Mbps pipe, you can e.g. have bit-torrent only use 0.5
Mbps (no hurry to get your movie...) and leave 1.5 Mbps at least not =
harmed
by the bit-torrent's bandwidth hunger.

Unfortunately WebRTC real-time traffic cannot tell all other file =
transfer
going on to restrict their bandwidth hunger. Instead we need to ask the
routers to prioritize the WebRTC real-time traffic (which then needs to =
be
marked for network elements to see). And WebRTC should not insert file
transfer type of traffic in its real-time flow, nor mark such traffic =
for
prioritization.


>
> Harald
>
>
>
> ------------------------------------ [Karl] I wrote this sometime=20
> before, regarding how QoS over the internet works in general - may be=20
> useful for understanding
>
> For better understanding of these QoS/QoE problems, and methods for=20
> improving (not specifically discussing the policies of sharing=20
> real-time traffic space), I would like to explain:
>
> Over an IP pipe only used for real-traffic (no TCP data traffic), it=20
> is sufficient that the pipe is wide enough for good QoS. That is often =

> used and implemented by separating IP pipes at a lower level using=20
> e.g. Ethernet VLAN, MPLS, Ethernet over ATM (std for ADSL modems).=20
> TURN servers can enforce real-traffic into such pipes and QoS is=20
> achieved. Let=92s call this Level 2 QoS (Network level)
>
> Over an IP pipe shared between quality requiring real-time traffic and =

> less demanding data or streaming (usually TCP) traffic, we have the=20
> sharing between these two traffic classes to consider. The main method =

> (and the mechanism making today=92s real-time QoE as good as it often=20
> is) is that TCP endpoints back-off and share their bandwidth usage.=20
> Real-time traffic using UDP transport do not back-off, the endpoints=20
> using UDP occupy the bandwidth needed. When an IP pipe gets filled, it =

> is all endpoint=92s TCP bandwidth usage that is back-off and shared=20
> between them, leaving room for the UDP traffic. This is mechanism we=20
> experience everyday over the Internet, using our =93Surf IP pipes=94.=20
> Let=92s call this Level 4 QoS (Transport level)
>
> However, this Level 4 QoS is based on that at congestion times (which=20
> happen every time we click =96 setting up a TCP flow and transferring=20
> some amount of data as quick as possible) the router handling the most =

> narrow part of the pipe (the congestion point) drop packets. It is=20
> this packet dropping that (i) signals to TCP endpoints to reduce their =

> bandwidth (via TCP=92s error correction/retransmission mechanism) and=20
> (ii) destroys the QoE of real-time traffic. (Both TCP and UDP packets=20
> are dropped in this process that is triggered by flow intensive TCP=20
> traffic.)
>
> We need (i) but don=92t want (ii) and to improve on this we can e.g.
> use diffserve, DSCP bits in IP packets to instruct routers to always=20
> forward the real-time traffic before any unmarked TCP traffic (which=20
> usually fills most of the pipe). Then QoS is then achieved for=20
> real-time traffic. This is Level 3 QoS (IP level). The method used is=20
> =93traffic shaping=94: backing off data traffic, leaving real-time =
traffic=20
> free passage without packet loss.
>
> _______________________________________________ rtcweb mailing list=20
> 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 24. apr. 2014, at 13:24, Karl Stahl wrote:

>=20
>=20
> -----Ursprungligt meddelande-----
> Fr=E5n: rtcweb [mailto:rtcweb-bounces@ietf.org] F=F6r Stefan H=E5kansson=
 LK
> Skickat: den 24 april 2014 11:57
> Till: Harald Alvestrand; rtcweb@ietf.org
> =C4mne: Re: [rtcweb] Pictures of congestion control on the Internet - =
which is
> more realistic?
>=20
> On 2014-04-23 14:31, Harald Alvestrand wrote:
>> I quote Karl's piece below, because I want other's opinion of whether=20=

>> it's a realistic picture.
>>=20
>> My impression is that it is not - in particular that:
>>=20
>> - Lots of interactive traffic goes over the Internet today without=20
>> layer 2 separation - All UDP endpoints that use the open Internet =
have=20
>> some form of backoff on congestion - using delay, packet drop or =
(most=20
>> likely) both to detect congestion
>>=20
>> I'm writing the transport document based on those assumptions - that=20=

>> we should specify something that will work as well as it can over an=20=

>> Internet that does not care, and try to allow for better things to=20
>> come along later.
>>=20
>> Am I wrong to write it like this?
>=20
> My impression (I have no facts, so I can be completely wrong; it would =
be
> nice if someone could show figures) is that most UDP traffic on the =
internet
> comes from modern (SW implementations) audiovisual communication, from
> torrent traffic and from other special transport built on top of UDP =
(quic
> is one example, but there are other used by e.g. games). I think all =
of them
> back off, so I think you are right.
>=20
> There is some older stuff that do not back off, but does it really =
generate
> enough traffic to be relevant?
>=20
> [Karl] If file transfer type of traffic (that wants max bandwidth =
during
> shortest time) run over UDP (like e.g. torrent) their endpoints need =
to
> back-off when seeing congestion - just like TCP has built-in. That is =
also
> one of the mechanisms build into SCTP (to "operate under adverse =
operational
> conditions" Quoting SCTP RFC4960: 7.  Congestion Control).
>=20
> To add level 3 QoS to the level 4 QoS we already had for WebRTC =
real-time
> traffic (NOT file transfer traffic type), the network need to see the
> traffic info in those packets and assure that those are prioritized =
instead
> of dropped at congestion in routers when fighting with file transfer =
traffic
> about the available bandwidth.=20
>=20
> If I remember correctly bit-torrent has also a setting of max =
bandwidth
> usage (maybe that is why torrent don't just use TCP's "take all you =
get" ),
> so if you have a 2 Mbps pipe, you can e.g. have bit-torrent only use =
0.5
> Mbps (no hurry to get your movie...) and leave 1.5 Mbps at least not =
harmed
> by the bit-torrent's bandwidth hunger.

Just as a data point, at least the official bit-torrent client uses a =
variant of LEDBAT ( RFC 6817 ) called uTP AFAIK, which is a mechanism =
that can better saturate a link than TCP in the absence of TCP, but is =
designed to back off fast to avoid getting in the way (by detecting =
growing queuing delay) when TCP is present.

Cheers,
Michael


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Subject: [rtcweb] Working Group Last Call for draft-ietf-rtcweb-rtp-usage-13
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This begins a working group last call on draft-ietf-rtcweb-rtp-usage-13;
please review the document and provide comments to the list by May 9, 2014.

thanks,

Ted, Cullen, Sean

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<div dir="ltr"><div class="gmail_default" style="font-family:georgia,serif">This begins a working group last call on draft-ietf-rtcweb-rtp-usage-13; please review the document and provide comments to the list by May 9, 2014.<br>
<br></div><div class="gmail_default" style="font-family:georgia,serif">thanks,<br><br></div><div class="gmail_default" style="font-family:georgia,serif">Ted, Cullen, Sean<br></div></div>

--001a11c30cc8350f8e04f7cb4ad9--


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Howdy,

Below is a rough sketch of the agenda for the upcoming RTCWEB Inteirm
(essentially, we have the morning sessions for the three days May 19, May
20, May 21 2014).  If you have not filled in your participation in the
doodle poll, please do so at:  http://doodle.com/qewq4xvszbc6d4sn.

Comments on the agenda timing, on other work, or mechanics welcome,

thanks,

Ted, Sean, Cullen

Interim Meeting RTCWEB May 2014



Day 1

RTP + Media  (2h)

draft-ietf-rtcweb-rtp-usage-13<http://datatracker.ietf.org/doc/draft-ietf-rtcweb-rtp-usage/>

draft-ietf-rtcweb-audio-05<http://datatracker.ietf.org/doc/draft-ietf-rtcweb-audio/>

draft-ietf-rtcweb-transports-03<http://datatracker.ietf.org/doc/draft-ietf-rtcweb-transports/>

JSEP (2h)

draft-ietf-rtcweb-jsep-06<http://datatracker.ietf.org/doc/draft-ietf-rtcweb-jsep/>

Day 2

Data Channel (4h)

draft-ietf-rtcweb-data-channel-08<http://datatracker.ietf.org/doc/draft-ietf-rtcweb-data-channel/>

draft-ietf-rtcweb-data-protocol-04<http://datatracker.ietf.org/doc/draft-ietf-rtcweb-data-protocol/>


Day 3

Security ( 2h )

draft-ietf-rtcweb-security-06<http://datatracker.ietf.org/doc/draft-ietf-rtcweb-security/>

draft-ietf-rtcweb-security-arch-09<http://datatracker.ietf.org/doc/draft-ietf-rtcweb-security-arch/>

ALPN ( 15m )

draft-thomson-rtcweb-alpn-00<http://datatracker.ietf.org/doc/draft-thomson-rtcweb-alpn/>

Consent Freshness ( 30m)

draft-ietf-rtcweb-stun-consent-freshness-02<http://datatracker.ietf.org/doc/draft-ietf-rtcweb-stun-consent-freshness/>

Matters arising from webrtc meetings ( remaining time)

