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From: Alissa Cooper <alissa@cooperw.in>
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Date: Fri, 1 Apr 2016 20:13:48 -0500
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To: Ted Hardie <ted.ietf@gmail.com>
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Subject: Re: [rtcweb] I-D Action: draft-ietf-rtcweb-ip-handling-01.txt
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> On Mar 22, 2016, at 6:10 PM, Ted Hardie <ted.ietf@gmail.com> wrote:
>=20
> Speaking as an individual, rather than chair:
>=20
> On Tue, Mar 22, 2016 at 3:31 AM, Drage, Keith (Nokia - GB) =
<keith.drage@nokia.com <mailto:keith.drage@nokia.com>> wrote:
> The header of this document says "standards track".
>=20
> The title of the document says "recommendations" which would equate to =
a normative document of SHOULD strength
>=20
> The abstract says "best practices"
>=20
> Throughout the document I detected lots of lower case usage of RFC =
2119 terms but did not see any upper case usage (but may have missed =
it).
>=20
> All of which leaves me confused.
>=20
> Would the chairs like to clarify the intent of this document in terms =
of document type?
>=20
> Regards
>=20
> Keith Drage
>=20
>=20
> I don't think this document describes any new protocol mechanisms, so =
the lack of upper case RFC 2119 terms seems to me personally fairly =
natural.  What it does instead is lay out the results of doing IP =
address handling in specific modes, along with a recommendation on which =
would make a reasonable default.=20
>=20
> Clearly it could have been part of JSEP, which would have made it part =
of a standards track doc.  The value to me of it being separate is that =
the advice can be revised independently of the core protocol =
specification.  To that extent, it is somewhere between "Applicability =
statement" and "BCP" in our formal terminology, with a thumb very =
slightly on the scale toward BCP. Ultimately, though, I am willing to be =
guided by our ADs or the community about the exact status.=20

I could see an argument for either standards track or BCP and don=E2=80=99=
t have a strong feeling either way. In either case some adjustments =
would be necessary to make clear the normativity or non-normativity of =
the guidance.

Alissa

> The high order bits are the advice and its decomposability.
>=20
> regards,
>=20
> Ted=20
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb


--Apple-Mail=_9D75D0A5-63F8-4B1F-B2A8-BB0C3975BF7C
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<html><head><meta http-equiv=3D"Content-Type" content=3D"text/html =
charset=3Dutf-8"></head><body style=3D"word-wrap: break-word; =
-webkit-nbsp-mode: space; -webkit-line-break: after-white-space;" =
class=3D""><br class=3D""><div><blockquote type=3D"cite" class=3D""><div =
class=3D"">On Mar 22, 2016, at 6:10 PM, Ted Hardie &lt;<a =
href=3D"mailto:ted.ietf@gmail.com" class=3D"">ted.ietf@gmail.com</a>&gt; =
wrote:</div><br class=3D"Apple-interchange-newline"><div class=3D""><div =
dir=3D"ltr" class=3D"">Speaking as an individual, rather than chair:<br =
class=3D""><div class=3D"gmail_extra"><br class=3D""><div =
class=3D"gmail_quote">On Tue, Mar 22, 2016 at 3:31 AM, Drage, Keith =
(Nokia - GB) <span dir=3D"ltr" class=3D"">&lt;<a =
href=3D"mailto:keith.drage@nokia.com" target=3D"_blank" =
class=3D"">keith.drage@nokia.com</a>&gt;</span> wrote:<br =
class=3D""><blockquote class=3D"gmail_quote" style=3D"margin:0px 0px 0px =
0.8ex;border-left:1px solid rgb(204,204,204);padding-left:1ex">The =
header of this document says "standards track".<br class=3D"">
<br class=3D"">
The title of the document says "recommendations" which would equate to a =
normative document of SHOULD strength<br class=3D"">
<br class=3D"">
The abstract says "best practices"<br class=3D"">
<br class=3D"">
Throughout the document I detected lots of lower case usage of RFC 2119 =
terms but did not see any upper case usage (but may have missed it).<br =
class=3D"">
<br class=3D"">
All of which leaves me confused.<br class=3D"">
<br class=3D"">
Would the chairs like to clarify the intent of this document in terms of =
document type?<br class=3D"">
<br class=3D"">
Regards<br class=3D"">
<span class=3D""><font color=3D"#888888" class=3D""><br class=3D"">
Keith Drage<br class=3D"">
</font></span><div class=3D""><div class=3D"h5"><br =
class=3D""></div></div></blockquote></div><br class=3D""></div><div =
class=3D"gmail_extra">I don't think this document describes any new =
protocol mechanisms, so the lack of upper case RFC 2119 terms seems to =
me personally fairly natural.&nbsp; What it does instead is lay out the =
results of doing IP address handling in specific modes, along with a =
recommendation on which would make a reasonable default. <br =
class=3D""><br class=3D"">Clearly it could have been part of JSEP, which =
would have made it part of a standards track doc.&nbsp; The value to me =
of it being separate is that the advice can be revised independently of =
the core protocol specification.&nbsp;  To that extent, it is somewhere =
between "Applicability statement" and "BCP" in our formal terminology, =
with a thumb very slightly on the scale toward BCP. Ultimately, though, =
I am willing to be guided by our ADs or the community about the exact =
status.&nbsp; </div></div></div></blockquote><div><br =
class=3D""></div><div>I could see an argument for either standards track =
or BCP and don=E2=80=99t have a strong feeling either way. In either =
case some adjustments would be necessary to make clear the normativity =
or non-normativity of the guidance.</div><div><br =
class=3D""></div><div>Alissa</div><br class=3D""><blockquote type=3D"cite"=
 class=3D""><div class=3D""><div dir=3D"ltr" class=3D""><div =
class=3D"gmail_extra">The high order bits are the advice and its =
decomposability.<br class=3D""><br class=3D""></div><div =
class=3D"gmail_extra">regards,<br class=3D""><br class=3D""></div><div =
class=3D"gmail_extra">Ted <br class=3D""></div></div>
_______________________________________________<br class=3D"">rtcweb =
mailing list<br class=3D""><a href=3D"mailto:rtcweb@ietf.org" =
class=3D"">rtcweb@ietf.org</a><br =
class=3D"">https://www.ietf.org/mailman/listinfo/rtcweb<br =
class=3D""></div></blockquote></div><br class=3D""></body></html>=

--Apple-Mail=_9D75D0A5-63F8-4B1F-B2A8-BB0C3975BF7C--


From nobody Sat Apr  2 15:52:11 2016
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To: "rtcweb@ietf.org" <rtcweb@ietf.org>
From: Harald Alvestrand <harald@alvestrand.no>
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Date: Sun, 3 Apr 2016 00:51:59 +0200
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Subject: [rtcweb] Positioning draft-ietf-rtcweb-ip-address as a standards-track document
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I've read through draft-ietf-rtcweb-ip-address again, with an eye to
seeing how it would read as a standards-track document.

As such, it seems to me to provide rather tentative language in a few
places, which could be strengthened by use of the CAPITALIZED WORDS.
I think this is a Good Thing; if this is standards-track, with
imperative, normative language, an implementation can claim support for
RFC XXXX and expect the statement to be understood.
(Remember - there is no protocol police. Nothing forces people to
support RFC XXXX.)

Changes needed:

- Add the RFC 2119 incantation, including Stefan's addition ("lowercase
words mean what they usually do")

- Replace all occurences of "We recommend that..." with "an
implementation MUST".

- In section 3, end of first paragraph: "   Specifically, WebRTC
should:" -> "Implementations of this specification will:"

- In section 4: "We recommend Mode 1 ... " -> "Mode 1 and mode 2 MUST be
supported. Mode 1 SHOULD be the default when the user has not granted
access to camera / microphone. Mode 2 SHOULD be the default when the
user has granted access to camera / microphone. The implementation MAY
provide means of letting the user or administrator select between modes.
These means MUST NOT be placed under the control of WebRTC applications."=


I think this would make the spec a more useful tool for specifying what
an application does.



--=20
Surveillance is pervasive. Go Dark.



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References: <20160320223116.8946.76840.idtracker@ietfa.amsl.com> <949EF20990823C4C85C18D59AA11AD8BADEAFFC7@FR712WXCHMBA11.zeu.alcatel-lucent.com> <CA+9kkMCsT43ZCSdq8gdKXu1k4pJgbf0ab5tE=dDiFfrTT2gtkA@mail.gmail.com> <949EF20990823C4C85C18D59AA11AD8BADEB0D16@FR712WXCHMBA11.zeu.alcatel-lucent.com> <56F79D05.8070004@alvestrand.no> <326E6502-28E5-4D09-BB99-4A5D80625EB0@stewe.org> <56F88E18.2060506@it.aoyama.ac.jp> <20160328104731.GO88304@verdi> <CALaySJ+hYMMsKE7Ws-NJbyqH55E-mQM-duTEcJGc0TWvTP88Ew@mail.gmail.com> <20160328132859.GP88304@verdi> <28975138-9EA1-4A9F-A6C0-BC1416B8EA44@sobco.com> <CALaySJJkNj2jfm0gJpuDzq8oFDjTNn-uQ5MHdmEOLwTiFZUyQQ@mail.gmail.com> <8975F15F-5C4C-4D02-98CD-BF4FDF104D35@sobco.com> <56F98CD1.10706@gmail.com>
Date: Sun, 3 Apr 2016 15:18:57 +0200
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From: Abdussalam Baryun <abdussalambaryun@gmail.com>
To: Brian E Carpenter <brian.e.carpenter@gmail.com>
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Cc: IETF discussion list <ietf@ietf.org>, "Heather Flanagan \(RFC Series Editor\)" <rse@rfc-editor.org>, "rtcweb@ietf.org" <rtcweb@ietf.org>, IESG <iesg@ietf.org>, Barry Leiba <barryleiba@computer.org>, "Scott O. Bradner" <sob@sobco.com>
Subject: Re: [rtcweb] Fuzzy words [was Uppercase question for RFC2119 words]
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The confusion if occured between English authors may mean that we can get
confusions among non-English authors as well, I recommend if a guidance
draft is written we will need many non-Enlish native speekers involved.


AB

On Mon, Mar 28, 2016 at 9:58 PM, Brian E Carpenter <
brian.e.carpenter@gmail.com> wrote:

> There are times when I think RFC2119 was a really bad idea, despite it
> having
> become probably the most frequently cited RFC (inside and outside the
> IETF).
> It seems to create as much confusion as it avoids.
>
> There are four words whose RFC2119 meaning is different from the dictiona=
ry
> meaning: should, recommended, may and optional. Having special typography
> for them is useful, because it signals the RFC2119 meanings. But if a spe=
c
> uses, for example, a mixture of SHOULD and should, who knows what the
> authors
> intended? To that extent, the proposed clarification is helpful.
>
> The other words (must, shall, required, not) mean what they always mean.
> The only argument for upper-casing them is aesthetic symmetry. If a spec
> uses alternatives like mandatory, necessary or forbidden, they are just a=
s
> powerful.
>
> So
> > these definitions are only meaningful if the words are capitalized
> can be applied to should, recommended, may and optional if we want,
> but strictly doesn't apply to must, shall, required, not, mandatory,
> necessary, forbidden, need, or any other such words.
>
> Where we can get into real trouble is if a spec contains should,
> recommended,
> may and optional *plus* other non-categorical (fuzzy) words like ought,
> encourage, suggest, can, might, allowed, permit (and I did not pull those
> words out of the air, but out of draft-hansen-nonkeywords-non2119). What =
do
> they mean? It can be very unclear. If a node receives a message containin=
g
> an element covered in the spec by "allowed" instead of "OPTIONAL", is the
> receiver supposed to interoperate or to reject the message?
>
> If we are issuing guidance, it should probably include a specific warning
> to use any such fuzzy words with extreme care.
>
>    Brian
> On 29/03/2016 03:13, Scott O. Bradner wrote:
> > one minor tweak
> >
> >> On Mar 28, 2016, at 10:09 AM, Barry Leiba <barryleiba@computer.org>
> wrote:
> >>
> >>> The wishy washy descriptive rather than proscriptive language in the
> abstract was because I,
> >>> the IESG and the community were not of one mind to say that the use o=
f
> such capitalized
> >>> terms should be mandatory - quite a few people felt that the english
> language was at
> >>> least good enough to convey  the writer=E2=80=99s intent without havi=
ng to
> aggrandize specific words.
> >>> Thus the abstract basically was saying: if you want to use capitalize=
d
> words here is a standard
> >>> way to say what they mean
> >>
> >> Ah.  Then perhaps the clarification needs to go a little further and
> >> make this clear:
> >> - We're defining specific terms that specifications can use.
> >> - These terms are always capitalized when these definitions are used.
> >
> > these definitions are only meaningful if the words are capitalized
> >
> >> - You don't have to use them.  If you do, they're capitalized and
> >> their meanings are as specified here.
> >> - There are similar-looking English words that are not capitalized,
> >> and they have their normal English meanings; this document has nothing
> >> to do with them.
> >>
> >> ...and I'd like to add one more, because so many people think that
> >> text isn't normative unless it has 2119 key words in all caps in it:
> >>
> >> - Normative text doesn't require the use of these key words.  They're
> >> used for clarity and consistency when you want that, but lots of
> >> normative text doesn't need to use them, and doesn't use them.
> >>
> >> Barry
> >
> >
>
>

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<div dir=3D"ltr"><div>The confusion if occured between English authors may =
mean that we can get confusions among non-English authors as well, I recomm=
end if a guidance draft is written we will need many non-Enlish native spee=
kers involved. <br><br><br></div><div>AB</div></div><div class=3D"gmail_ext=
ra"><br><div class=3D"gmail_quote">On Mon, Mar 28, 2016 at 9:58 PM, Brian E=
 Carpenter <span dir=3D"ltr">&lt;<a href=3D"mailto:brian.e.carpenter@gmail.=
com" target=3D"_blank">brian.e.carpenter@gmail.com</a>&gt;</span> wrote:<br=
><blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1=
px #ccc solid;padding-left:1ex">There are times when I think RFC2119 was a =
really bad idea, despite it having<br>
become probably the most frequently cited RFC (inside and outside the IETF)=
.<br>
It seems to create as much confusion as it avoids.<br>
<br>
There are four words whose RFC2119 meaning is different from the dictionary=
<br>
meaning: should, recommended, may and optional. Having special typography<b=
r>
for them is useful, because it signals the RFC2119 meanings. But if a spec<=
br>
uses, for example, a mixture of SHOULD and should, who knows what the autho=
rs<br>
intended? To that extent, the proposed clarification is helpful.<br>
<br>
The other words (must, shall, required, not) mean what they always mean.<br=
>
The only argument for upper-casing them is aesthetic symmetry. If a spec<br=
>
uses alternatives like mandatory, necessary or forbidden, they are just as<=
br>
powerful.<br>
<br>
So<br>
&gt; these definitions are only meaningful if the words are capitalized<br>
can be applied to should, recommended, may and optional if we want,<br>
but strictly doesn&#39;t apply to must, shall, required, not, mandatory,<br=
>
necessary, forbidden, need, or any other such words.<br>
<br>
Where we can get into real trouble is if a spec contains should, recommende=
d,<br>
may and optional *plus* other non-categorical (fuzzy) words like ought,<br>
encourage, suggest, can, might, allowed, permit (and I did not pull those<b=
r>
words out of the air, but out of draft-hansen-nonkeywords-non2119). What do=
<br>
they mean? It can be very unclear. If a node receives a message containing<=
br>
an element covered in the spec by &quot;allowed&quot; instead of &quot;OPTI=
ONAL&quot;, is the<br>
receiver supposed to interoperate or to reject the message?<br>
<br>
If we are issuing guidance, it should probably include a specific warning<b=
r>
to use any such fuzzy words with extreme care.<br>
<br>
=C2=A0 =C2=A0Brian<br>
On 29/03/2016 03:13, Scott O. Bradner wrote:<br>
&gt; one minor tweak<br>
&gt;<br>
&gt;&gt; On Mar 28, 2016, at 10:09 AM, Barry Leiba &lt;<a href=3D"mailto:ba=
rryleiba@computer.org">barryleiba@computer.org</a>&gt; wrote:<br>
&gt;&gt;<br>
&gt;&gt;&gt; The wishy washy descriptive rather than proscriptive language =
in the abstract was because I,<br>
&gt;&gt;&gt; the IESG and the community were not of one mind to say that th=
e use of such capitalized<br>
&gt;&gt;&gt; terms should be mandatory - quite a few people felt that the e=
nglish language was at<br>
&gt;&gt;&gt; least good enough to convey=C2=A0 the writer=E2=80=99s intent =
without having to aggrandize specific words.<br>
&gt;&gt;&gt; Thus the abstract basically was saying: if you want to use cap=
italized words here is a standard<br>
&gt;&gt;&gt; way to say what they mean<br>
&gt;&gt;<br>
&gt;&gt; Ah.=C2=A0 Then perhaps the clarification needs to go a little furt=
her and<br>
&gt;&gt; make this clear:<br>
&gt;&gt; - We&#39;re defining specific terms that specifications can use.<b=
r>
&gt;&gt; - These terms are always capitalized when these definitions are us=
ed.<br>
&gt;<br>
&gt; these definitions are only meaningful if the words are capitalized<br>
&gt;<br>
&gt;&gt; - You don&#39;t have to use them.=C2=A0 If you do, they&#39;re cap=
italized and<br>
&gt;&gt; their meanings are as specified here.<br>
&gt;&gt; - There are similar-looking English words that are not capitalized=
,<br>
&gt;&gt; and they have their normal English meanings; this document has not=
hing<br>
&gt;&gt; to do with them.<br>
&gt;&gt;<br>
&gt;&gt; ...and I&#39;d like to add one more, because so many people think =
that<br>
&gt;&gt; text isn&#39;t normative unless it has 2119 key words in all caps =
in it:<br>
&gt;&gt;<br>
&gt;&gt; - Normative text doesn&#39;t require the use of these key words.=
=C2=A0 They&#39;re<br>
&gt;&gt; used for clarity and consistency when you want that, but lots of<b=
r>
&gt;&gt; normative text doesn&#39;t need to use them, and doesn&#39;t use t=
hem.<br>
&gt;&gt;<br>
&gt;&gt; Barry<br>
&gt;<br>
&gt;<br>
<br>
</blockquote></div><br></div>

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From: Eric Rescorla <ekr@rtfm.com>
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Subject: [rtcweb] Tunnelling DTLS in SDP
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Hi folks,

I wanted to call your attention to a draft I just published with a possibly
stupid
idea.

https://tools.ietf.org/html/draft-rescorla-dtls-in-sdp-00

A nontrivial fraction of call setup time in WebRTC is the DTLS handshake.
This document describes how to piggyback the first few handshake messages
in the SDP offer/answer exchange, thus reducing latency.

Comments welcome.

-Ekr

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<div dir=3D"ltr">Hi folks,<div><br></div><div>I wanted to call your attenti=
on to a draft I just published with a possibly stupid</div><div>idea.</div>=
<div><br></div><div><a href=3D"https://tools.ietf.org/html/draft-rescorla-d=
tls-in-sdp-00">https://tools.ietf.org/html/draft-rescorla-dtls-in-sdp-00</a=
><br></div><div><br></div><div>A nontrivial fraction of call setup time in =
WebRTC is the DTLS handshake.</div><div>This document describes how to pigg=
yback the first few handshake messages</div><div>in the SDP offer/answer ex=
change, thus reducing latency.</div><div><br></div><div>Comments welcome.</=
div><div><br></div><div>-Ekr</div><div><br></div><div><br></div><div><br></=
div></div>

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Subject: Re: [rtcweb] Tunnelling DTLS in SDP
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> On 4 Apr 2016, at 14:10, Eric Rescorla <ekr@rtfm.com> wrote:
>=20
> Hi folks,
>=20
> I wanted to call your attention to a draft I just published with a =
possibly stupid
> idea.
>=20
> https://tools.ietf.org/html/draft-rescorla-dtls-in-sdp-00 =
<https://tools.ietf.org/html/draft-rescorla-dtls-in-sdp-00>
>=20
> A nontrivial fraction of call setup time in WebRTC is the DTLS =
handshake.
> This document describes how to piggyback the first few handshake =
messages
> in the SDP offer/answer exchange, thus reducing latency.

It strikes me we could get the same reduced latency benefits by =
piggybacking on ICE
rather than SDP, e.g. embedding the DTLS packet as data in a new STUN =
attribute type.