--20cf30223a23a1d0ea04f7cbaa1d
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<div dir=3D"ltr"><div class=3D"gmail_default" style=3D"font-family:georgia,=
serif">Howdy,<br><br></div><div class=3D"gmail_default" style=3D"font-famil=
y:georgia,serif">Below is a rough sketch of the agenda for the upcoming RTC=
WEB Inteirm (essentially, we have the morning sessions for the three days M=
ay 19, May 20, May 21 2014).=C2=A0 If you have not filled in your participa=
tion in the doodle poll, please do so at:=C2=A0 <a href=3D"http://doodle.co=
m/qewq4xvszbc6d4sn">http://doodle.com/qewq4xvszbc6d4sn</a>.<br>
<br></div><div class=3D"gmail_default" style=3D"font-family:georgia,serif">=
Comments on the agenda timing, on other work, or mechanics welcome,<br><br>=
thanks,<br><br>Ted, Sean, Cullen<br><br><p dir=3D"ltr" style=3D"line-height=
:1.15;margin-top:0pt;margin-bottom:0pt" id=3D"docs-internal-guid-8d0976b3-9=
464-7b1e-5602-9fab276d8e39">
<span style=3D"font-size:15px;font-family:Arial;color:rgb(0,0,0);background=
-color:transparent;font-weight:normal;font-style:normal;font-variant:normal=
;text-decoration:none;vertical-align:baseline">Interim Meeting RTCWEB May 2=
014</span></p>
<p dir=3D"ltr" style=3D"line-height:1.15;margin-top:0pt;margin-bottom:0pt">=
<span style=3D"font-size:15px;font-family:Arial;color:rgb(17,85,204);backgr=
ound-color:transparent;font-weight:normal;font-style:normal;font-variant:no=
rmal;text-decoration:underline;vertical-align:baseline"><br>
</span></p><br><p dir=3D"ltr" style=3D"line-height:1.15;margin-top:0pt;marg=
in-bottom:0pt"><span style=3D"font-size:15px;font-family:Arial;color:rgb(0,=
0,0);background-color:transparent;font-weight:normal;font-style:normal;font=
-variant:normal;text-decoration:none;vertical-align:baseline">Day 1</span><=
/p>
<br><p dir=3D"ltr" style=3D"line-height:1.15;margin-top:0pt;margin-bottom:0=
pt"><span style=3D"font-size:15px;font-family:Arial;color:rgb(0,0,0);backgr=
ound-color:transparent;font-weight:normal;font-style:normal;font-variant:no=
rmal;text-decoration:none;vertical-align:baseline">RTP + Media =C2=A0(2h)</=
span></p>
<p dir=3D"ltr" style=3D"line-height:1.15;margin-top:0pt;margin-bottom:0pt">=
<a href=3D"http://datatracker.ietf.org/doc/draft-ietf-rtcweb-rtp-usage/" st=
yle=3D"text-decoration:none"><span style=3D"font-size:13px;font-family:Aria=
l;color:rgb(17,85,204);background-color:rgb(255,255,255);font-weight:normal=
;font-style:normal;font-variant:normal;text-decoration:underline;vertical-a=
lign:baseline">draft-ietf-rtcweb-rtp-usage-13</span></a></p>
<p dir=3D"ltr" style=3D"line-height:1.15;margin-top:0pt;margin-bottom:0pt">=
<a href=3D"http://datatracker.ietf.org/doc/draft-ietf-rtcweb-audio/" style=
=3D"text-decoration:none"><span style=3D"font-size:13px;font-family:Arial;c=
olor:rgb(17,85,204);background-color:rgb(237,245,255);font-weight:normal;fo=
nt-style:normal;font-variant:normal;text-decoration:underline;vertical-alig=
n:baseline">draft-ietf-rtcweb-audio-05</span></a></p>
<p dir=3D"ltr" style=3D"line-height:1.15;margin-top:0pt;margin-bottom:0pt">=
<a href=3D"http://datatracker.ietf.org/doc/draft-ietf-rtcweb-transports/" s=
tyle=3D"text-decoration:none"><span style=3D"font-size:13px;font-family:Ari=
al;color:rgb(17,85,204);background-color:rgb(255,255,255);font-weight:norma=
l;font-style:normal;font-variant:normal;text-decoration:underline;vertical-=
align:baseline">draft-ietf-rtcweb-transports-03</span></a></p>
<br><p dir=3D"ltr" style=3D"line-height:1.15;margin-top:0pt;margin-bottom:0=
pt"><span style=3D"font-size:15px;font-family:Arial;color:rgb(0,0,0);backgr=
ound-color:transparent;font-weight:normal;font-style:normal;font-variant:no=
rmal;text-decoration:none;vertical-align:baseline">JSEP (2h)</span></p>
<p dir=3D"ltr" style=3D"line-height:1.15;margin-top:0pt;margin-bottom:0pt">=
<a href=3D"http://datatracker.ietf.org/doc/draft-ietf-rtcweb-jsep/" style=
=3D"text-decoration:none"><span style=3D"font-size:13px;font-family:Arial;c=
olor:rgb(17,85,204);background-color:rgb(255,255,255);font-weight:normal;fo=
nt-style:normal;font-variant:normal;text-decoration:underline;vertical-alig=
n:baseline">draft-ietf-rtcweb-jsep-06</span></a></p>
<br><p dir=3D"ltr" style=3D"line-height:1.15;margin-top:0pt;margin-bottom:0=
pt"><span style=3D"font-size:15px;font-family:Arial;color:rgb(0,0,0);backgr=
ound-color:transparent;font-weight:normal;font-style:normal;font-variant:no=
rmal;text-decoration:none;vertical-align:baseline">Day 2</span></p>
<br><p dir=3D"ltr" style=3D"line-height:1.15;margin-top:0pt;margin-bottom:0=
pt"><span style=3D"font-size:15px;font-family:Arial;color:rgb(0,0,0);backgr=
ound-color:transparent;font-weight:normal;font-style:normal;font-variant:no=
rmal;text-decoration:none;vertical-align:baseline">Data Channel (4h)</span>=
</p>
<p dir=3D"ltr" style=3D"line-height:1.15;margin-top:0pt;margin-bottom:0pt">=
<a href=3D"http://datatracker.ietf.org/doc/draft-ietf-rtcweb-data-channel/"=
 style=3D"text-decoration:none"><span style=3D"font-size:13px;font-family:A=
rial;color:rgb(17,85,204);background-color:rgb(255,255,255);font-weight:nor=
mal;font-style:normal;font-variant:normal;text-decoration:underline;vertica=
l-align:baseline">draft-ietf-rtcweb-data-channel-08</span></a></p>
<p dir=3D"ltr" style=3D"line-height:1.15;margin-top:0pt;margin-bottom:0pt">=
<a href=3D"http://datatracker.ietf.org/doc/draft-ietf-rtcweb-data-protocol/=
" style=3D"text-decoration:none"><span style=3D"font-size:13px;font-family:=
Arial;color:rgb(17,85,204);background-color:rgb(237,245,255);font-weight:no=
rmal;font-style:normal;font-variant:normal;text-decoration:underline;vertic=
al-align:baseline">draft-ietf-rtcweb-data-protocol-04</span></a></p>
<br><br><p dir=3D"ltr" style=3D"line-height:1.15;margin-top:0pt;margin-bott=
om:0pt"><span style=3D"font-size:15px;font-family:Arial;color:rgb(0,0,0);ba=
ckground-color:transparent;font-weight:normal;font-style:normal;font-varian=
t:normal;text-decoration:none;vertical-align:baseline">Day 3</span></p>
<br><p dir=3D"ltr" style=3D"line-height:1.15;margin-top:0pt;margin-bottom:0=
pt"><span style=3D"font-size:15px;font-family:Arial;color:rgb(0,0,0);backgr=
ound-color:transparent;font-weight:normal;font-style:normal;font-variant:no=
rmal;text-decoration:none;vertical-align:baseline">Security ( 2h )</span></=
p>
<p dir=3D"ltr" style=3D"line-height:1.15;margin-top:0pt;margin-bottom:0pt">=
<a href=3D"http://datatracker.ietf.org/doc/draft-ietf-rtcweb-security/" sty=
le=3D"text-decoration:none"><span style=3D"font-size:13px;font-family:Arial=
;color:rgb(17,85,204);background-color:rgb(237,245,255);font-weight:normal;=
font-style:normal;font-variant:normal;text-decoration:underline;vertical-al=
ign:baseline">draft-ietf-rtcweb-security-06</span></a></p>
<p dir=3D"ltr" style=3D"line-height:1.15;margin-top:0pt;margin-bottom:0pt">=
<a href=3D"http://datatracker.ietf.org/doc/draft-ietf-rtcweb-security-arch/=
" style=3D"text-decoration:none"><span style=3D"font-size:13px;font-family:=
Arial;color:rgb(17,85,204);background-color:rgb(255,255,255);font-weight:no=
rmal;font-style:normal;font-variant:normal;text-decoration:underline;vertic=
al-align:baseline">draft-ietf-rtcweb-security-arch-09</span></a></p>
<br><p dir=3D"ltr" style=3D"line-height:1.15;margin-top:0pt;margin-bottom:0=
pt"><span style=3D"font-size:15px;font-family:Arial;color:rgb(0,0,0);backgr=
ound-color:transparent;font-weight:normal;font-style:normal;font-variant:no=
rmal;text-decoration:none;vertical-align:baseline">ALPN ( 15m )</span></p>
<p dir=3D"ltr" style=3D"line-height:1.15;margin-top:0pt;margin-bottom:0pt">=
<a href=3D"http://datatracker.ietf.org/doc/draft-thomson-rtcweb-alpn/" styl=
e=3D"text-decoration:none"><span style=3D"font-size:13px;font-family:Arial;=
color:rgb(17,85,204);background-color:rgb(255,255,255);font-weight:normal;f=
ont-style:normal;font-variant:normal;text-decoration:underline;vertical-ali=
gn:baseline">draft-thomson-rtcweb-alpn-00</span></a></p>
<br><p dir=3D"ltr" style=3D"line-height:1.15;margin-top:0pt;margin-bottom:1=
2pt"><span style=3D"font-size:15px;font-family:Arial;color:rgb(0,0,0);backg=
round-color:transparent;font-weight:normal;font-style:normal;font-variant:n=
ormal;text-decoration:none;vertical-align:baseline">Consent Freshness ( 30m=
)</span></p>
<p dir=3D"ltr" style=3D"line-height:1.15;margin-top:0pt;margin-bottom:12pt"=
><a href=3D"http://datatracker.ietf.org/doc/draft-ietf-rtcweb-stun-consent-=
freshness/" style=3D"text-decoration:none"><span style=3D"font-size:13px;fo=
nt-family:Arial;color:rgb(17,85,204);background-color:rgb(237,245,255);font=
-weight:normal;font-style:normal;font-variant:normal;text-decoration:underl=
ine;vertical-align:baseline">draft-ietf-rtcweb-stun-consent-freshness-02</s=
pan></a></p>
<br><p dir=3D"ltr" style=3D"line-height:1.15;margin-top:0pt;margin-bottom:1=
2pt"><span style=3D"font-size:15px;font-family:Arial;color:rgb(0,0,0);backg=
round-color:transparent;font-weight:normal;font-style:normal;font-variant:n=
ormal;text-decoration:none;vertical-align:baseline">Matters arising from we=
brtc meetings ( remaining time) </span></p>
<br><br></div></div>

--20cf30223a23a1d0ea04f7cbaa1d--


From nobody Thu Apr 24 12:08:53 2014
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From: Dave Taht <dave.taht@gmail.com>
To: John Leslie <john@jlc.net>
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Subject: Re: [rtcweb] Pictures of congestion control on the Internet - which is more realistic?
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On Wed, Apr 23, 2014 at 9:37 AM, John Leslie <john@jlc.net> wrote:
> Dave Taht <dave.taht@gmail.com> wrote:
>>
>> I'm not sure if we have a shared definition of "congestion".  Mine is
>> network delays exceeding that of what human factors research notes
>> as "perceptible" with about 20ms as the outer bound.

>    I, OTOH, am quite sure we don't have a shared definition of congestion=
.
>
>    My preferred definition is rate-of-arrival exceeding rate-of-forwardin=
g.

That's not bad. But I would call that instantaneous congestion:

             exit
            --------
              1
              2
              3 4
              5
              6 7
              8
           --------
             arrival at a slightly higher rate

when two packets arrive so that they must be queued or dropped.
Without clearing the
delay caused by packet 4, we induce a 1 packet delay, and without clearing =
the
delay caused by packet 7, we are now at two packets of delay. And so on.

When you got to defining congestion, the "rate" term you used is actually a
measurement over something else, usually an interval. Where I mentioned
X*1.1*RTT was trying to get to where there was a definition of interval tha=
t
made sense.


>    But, of course, you don't have to agree with mine -- "shared" merely
> implies _some_ overlap of the sets of definitions we use...

I should probably have said "induced network delays", although I have to ad=
mit
that ANY delay - whether it be physical RTT, switching overhead, or
encoding overhead,
or congestion, bothers me. I've spent a lot of time working on RT
systems (see ardour.org)
where 2.7ms delay was too much delay, and others with much tighter constrai=
nts.

So me saying it above was not a definition of congestion but my
subconscious making
a twitchy reflexive swing at delay inducing techniques in general.

>    Dave's, I fear, isn't terribly useful. There is a base latency (in the
> absence of any traffic) which exceeds 20 milliseconds for many paths
> of interest.

Well, in the presence of some traffic, it's quite possible to get zero dela=
y,
for special traffic that can be slipped in when the grant arrives.

Cablelabs did this with their SFQ sim, and it's possible
with any request/grant architecture.

http://www.igvita.com/2014/04/21/uplink-scheduling-in-4G-networks/

As for baseline physical RTTs, well, in major cities, data centers are with=
in
2ms of nearly everyone.

I'd like to see a multi-industry-wide effort to remove excessive latency
from enodeBs, and wifi APs, and from future layer 2 standards.

Stuart Cheshire and I did a preso at ietf last time showing a 170ms latency
on data delivery on a 10Mbit/1ms path, first chopping it down to <10ms
with codel, then
to 0 with codel + ecn.

http://www.ietf.org/proceedings/89/slides/slides-89-tsvarea-1.pdf

It would be nice if every last mile technology could be as good as ethernet=
,
circa 1980.

>
>    To tell truth, though, most users probably only notice the total delay=
,

and jitter.

> not the components of it; so I think Dave's metric is useful -- it's just
> not helpful as a measure of "congestion".

Well, I suggested several possible metrics.

>
>    I'd like to follow up a bit on Dave's metric, regardless.
>
>    There's a lot of talk about "buffer-bloat" where I believe we all
> agree the situation would be better with less buffering.