The downside of piggybacking on the SDP is that you are increasing the =
trust you have to place in the
signalling server undermining the elegant decoupling we have at the =
moment between signalling and=20
media. (The SDES issues of logging keys in the web servers apply to the =
public certs as well).

It also significantly clutters the SDP (even more) !

As you point out, it weakens the usefulness of longer term certs, which
would be a major nuisance IMHO.

Tim.


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<html><head><meta http-equiv=3D"Content-Type" content=3D"text/html =
charset=3Dus-ascii"></head><body style=3D"word-wrap: break-word; =
-webkit-nbsp-mode: space; -webkit-line-break: after-white-space;" =
class=3D""><br class=3D""><div><blockquote type=3D"cite" class=3D""><div =
class=3D"">On 4 Apr 2016, at 14:10, Eric Rescorla &lt;<a =
href=3D"mailto:ekr@rtfm.com" class=3D"">ekr@rtfm.com</a>&gt; =
wrote:</div><br class=3D"Apple-interchange-newline"><div class=3D""><div =
dir=3D"ltr" class=3D"">Hi folks,<div class=3D""><br class=3D""></div><div =
class=3D"">I wanted to call your attention to a draft I just published =
with a possibly stupid</div><div class=3D"">idea.</div><div class=3D""><br=
 class=3D""></div><div class=3D""><a =
href=3D"https://tools.ietf.org/html/draft-rescorla-dtls-in-sdp-00" =
class=3D"">https://tools.ietf.org/html/draft-rescorla-dtls-in-sdp-00</a><b=
r class=3D""></div><div class=3D""><br class=3D""></div><div class=3D"">A =
nontrivial fraction of call setup time in WebRTC is the DTLS =
handshake.</div><div class=3D"">This document describes how to piggyback =
the first few handshake messages</div><div class=3D"">in the SDP =
offer/answer exchange, thus reducing =
latency.</div></div></div></blockquote><div><br class=3D""></div><div>It =
strikes me we could get the same reduced latency benefits by =
piggybacking on ICE</div><div>rather than SDP, e.g. embedding the DTLS =
packet as data in a new STUN attribute type.</div><div><br =
class=3D""></div><div>The downside of piggybacking on the SDP is that =
you are increasing the trust you have to place in =
the</div><div>signalling server undermining the elegant decoupling we =
have at the moment between signalling and&nbsp;</div><div>media. (The =
SDES issues of logging keys in the web servers apply to the public certs =
as well).</div><div><br class=3D""></div><div>It also significantly =
clutters the SDP (even more) !</div><div><br class=3D""></div><div>As =
you point out, it weakens the usefulness of longer term certs, =
which</div><div>would be a major nuisance IMHO.</div><div><br =
class=3D""></div><div>Tim.</div></div><br class=3D""></body></html>=

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From: Roman Shpount <roman@telurix.com>
To: Eric Rescorla <ekr@rtfm.com>
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Subject: Re: [rtcweb] Tunnelling DTLS in SDP
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The proposal makes sense to me.

1. Why do you need to send both fingerprints and certificates in answer? If
certificate is sent, fingerprint can be calculated from it and will be
completely redundant.

2. Can you show where and how the DTLS-SRTP keys are going to be sent? I
assume they will be transmitted together with ChangeCipherSpec.

I think sending certificate in the answer instead of data path does not
compromise security since if signaling path is compromised, data traffic
can be redirected to MITM agent which can collect exactly the same data.

This has the benefit over sending DTLS handshake using ICE/STUN messages
that it does not need to deal with packet fragmentation and reassembly.

Regards,
_____________
Roman Shpount

On Mon, Apr 4, 2016 at 9:10 AM, Eric Rescorla <ekr@rtfm.com> wrote:

> Hi folks,
>
> I wanted to call your attention to a draft I just published with a
> possibly stupid
> idea.
>
> https://tools.ietf.org/html/draft-rescorla-dtls-in-sdp-00
>
> A nontrivial fraction of call setup time in WebRTC is the DTLS handshake.
> This document describes how to piggyback the first few handshake messages
> in the SDP offer/answer exchange, thus reducing latency.
>
> Comments welcome.
>
> -Ekr
>
>
>
>
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb
>
>

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<div dir=3D"ltr"><div>The proposal makes sense to me.</div><div><br></div>1=
. Why do you need to send both fingerprints and certificates in answer? If =
certificate is sent, fingerprint can be calculated from it and will be comp=
letely redundant.<div><br></div><div>2. Can you show where and how the DTLS=
-SRTP keys are going to be sent? I assume they will be transmitted together=
 with=C2=A0<span style=3D"color:rgb(0,0,0);font-size:13.3333px">ChangeCiphe=
rSpec.</span></div><div><font color=3D"#000000"><span style=3D"font-size:13=
.3333px"><br></span></font></div><div><font color=3D"#000000"><span style=
=3D"font-size:13.3333px">I think sending certificate in the answer instead =
of data path does not compromise security since if signaling path is compro=
mised, data traffic can be redirected to MITM agent which can collect exact=
ly the same data.</span></font></div><div><font color=3D"#000000"><span sty=
le=3D"font-size:13.3333px"><br></span></font></div><div><font color=3D"#000=
000"><span style=3D"font-size:13.3333px">This has the benefit over sending =
DTLS handshake using ICE/STUN messages that it does not need to deal with p=
acket fragmentation and reassembly.</span></font></div><div><font color=3D"=
#000000"><span style=3D"font-size:13.3333px"><br></span></font></div><div><=
font color=3D"#000000"><span style=3D"font-size:13.3333px">Regards,</span><=
/font><div class=3D"gmail_extra"><div><div class=3D"gmail_signature">______=
_______<br>Roman Shpount</div></div>
<br><div class=3D"gmail_quote">On Mon, Apr 4, 2016 at 9:10 AM, Eric Rescorl=
a <span dir=3D"ltr">&lt;<a href=3D"mailto:ekr@rtfm.com" target=3D"_blank">e=
kr@rtfm.com</a>&gt;</span> wrote:<br><blockquote class=3D"gmail_quote" styl=
e=3D"margin:0px 0px 0px 0.8ex;border-left-width:1px;border-left-style:solid=
;border-left-color:rgb(204,204,204);padding-left:1ex"><div dir=3D"ltr">Hi f=
olks,<div><br></div><div>I wanted to call your attention to a draft I just =
published with a possibly stupid</div><div>idea.</div><div><br></div><div><=
a href=3D"https://tools.ietf.org/html/draft-rescorla-dtls-in-sdp-00" target=
=3D"_blank">https://tools.ietf.org/html/draft-rescorla-dtls-in-sdp-00</a><b=
r></div><div><br></div><div>A nontrivial fraction of call setup time in Web=
RTC is the DTLS handshake.</div><div>This document describes how to piggyba=
ck the first few handshake messages</div><div>in the SDP offer/answer excha=
nge, thus reducing latency.</div><div><br></div><div>Comments welcome.</div=
><div><br></div><div>-Ekr</div><div><br></div><div><br></div><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" rel=3D"noreferrer"=
 target=3D"_blank">https://www.ietf.org/mailman/listinfo/rtcweb</a><br>
<br></blockquote></div><br></div></div></div>

--001a1140d71e6f72a6052faab865--


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Subject: Re: [rtcweb] Tunnelling DTLS in SDP
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> On 4 Apr 2016, at 16:52, Roman Shpount <roman@telurix.com> wrote:
>=20
>=20
> I think sending certificate in the answer instead of data path does =
not compromise security since if signaling path is compromised, data =
traffic can be redirected to MITM agent which can collect exactly the =
same data.

That isn=E2=80=99t quite the same, currently in a MITM=E2=80=99d =
scenario you have the option of not completing the DTLS handshake =
(assuming you had a way
to verify the fingerprints.). With this change you=E2=80=99ll have all =
the certs logged in a web server somewhere.

>=20
> This has the benefit over sending DTLS handshake using ICE/STUN =
messages that it does not need to deal with packet fragmentation and =
reassembly.

DTLS does have support for packet assembly/disassembly. I=E2=80=99d be =
surprised if the DTLS handshake packets exceeded a reasonable MTU,
unless you are sending a full chain of certificates or stuffing a logo =
into the x509 :-)=20

Stepping back from the detail, (D)TLS implementations are brittle enough =
as it is, we should avoid introducing a whole new level
of complexity if we can. Implementing this piggybacking in ICE at least =
keeps it in the media path, rather than spreading it through the
signalling and web servers too.

T.


>=20
> Regards,
> _____________
> Roman Shpount
>=20
> On Mon, Apr 4, 2016 at 9:10 AM, Eric Rescorla <ekr@rtfm.com =
<mailto:ekr@rtfm.com>> wrote:
> Hi folks,
>=20
> I wanted to call your attention to a draft I just published with a =
possibly stupid
> idea.
>=20
> https://tools.ietf.org/html/draft-rescorla-dtls-in-sdp-00 =
<https://tools.ietf.org/html/draft-rescorla-dtls-in-sdp-00>
>=20
> A nontrivial fraction of call setup time in WebRTC is the DTLS =
handshake.
> This document describes how to piggyback the first few handshake =
messages
> in the SDP offer/answer exchange, thus reducing latency.
>=20
> Comments welcome.
>=20
> -Ekr
>=20
>=20
>=20
>=20
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org <mailto:rtcweb@ietf.org>
> https://www.ietf.org/mailman/listinfo/rtcweb =
<https://www.ietf.org/mailman/listinfo/rtcweb>
>=20
>=20
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb


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<html><head><meta http-equiv=3D"Content-Type" content=3D"text/html =
charset=3Dutf-8"></head><body style=3D"word-wrap: break-word; =
-webkit-nbsp-mode: space; -webkit-line-break: after-white-space;" =
class=3D""><br class=3D""><div><blockquote type=3D"cite" class=3D""><div =
class=3D"">On 4 Apr 2016, at 16:52, Roman Shpount &lt;<a =
href=3D"mailto:roman@telurix.com" class=3D"">roman@telurix.com</a>&gt; =
wrote:</div><br class=3D"Apple-interchange-newline"><div class=3D""><div =
dir=3D"ltr" class=3D""><div class=3D""><br class=3D""></div><div =
class=3D""><font class=3D""><span style=3D"font-size:13.3333px" =
class=3D"">I think sending certificate in the answer instead of data =
path does not compromise security since if signaling path is =
compromised, data traffic can be redirected to MITM agent which can =
collect exactly the same =
data.</span></font></div></div></div></blockquote><div><br =
class=3D""></div><div>That isn=E2=80=99t quite the same, currently in a =
MITM=E2=80=99d scenario you have the option of not completing the DTLS =
handshake (assuming you had a way</div><div>to verify the =
fingerprints.). With this change you=E2=80=99ll have all the certs =
logged in a web server somewhere.</div><br class=3D""><blockquote =
type=3D"cite" class=3D""><div class=3D""><div dir=3D"ltr" class=3D""><div =
class=3D""><font class=3D""><span style=3D"font-size:13.3333px" =
class=3D""><br class=3D""></span></font></div><div class=3D""><font =
class=3D""><span style=3D"font-size:13.3333px" class=3D"">This has the =
benefit over sending DTLS handshake using ICE/STUN messages that it does =
not need to deal with packet fragmentation and =
reassembly.</span></font></div></div></div></blockquote><div><br =
class=3D""></div><div>DTLS does have support for packet =
assembly/disassembly. I=E2=80=99d be surprised if the DTLS handshake =
packets exceeded a reasonable MTU,</div><div>unless you are sending a =
full chain of certificates or stuffing a logo into the x509 =
:-)&nbsp;</div><div><br class=3D""></div><div>Stepping back from the =
detail, (D)TLS implementations are brittle enough as it is, we should =
avoid introducing a whole new level</div><div>of complexity if we can. =
Implementing this piggybacking in ICE at least keeps it in the media =
path, rather than spreading it through the</div><div>signalling and web =
servers too.</div><div><br class=3D""></div><div>T.</div><div><br =
class=3D""></div><br class=3D""><blockquote type=3D"cite" class=3D""><div =
class=3D""><div dir=3D"ltr" class=3D""><div class=3D""><font =
class=3D""><span style=3D"font-size:13.3333px" class=3D""><br =
class=3D""></span></font></div><div class=3D""><font class=3D""><span =
style=3D"font-size:13.3333px" class=3D"">Regards,</span></font><div =
class=3D"gmail_extra"><div class=3D""><div =
class=3D"gmail_signature">_____________<br class=3D"">Roman =
Shpount</div></div>
<br class=3D""><div class=3D"gmail_quote">On Mon, Apr 4, 2016 at 9:10 =
AM, Eric Rescorla <span dir=3D"ltr" class=3D"">&lt;<a =
href=3D"mailto:ekr@rtfm.com" target=3D"_blank" =
class=3D"">ekr@rtfm.com</a>&gt;</span> wrote:<br class=3D""><blockquote =
class=3D"gmail_quote" style=3D"margin:0px 0px 0px =
0.8ex;border-left-width:1px;border-left-style:solid;border-left-color:rgb(=
204,204,204);padding-left:1ex"><div dir=3D"ltr" class=3D"">Hi folks,<div =
class=3D""><br class=3D""></div><div class=3D"">I wanted to call your =
attention to a draft I just published with a possibly stupid</div><div =
class=3D"">idea.</div><div class=3D""><br class=3D""></div><div =
class=3D""><a =
href=3D"https://tools.ietf.org/html/draft-rescorla-dtls-in-sdp-00" =
target=3D"_blank" =
class=3D"">https://tools.ietf.org/html/draft-rescorla-dtls-in-sdp-00</a><b=
r class=3D""></div><div class=3D""><br class=3D""></div><div class=3D"">A =
nontrivial fraction of call setup time in WebRTC is the DTLS =
handshake.</div><div class=3D"">This document describes how to piggyback =
the first few handshake messages</div><div class=3D"">in the SDP =
offer/answer exchange, thus reducing latency.</div><div class=3D""><br =
class=3D""></div><div class=3D"">Comments welcome.</div><div =
class=3D""><br class=3D""></div><div class=3D"">-Ekr</div><div =
class=3D""><br class=3D""></div><div class=3D""><br class=3D""></div><div =
class=3D""><br class=3D""></div></div>
<br class=3D"">_______________________________________________<br =
class=3D"">
rtcweb mailing list<br class=3D"">
<a href=3D"mailto:rtcweb@ietf.org" class=3D"">rtcweb@ietf.org</a><br =
class=3D"">
<a href=3D"https://www.ietf.org/mailman/listinfo/rtcweb" =
rel=3D"noreferrer" target=3D"_blank" =
class=3D"">https://www.ietf.org/mailman/listinfo/rtcweb</a><br class=3D"">=

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

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From: Roman Shpount <roman@telurix.com>
To: pfh <tim@phonefromhere.com>
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Subject: Re: [rtcweb] Tunnelling DTLS in SDP
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On Mon, Apr 4, 2016 at 12:14 PM, pfh <tim@phonefromhere.com> wrote:

>
> On 4 Apr 2016, at 16:52, Roman Shpount <roman@telurix.com> wrote:
>
>
> I think sending certificate in the answer instead of data path does not
> compromise security since if signaling path is compromised, data traffic
> can be redirected to MITM agent which can collect exactly the same data.
>
>
> That isn=E2=80=99t quite the same, currently in a MITM=E2=80=99d scenario=
 you have the
> option of not completing the DTLS handshake (assuming you had a way
> to verify the fingerprints.). With this change you=E2=80=99ll have all th=
e certs
> logged in a web server somewhere.
>

In the scenario I had in mind MTIM will not modify fingerprints, it will
modify the transport addresses or put itself on the media path using a
compromised router so that it can log the DTLS handshake (and subsequent
media). This will provide exactly the same security risk as logging
signaling exchange when certificate is sent in the signaling path. In order
to compromise media, attacker needs access to the media path. Once it gets
access to the media path, it gets access to DTLS handshake and
certificates. If one of those certificates is sent over signaling channel,
it does not make the whole flow any less secure. Actually you might argue
it makes it more secure, since now attacker needs to compromise both media
and signaling paths in order to decode.
_____________
Roman Shpount

--001a1140d71eca2468052fab7bd4
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<div dir=3D"ltr"><div class=3D"gmail_extra"><div><div class=3D"gmail_signat=
ure">On Mon, Apr 4, 2016 at 12:14 PM, pfh <span dir=3D"ltr">&lt;<a href=3D"=
mailto:tim@phonefromhere.com" target=3D"_blank">tim@phonefromhere.com</a>&g=
t;</span> wrote:<br></div></div><div class=3D"gmail_quote"><blockquote clas=
s=3D"gmail_quote" style=3D"margin:0px 0px 0px 0.8ex;border-left-width:1px;b=
order-left-style:solid;border-left-color:rgb(204,204,204);padding-left:1ex"=
><div style=3D"word-wrap:break-word"><br><div><span class=3D""><blockquote =
type=3D"cite"><div>On 4 Apr 2016, at 16:52, Roman Shpount &lt;<a href=3D"ma=
ilto:roman@telurix.com" target=3D"_blank">roman@telurix.com</a>&gt; wrote:<=
/div><br><div><div dir=3D"ltr"><div><br></div><div><font><span style=3D"fon=
t-size:13.3333px">I think sending certificate in the answer instead of data=
 path does not compromise security since if signaling path is compromised, =
data traffic can be redirected to MITM agent which can collect exactly the =
same data.</span></font></div></div></div></blockquote><div><br></div></spa=
n><div>That isn=E2=80=99t quite the same, currently in a MITM=E2=80=99d sce=
nario you have the option of not completing the DTLS handshake (assuming yo=
u had a way</div><div>to verify the fingerprints.). With this change you=E2=
=80=99ll have all the certs logged in a web server somewhere.</div></div></=
div></blockquote></div><br></div><div class=3D"gmail_extra">In the scenario=
 I had in mind MTIM will not modify fingerprints, it will modify the transp=
ort addresses or put itself on the media path using a compromised router so=
 that it can log the DTLS handshake (and subsequent media). This will provi=
de exactly the same security risk as logging signaling exchange when certif=
icate is sent in the signaling path. In order to compromise media, attacker=
 needs access to the media path. Once it gets access to the media path, it =
gets access to DTLS handshake and certificates. If one of those certificate=
s is sent over signaling channel, it does not make the whole flow any less =
secure. Actually you might argue it makes it more secure, since now attacke=
r needs to compromise both media and signaling paths in order to decode.</d=
iv><div class=3D"gmail_extra"><div><div class=3D"gmail_signature">_________=
____<br>Roman Shpount</div></div><div><br></div></div></div>

--001a1140d71eca2468052fab7bd4--


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From: Martin Thomson <martin.thomson@gmail.com>
To: pfh <tim@phonefromhere.com>
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Subject: Re: [rtcweb] Tunnelling DTLS in SDP
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On 4 April 2016 at 12:39, pfh <tim@phonefromhere.com> wrote:
> It strikes me we could get the same reduced latency benefits by piggybacking
> on ICE
> rather than SDP, e.g. embedding the DTLS packet as data in a new STUN
> attribute type.

One of the goals for connectivity checks is to keep them small.
Cramming DTLS in is not compatible with this.


From nobody Mon Apr  4 10:35:00 2016
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From: Martin Thomson <martin.thomson@gmail.com>
To: pfh <tim@phonefromhere.com>
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Subject: Re: [rtcweb] Tunnelling DTLS in SDP
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On 4 April 2016 at 13:14, pfh <tim@phonefromhere.com> wrote:
> I=E2=80=99d be surprised if the DTLS handshake packets exceeded a reasona=
ble MTU,
> unless you are sending a full chain of certificates or stuffing a logo in=
to
> the x509 :-)


Fragmentation is common.


From nobody Mon Apr  4 11:02:40 2016
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From: Christer Holmberg <christer.holmberg@ericsson.com>
To: Eric Rescorla <ekr@rtfm.com>, mmusic WG <mmusic@ietf.org>, "rtcweb@ietf.org" <rtcweb@ietf.org>
Thread-Topic: [rtcweb] Tunnelling DTLS in SDP
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Subject: Re: [rtcweb] Tunnelling DTLS in SDP
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From: Harald Alvestrand <harald@alvestrand.no>
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Date: Mon, 4 Apr 2016 22:42:07 +0200
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Subject: Re: [rtcweb] Tunnelling DTLS in SDP
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On 04/04/2016 05:39 PM, pfh wrote:
>
>> On 4 Apr 2016, at 14:10, Eric Rescorla <ekr@rtfm.com
>> <mailto:ekr@rtfm.com>> wrote:
>>
>> Hi folks,
>>
>> I wanted to call your attention to a draft I just published with a
>> possibly stupid
>> idea.
>>
>> https://tools.ietf.org/html/draft-rescorla-dtls-in-sdp-00
>>
>> A nontrivial fraction of call setup time in WebRTC is the DTLS handshake.
>> This document describes how to piggyback the first few handshake messages
>> in the SDP offer/answer exchange, thus reducing latency.
>
> It strikes me we could get the same reduced latency benefits by
> piggybacking on ICE
> rather than SDP, e.g. embedding the DTLS packet as data in a new STUN
> attribute type.