>    But can we talk about the other end of that spectrum?
>
>    Surely there is some minimal amount of buffering, for any particular
> traffic, which is needed in order to maintain a reasonable flow rate.
> Most likely, that minimal buffering varies with factors such as RTT. Do
> we even have a shared concept of what those factors are?
>
>    I'd like to see a thread on what is needed as a minimum buffering
> for various kinds of traffic. I suspect that minimum can be fairly small
> except for TCP transport of "large" files...
>> http://gettys.wordpress.com/2013/07/10/low-latency-requires-smart-queuin=
g-traditional-aqm-is-not-enough/
>
>    Excellent as food for thought...
>
> --
> John Leslie <john@jlc.net>



--=20
Dave T=C3=A4ht

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


From nobody Fri Apr 25 01:47:29 2014
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A New Internet-Draft is available from the on-line Internet-Drafts directories.
 This draft is a work item of the Real-Time Communication in WEB-browsers Working Group of the IETF.

        Title           : Transports for RTCWEB
        Author          : Harald Alvestrand
	Filename        : draft-ietf-rtcweb-transports-04.txt
	Pages           : 13
	Date            : 2014-04-25

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


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

There's also a htmlized version available at:
http://tools.ietf.org/html/draft-ietf-rtcweb-transports-04

A diff from the previous version is available at:
http://www.ietf.org/rfcdiff?url2=draft-ietf-rtcweb-transports-04


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

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


From nobody Fri Apr 25 01:50:58 2014
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For easy access, here's the changelog section:


A.4.  Changes from -03 to -04

    o  Added a section on prioritization, moved the DSCP section into it,
       and added a section on local prioritization, giving a specific
       algorithm for interpreting "priority" in local prioritization.

    o  ICE-TCP candidates was changed from MAY to MUST, in recognition of
       the sense of the room at the London IETF.

There are also some attempts at language improvements, of course, but 
those should be the main changes.


On 04/25/2014 10:47 AM, internet-drafts@ietf.org wrote:
> A New Internet-Draft is available from the on-line Internet-Drafts directories.
>   This draft is a work item of the Real-Time Communication in WEB-browsers Working Group of the IETF.
>
>          Title           : Transports for RTCWEB
>          Author          : Harald Alvestrand
> 	Filename        : draft-ietf-rtcweb-transports-04.txt
> 	Pages           : 13
> 	Date            : 2014-04-25
>
> Abstract:
>     This document describes the data transport protocols used by RTCWEB,
>     including the protocols used for interaction with intermediate boxes
>     such as firewalls, relays and NAT boxes.
>
>
> The IETF datatracker status page for this draft is:
> https://datatracker.ietf.org/doc/draft-ietf-rtcweb-transports/
>
> There's also a htmlized version available at:
> http://tools.ietf.org/html/draft-ietf-rtcweb-transports-04
>
> A diff from the previous version is available at:
> http://www.ietf.org/rfcdiff?url2=draft-ietf-rtcweb-transports-04
>
>
> Please note that it may take a couple of minutes from the time of submission
> until the htmlized version and diff are available at tools.ietf.org.
>
> Internet-Drafts are also available by anonymous FTP at:
> ftp://ftp.ietf.org/internet-drafts/
>
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb


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Archived-At: http://mailarchive.ietf.org/arch/msg/rtcweb/awybah3uG7LhtiJgp89ltVsmBzE
Subject: [rtcweb] Query Regarding Mandatory audio codecs draft
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-----BEGIN PGP SIGNED MESSAGE-----
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On 25/04/14 06:21 AM, Kiran Kumar Guduru wrote:
> Is that means spec is recommending G.711 (PCMA/PCMU) for sampling
> rate less than or equal to 8 kHz?

No. Opus is still a good idea for 8 kHz, but it's not 2119 RECOMMENDED
since there are valid (mostly complexity-related) reasons to use G.711
despite the higher bit-rate.

	Jean-Marc
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Subject: Re: [rtcweb] Query Regarding Mandatory audio codecs draft
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On 25/04/14 06:38 AM, Kiran Kumar Guduru wrote:
> Spec prefers G.711 at 8 kHz sampling rate and RECOMMENDS Opus above
> 8 kHz sampling rate.
> 
> Is my understanding correct?

No. The spec does not prefer anything and does not make any
recommendation at 8 kHz. Just like the statement "if it rains, bring
an umbrella" does not say anything about the case where it doesn't rain.

	Jean-Marc
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From nobody Fri Apr 25 04:54:33 2014
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Date: Fri, 25 Apr 2014 13:54:23 +0200
From: Sergio Garcia Murillo <sergio.garcia.murillo@gmail.com>
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Subject: Re: [rtcweb] Pictures of congestion control on the Internet - which is more realistic?
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El 23/04/2014 17:46, Stephan Wenger escribió:
> In both cases, the typical reaction of an endpoint is to expose the 
> user to crappy quality, until the user hangs up.  Sometimes, slightly 
> smarter implementations hang up themselves if the packet loss rate 
> becomes excessive.I would suggest not to paint a rosier picture in the 
> draft than what is warranted based on real-world observations.
>
>     Based on this, most UDP endpoints do not deal well with
>     congestion. Ideally this should change to support better bandwidth
>     management. Futhermore, reliably transmitting real time media over
>     public internet would also require changes to biffer management
>     algorithms in routers, which will, in turn, affect congestion
>     handling strategies in the end points.
>

+1, but also from my real life experience, the most problems I am having 
with quality are related to bad Wifi coverage or cheap/crappy XDSL 
routers/cable modems. Almost 100% of problems with quality I had found 
were solved just by plugging the ethernet cable.

Best regards
Sergio

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    <div class="moz-cite-prefix">El 23/04/2014 17:46, Stephan Wenger
      escribi&oacute;:<br>
    </div>
    <blockquote cite="mid:CF7D2A6D.464A8%25stewe@stewe.org" type="cite">
      <div>In both cases, the typical reaction of an endpoint is to
        expose the user to crappy quality, until the user hangs up.
        &nbsp;Sometimes, slightly smarter implementations hang up themselves
        if the packet loss rate becomes excessive.I would suggest not to
        paint a rosier picture in the draft than what is warranted based
        on real-world observations. &nbsp;</div>
      <span id="OLK_SRC_BODY_SECTION">
        <div>
          <div>
            <blockquote style="margin:0 0 0 40px; border:none;
              padding:0px;">
              <p dir="ltr">Based on this, most UDP endpoints do not deal
                well with congestion. Ideally this should change to
                support better bandwidth management. Futhermore,
                reliably transmitting real time media over public
                internet would also require changes to biffer management
                algorithms in routers, which will, in turn, affect
                congestion handling strategies in the end points.</p>
            </blockquote>
          </div>
        </div>
      </span></blockquote>
    <br>
    +1, but also from my real life experience, the most problems I am
    having with quality are related to bad Wifi coverage or cheap/crappy
    XDSL routers/cable modems. Almost 100% of problems with quality I
    had found were solved just by plugging the ethernet cable.&nbsp; <br>
    <br>
    Best regards<br>
    Sergio<br>
  </body>
</html>

--------------070304070203000802070708--


From nobody Fri Apr 25 05:04:39 2014
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To: Ted Hardie <ted.ietf@gmail.com>, "rtcweb@ietf.org" <rtcweb@ietf.org>, "Sean Turner" <turners@ieca.com>, Cullen Jennings <fluffy@cisco.com>
Thread-Topic: [rtcweb] Sketch of agenda for upcoming RTCWEB Interim
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--_000_7594FB04B1934943A5C02806D1A2204B1D2DC366ESESSMB209erics_--


From nobody Fri Apr 25 05:53:44 2014
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Le 2014-04-25 04:50, Harald Alvestrand a écrit :
> A.4.  Changes from -03 to -04
> 
>    o  Added a section on prioritization, moved the DSCP section into it,
>       and added a section on local prioritization, giving a specific
>       algorithm for interpreting "priority" in local prioritization.

Harald,

The new text looks good. A comment about this part:

> 			When an RTCWEB implementation has packets to send on multiple streams	
> 			that are congestion-controlled under the same congestion controller,	
> 			the RTCWEB implementation SHOULD serve the streams in a weighted	
> 			round-robin fashion, with each stream at each level of priority being	
> 			given approximately twice the transmission capacity (measured in	
> 			payload bytes) of the level below.

This looks like a QoS algorithm that might get quickly obsoleted by
better ones. We don't want to prevent innovation in this matter by
prescribing an obsolete QoS algorithm.

Suggestion: leave it up to implementations to interpret priority levels
however they want. Reword the current text so that it becomes an
*example* of what an implementation might do.

Simon
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On 04/25/2014 02:53 PM, Simon Perreault wrote:
> Le 2014-04-25 04:50, Harald Alvestrand a écrit :
>> A.4.  Changes from -03 to -04
>>
>>     o  Added a section on prioritization, moved the DSCP section into it,
>>        and added a section on local prioritization, giving a specific
>>        algorithm for interpreting "priority" in local prioritization.
> Harald,
>
> The new text looks good. A comment about this part:
>
>> 			When an RTCWEB implementation has packets to send on multiple streams	
>> 			that are congestion-controlled under the same congestion controller,	
>> 			the RTCWEB implementation SHOULD serve the streams in a weighted	
>> 			round-robin fashion, with each stream at each level of priority being	
>> 			given approximately twice the transmission capacity (measured in	
>> 			payload bytes) of the level below.
> This looks like a QoS algorithm that might get quickly obsoleted by
> better ones. We don't want to prevent innovation in this matter by
> prescribing an obsolete QoS algorithm.
>
> Suggestion: leave it up to implementations to interpret priority levels
> however they want. Reword the current text so that it becomes an
> *example* of what an implementation might do.

The problem with doing that is that it leads to completely inconsistent 
behaviour.

The reason I wrote it this way is because the bandwidth estimation 
problem breaks down into two parts:

- Telling if it's a good idea to send a packet (congestion control)
- Deciding which packet to send (prioritization)

Congestion control is an active area of research and innovation. I said 
very little about it.

Prioritization is a different matter.

What I was trying to encapsulate was:

- Higher priority means more packets get sent
- "More" has a numeric value, so behaviour is predictable (that's where 
the x2 figure came from)
- Small packet flows get more packets than large-packet flows (counting 
bytes not packets)
- Lower priority flows can't be starved completely

This text captures that. I'd like to continue to capture that.


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Le 2014-04-25 09:21, Harald Alvestrand a écrit :
>> Suggestion: leave it up to implementations to interpret priority levels
>> however they want. Reword the current text so that it becomes an
>> *example* of what an implementation might do.
> 
> The problem with doing that is that it leads to completely inconsistent
> behaviour.

Understood. But is that something we should care about? I mean, if it
does not lead to application developers doing "if (chrome) ... else if
(firefox) ..." then behaviour inconsistency is of no consequence. I
can't imagine a situation where that would happen if we leave it open.

Simon
-- 
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Harald and WG,

I am catching up on the changes and improvements in the two latest
versions. I have tried to do a complete review through and match with my
previous comments.

First of all, I think the two revisions have improved a number of things.

1. Section 3.1:
   Platforms that do not give access to these interfaces will not be
   able to support a conforming RTCWEB implementation.

I am very split over this statement. We do want platforms to support per
packet DSCP markings, while at the same time we know we might not be
able to set DSCP at all for a transport flow (5-tuple). Thus, I wonder
if calling the implementations non-conformant due to the underlying
platform is a bit though.

2. Section 3.1:
   This specification does not assume that the implementation will have
   access to ICMP or raw IP.

But, to my understanding we have things that could benefit from getting
the information from ICMP, e.g. the SCTP MTU discovery mechanism. Thus,
I wonder if this statement should be changed to:

   This specification does not assume that the implementation will have
   access to ICMP or raw IP, however access to ICMP information can
   benefit the WebRTC implementation and its performance.

3. Section 3.4:
   TURN TCP candidates [RFC6062] MAY be supported.

   However, such candidates are not seen as providing any significant
   benefit.  First, use of TURN TCP would only be relevant in cases
   which both peers are required to use TCP to establish a
   PeerConnection.  Secondly, that use case is anyway supported by both
   sides establishing UDP relay candidates using TURN over TCP to
   connect to the relay server.  Thirdly, using TCP only between the
   endpoint and its relay may result in less issues with TCP in regards
   to real-time constraints, e.g. due to head of line blocking.

I think these two paragraphs should be moved to after the following
paragraph:

   ICE-TCP candidates [RFC6544] MUST be supported; this may allow
   applications to communicate to peers with public IP addresses across
   UDP-blocking firewalls without using a TURN server.

>From a discussion of WebRTC RTP or DTLS packets over TCP, it makes
better sense to first discuss the ICE TCP candidates, then discuss why
TCP candidates from TURN makes less so.

4. Section 3.4:
   If TCP connections are used, RTP framing according to [RFC4571] MUST
   be used, both for the RTP packets and for the DTLS packets used to
   carry data channels.

I think this paragraph forgets the encapsulation of the STUN messages
that ICE emits for initial path verification and nomination, as well as
for keep-alive and consent freshness.

5. section 3.4:

   NOTE IN DRAFT: This may be added.

I don't understand this note.

6. Section 3.5:

   The setup protocol for RTCWEB data channels is described in
   [I-D.jesup-rtcweb-data-protocol].

First of all, this pointer is the individual version rather than WG draft.

Secondly, I think the DCEP shall be mandated to be supported by an
WebRTC implementation in this specification, and thus become a normative
reference also.

7. Section 4:

   The RTCWEB prioritization model is that the application tells the
   RTCWEB implementation about the priority of media and data flows
   through an API.

As we have discussed before in regards to DART as well as
draft-ietf-tsvwg-rtcweb-qos has needs for clear definitions what is
meant with media and data flows here. But, reading this section it is
clear that the document itself has need for clear definitions.

A data flow appear to be the one of the SCTP streams that forms a single
data channel, as this is the level where we assign data priority.

A WebRTC media flows is a more complex beast. It is one or more RTP
packet streams (draft-ietf-avtext-grouping-taxonomy), i.e. the RTP
packets associated with an SSRC. However, due to FEC and RTP
retransmission as well as scalable codecs, there could be multiple SSRCs
that carries data associated and possibly necessary to be able to
perform media decoding. Thus, I think we are talking about the RTP
packet streams that are the result of a single MediaStreamTrack within
one RTCPeerConnection.

I would also note that Section 4.1 uses Media Stream instead of media
flow. There are need for alignment here.

8. Section 4.1:

I think this section is a good start. It clearly needs some more input.
Especially about the notes made.

In addition to the first three configurations, I think the one where one
has different transport flows per priority level for the media flows
appears to be one most likely to require mandating implementation
support. Yes this means that I also would like to mandate implementation
support for the one transport flow per media type configuration.

9. Section 4.1:

A receiving implementation MUST be able to receive media and data in
   all these configurations.

It is very unclear what "all" in the above sentence includes.

10. Section 4.2:

First of all I do wonder if this is the right solution. I haven't yet
spent enough time on the issue to know if the proposal is useful and
good enough, or if we need something else. I do encourage people to
consider this. However, considering Simon Perrault's comment, I do want
to state that I share Harald's view that this prioritization behavior
needs to be consistent over the implementations.

11. Section 4.2:
   This prioritization is independent of the
   media type, but will lead to packet loss due to full send buffers
   occuring first on the highest volume flows at any given priority
   level.

I think I have two questions here. First, I appears that you assume that
you have an API model where the framework will try to send and fail
rather than creating the data based on allowance or with the knowledge
that you can send a particular data amount. I think creating sender side
buffer overflow is a poor regulation model.

Secondly, I would not that this issue has a higher probability of
effecting the flow with most data first.

Cheers

Magnus Westerlund

----------------------------------------------------------------------
Services, Media and Network features, Ericsson Research EAB/TXM
----------------------------------------------------------------------
Ericsson AB                 | Phone  +46 10 7148287
Färögatan 6                 | Mobile +46 73 0949079
SE-164 80 Stockholm, Sweden | mailto: magnus.westerlund@ericsson.com
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On 04/25/2014 03:53 PM, Simon Perreault wrote:
> Le 2014-04-25 09:21, Harald Alvestrand a écrit :
>>> Suggestion: leave it up to implementations to interpret priority levels
>>> however they want. Reword the current text so that it becomes an
>>> *example* of what an implementation might do.
>> The problem with doing that is that it leads to completely inconsistent
>> behaviour.
> Understood. But is that something we should care about? I mean, if it
> does not lead to application developers doing "if (chrome) ... else if
> (firefox) ..." then behaviour inconsistency is of no consequence. I
> can't imagine a situation where that would happen if we leave it open.

I can imagine that happening. Consider a scenario where complete 
starvation is considered a Really Bad Thing.

if (firefox) {
    // higher priority channels will completely starve out lower 
priority ones. Don't.
   priority = normal
} else if (chrome) {
    // lower priority channels will get some packets through. That's OK.
    priority = high
} else {
    // I have no idea what will happen, let's play it safe.
    priority = normal
}


>
> Simon


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Le 2014-04-25 11:02, Harald Alvestrand a écrit :
> On 04/25/2014 03:53 PM, Simon Perreault wrote:
>> Le 2014-04-25 09:21, Harald Alvestrand a écrit :
>>>> Suggestion: leave it up to implementations to interpret priority levels
>>>> however they want. Reword the current text so that it becomes an
>>>> *example* of what an implementation might do.
>>> The problem with doing that is that it leads to completely inconsistent
>>> behaviour.
>> Understood. But is that something we should care about? I mean, if it
>> does not lead to application developers doing "if (chrome) ... else if
>> (firefox) ..." then behaviour inconsistency is of no consequence. I
>> can't imagine a situation where that would happen if we leave it open.
> 
> I can imagine that happening. Consider a scenario where complete
> starvation is considered a Really Bad Thing.
> 
> if (firefox) {
>    // higher priority channels will completely starve out lower priority
> ones. Don't.
>   priority = normal
> } else if (chrome) {
>    // lower priority channels will get some packets through. That's OK.
>    priority = high
> } else {
>    // I have no idea what will happen, let's play it safe.
>    priority = normal
> }

Ok, then how about "MUST NOT do starvation"?

I'll shut up now. I see your point. If nobody else feels this issue is
important, feel free to ignore my comment.

Simon
-- 
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On 25 April 2014 08:17, Simon Perreault <simon.perreault@viagenie.ca> wrote:
> I'll shut up now. I see your point. If nobody else feels this issue is
> important, feel free to ignore my comment.

Honestly, I was originally inclined to agree with you, but the SHOULD
leaves plenty of wiggle room for those who think they know better.

I think that providing motivation in the document - something like
"This ensures that lower priority streams receive fewer resources, but
cannot be completely blocked by those at higher priority." - would
help to ensure that people who think they know better - and don't -
can't screw things up too much.


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If a "lower priority" packet is dispatched before a "higher priority" packe=
t in order to "prevent starvation", then what does "higher priority" mean?

Streams that are deliberately set to take resources from other streams shou=
ld behave that way. Otherwise the knob shouldn't be labeled the way it is.

Matthew Kaufman

> -----Original Message-----
> From: rtcweb [mailto:rtcweb-bounces@ietf.org] On Behalf Of Martin
> Thomson
> Sent: Friday, April 25, 2014 10:01 AM
> To: Simon Perreault
> Cc: rtcweb@ietf.org
> Subject: Re: [rtcweb] Prioritization
>=20
> On 25 April 2014 08:17, Simon Perreault <simon.perreault@viagenie.ca>
> wrote:
> > I'll shut up now. I see your point. If nobody else feels this issue is
> > important, feel free to ignore my comment.
>=20
> Honestly, I was originally inclined to agree with you, but the SHOULD lea=
ves
> plenty of wiggle room for those who think they know better.
>=20
> I think that providing motivation in the document - something like "This
> ensures that lower priority streams receive fewer resources, but cannot b=
e
> completely blocked by those at higher priority." - would help to ensure t=
hat
> people who think they know better - and don't - can't screw things up too
> much.
>=20
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb


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Harald Alvestrand:
> Prioritization is a different matter.
>=20
> What I was trying to encapsulate was:
>=20
> - Higher priority means more packets get sent
> - "More" has a numeric value, so behaviour is predictable (that's where t=
he
> x2 figure came from)
> - Small packet flows get more packets than large-packet flows (counting
> bytes not packets)
> - Lower priority flows can't be starved completely
>=20
> This text captures that. I'd like to continue to capture that.

The above is simply wrong, so the text shouldn't capture it at all.

If I say that audio is more important than video, and I try to send 100kbps=
 of audio and 3 Mbps of video over a 101kbps channel, how much audio should=
 I drop at send time to let video through?

Matthew Kaufman


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On 04/25/2014 07:06 PM, Matthew Kaufman (SKYPE) wrote:
> Harald Alvestrand:
>> Prioritization is a different matter.
>>
>> What I was trying to encapsulate was:
>>
>> - Higher priority means more packets get sent
>> - "More" has a numeric value, so behaviour is predictable (that's where the
>> x2 figure came from)
>> - Small packet flows get more packets than large-packet flows (counting
>> bytes not packets)
>> - Lower priority flows can't be starved completely
>>
>> This text captures that. I'd like to continue to capture that.
> The above is simply wrong, so the text shouldn't capture it at all.
>
> If I say that audio is more important than video, and I try to send 100kbps of audio and 3 Mbps of video over a 101kbps channel, how much audio should I drop at send time to let video through?
>
> Matthew Kaufman
Is either stream adaptive?

If I send high quality OPUS (100 Kbits/sec) and VP8 in HD (3 Mbits/sec), 
the suggested result would likely be OPUS scaled down to ~60 Kbits/sec 
and VP8 scaled down to ~30 Kbits/sec.

The latter is probably useful for recognizing that there was an 
intention to send a picture.
This MAY be a fine result, depending on the application's requirements.



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From: Paul Coverdale <coverdale@sympatico.ca>
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>>On 25/04/14 06:21 AM, Kiran Kumar Guduru wrote:
>> Is that means spec is recommending G.711 (PCMA/PCMU) for sampling rate
>> less than or equal to 8 kHz?
>
>No. Opus is still a good idea for 8 kHz, but it's not 2119 RECOMMENDED
>since there are valid (mostly complexity-related) reasons to use G.711
>despite the higher bit-rate.
>
>	Jean-Marc

Opus isn't a good idea for 8 KHz if you are trying to inter-operate with
another end-point using G.711.

...Paul


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--001a11c12366f4922a04f812e8fa
Content-Type: text/plain; charset=UTF-8

Paul,
What do you say, if another end point is using both G.711 and Opus at 8kHz
and why ?

Steev.



>> Is that means spec is recommending G.711 (PCMA/PCMU) for sampling rate
> >> less than or equal to 8 kHz?
> >
> >No. Opus is still a good idea for 8 kHz, but it's not 2119 RECOMMENDED
> >since there are valid (mostly complexity-related) reasons to use G.711
> >despite the higher bit-rate.
> >
> >       Jean-Marc
>
> Opus isn't a good idea for 8 KHz if you are trying to inter-operate with
> another end-point using G.711.
>
> ...Paul
>
>
>

--001a11c12366f4922a04f812e8fa
Content-Type: text/html; charset=UTF-8
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<div dir=3D"ltr"><div><div>Paul,<br></div>What do you say, if another end p=
oint is using both G.711 and Opus at 8kHz and why ?<br><br></div>Steev.<br>=
<div><div class=3D"gmail_extra"><br><br><div class=3D"gmail_quote"><br><blo=
ckquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1px #c=
cc solid;padding-left:1ex">

&gt;&gt; Is that means spec is recommending G.711 (PCMA/PCMU) for sampling =
rate<br>
&gt;&gt; less than or equal to 8 kHz?<br>
&gt;<br>
&gt;No. Opus is still a good idea for 8 kHz, but it&#39;s not 2119 RECOMMEN=
DED<br>
&gt;since there are valid (mostly complexity-related) reasons to use G.711<=
br>
&gt;despite the higher bit-rate.<br>
&gt;<br>
&gt; =C2=A0 =C2=A0 =C2=A0 Jean-Marc<br>
<br>
Opus isn&#39;t a good idea for 8 KHz if you are trying to inter-operate wit=
h<br>
another end-point using G.711.<br>
<br>
...Paul<br>
<br>
<br></blockquote></div><br></div></div></div>

--001a11c12366f4922a04f812e8fa--


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

So, if the other endpoint only supports G.711 (legacy), then it
doesn't matter which codec you offer first. The the other endpoint
supports both Opus and G.711, but only at 8 kHz, then it comes down to
which of complexity or bandwidth is more important. On a corporate LAN
you'd likely want to use G.711 while for chatting from your home,
you'd usually want Opus. This is why the draft makes has no SHOULD
when it comes to narrowband.