Piggybacking in ICE means that you have to repeat it in every probe
packet, since you don't know which is going to make it through.

And you can't start ICE probing until the handshake has completed.


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<html>
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    <div class="moz-cite-prefix">On 04/04/2016 05:39 PM, pfh wrote:<br>
    </div>
    <blockquote
      cite="mid:D5416C24-0032-48CB-8CC6-FD5D4E046C0D@phonefromhere.com"
      type="cite">
      <meta http-equiv="Content-Type" content="text/html;
        charset=windows-1252">
      <br class="">
      <div>
        <blockquote type="cite" class="">
          <div class="">On 4 Apr 2016, at 14:10, Eric Rescorla &lt;<a
              moz-do-not-send="true" href="mailto:ekr@rtfm.com" class=""><a class="moz-txt-link-abbreviated" href="mailto:ekr@rtfm.com">ekr@rtfm.com</a></a>&gt;
            wrote:</div>
          <br class="Apple-interchange-newline">
          <div class="">
            <div dir="ltr" class="">Hi folks,
              <div class=""><br class="">
              </div>
              <div class="">I wanted to call your attention to a draft I
                just published with a possibly stupid</div>
              <div class="">idea.</div>
              <div class=""><br class="">
              </div>
              <div class=""><a moz-do-not-send="true"
                  href="https://tools.ietf.org/html/draft-rescorla-dtls-in-sdp-00"
                  class="">https://tools.ietf.org/html/draft-rescorla-dtls-in-sdp-00</a><br
                  class="">
              </div>
              <div class=""><br class="">
              </div>
              <div class="">A nontrivial fraction of call setup time in
                WebRTC is the DTLS handshake.</div>
              <div class="">This document describes how to piggyback the
                first few handshake messages</div>
              <div class="">in the SDP offer/answer exchange, thus
                reducing latency.</div>
            </div>
          </div>
        </blockquote>
        <div><br class="">
        </div>
        <div>It strikes me we could get the same reduced latency
          benefits by piggybacking on ICE</div>
        <div>rather than SDP, e.g. embedding the DTLS packet as data in
          a new STUN attribute type.</div>
      </div>
    </blockquote>
    <br>
    Piggybacking in ICE means that you have to repeat it in every probe
    packet, since you don't know which is going to make it through.<br>
    <br>
    And you can't start ICE probing until the handshake has completed.<br>
    <br>
  </body>
</html>

--------------040307010908010001010300--


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From: Justin Uberti <juberti@google.com>
Date: Mon, 4 Apr 2016 19:06:50 -0700
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Subject: Re: [rtcweb] Tunnelling DTLS in SDP
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On Mon, Apr 4, 2016 at 8:39 AM, pfh <tim@phonefromhere.com> wrote:

>
> On 4 Apr 2016, at 14:10, Eric Rescorla <ekr@rtfm.com> wrote:
>
> Hi folks,
>
> I wanted to call your attention to a draft I just published with a
> possibly stupid
> idea.
>
> https://tools.ietf.org/html/draft-rescorla-dtls-in-sdp-00
>
> A nontrivial fraction of call setup time in WebRTC is the DTLS handshake.
> This document describes how to piggyback the first few handshake messages
> in the SDP offer/answer exchange, thus reducing latency.
>
>
> It strikes me we could get the same reduced latency benefits by
> piggybacking on ICE
> rather than SDP, e.g. embedding the DTLS packet as data in a new STUN
> attribute type.
>
> The downside of piggybacking on the SDP is that you are increasing the
> trust you have to place in the
> signalling server undermining the elegant decoupling we have at the moment
> between signalling and
> media. (The SDES issues of logging keys in the web servers apply to the
> public certs as well).
>

I don't think the logging issue applies here. Only the public key would be
logged, and it's already transmitted in cleartext during a normal DTLS
handshake (it's a public key, after all).

>
> It also significantly clutters the SDP (even more) !
>

Please see Jonathan Lennox's comment about garbage piles. I think the only
truly unattractive part here is the fact that the messages will need to be
base64ed, resulting in minor blowup. But that seems like a reasonable
downside for 1+ RTT upside.


> As you point out, it weakens the usefulness of longer term certs, which
> would be a major nuisance IMHO.
>
> Tim.
>
>
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb
>
>

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<div dir=3D"ltr"><br><div class=3D"gmail_extra"><br><div class=3D"gmail_quo=
te">On Mon, Apr 4, 2016 at 8:39 AM, pfh <span dir=3D"ltr">&lt;<a href=3D"ma=
ilto:tim@phonefromhere.com" target=3D"_blank">tim@phonefromhere.com</a>&gt;=
</span> wrote:<br><blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .=
8ex;border-left:1px #ccc solid;padding-left:1ex"><div style=3D"word-wrap:br=
eak-word"><br><div><span class=3D""><blockquote type=3D"cite"><div>On 4 Apr=
 2016, at 14:10, Eric Rescorla &lt;<a href=3D"mailto:ekr@rtfm.com" target=
=3D"_blank">ekr@rtfm.com</a>&gt; wrote:</div><br><div><div dir=3D"ltr">Hi f=
olks,<div><br></div><div>I wanted to call your attention to a draft I just =
published with a possibly stupid</div><div>idea.</div><div><br></div><div><=
a href=3D"https://tools.ietf.org/html/draft-rescorla-dtls-in-sdp-00" target=
=3D"_blank">https://tools.ietf.org/html/draft-rescorla-dtls-in-sdp-00</a><b=
r></div><div><br></div><div>A nontrivial fraction of call setup time in Web=
RTC is the DTLS handshake.</div><div>This document describes how to piggyba=
ck the first few handshake messages</div><div>in the SDP offer/answer excha=
nge, thus reducing latency.</div></div></div></blockquote><div><br></div></=
span><div>It strikes me we could get the same reduced latency benefits by p=
iggybacking on ICE</div><div>rather than SDP, e.g. embedding the DTLS packe=
t as data in a new STUN attribute type.</div><div><br></div><div>The downsi=
de of piggybacking on the SDP is that you are increasing the trust you have=
 to place in the</div><div>signalling server undermining the elegant decoup=
ling we have at the moment between signalling and=C2=A0</div><div>media. (T=
he SDES issues of logging keys in the web servers apply to the public certs=
 as well).</div></div></div></blockquote><div><br></div><div>I don&#39;t th=
ink the logging issue applies here. Only the public key would be logged, an=
d it&#39;s already transmitted in cleartext during a normal DTLS handshake =
(it&#39;s a public key, after all).=C2=A0</div><blockquote class=3D"gmail_q=
uote" style=3D"margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1e=
x"><div style=3D"word-wrap:break-word"><div><div><br></div><div>It also sig=
nificantly clutters the SDP (even more) !</div></div></div></blockquote><di=
v><br></div><div>Please see Jonathan Lennox&#39;s comment about garbage pil=
es. I think the only truly unattractive part here is the fact that the mess=
ages will need to be base64ed, resulting in minor blowup. But that seems li=
ke a reasonable downside for 1+ RTT upside.</div><div><br></div><blockquote=
 class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1px #ccc soli=
d;padding-left:1ex"><div style=3D"word-wrap:break-word"><div><div><br></div=
><div>As you point out, it weakens the usefulness of longer term certs, whi=
ch</div><div>would be a major nuisance IMHO.</div><div><br></div><div>Tim.<=
/div></div><br></div><br>_______________________________________________<br=
>
rtcweb mailing list<br>
<a href=3D"mailto:rtcweb@ietf.org">rtcweb@ietf.org</a><br>
<a href=3D"https://www.ietf.org/mailman/listinfo/rtcweb" rel=3D"noreferrer"=
 target=3D"_blank">https://www.ietf.org/mailman/listinfo/rtcweb</a><br>
<br></blockquote></div><br></div></div>

--047d7b86cc1e88d08e052fb34d33--


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From: Justin Uberti <juberti@google.com>
Date: Mon, 4 Apr 2016 19:08:31 -0700
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Subject: Re: [rtcweb] Positioning draft-ietf-rtcweb-ip-address as a standards-track document
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Thanks for the concrete feedback. I agree with the proposed direction.