	Jean-Marc

On 28/04/14 12:40 AM, Steev James wrote:
> Paul, What do you say, if another end point is using both G.711 and
> Opus at 8kHz and why ?
> 
> Steev.
> 
> 
> 
>>> Is that means spec is recommending G.711 (PCMA/PCMU) for
>>> sampling
> rate
>>> less than or equal to 8 kHz?
>> 
>> No. Opus is still a good idea for 8 kHz, but it's not 2119
>> RECOMMENDED since there are valid (mostly complexity-related)
>> reasons to use G.711 despite the higher bit-rate.
>> 
>> Jean-Marc
> 
> Opus isn't a good idea for 8 KHz if you are trying to inter-operate
> with another end-point using G.711.
> 
> ...Paul
> 
> 
> 
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--001a11c300a608470404f816e729
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Means you say that G.711 uses less bandwidth than Opus at the expense of
complexity.

Am I right ?


On Mon, Apr 28, 2014 at 11:23 AM, Jean-Marc Valin <jmvalin@mozilla.com>wrote:

> -----BEGIN PGP SIGNED MESSAGE-----
> Hash: SHA1
>
> So, if the other endpoint only supports G.711 (legacy), then it
> doesn't matter which codec you offer first. The the other endpoint
> supports both Opus and G.711, but only at 8 kHz, then it comes down to
> which of complexity or bandwidth is more important. On a corporate LAN
> you'd likely want to use G.711 while for chatting from your home,
> you'd usually want Opus. This is why the draft makes has no SHOULD
> when it comes to narrowband.
>
>         Jean-Marc
>
> On 28/04/14 12:40 AM, Steev James wrote:
> > Paul, What do you say, if another end point is using both G.711 and
> > Opus at 8kHz and why ?
> >
> > Steev.
> >
> >
> >
> >>> Is that means spec is recommending G.711 (PCMA/PCMU) for
> >>> sampling
> > rate
> >>> less than or equal to 8 kHz?
> >>
> >> No. Opus is still a good idea for 8 kHz, but it's not 2119
> >> RECOMMENDED since there are valid (mostly complexity-related)
> >> reasons to use G.711 despite the higher bit-rate.
> >>
> >> Jean-Marc
> >
> > Opus isn't a good idea for 8 KHz if you are trying to inter-operate
> > with another end-point using G.711.
> >
> > ...Paul
> >
> >
> >
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<div dir=3D"ltr"><div>Means you say that G.711 uses less bandwidth than Opu=
s at the expense of complexity.<br><br></div>Am I right ?<br><div><div><div=
 class=3D"gmail_extra"><br><br><div class=3D"gmail_quote">On Mon, Apr 28, 2=
014 at 11:23 AM, Jean-Marc Valin <span dir=3D"ltr">&lt;<a href=3D"mailto:jm=
valin@mozilla.com" target=3D"_blank">jmvalin@mozilla.com</a>&gt;</span> wro=
te:<br>
<blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1p=
x #ccc solid;padding-left:1ex">-----BEGIN PGP SIGNED MESSAGE-----<br>
Hash: SHA1<br>
<br>
So, if the other endpoint only supports G.711 (legacy), then it<br>
doesn&#39;t matter which codec you offer first. The the other endpoint<br>
supports both Opus and G.711, but only at 8 kHz, then it comes down to<br>
which of complexity or bandwidth is more important. On a corporate LAN<br>
you&#39;d likely want to use G.711 while for chatting from your home,<br>
you&#39;d usually want Opus. This is why the draft makes has no SHOULD<br>
when it comes to narrowband.<br>
<br>
=C2=A0 =C2=A0 =C2=A0 =C2=A0 Jean-Marc<br>
<div><div class=3D"h5"><br>
On 28/04/14 12:40 AM, Steev James wrote:<br>
&gt; Paul, What do you say, if another end point is using both G.711 and<br=
>
&gt; Opus at 8kHz and why ?<br>
&gt;<br>
&gt; Steev.<br>
&gt;<br>
&gt;<br>
&gt;<br>
&gt;&gt;&gt; Is that means spec is recommending G.711 (PCMA/PCMU) for<br>
&gt;&gt;&gt; sampling<br>
&gt; rate<br>
&gt;&gt;&gt; less than or equal to 8 kHz?<br>
&gt;&gt;<br>
&gt;&gt; No. Opus is still a good idea for 8 kHz, but it&#39;s not 2119<br>
&gt;&gt; RECOMMENDED since there are valid (mostly complexity-related)<br>
&gt;&gt; reasons to use G.711 despite the higher bit-rate.<br>
&gt;&gt;<br>
&gt;&gt; Jean-Marc<br>
&gt;<br>
&gt; Opus isn&#39;t a good idea for 8 KHz if you are trying to inter-operat=
e<br>
&gt; with another end-point using G.711.<br>
&gt;<br>
&gt; ...Paul<br>
&gt;<br>
&gt;<br>
&gt;<br>
</div></div>-----BEGIN PGP SIGNATURE-----<br>
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</blockquote></div><br></div></div></div></div>

--001a11c300a608470404f816e729--


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Subject: Re: [rtcweb] Query Regarding Mandatory audio codecs draft
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On 04/28/2014 11:25 AM, Steev James wrote:
> Means you say that G.711 uses less bandwidth than Opus at the expense 
> of complexity.

G.711 gives very low audio quality. The larger complexity of Opus means 
that it is able to give a much better quality at the same bandwidth.

Personal opinion: The only time I'd be wanting to use G.711 if Opus is 
available is when I'm on a severely underpowered mobile phone. That 
platform is likely also so weak that I couldn't even imagine using video 
on it (video is order-of-magnitude more complex to encode/decode than Opus).


From nobody Mon Apr 28 05:51:24 2014
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--001a1136526a560c5f04f819c5f0
Content-Type: text/plain; charset=UTF-8

No.  G.711 uses more network bandwidth than OPUS, but OPUS is more complex.

The main benefit of using G.711 is that it can avoid the use of transcoding
in some scenarios.

My view is that the approach in the current text is correct - there is no
need to specify a SHOULD on which codec is offered first.

Stephen


On Mon, Apr 28, 2014 at 5:25 AM, Steev James <steev.a.james@gmail.com>wrote:

> Means you say that G.711 uses less bandwidth than Opus at the expense of
> complexity.
>
> Am I right ?
>
>
> On Mon, Apr 28, 2014 at 11:23 AM, Jean-Marc Valin <jmvalin@mozilla.com>wrote:
>
>> -----BEGIN PGP SIGNED MESSAGE-----
>> Hash: SHA1
>>
>> So, if the other endpoint only supports G.711 (legacy), then it
>> doesn't matter which codec you offer first. The the other endpoint
>> supports both Opus and G.711, but only at 8 kHz, then it comes down to
>> which of complexity or bandwidth is more important. On a corporate LAN
>> you'd likely want to use G.711 while for chatting from your home,
>> you'd usually want Opus. This is why the draft makes has no SHOULD
>> when it comes to narrowband.
>>
>>         Jean-Marc
>>
>> On 28/04/14 12:40 AM, Steev James wrote:
>> > Paul, What do you say, if another end point is using both G.711 and
>> > Opus at 8kHz and why ?
>> >
>> > Steev.
>>
>> >
>> >
>> >
>> >>> Is that means spec is recommending G.711 (PCMA/PCMU) for
>> >>> sampling
>> > rate
>> >>> less than or equal to 8 kHz?
>> >>
>> >> No. Opus is still a good idea for 8 kHz, but it's not 2119
>> >> RECOMMENDED since there are valid (mostly complexity-related)
>> >> reasons to use G.711 despite the higher bit-rate.
>> >>
>> >> Jean-Marc
>> >
>> > Opus isn't a good idea for 8 KHz if you are trying to inter-operate
>> > with another end-point using G.711.
>> >
>> > ...Paul
>> >
>> >
>> >
>> -----BEGIN PGP SIGNATURE-----
>> Version: GnuPG v1
>> Comment: Using GnuPG with Thunderbird - http://www.enigmail.net/
>>
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>>
>
>
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb
>
>

--001a1136526a560c5f04f819c5f0
Content-Type: text/html; charset=UTF-8
Content-Transfer-Encoding: quoted-printable

<div dir=3D"ltr">No. =C2=A0G.711 uses more network bandwidth than OPUS, but=
 OPUS is more complex.<div><br></div><div>The main benefit of using G.711 i=
s that it can avoid the use of transcoding in some scenarios.<br><div><div>=
<br>
</div><div>My view is that the approach in the current text is correct - th=
ere is no need to specify a SHOULD on which codec is offered first.</div></=
div></div><div><br></div><div>Stephen</div><div class=3D"gmail_extra"><br>
<br><div class=3D"gmail_quote">On Mon, Apr 28, 2014 at 5:25 AM, Steev James=
 <span dir=3D"ltr">&lt;<a href=3D"mailto:steev.a.james@gmail.com" target=3D=
"_blank">steev.a.james@gmail.com</a>&gt;</span> wrote:<br><blockquote class=
=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1px #ccc solid;padd=
ing-left:1ex">
<div dir=3D"ltr"><div>Means you say that G.711 uses less bandwidth than Opu=
s at the expense of complexity.<br><br></div>Am I right ?<br><div><div><div=
 class=3D"gmail_extra"><br><br><div class=3D"gmail_quote"><div class=3D"">O=
n Mon, Apr 28, 2014 at 11:23 AM, Jean-Marc Valin <span dir=3D"ltr">&lt;<a h=
ref=3D"mailto:jmvalin@mozilla.com" target=3D"_blank">jmvalin@mozilla.com</a=
>&gt;</span> wrote:<br>

</div><blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-l=
eft:1px #ccc solid;padding-left:1ex"><div class=3D"">-----BEGIN PGP SIGNED =
MESSAGE-----<br>
Hash: SHA1<br>
<br></div>
So, if the other endpoint only supports G.711 (legacy), then it<br>
doesn&#39;t matter which codec you offer first. The the other endpoint<br>
supports both Opus and G.711, but only at 8 kHz, then it comes down to<br>
which of complexity or bandwidth is more important. On a corporate LAN<br>
you&#39;d likely want to use G.711 while for chatting from your home,<br>
you&#39;d usually want Opus. This is why the draft makes has no SHOULD<br>
when it comes to narrowband.<br>
<br>
=C2=A0 =C2=A0 =C2=A0 =C2=A0 Jean-Marc<br>
<div><div><br>
On 28/04/14 12:40 AM, Steev James wrote:<br>
&gt; Paul, What do you say, if another end point is using both G.711 and<br=
>
&gt; Opus at 8kHz and why ?<br>
&gt;<br>
&gt; Steev.<div class=3D""><br>
&gt;<br>
&gt;<br>
&gt;<br>
&gt;&gt;&gt; Is that means spec is recommending G.711 (PCMA/PCMU) for<br>
&gt;&gt;&gt; sampling<br>
&gt; rate<br>
&gt;&gt;&gt; less than or equal to 8 kHz?<br>
&gt;&gt;<br>
&gt;&gt; No. Opus is still a good idea for 8 kHz, but it&#39;s not 2119<br>
&gt;&gt; RECOMMENDED since there are valid (mostly complexity-related)<br>
&gt;&gt; reasons to use G.711 despite the higher bit-rate.<br>
&gt;&gt;<br>
&gt;&gt; Jean-Marc<br>
&gt;<br>
&gt; Opus isn&#39;t a good idea for 8 KHz if you are trying to inter-operat=
e<br>
&gt; with another end-point using G.711.<br>
&gt;<br>
&gt; ...Paul<br>
&gt;<br>
&gt;<br>
&gt;<br>
</div></div></div><div class=3D"">-----BEGIN PGP SIGNATURE-----<br>
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</blockquote></div><br></div></div></div></div>
<br>_______________________________________________<br>
rtcweb mailing list<br>
<a href=3D"mailto:rtcweb@ietf.org">rtcweb@ietf.org</a><br>
<a href=3D"https://www.ietf.org/mailman/listinfo/rtcweb" target=3D"_blank">=
https://www.ietf.org/mailman/listinfo/rtcweb</a><br>
<br></blockquote></div><br></div></div>

--001a1136526a560c5f04f819c5f0--


From nobody Mon Apr 28 06:46:27 2014
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From: Paul Coverdale <coverdale@sympatico.ca>
To: "'Steev James'" <steev.a.james@gmail.com>, "'Jean-Marc Valin'" <jmvalin@mozilla.com>
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Subject: Re: [rtcweb] Query Regarding Mandatory audio codecs draft
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I think we=E2=80=99re mixing up sampling frequency and bit-rate here.

=20

By definition, G.711 uses a sampling frequency of 8 KHz and encodes each =
sample with an 8-bit logarithmically companded, encoding law (A or Mu), =
resulting in an output bit-rate of 64 kbit/s.=20

=20

Opus can operate with a number of different sampling frequencies =
(depending on the desired audio bandwidth) and output bit-rates, =
including 64 kbit/s.

=20

The point I was trying to make is that, even if both codecs provide the =
same numerical bit-rate, they are not compatible between end-points. If =
a legacy end-point can only support G.711, it=E2=80=99s no good sending =
it anything else but G.711.

=20

...Paul

=20

From: Steev James [mailto:steev.a.james@gmail.com]=20
Sent: Monday, April 28, 2014 5:26 AM
To: Jean-Marc Valin
Cc: rtcweb@ietf.org; coverdale@sympatico.ca; kiran.guduru@samsung.com; =
cary.bran@plantronics.com
Subject: Re:Query Regarding Mandatory audio codecs draft

=20

Means you say that G.711 uses less bandwidth than Opus at the expense of =
complexity.

Am I right ?

=20

On Mon, Apr 28, 2014 at 11:23 AM, Jean-Marc Valin <jmvalin@mozilla.com> =
wrote:

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

So, if the other endpoint only supports G.711 (legacy), then it
doesn't matter which codec you offer first. The the other endpoint
supports both Opus and G.711, but only at 8 kHz, then it comes down to
which of complexity or bandwidth is more important. On a corporate LAN
you'd likely want to use G.711 while for chatting from your home,
you'd usually want Opus. This is why the draft makes has no SHOULD
when it comes to narrowband.

        Jean-Marc


On 28/04/14 12:40 AM, Steev James wrote:
> Paul, What do you say, if another end point is using both G.711 and
> Opus at 8kHz and why ?
>
> Steev.
>
>
>
>>> Is that means spec is recommending G.711 (PCMA/PCMU) for
>>> sampling
> rate
>>> less than or equal to 8 kHz?
>>
>> No. Opus is still a good idea for 8 kHz, but it's not 2119
>> RECOMMENDED since there are valid (mostly complexity-related)
>> reasons to use G.711 despite the higher bit-rate.
>>
>> Jean-Marc
>
> Opus isn't a good idea for 8 KHz if you are trying to inter-operate
> with another end-point using G.711.
>
> ...Paul
>
>
>

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vlink=3Dpurple><div class=3DWordSection1><p class=3DMsoNormal><span =
style=3D'font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497=
D'>I think we=E2=80=99re mixing up sampling frequency and bit-rate =
here.<o:p></o:p></span></p><p class=3DMsoNormal><span =
style=3D'font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497=
D'><o:p>&nbsp;</o:p></span></p><p class=3DMsoNormal><span =
style=3D'font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497=
D'>By definition, G.711 uses a sampling frequency of 8 KHz and encodes =
each sample with an 8-bit logarithmically companded, encoding law (A or =
Mu), resulting in an output bit-rate of 64 kbit/s. =
<o:p></o:p></span></p><p class=3DMsoNormal><span =
style=3D'font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497=
D'><o:p>&nbsp;</o:p></span></p><p class=3DMsoNormal><span =
style=3D'font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497=
D'>Opus can operate with a number of different sampling frequencies =
(depending on the desired audio bandwidth) and output bit-rates, =
including 64 kbit/s.<o:p></o:p></span></p><p class=3DMsoNormal><span =
style=3D'font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497=
D'><o:p>&nbsp;</o:p></span></p><p class=3DMsoNormal><span =
style=3D'font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497=
D'>The point I was trying to make is that, even if both codecs provide =
the same numerical bit-rate, they are not compatible between end-points. =
If a legacy end-point can only support G.711, it=E2=80=99s no good =
sending it anything else but G.711.<o:p></o:p></span></p><p =
class=3DMsoNormal><span =
style=3D'font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497=
D'><o:p>&nbsp;</o:p></span></p><p class=3DMsoNormal><span =
style=3D'font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497=
D'>...Paul<o:p></o:p></span></p><p class=3DMsoNormal><span =
style=3D'font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497=
D'><o:p>&nbsp;</o:p></span></p><div =
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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"'> =
Steev James [mailto:steev.a.james@gmail.com] <br><b>Sent:</b> Monday, =
April 28, 2014 5:26 AM<br><b>To:</b> Jean-Marc Valin<br><b>Cc:</b> =
rtcweb@ietf.org; coverdale@sympatico.ca; kiran.guduru@samsung.com; =
cary.bran@plantronics.com<br><b>Subject:</b> Re:Query Regarding =
Mandatory audio codecs draft<o:p></o:p></span></p></div></div><p =
class=3DMsoNormal><o:p>&nbsp;</o:p></p><div><div><p class=3DMsoNormal =
style=3D'margin-bottom:12.0pt'>Means you say that G.711 uses less =
bandwidth than Opus at the expense of complexity.<o:p></o:p></p></div><p =
class=3DMsoNormal>Am I right ?<o:p></o:p></p><div><div><div><p =
class=3DMsoNormal =
style=3D'margin-bottom:12.0pt'><o:p>&nbsp;</o:p></p><div><p =
class=3DMsoNormal>On Mon, Apr 28, 2014 at 11:23 AM, Jean-Marc Valin =
&lt;<a href=3D"mailto:jmvalin@mozilla.com" =
target=3D"_blank">jmvalin@mozilla.com</a>&gt; wrote:<o:p></o:p></p><p =
class=3DMsoNormal>-----BEGIN PGP SIGNED MESSAGE-----<br>Hash: =
SHA1<br><br>So, if the other endpoint only supports G.711 (legacy), then =
it<br>doesn't matter which codec you offer first. The the other =
endpoint<br>supports both Opus and G.711, but only at 8 kHz, then it =
comes down to<br>which of complexity or bandwidth is more important. On =
a corporate LAN<br>you'd likely want to use G.711 while for chatting =
from your home,<br>you'd usually want Opus. This is why the draft makes =
has no SHOULD<br>when it comes to narrowband.<br><br>&nbsp; &nbsp; =
&nbsp; &nbsp; Jean-Marc<o:p></o:p></p><div><div><p =
class=3DMsoNormal><br>On 28/04/14 12:40 AM, Steev James wrote:<br>&gt; =
Paul, What do you say, if another end point is using both G.711 =
and<br>&gt; Opus at 8kHz and why ?<br>&gt;<br>&gt; =
Steev.<br>&gt;<br>&gt;<br>&gt;<br>&gt;&gt;&gt; Is that means spec is =
recommending G.711 (PCMA/PCMU) for<br>&gt;&gt;&gt; sampling<br>&gt; =
rate<br>&gt;&gt;&gt; less than or equal to 8 =
kHz?<br>&gt;&gt;<br>&gt;&gt; No. Opus is still a good idea for 8 kHz, =
but it's not 2119<br>&gt;&gt; RECOMMENDED since there are valid (mostly =
complexity-related)<br>&gt;&gt; reasons to use G.711 despite the higher =
bit-rate.<br>&gt;&gt;<br>&gt;&gt; Jean-Marc<br>&gt;<br>&gt; Opus isn't a =
good idea for 8 KHz if you are trying to inter-operate<br>&gt; with =
another end-point using G.711.<br>&gt;<br>&gt; =
...Paul<br>&gt;<br>&gt;<br>&gt;<o:p></o:p></p></div></div><p =
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From nobody Mon Apr 28 08:16:51 2014
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From: Ted Hardie <ted.ietf@gmail.com>
To: "rtcweb@ietf.org" <rtcweb@ietf.org>
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Subject: [rtcweb] Document mechanics for JSEP
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--089e0112c3b0d1d4c204f81bcd31
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Howdy,

In order to increase the velocity of our work, the chairs would like to
make some changes to how we approach the document editing for JSEP.  First,
we have asked Eric Rescorla to act as an editor of the document.  Cullen
and Justin will, of course, continue to be listed as authors, but the
day-to-day editorial work will be done by ekr.  Second, we would like to
work using github as the repository of the working copy, with pull requests
for specific text changes there.  The repository is at:

https://github.com/rtcweb-wg/jsep

Note that issues can be raised there, but that non-editorial issues must
then be submitted to the mailing list, and  substantive discussion of those
issues should remain on the mailing list.

This approach has worked well in the HTTP and TLS working groups, and we
hope it can help us deliver in a timely way.

Let us know if there are questions or concerns,

thanks,

Ted and Sean (Cullen recused)

--089e0112c3b0d1d4c204f81bcd31
Content-Type: text/html; charset=UTF-8
Content-Transfer-Encoding: quoted-printable

<div dir=3D"ltr"><div class=3D"gmail_default" style=3D"font-family:georgia,=
serif"><div class=3D"gmail_default" style=3D"font-family:georgia,serif">How=
dy,<br><br></div><div class=3D"gmail_default" style=3D"font-family:georgia,=
serif">In
 order to increase the velocity of our work, the chairs would like to=20
make some changes to how we approach the document editing for JSEP.=C2=A0=
=20
First, we have asked Eric Rescorla to act as an editor of the document.=C2=
=A0
 Cullen and Justin will, of course, continue to be listed as authors,=20
but the day-to-day editorial work will be done by ekr.=C2=A0 Second, we wou=
ld
 like to work using github as the repository of the working copy, with=20
pull requests for specific text changes there.=C2=A0 The repository is at:<=
br>
<br><a href=3D"https://github.com/rtcweb-wg/jsep" target=3D"_blank">https:/=
/github.com/rtcweb-wg/jsep</a><br><br></div><div class=3D"gmail_default" st=
yle=3D"font-family:georgia,serif">Note
 that issues can be raised there, but that non-editorial issues must=20
then be submitted to the mailing list, and=C2=A0 substantive discussion of=
=20
those issues should remain on the mailing list.=C2=A0 <br>
<br></div><div class=3D"gmail_default" style=3D"font-family:georgia,serif">=
This approach has worked well in the HTTP and TLS working groups, and we ho=
pe it can help us deliver in a timely way.<br><br></div>
Let us know if there are questions or concerns,<br><br>thanks,<br><br>Ted a=
nd Sean (Cullen recused)</div></div>

--089e0112c3b0d1d4c204f81bcd31--


From nobody Mon Apr 28 08:19:46 2014
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Date: Mon, 28 Apr 2014 09:19:51 -0600
From: Jessie Morris <jmorris@getjive.com>
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Subject: Re: [rtcweb] Query Regarding Mandatory audio codecs draft
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The only situation that it would make sense to use G.711 and video is on 
a weak platform that happens to have hardware decoding capabilities for 
H.264 or VP8 or whatever the video codec is. I don't think there are a 
lot of these currently, but it is possible.