On Sat, Apr 2, 2016 at 3:51 PM, Harald Alvestrand <harald@alvestrand.no>
wrote:

> I've read through draft-ietf-rtcweb-ip-address again, with an eye to
> seeing how it would read as a standards-track document.
>
> As such, it seems to me to provide rather tentative language in a few
> places, which could be strengthened by use of the CAPITALIZED WORDS.
> I think this is a Good Thing; if this is standards-track, with
> imperative, normative language, an implementation can claim support for
> RFC XXXX and expect the statement to be understood.
> (Remember - there is no protocol police. Nothing forces people to
> support RFC XXXX.)
>
> Changes needed:
>
> - Add the RFC 2119 incantation, including Stefan's addition ("lowercase
> words mean what they usually do")
>
> - Replace all occurences of "We recommend that..." with "an
> implementation MUST".
>
> - In section 3, end of first paragraph: "   Specifically, WebRTC
> should:" -> "Implementations of this specification will:"
>
> - In section 4: "We recommend Mode 1 ... " -> "Mode 1 and mode 2 MUST be
> supported. Mode 1 SHOULD be the default when the user has not granted
> access to camera / microphone. Mode 2 SHOULD be the default when the
> user has granted access to camera / microphone. The implementation MAY
> provide means of letting the user or administrator select between modes.
> These means MUST NOT be placed under the control of WebRTC applications."
>
> I think this would make the spec a more useful tool for specifying what
> an application does.
>
>
>
> --
> Surveillance is pervasive. Go Dark.
>
>
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb
>

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<div dir=3D"ltr">Thanks for the concrete feedback. I agree with the propose=
d direction.</div><div class=3D"gmail_extra"><br><div class=3D"gmail_quote"=
>On Sat, Apr 2, 2016 at 3:51 PM, Harald Alvestrand <span dir=3D"ltr">&lt;<a=
 href=3D"mailto:harald@alvestrand.no" target=3D"_blank">harald@alvestrand.n=
o</a>&gt;</span> wrote:<br><blockquote class=3D"gmail_quote" style=3D"margi=
n:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">I&#39;ve read thr=
ough draft-ietf-rtcweb-ip-address again, with an eye to<br>
seeing how it would read as a standards-track document.<br>
<br>
As such, it seems to me to provide rather tentative language in a few<br>
places, which could be strengthened by use of the CAPITALIZED WORDS.<br>
I think this is a Good Thing; if this is standards-track, with<br>
imperative, normative language, an implementation can claim support for<br>
RFC XXXX and expect the statement to be understood.<br>
(Remember - there is no protocol police. Nothing forces people to<br>
support RFC XXXX.)<br>
<br>
Changes needed:<br>
<br>
- Add the RFC 2119 incantation, including Stefan&#39;s addition (&quot;lowe=
rcase<br>
words mean what they usually do&quot;)<br>
<br>
- Replace all occurences of &quot;We recommend that...&quot; with &quot;an<=
br>
implementation MUST&quot;.<br>
<br>
- In section 3, end of first paragraph: &quot;=C2=A0 =C2=A0Specifically, We=
bRTC<br>
should:&quot; -&gt; &quot;Implementations of this specification will:&quot;=
<br>
<br>
- In section 4: &quot;We recommend Mode 1 ... &quot; -&gt; &quot;Mode 1 and=
 mode 2 MUST be<br>
supported. Mode 1 SHOULD be the default when the user has not granted<br>
access to camera / microphone. Mode 2 SHOULD be the default when the<br>
user has granted access to camera / microphone. The implementation MAY<br>
provide means of letting the user or administrator select between modes.<br=
>
These means MUST NOT be placed under the control of WebRTC applications.&qu=
ot;<br>
<br>
I think this would make the spec a more useful tool for specifying what<br>
an application does.<br>
<span class=3D"HOEnZb"><font color=3D"#888888"><br>
<br>
<br>
--<br>
Surveillance is pervasive. Go Dark.<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" rel=3D"noreferrer"=
 target=3D"_blank">https://www.ietf.org/mailman/listinfo/rtcweb</a><br>
</font></span></blockquote></div><br></div>

--047d7bd9172494d042052fb35319--


From nobody Mon Apr  4 20:41:46 2016
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Cc: "rtcweb@ietf.org" <rtcweb@ietf.org>, mmusic WG <mmusic@ietf.org>
Subject: Re: [rtcweb] [MMUSIC]  Tunnelling DTLS in SDP
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As someone who designed a protocol that overlaps NAT traversal / consent and=
 key agreement, I agree that putting it in the ICE exchange would be benefic=
ial.

Matthew Kaufman

(Sent from my iPhone)

> On Apr 4, 2016, at 8:39 AM, pfh <tim@phonefromhere.com> wrote:
>=20
>=20
>> On 4 Apr 2016, at 14:10, Eric Rescorla <ekr@rtfm.com> wrote:
>>=20
>> Hi folks,
>>=20
>> I wanted to call your attention to a draft I just published with a possib=
ly stupid
>> idea.
>>=20
>> https://tools.ietf.org/html/draft-rescorla-dtls-in-sdp-00
>>=20
>> A nontrivial fraction of call setup time in WebRTC is the DTLS handshake.=

>> This document describes how to piggyback the first few handshake messages=

>> in the SDP offer/answer exchange, thus reducing latency.
>=20
> It strikes me we could get the same reduced latency benefits by piggybacki=
ng on ICE
> rather than SDP, e.g. embedding the DTLS packet as data in a new STUN attr=
ibute type.
>=20
> The downside of piggybacking on the SDP is that you are increasing the tru=
st you have to place in the
> signalling server undermining the elegant decoupling we have at the moment=
 between signalling and=20
> media. (The SDES issues of logging keys in the web servers apply to the pu=
blic certs as well).
>=20
> It also significantly clutters the SDP (even more) !
>=20
> As you point out, it weakens the usefulness of longer term certs, which
> would be a major nuisance IMHO.
>=20
> Tim.
>=20
> _______________________________________________
> mmusic mailing list
> mmusic@ietf.org
> https://www.ietf.org/mailman/listinfo/mmusic

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<html><head><meta http-equiv=3D"content-type" content=3D"text/html; charset=3D=
utf-8"></head><body dir=3D"auto"><div>As someone who designed a protocol tha=
t overlaps NAT traversal / consent and key agreement, I agree that putting i=
t in the ICE exchange would be beneficial.<br><br>Matthew Kaufman<div><br>(S=
ent from my iPhone)</div></div><div><br>On Apr 4, 2016, at 8:39 AM, pfh &lt;=
<a href=3D"mailto:tim@phonefromhere.com">tim@phonefromhere.com</a>&gt; wrote=
:<br><br></div><blockquote type=3D"cite"><div><meta http-equiv=3D"Content-Ty=
pe" content=3D"text/html charset=3Dus-ascii"><br class=3D""><div><blockquote=
 type=3D"cite" class=3D""><div class=3D"">On 4 Apr 2016, at 14:10, Eric Resc=
orla &lt;<a href=3D"mailto:ekr@rtfm.com" class=3D"">ekr@rtfm.com</a>&gt; wro=
te:</div><br class=3D"Apple-interchange-newline"><div class=3D""><div dir=3D=
"ltr" class=3D"">Hi folks,<div class=3D""><br class=3D""></div><div class=3D=
"">I wanted to call your attention to a draft I just published with a possib=
ly stupid</div><div class=3D"">idea.</div><div class=3D""><br class=3D""></d=
iv><div class=3D""><a href=3D"https://tools.ietf.org/html/draft-rescorla-dtl=
s-in-sdp-00" class=3D"">https://tools.ietf.org/html/draft-rescorla-dtls-in-s=
dp-00</a><br class=3D""></div><div class=3D""><br class=3D""></div><div clas=
s=3D"">A nontrivial fraction of call setup time in WebRTC is the DTLS handsh=
ake.</div><div class=3D"">This document describes how to piggyback the first=
 few handshake messages</div><div class=3D"">in the SDP offer/answer exchang=
e, thus reducing latency.</div></div></div></blockquote><div><br class=3D"">=
</div><div>It strikes me we could get the same reduced latency benefits by p=
iggybacking on ICE</div><div>rather than SDP, e.g. embedding the DTLS packet=
 as data in a new STUN attribute type.</div><div><br class=3D""></div><div>T=
he downside of piggybacking on the SDP is that you are increasing the trust y=
ou have to place in the</div><div>signalling server undermining the elegant d=
ecoupling we have at the moment between signalling and&nbsp;</div><div>media=
. (The SDES issues of logging keys in the web servers apply to the public ce=
rts as well).</div><div><br class=3D""></div><div>It also significantly clut=
ters the SDP (even more) !</div><div><br class=3D""></div><div>As you point o=
ut, it weakens the usefulness of longer term certs, which</div><div>would be=
 a major nuisance IMHO.</div><div><br class=3D""></div><div>Tim.</div></div>=
<br class=3D""></div></blockquote><blockquote type=3D"cite"><div><span>_____=
__________________________________________</span><br><span>mmusic mailing li=
st</span><br><span><a href=3D"mailto:mmusic@ietf.org">mmusic@ietf.org</a></s=
pan><br><span><a href=3D"https://www.ietf.org/mailman/listinfo/mmusic">https=
://www.ietf.org/mailman/listinfo/mmusic</a></span><br></div></blockquote></b=
ody></html>=

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> On 5 Apr 2016, at 03:06, Justin Uberti <juberti@google.com> wrote:
>=20
>=20
>=20
> On Mon, Apr 4, 2016 at 8:39 AM, pfh <tim@phonefromhere.com =
<mailto:tim@phonefromhere.com>> wrote:
>=20
>> On 4 Apr 2016, at 14:10, Eric Rescorla <ekr@rtfm.com =
<mailto:ekr@rtfm.com>> wrote:
>>=20
>> Hi folks,
>>=20
>> I wanted to call your attention to a draft I just published with a =
possibly stupid
>> idea.
>>=20
>> https://tools.ietf.org/html/draft-rescorla-dtls-in-sdp-00 =
<https://tools.ietf.org/html/draft-rescorla-dtls-in-sdp-00>
>>=20
>> A nontrivial fraction of call setup time in WebRTC is the DTLS =
handshake.
>> This document describes how to piggyback the first few handshake =
messages
>> in the SDP offer/answer exchange, thus reducing latency.
>=20
> It strikes me we could get the same reduced latency benefits by =
piggybacking on ICE
> rather than SDP, e.g. embedding the DTLS packet as data in a new STUN =
attribute type.
>=20
> The downside of piggybacking on the SDP is that you are increasing the =
trust you have to place in the
> signalling server undermining the elegant decoupling we have at the =
moment between signalling and=20
> media. (The SDES issues of logging keys in the web servers apply to =
the public certs as well).
>=20
> I don't think the logging issue applies here. Only the public key =
would be logged, and it's already transmitted in cleartext during a =
normal DTLS handshake (it's a public key, after all).=20

True, but over a different path.=20

>=20
> It also significantly clutters the SDP (even more) !
>=20
> Please see Jonathan Lennox's comment about garbage piles. I think the =
only truly unattractive part here is the fact that the messages will =
need to be base64ed, resulting in minor blowup. But that seems like a =
reasonable downside for 1+ RTT upside.

Just when the WebRTC NG was freeing us from Offer/Answer and SDP=E2=80=A6.=


Sigh.