On 2014-04-28, 06:14, Harald Alvestrand wrote:
> On 04/28/2014 11:25 AM, Steev James wrote:
>> Means you say that G.711 uses less bandwidth than Opus at the expense 
>> of complexity.
>
> G.711 gives very low audio quality. The larger complexity of Opus 
> means that it is able to give a much better quality at the same 
> bandwidth.
>
> Personal opinion: The only time I'd be wanting to use G.711 if Opus is 
> available is when I'm on a severely underpowered mobile phone. That 
> platform is likely also so weak that I couldn't even imagine using 
> video on it (video is order-of-magnitude more complex to encode/decode 
> than Opus).
>
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb
--
Jessie A. Morris
Jive Communications, Inc.
jmorris@getjive.com


From nobody Mon Apr 28 09:19:24 2014
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On 28 April 2014 08:16, Ted Hardie <ted.ietf@gmail.com> wrote:
> The repository is at:
>
> https://github.com/rtcweb-wg/jsep

Good.


From nobody Mon Apr 28 11:23:28 2014
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--001a1136526a2e0a2604f81e69b7
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The mandatory codec question for audio is long settled, and personally I am
thinking we really don't want to reopen that discussion.

The original poster was wanting to know if G.711 was preferred over OPUS
for narrow band.  The answer given was "no".  Is there really a need to
discuss this more?

Stephen


On Mon, Apr 28, 2014 at 11:19 AM, Jessie Morris <jmorris@getjive.com> wrote:

> The only situation that it would make sense to use G.711 and video is on a
> weak platform that happens to have hardware decoding capabilities for H.264
> or VP8 or whatever the video codec is. I don't think there are a lot of
> these currently, but it is possible.
>
>
> On 2014-04-28, 06:14, Harald Alvestrand wrote:
>
>> On 04/28/2014 11:25 AM, Steev James wrote:
>>
>>> Means you say that G.711 uses less bandwidth than Opus at the expense of
>>> complexity.
>>>
>>
>> G.711 gives very low audio quality. The larger complexity of Opus means
>> that it is able to give a much better quality at the same bandwidth.
>>
>> Personal opinion: The only time I'd be wanting to use G.711 if Opus is
>> available is when I'm on a severely underpowered mobile phone. That
>> platform is likely also so weak that I couldn't even imagine using video on
>> it (video is order-of-magnitude more complex to encode/decode than Opus).
>>
>> _______________________________________________
>> rtcweb mailing list
>> rtcweb@ietf.org
>> https://www.ietf.org/mailman/listinfo/rtcweb
>>
> --
> Jessie A. Morris
> Jive Communications, Inc.
> jmorris@getjive.com
>
>
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb
>

--001a1136526a2e0a2604f81e69b7
Content-Type: text/html; charset=UTF-8
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<div dir=3D"ltr"><div>The mandatory codec question for audio is long settle=
d, and personally I am thinking we really don&#39;t want to reopen that dis=
cussion. =C2=A0<br></div><div><br></div><div>The original poster was wantin=
g to know if G.711 was preferred over OPUS for narrow band. =C2=A0The answe=
r given was &quot;no&quot;. =C2=A0Is there really a need to discuss this mo=
re?</div>
<div><br></div><div>Stephen =C2=A0</div><div class=3D"gmail_extra"><br><br>=
<div class=3D"gmail_quote">On Mon, Apr 28, 2014 at 11:19 AM, Jessie Morris =
<span dir=3D"ltr">&lt;<a href=3D"mailto:jmorris@getjive.com" target=3D"_bla=
nk">jmorris@getjive.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">The only situation that it would make sense =
to use G.711 and video is on a weak platform that happens to have hardware =
decoding capabilities for H.264 or VP8 or whatever the video codec is. I do=
n&#39;t think there are a lot of these currently, but it is possible.<div>
<div class=3D"h5"><br>
<br>
On 2014-04-28, 06:14, Harald Alvestrand wrote:<br>
<blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1p=
x #ccc solid;padding-left:1ex">
On 04/28/2014 11:25 AM, Steev James wrote:<br>
<blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1p=
x #ccc solid;padding-left:1ex">
Means you say that G.711 uses less bandwidth than Opus at the expense of co=
mplexity.<br>
</blockquote>
<br>
G.711 gives very low audio quality. The larger complexity of Opus means tha=
t it is able to give a much better quality at the same bandwidth.<br>
<br>
Personal opinion: The only time I&#39;d be wanting to use G.711 if Opus is =
available is when I&#39;m on a severely underpowered mobile phone. That pla=
tform is likely also so weak that I couldn&#39;t even imagine using video o=
n it (video is order-of-magnitude more complex to encode/decode than Opus).=
<br>

<br>
______________________________<u></u>_________________<br>
rtcweb mailing list<br>
<a href=3D"mailto:rtcweb@ietf.org" target=3D"_blank">rtcweb@ietf.org</a><br=
>
<a href=3D"https://www.ietf.org/mailman/listinfo/rtcweb" target=3D"_blank">=
https://www.ietf.org/mailman/<u></u>listinfo/rtcweb</a><br>
</blockquote></div></div>
--<br>
Jessie A. Morris<br>
Jive Communications, Inc.<br>
<a href=3D"mailto:jmorris@getjive.com" target=3D"_blank">jmorris@getjive.co=
m</a><div class=3D"HOEnZb"><div class=3D"h5"><br>
<br>
______________________________<u></u>_________________<br>
rtcweb mailing list<br>
<a href=3D"mailto:rtcweb@ietf.org" target=3D"_blank">rtcweb@ietf.org</a><br=
>
<a href=3D"https://www.ietf.org/mailman/listinfo/rtcweb" target=3D"_blank">=
https://www.ietf.org/mailman/<u></u>listinfo/rtcweb</a><br>
</div></div></blockquote></div><br></div></div>

--001a1136526a2e0a2604f81e69b7--


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Archived-At: http://mailarchive.ietf.org/arch/msg/rtcweb/SBBqERLr-nhsYxPMe4mnOAE8IwM
Subject: Re: [rtcweb] Document mechanics for JSEP
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From nobody Wed Apr 30 14:09:26 2014
Return-Path: <fluffy@cisco.com>
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From: "Cullen Jennings (fluffy)" <fluffy@cisco.com>
To: Matthew Kaufman <matthew.kaufman@skype.net>
Thread-Topic: [rtcweb] Prioritization
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Date: Wed, 30 Apr 2014 21:09:19 +0000
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Subject: Re: [rtcweb] Prioritization
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On Apr 25, 2014, at 11:03 AM, Matthew Kaufman (SKYPE) <matthew.kaufman@skyp=
e.net> wrote:

> If a "lower priority" packet is dispatched before a "higher priority" pac=
ket in order to "prevent starvation", then what does "higher priority" mean=
?

I think the labels reflect what "might" happen on average and not for any p=
articular packet. I say "might" because in some deployments it possible the=
 values will have no effect at all and it is possible to construct bizarre =
environments where you could get the opposite of the desired behavior in so=
me cases. None of that bothers me as the label still seem like the closest =
to helping developers request the desired treatment. With all things qos li=
ke, the best we can for is desired outcome and not what will necessarily ha=
ppen on internet.=20

priority might be better called "desired priority" and the values of "very =
low, low, med, high" might better be called "broccoli, apple, orange, cherr=
y" - they are really just some labels that describe how the packets *might*=
 be queued by systems they pass through.  I think the fruit labels would be=
 more confusing to web programmers.=20



From nobody Wed Apr 30 14:22:13 2014
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From: "Cullen Jennings (fluffy)" <fluffy@cisco.com>
To: Roman Shpount <roman@telurix.com>
Thread-Topic: [rtcweb] Pictures of congestion control on the Internet - which is more realistic?
Thread-Index: AQHPZLoqT0r3kE244UWS8vLGoCPh4w==
Date: Wed, 30 Apr 2014 21:21:35 +0000
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Subject: Re: [rtcweb] Pictures of congestion control on the Internet - which is more realistic?
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On Apr 23, 2014, at 9:02 AM, Roman Shpount <roman@telurix.com> wrote:

> I would say that some video end points have congestion back off.  Almost =
all audio end points have none. Based on this, most UDP endpoints do not de=
al well with congestion.

Well =85 I sort of agree with you and Wenger and sort of don=92t. They have=
 an upper layer congestion control. Basically when the congestion gets bad,=
 the humans on both ends hang up the call and go call each other on their c=
ell phones.=20

But the bottom line of all this is we all wish we had better congestion con=
trol for interactive voice and video and RMCAT is working on that and it is=
 hard and going to take awhile to figure out and probably longer to deploy.=
=20



From nobody Wed Apr 30 14:23:40 2014
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References: <20140425084726.8812.24604.idtracker@ietfa.amsl.com> <535A21E3.7070008@alvestrand.no> <535A5ACC.9070700@viagenie.ca> <535A6151.1060501@alvestrand.no> <535A68E1.9090901@viagenie.ca> <535A78FF.20700@alvestrand.no> <535A7C73.6050701@viagenie.ca> <CABkgnnWkOGdSzP42rZ-aGjFkGDOOGOfk64rq-80GjeVPZJAqaw@mail.gmail.com> <AE1A6B5FD507DC4FB3C5166F3A05A484504DFEA3@TK5EX14MBXC298.redmond.corp.microsoft.com> <F60C5C26-CFFE-45D1-BF1A-D1C320835C8A@cisco.com>
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From: Martin Thomson <martin.thomson@gmail.com>
To: "Cullen Jennings (fluffy)" <fluffy@cisco.com>
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Subject: Re: [rtcweb] Prioritization
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On 30 April 2014 14:09, Cullen Jennings (fluffy) <fluffy@cisco.com> wrote:
> On Apr 25, 2014, at 11:03 AM, Matthew Kaufman (SKYPE) <matthew.kaufman@skype.net> wrote:
>
>> If a "lower priority" packet is dispatched before a "higher priority" packet in order to "prevent starvation", then what does "higher priority" mean?
>
> I think the labels reflect what "might" happen on average and not for any particular packet.

I think that Matthew was referring to the part where the browser is
involved.  That is, the bit where, when presented with the option to
send just one packet from buckets A through D, how does it choose from
those buckets.

The implication was that if A is more important than B, then if A
wants to send, it gets to send.  Period.  The "prevents starvation"
view of the world says that work is shared between A-D, with
increasingly large proportions of the available capacity given to
higher priority streams.  The problem with both these models is that
they are crap in various ways.  In one, you get cases where lower
priority stuff never happens, even if that isn't what you wanted; in
another, lower priority stuff can get resources, and that wasn't what
you wanted.

The DSCP markings and how they might interact with this are just an
additional layer of uncertainty, primarily.


From nobody Wed Apr 30 14:37:25 2014
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From: "Matthew Kaufman (SKYPE)" <matthew.kaufman@skype.net>
To: Martin Thomson <martin.thomson@gmail.com>, "Cullen Jennings (fluffy)" <fluffy@cisco.com>
Thread-Topic: [rtcweb] Prioritization
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References: <20140425084726.8812.24604.idtracker@ietfa.amsl.com> <535A21E3.7070008@alvestrand.no>	<535A5ACC.9070700@viagenie.ca> <535A6151.1060501@alvestrand.no>	<535A68E1.9090901@viagenie.ca> <535A78FF.20700@alvestrand.no>	<535A7C73.6050701@viagenie.ca> <CABkgnnWkOGdSzP42rZ-aGjFkGDOOGOfk64rq-80GjeVPZJAqaw@mail.gmail.com> <AE1A6B5FD507DC4FB3C5166F3A05A484504DFEA3@TK5EX14MBXC298.redmond.corp.microsoft.com> <F60C5C26-CFFE-45D1-BF1A-D1C320835C8A@cisco.com> <CABkgnnWcE+KaDk4OnHo0wDwK_gz_4gSr_F5FRe-X1gf41hKotQ@mail.gmail.com>
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From nobody Wed Apr 30 14:46:09 2014
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Date: Wed, 30 Apr 2014 14:46:01 -0700
Message-ID: <CAA93jw51zt-xfFjbn13WAj+gOfWmef+qnUZ4Ez=KHJO=mM9Akg@mail.gmail.com>
From: Dave Taht <dave.taht@gmail.com>
To: "Matthew Kaufman (SKYPE)" <matthew.kaufman@skype.net>
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Cc: "Cullen Jennings \(fluffy\)" <fluffy@cisco.com>, "rtcweb@ietf.org" <rtcweb@ietf.org>
Subject: Re: [rtcweb] Prioritization
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On Wed, Apr 30, 2014 at 2:36 PM, Matthew Kaufman (SKYPE)
<matthew.kaufman@skype.net> wrote:
>
>> -----Original Message-----
>> From: Martin Thomson [mailto:martin.thomson@gmail.com]
>> Sent: Wednesday, April 30, 2014 2:23 PM
>> To: Cullen Jennings (fluffy)
>> Cc: Matthew Kaufman (SKYPE); Simon Perreault; rtcweb@ietf.org
>> Subject: Re: [rtcweb] Prioritization
>>
>> On 30 April 2014 14:09, Cullen Jennings (fluffy) <fluffy@cisco.com> wrot=
e:
>> > On Apr 25, 2014, at 11:03 AM, Matthew Kaufman (SKYPE)
>> <matthew.kaufman@skype.net> wrote:
>> >
>> >> If a "lower priority" packet is dispatched before a "higher priority"=
 packet
>> in order to "prevent starvation", then what does "higher priority" mean?
>> >
>> > I think the labels reflect what "might" happen on average and not for =
any
>> particular packet.
>>
>> I think that Matthew was referring to the part where the browser is invo=
lved.
>> That is, the bit where, when presented with the option to send just one
>> packet from buckets A through D, how does it choose from those buckets.
>>
>> The implication was that if A is more important than B, then if A wants =
to
>> send, it gets to send.  Period.  The "prevents starvation"
>> view of the world says that work is shared between A-D, with increasingl=
y
>> large proportions of the available capacity given to higher priority str=
eams.
>> The problem with both these models is that they are crap in various ways=
.  In
>> one, you get cases where lower priority stuff never happens, even if tha=
t
>> isn't what you wanted; in another, lower priority stuff can get resource=
s, and
>> that wasn't what you wanted.

It is extremely rare that you want a starvation queue. Two examples are
holding a port open through a natted device, and your typical keepalives.

If you want to support near-starvation queues, you could specify an outer
bound for the maximum time a lower priority queue can be starved.

otherwise a system of weights seems more appropo.

>> The DSCP markings and how they might interact with this are just an
>> additional layer of uncertainty, primarily.
>
> Accurate summary of my position.
>
> Matthew Kaufman
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb



--=20
Dave T=C3=A4ht

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


From nobody Wed Apr 30 14:56:47 2014
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From: "Cullen Jennings (fluffy)" <fluffy@cisco.com>
To: Magnus Westerlund <magnus.westerlund@ericsson.com>
Thread-Topic: [rtcweb] Draft proposal for updating Multiparty topologies in draft-ietf-rtcweb-rtp-usage
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On Apr 15, 2014, at 9:14 AM, Magnus Westerlund <magnus.westerlund@ericsson.=
com> wrote:

> This limitation means that
>   some of the RTP middlebox-based topologies are not suitable for use
>   in the WebRTC environment.  Specifically:
>=20
>   o  Video switching MCUs (Topo-Video-switch-MCU) SHOULD NOT be used,
>      since they make the use of RTCP for congestion control and quality
>      of service reports problematic (see Section 3.8 of
>      [I-D.ietf-avtcore-rtp-topologies-update]).

I think this is deserving of some WG discussion as people may not be up to =
speed of what this is allowing or not allowing. My understanding was severa=
l companies at the last WebRTC Expo conference were demonstrating system th=
at used this type of MCU.=20

If SRTP were more flexible and there was a way to a mixer work without givi=
ng it the keys to the decrypt the media, I think people would be keener on =
mixers but right it seems like the pro / cons invovle a trade off between s=
ignificant security functionally loss and possible loss of some RTCP data w=
hich many systems totally ignore.  Anyway, not taking any opinion other tha=
n this seems like a significant enough change to have some discussion on it=
.=20






From nobody Wed Apr 30 16:01:43 2014
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Date: Wed, 30 Apr 2014 19:01:24 -0400
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From: Roman Shpount <roman@telurix.com>
To: "Cullen Jennings (fluffy)" <fluffy@cisco.com>
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Subject: Re: [rtcweb] Pictures of congestion control on the Internet - which is more realistic?
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On Wed, Apr 30, 2014 at 5:21 PM, Cullen Jennings (fluffy)
<fluffy@cisco.com>wrote:

>
> On Apr 23, 2014, at 9:02 AM, Roman Shpount <roman@telurix.com> wrote:
>
> > I would say that some video end points have congestion back off.  Almos=
t
> all audio end points have none. Based on this, most UDP endpoints do not
> deal well with congestion.
>
> Well =E2=80=A6 I sort of agree with you and Wenger and sort of don=E2=80=
=99t. They have an
> upper layer congestion control. Basically when the congestion gets bad, t=
he
> humans on both ends hang up the call and go call each other on their cell
> phones.


I know we are deep in the discussion but originally I was addressing the
following point:

On 2014-04-23 14:31, Harald Alvestrand wrote:
>  All UDP endpoints that use the open Internet have some form of backoff
on congestion - using delay, packet drop or
> (most likely) both to detect congestion

>From my experience this is not true for almost all voice end points and it
is not true for majority of video end points. So, Harald's assumption does
not hold.

I think based on this assumption Harald was trying to decide if QOS support
needs to be exposed in WebRTC. My second point was that if congestion is
introduced by something other then the real time media client (such as
loading a large document over HTTP from public internet or using a separate
screen sharing program which uses TCP to upload the screen on a slide
change), there is little that can be done by the real time media end point
to deal with this congestion except implement a good packet loss recovery
policy. Better handling of such congestion will require either changes to
router queuing policies (using PIE or Codel queuing policies at the point
of congestion can help a lot) or better yet implementing QOS. Such router
policy changes are outside of this groups charter, but exposing QOS support
to WebRTC is definitely something that should be discussed.
_____________
Roman Shpount

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<div dir=3D"ltr"><div class=3D"gmail_extra"><div class=3D"gmail_quote">On W=
ed, Apr 30, 2014 at 5:21 PM, Cullen Jennings (fluffy) <span dir=3D"ltr">&lt=
;<a href=3D"mailto:fluffy@cisco.com" target=3D"_blank">fluffy@cisco.com</a>=
&gt;</span> wrote:<br>
<blockquote class=3D"gmail_quote" style=3D"margin:0px 0px 0px 0.8ex;border-=
left-width:1px;border-left-color:rgb(204,204,204);border-left-style:solid;p=
adding-left:1ex"><br>
On Apr 23, 2014, at 9:02 AM, Roman Shpount &lt;<a href=3D"mailto:roman@telu=
rix.com">roman@telurix.com</a>&gt; wrote:<br>
<br>
&gt; I would say that some video end points have congestion back off. =C2=
=A0Almost all audio end points have none. Based on this, most UDP endpoints=
 do not deal well with congestion.<br>
<br>
Well =E2=80=A6 I sort of agree with you and Wenger and sort of don=E2=80=99=
t. They have an upper layer congestion control. Basically when the congesti=
on gets bad, the humans on both ends hang up the call and go call each othe=
r on their cell phones.</blockquote>
<div><br></div>I know we are deep in the discussion but originally I was ad=
dressing the following point:<br><br></div><div class=3D"gmail_quote">On 20=
14-04-23 14:31, Harald Alvestrand wrote:<br>&gt; =C2=A0All UDP endpoints th=
at use the open Internet have some form of backoff on congestion - using de=
lay, packet drop or<br>
&gt; (most likely) both to detect congestion <div><br></div><div>From my ex=
perience this is not true for almost all voice end points and it is not tru=
e for majority of video end points. So, Harald&#39;s assumption does not ho=
ld.</div>
<div><br></div><div>I think based on this assumption Harald was trying to d=
ecide if QOS support needs to be exposed in WebRTC. My second point was tha=
t if congestion is introduced by something other then the real time media c=
lient (such as loading a large document over HTTP from public internet or u=
sing a separate screen sharing program which uses TCP to upload the screen =
on a slide change), there is little that can be done by the real time media=
 end point to deal with this congestion except implement a good packet loss=
 recovery policy. Better handling of such congestion will require either ch=
anges to router queuing policies (using PIE or Codel queuing policies at th=
e point of congestion can help a lot) or better yet implementing QOS. Such =
router policy changes are outside of this groups charter, but exposing QOS =
support to WebRTC is definitely something that should be discussed.</div>
<div><div>_____________<br>Roman Shpount</div></div><div><br></div></div></=
div></div>

--001a11346f6a197c9004f84a87b0--


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From: "DRUTA, DAN" <dd5826@att.com>
To: "Cullen Jennings (fluffy)" <fluffy@cisco.com>, Matthew Kaufman <matthew.kaufman@skype.net>
Thread-Topic: [rtcweb] Prioritization
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Subject: Re: [rtcweb] Prioritization
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We discussed this a while back and to a certain extent is a split problem b=
etween the API and the protocol.

If we look from developer's point of view, they are the ones that know with=
in their app what's more important.
For a game data channel might be more important than video where for a cust=
omer service app audio might be more important than video (could be the opp=
osite if it's a furniture store and showing how to assemble a piece of furn=
iture is far more important).
The point is that it starts with the API and hopefully we will get somethin=
g that will give the browser a hint on the track priority.
This indication should be used for two purposes:
	1. Plug in a common congestion control between media and data within the b=
rowser implementation, try to honor this application demand and potentially=
 balance it with other web applications competing for resources and even ot=
her applications on the same device.
	2. Allow the browser to mark the traffic accordingly so with additional ne=
twork support it can be prioritized where possible e2e. It might require an=
 out of band negotiation between the application and the network in order t=
o request and honor the fulfilment of this kind of prioritization where pos=
sible.
The relative priority within an app is critical and I'm sure no app develop=
er would mark all of their tracks as high knowing that it would not do any =
good.
Ideally we would have an algorithm implemented consistently across the boar=
d within browsers so developers would know what to expect but the algorithm=
 must take input from the app as a factor into its decision.

Dan

-----Original Message-----
From: rtcweb [mailto:rtcweb-bounces@ietf.org] On Behalf Of Cullen Jennings =
(fluffy)
Sent: Wednesday, April 30, 2014 2:09 PM
To: Matthew Kaufman
Cc: rtcweb@ietf.org
Subject: Re: [rtcweb] Prioritization


On Apr 25, 2014, at 11:03 AM, Matthew Kaufman (SKYPE) <matthew.kaufman@skyp=
e.net> wrote:

> If a "lower priority" packet is dispatched before a "higher priority" pac=
ket in order to "prevent starvation", then what does "higher priority" mean=
?

I think the labels reflect what "might" happen on average and not for any p=
articular packet. I say "might" because in some deployments it possible the=
 values will have no effect at all and it is possible to construct bizarre =
environments where you could get the opposite of the desired behavior in so=
me cases. None of that bothers me as the label still seem like the closest =
to helping developers request the desired treatment. With all things qos li=
ke, the best we can for is desired outcome and not what will necessarily ha=
ppen on internet.=20

priority might be better called "desired priority" and the values of "very =
low, low, med, high" might better be called "broccoli, apple, orange, cherr=
y" - they are really just some labels that describe how the packets *might*=
 be queued by systems they pass through.  I think the fruit labels would be=
 more confusing to web programmers.=20


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rtcweb@ietf.org
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