>=20
>=20
> As you point out, it weakens the usefulness of longer term certs, =
which
> would be a major nuisance IMHO.
>=20
> Tim.
>=20
>=20
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org <mailto:rtcweb@ietf.org>
> https://www.ietf.org/mailman/listinfo/rtcweb =
<https://www.ietf.org/mailman/listinfo/rtcweb>
>=20
>=20


--Apple-Mail=_73971B90-CF02-43F3-83D4-3475486FCF6D
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	charset=utf-8

<html><head><meta http-equiv=3D"Content-Type" content=3D"text/html =
charset=3Dutf-8"></head><body style=3D"word-wrap: break-word; =
-webkit-nbsp-mode: space; -webkit-line-break: after-white-space;" =
class=3D""><br class=3D""><div><blockquote type=3D"cite" class=3D""><div =
class=3D"">On 5 Apr 2016, at 03:06, Justin Uberti &lt;<a =
href=3D"mailto:juberti@google.com" class=3D"">juberti@google.com</a>&gt; =
wrote:</div><br class=3D"Apple-interchange-newline"><div class=3D""><div =
dir=3D"ltr" class=3D""><br class=3D""><div class=3D"gmail_extra"><br =
class=3D""><div class=3D"gmail_quote">On Mon, Apr 4, 2016 at 8:39 AM, =
pfh <span dir=3D"ltr" class=3D"">&lt;<a =
href=3D"mailto:tim@phonefromhere.com" target=3D"_blank" =
class=3D"">tim@phonefromhere.com</a>&gt;</span> wrote:<br =
class=3D""><blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 =
.8ex;border-left:1px #ccc solid;padding-left:1ex"><div =
style=3D"word-wrap:break-word" class=3D""><br class=3D""><div =
class=3D""><span class=3D""><blockquote type=3D"cite" class=3D""><div =
class=3D"">On 4 Apr 2016, at 14:10, Eric Rescorla &lt;<a =
href=3D"mailto:ekr@rtfm.com" target=3D"_blank" =
class=3D"">ekr@rtfm.com</a>&gt; wrote:</div><br class=3D""><div =
class=3D""><div dir=3D"ltr" class=3D"">Hi folks,<div class=3D""><br =
class=3D""></div><div class=3D"">I wanted to call your attention to a =
draft I just published with a possibly stupid</div><div =
class=3D"">idea.</div><div class=3D""><br class=3D""></div><div =
class=3D""><a =
href=3D"https://tools.ietf.org/html/draft-rescorla-dtls-in-sdp-00" =
target=3D"_blank" =
class=3D"">https://tools.ietf.org/html/draft-rescorla-dtls-in-sdp-00</a><b=
r class=3D""></div><div class=3D""><br class=3D""></div><div class=3D"">A =
nontrivial fraction of call setup time in WebRTC is the DTLS =
handshake.</div><div class=3D"">This document describes how to piggyback =
the first few handshake messages</div><div class=3D"">in the SDP =
offer/answer exchange, thus reducing =
latency.</div></div></div></blockquote><div class=3D""><br =
class=3D""></div></span><div class=3D"">It strikes me we could get the =
same reduced latency benefits by piggybacking on ICE</div><div =
class=3D"">rather than SDP, e.g. embedding the DTLS packet as data in a =
new STUN attribute type.</div><div class=3D""><br class=3D""></div><div =
class=3D"">The downside of piggybacking on the SDP is that you are =
increasing the trust you have to place in the</div><div =
class=3D"">signalling server undermining the elegant decoupling we have =
at the moment between signalling and&nbsp;</div><div class=3D"">media. =
(The SDES issues of logging keys in the web servers apply to the public =
certs as well).</div></div></div></blockquote><div class=3D""><br =
class=3D""></div><div class=3D"">I don't think the logging issue applies =
here. Only the public key would be logged, and it's already transmitted =
in cleartext during a normal DTLS handshake (it's a public key, after =
all).&nbsp;</div></div></div></div></div></blockquote><div><br =
class=3D""></div><div>True, but over a different path.&nbsp;</div><br =
class=3D""><blockquote type=3D"cite" class=3D""><div class=3D""><div =
dir=3D"ltr" class=3D""><div class=3D"gmail_extra"><div =
class=3D"gmail_quote"><blockquote class=3D"gmail_quote" style=3D"margin:0 =
0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex"><div =
style=3D"word-wrap:break-word" class=3D""><div class=3D""><div =
class=3D""><br class=3D""></div><div class=3D"">It also significantly =
clutters the SDP (even more) !</div></div></div></blockquote><div =
class=3D""><br class=3D""></div><div class=3D"">Please see Jonathan =
Lennox's comment about garbage piles. I think the only truly =
unattractive part here is the fact that the messages will need to be =
base64ed, resulting in minor blowup. But that seems like a reasonable =
downside for 1+ RTT =
upside.</div></div></div></div></div></blockquote><div><br =
class=3D""></div>Just when the WebRTC NG was freeing us from =
Offer/Answer and SDP=E2=80=A6.</div><div><br =
class=3D""></div><div>Sigh.</div><div><br class=3D""><br =
class=3D""><blockquote type=3D"cite" class=3D""><div class=3D""><div =
dir=3D"ltr" class=3D""><div class=3D"gmail_extra"><div =
class=3D"gmail_quote"><div class=3D""><br class=3D""></div><blockquote =
class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1px #ccc =
solid;padding-left:1ex"><div style=3D"word-wrap:break-word" =
class=3D""><div class=3D""><div class=3D""><br class=3D""></div><div =
class=3D"">As you point out, it weakens the usefulness of longer term =
certs, which</div><div class=3D"">would be a major nuisance =
IMHO.</div><div class=3D""><br class=3D""></div><div =
class=3D"">Tim.</div></div><br class=3D""></div><br =
class=3D"">_______________________________________________<br class=3D"">
rtcweb mailing list<br class=3D"">
<a href=3D"mailto:rtcweb@ietf.org" class=3D"">rtcweb@ietf.org</a><br =
class=3D"">
<a href=3D"https://www.ietf.org/mailman/listinfo/rtcweb" =
rel=3D"noreferrer" target=3D"_blank" =
class=3D"">https://www.ietf.org/mailman/listinfo/rtcweb</a><br class=3D"">=

<br class=3D""></blockquote></div><br class=3D""></div></div>
</div></blockquote></div><br class=3D""></body></html>=

--Apple-Mail=_73971B90-CF02-43F3-83D4-3475486FCF6D--


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From: Harald Alvestrand <harald@alvestrand.no>
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Subject: Re: [rtcweb] Tunnelling DTLS in SDP
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On first read, this makes sense to me.

I wonder if it could/should be made into a general concept, to fit with
the tendency in WebRTC:NG to separate signalling format even more from
operation?

We could call it "out of band DTLS setup", say that in general, a DTLS
session can be started in one medium (SDP signalling, in this case), and
continued in another medium (the DTLS-protected media channel), and have
a section describing the details of carrying DTLS-over-SDP.

When viewing it in this way, using the same technique with Jabber or
proprietary signalling becomes a reasonably obvious exercise. There are
some other twists that seem obvious too - for instance, one could
continue the setup over the SDP channel in subsequent offer/answers if
the first exchange failed to set up a media channel. I'm not sure that
makes sense, though.

One SDP twist: If forking happens, it could be treated like any other
attempt to generate multiple answers to a ClientHello, I think. I'm sure
it's well defined how to respond to that - it's an obvious attack. Only
one leg of the fork would ever succeed, I assume.


On 04/04/2016 03:10 PM, Eric Rescorla wrote:
> Hi folks,
>
> I wanted to call your attention to a draft I just published with a
> possibly stupid
> idea.
>
> https://tools.ietf.org/html/draft-rescorla-dtls-in-sdp-00
>
> A nontrivial fraction of call setup time in WebRTC is the DTLS handshak=
e.
> This document describes how to piggyback the first few handshake messag=
es
> in the SDP offer/answer exchange, thus reducing latency.
>
> Comments welcome.
>
> -Ekr
>
>
>
>
>
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb


--=20
Surveillance is pervasive. Go Dark.


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    <div class="moz-cite-prefix">On first read, this makes sense to me.<br>
      <br>
      I wonder if it could/should be made into a general concept, to fit
      with the tendency in WebRTC:NG to separate signalling format even
      more from operation?<br>
      <br>
      We could call it "out of band DTLS setup", say that in general, a
      DTLS session can be started in one medium (SDP signalling, in this
      case), and continued in another medium (the DTLS-protected media
      channel), and have a section describing the details of carrying
      DTLS-over-SDP.<br>
      <br>
      When viewing it in this way, using the same technique with Jabber
      or proprietary signalling becomes a reasonably obvious exercise.
      There are some other twists that seem obvious too - for instance,
      one could continue the setup over the SDP channel in subsequent
      offer/answers if the first exchange failed to set up a media
      channel. I'm not sure that makes sense, though.<br>
      <br>
      One SDP twist: If forking happens, it could be treated like any
      other attempt to generate multiple answers to a ClientHello, I
      think. I'm sure it's well defined how to respond to that - it's an
      obvious attack. Only one leg of the fork would ever succeed, I
      assume.<br>
      <br>
      <br>
      On 04/04/2016 03:10 PM, Eric Rescorla wrote:<br>
    </div>
    <blockquote
cite="mid:CABcZeBOM1KoXpXFhvjS753EVpsMENWVen3CCdFj8ry36vPH0dg@mail.gmail.com"
      type="cite">
      <div dir="ltr">Hi folks,
        <div><br>
        </div>
        <div>I wanted to call your attention to a draft I just published
          with a possibly stupid</div>
        <div>idea.</div>
        <div><br>
        </div>
        <div><a moz-do-not-send="true"
            href="https://tools.ietf.org/html/draft-rescorla-dtls-in-sdp-00">https://tools.ietf.org/html/draft-rescorla-dtls-in-sdp-00</a><br>
        </div>
        <div><br>
        </div>
        <div>A nontrivial fraction of call setup time in WebRTC is the
          DTLS handshake.</div>
        <div>This document describes how to piggyback the first few
          handshake messages</div>
        <div>in the SDP offer/answer exchange, thus reducing latency.</div>
        <div><br>
        </div>
        <div>Comments welcome.</div>
        <div><br>
        </div>
        <div>-Ekr</div>
        <div><br>
        </div>
        <div><br>
        </div>
        <div><br>
        </div>
      </div>
      <br>
      <fieldset class="mimeAttachmentHeader"></fieldset>
      <br>
      <pre wrap="">_______________________________________________
rtcweb mailing list
<a class="moz-txt-link-abbreviated" href="mailto:rtcweb@ietf.org">rtcweb@ietf.org</a>
<a class="moz-txt-link-freetext" href="https://www.ietf.org/mailman/listinfo/rtcweb">https://www.ietf.org/mailman/listinfo/rtcweb</a>
</pre>
    </blockquote>
    <br>
    <br>
    <pre class="moz-signature" cols="72">-- 
Surveillance is pervasive. Go Dark.
</pre>
  </body>
</html>

--------------000008020101080200000803--


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Subject: Re: [rtcweb] Tunnelling DTLS in SDP
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> On 4 Apr 2016, at 18:34, Martin Thomson <martin.thomson@gmail.com> =
wrote:
>=20
> On 4 April 2016 at 12:39, pfh <tim@phonefromhere.com> wrote:
>> It strikes me we could get the same reduced latency benefits by =
piggybacking
>> on ICE
>> rather than SDP, e.g. embedding the DTLS packet as data in a new STUN
>> attribute type.
>=20
> One of the goals for connectivity checks is to keep them small.
> Cramming DTLS in is not compatible with this.

It wouldn=E2=80=99t be in every ICE packet, just in ones with (say) =
USE-CANDIDATE set.

T.



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From: Tim Panton <tim@phonefromhere.com>
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To: Harald Alvestrand <harald@alvestrand.no>
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Subject: Re: [rtcweb] Tunnelling DTLS in SDP
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> On 5 Apr 2016, at 10:53, Harald Alvestrand <harald@alvestrand.no =
<mailto:harald@alvestrand.no>> wrote:
>=20
> On first read, this makes sense to me.
>=20
> I wonder if it could/should be made into a general concept, to fit =
with the tendency in WebRTC:NG to separate signalling format even more =
from operation?
>=20
> We could call it "out of band DTLS setup", say that in general, a DTLS =
session can be started in one medium (SDP signalling, in this case), and =
continued in another medium (the DTLS-protected media channel), and have =
a section describing the details of carrying DTLS-over-SDP.
>=20
> When viewing it in this way, using the same technique with Jabber or =
proprietary signalling becomes a reasonably obvious exercise. There are =
some other twists that seem obvious too - for instance, one could =
continue the setup over the SDP channel in subsequent offer/answers if =
the first exchange failed to set up a media channel. I'm not sure that =
makes sense, though.
>=20
> One SDP twist: If forking happens, it could be treated like any other =
attempt to generate multiple answers to a ClientHello, I think. I'm sure =
it's well defined how to respond to that - it's an obvious attack. Only =
one leg of the fork would ever succeed, I assume.

Doesn=92t passing the DTLS Hellos through javascript open us to a whole =
pile of bid-down attacks where
the lists of supported crypto suites and extensions are manipulated in =
the js ? E.G It  might allow the javascript to insert=20
the magic extension that says "this isn=92t being recorded=94  without =
the browser actually being in that mode.=20
Or is there a way that DTLS can detect this ?

T.



>=20
>=20
> On 04/04/2016 03:10 PM, Eric Rescorla wrote:
>> Hi folks,
>>=20
>> I wanted to call your attention to a draft I just published with a =
possibly stupid
>> idea.
>>=20
>> https://tools.ietf.org/html/draft-rescorla-dtls-in-sdp-00 =
<https://tools.ietf.org/html/draft-rescorla-dtls-in-sdp-00>
>>=20
>> A nontrivial fraction of call setup time in WebRTC is the DTLS =
handshake.
>> This document describes how to piggyback the first few handshake =
messages
>> in the SDP offer/answer exchange, thus reducing latency.
>>=20
>> Comments welcome.
>>=20
>> -Ekr
>>=20
>>=20
>>=20
>>=20
>>=20
>> _______________________________________________
>> rtcweb mailing list
>> rtcweb@ietf.org <mailto:rtcweb@ietf.org>
>> https://www.ietf.org/mailman/listinfo/rtcweb =
<https://www.ietf.org/mailman/listinfo/rtcweb>
>=20
>=20
> --=20
> Surveillance is pervasive. Go Dark.
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org <mailto:rtcweb@ietf.org>
> https://www.ietf.org/mailman/listinfo/rtcweb

Tim Panton - Web/VoIP consultant and implementor
www.westhawk.co.uk <http://www.westhawk.co.uk/>




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<html><head><meta http-equiv=3D"Content-Type" content=3D"text/html =
charset=3Dwindows-1252"><meta http-equiv=3D"Content-Type" =
content=3D"text/html charset=3Dwindows-1252"></head><body =
style=3D"word-wrap: break-word; -webkit-nbsp-mode: space; =
-webkit-line-break: after-white-space;" class=3D""><br class=3D""><div =
class=3D""><blockquote type=3D"cite" class=3D""><div class=3D"">On 5 Apr =
2016, at 10:53, Harald Alvestrand &lt;<a =
href=3D"mailto:harald@alvestrand.no" =
class=3D"">harald@alvestrand.no</a>&gt; wrote:</div><br =
class=3D"Apple-interchange-newline"><div class=3D"">
 =20
    <meta content=3D"text/html; charset=3Dwindows-1252" =
http-equiv=3D"Content-Type" class=3D"">
 =20
  <div text=3D"#000000" bgcolor=3D"#FFFFFF" class=3D"">
    <div class=3D"moz-cite-prefix">On first read, this makes sense to =
me.<br class=3D"">
      <br class=3D"">
      I wonder if it could/should be made into a general concept, to fit
      with the tendency in WebRTC:NG to separate signalling format even
      more from operation?<br class=3D"">
      <br class=3D"">
      We could call it "out of band DTLS setup", say that in general, a
      DTLS session can be started in one medium (SDP signalling, in this
      case), and continued in another medium (the DTLS-protected media
      channel), and have a section describing the details of carrying
      DTLS-over-SDP.<br class=3D"">
      <br class=3D"">
      When viewing it in this way, using the same technique with Jabber
      or proprietary signalling becomes a reasonably obvious exercise.
      There are some other twists that seem obvious too - for instance,
      one could continue the setup over the SDP channel in subsequent
      offer/answers if the first exchange failed to set up a media
      channel. I'm not sure that makes sense, though.<br class=3D"">
      <br class=3D"">
      One SDP twist: If forking happens, it could be treated like any
      other attempt to generate multiple answers to a ClientHello, I
      think. I'm sure it's well defined how to respond to that - it's an
      obvious attack. Only one leg of the fork would ever succeed, I
      assume.<br class=3D""></div></div></div></blockquote><div =
class=3D""><br class=3D""></div><div class=3D"">Doesn=92t passing the =
DTLS Hellos through javascript open us to a whole pile of bid-down =
attacks where</div><div class=3D"">the lists of supported crypto suites =
and extensions are manipulated in the js ? E.G It &nbsp;might allow the =
javascript to insert&nbsp;</div><div class=3D"">the magic extension that =
says "this isn=92t being recorded=94 &nbsp;without the browser actually =
being in that mode.&nbsp;</div><div class=3D"">Or is there a way that =
DTLS can detect this ?</div><div class=3D""><br class=3D""></div><div =
class=3D"">T.</div><div class=3D""><br class=3D""></div><div =
class=3D""><br class=3D""></div><br class=3D""><blockquote type=3D"cite" =
class=3D""><div class=3D""><div text=3D"#000000" bgcolor=3D"#FFFFFF" =
class=3D""><div class=3D"moz-cite-prefix">
      <br class=3D"">
      <br class=3D"">
      On 04/04/2016 03:10 PM, Eric Rescorla wrote:<br class=3D"">
    </div>
    <blockquote =
cite=3D"mid:CABcZeBOM1KoXpXFhvjS753EVpsMENWVen3CCdFj8ry36vPH0dg@mail.gmail=
.com" type=3D"cite" class=3D"">
      <div dir=3D"ltr" class=3D"">Hi folks,
        <div class=3D""><br class=3D"">
        </div>
        <div class=3D"">I wanted to call your attention to a draft I =
just published
          with a possibly stupid</div>
        <div class=3D"">idea.</div>
        <div class=3D""><br class=3D"">
        </div>
        <div class=3D""><a moz-do-not-send=3D"true" =
href=3D"https://tools.ietf.org/html/draft-rescorla-dtls-in-sdp-00" =
class=3D"">https://tools.ietf.org/html/draft-rescorla-dtls-in-sdp-00</a><b=
r class=3D"">
        </div>
        <div class=3D""><br class=3D"">
        </div>
        <div class=3D"">A nontrivial fraction of call setup time in =
WebRTC is the
          DTLS handshake.</div>
        <div class=3D"">This document describes how to piggyback the =
first few
          handshake messages</div>
        <div class=3D"">in the SDP offer/answer exchange, thus reducing =
latency.</div>
        <div class=3D""><br class=3D"">
        </div>
        <div class=3D"">Comments welcome.</div>
        <div class=3D""><br class=3D"">
        </div>
        <div class=3D"">-Ekr</div>
        <div class=3D""><br class=3D"">
        </div>
        <div class=3D""><br class=3D"">
        </div>
        <div class=3D""><br class=3D"">
        </div>
      </div>
      <br class=3D"">
      <fieldset class=3D"mimeAttachmentHeader"></fieldset>
      <br class=3D"">
      <pre wrap=3D"" =
class=3D"">_______________________________________________
rtcweb mailing list
<a class=3D"moz-txt-link-abbreviated" =
href=3D"mailto:rtcweb@ietf.org">rtcweb@ietf.org</a>
<a class=3D"moz-txt-link-freetext" =
href=3D"https://www.ietf.org/mailman/listinfo/rtcweb">https://www.ietf.org=
/mailman/listinfo/rtcweb</a>
</pre>
    </blockquote>
    <br class=3D"">
    <br class=3D"">
    <pre class=3D"moz-signature" cols=3D"72">--=20
Surveillance is pervasive. Go Dark.
</pre>
  </div>

_______________________________________________<br class=3D"">rtcweb =
mailing list<br class=3D""><a href=3D"mailto:rtcweb@ietf.org" =
class=3D"">rtcweb@ietf.org</a><br class=3D""><a =
href=3D"https://www.ietf.org/mailman/listinfo/rtcweb" =
class=3D"">https://www.ietf.org/mailman/listinfo/rtcweb</a><br =
class=3D""></div></blockquote></div><br class=3D""><div =
apple-content-edited=3D"true" class=3D"">
<span class=3D"Apple-style-span" style=3D"border-collapse: separate; =
color: rgb(0, 0, 0); font-family: Helvetica; font-size: 12px; =
font-style: normal; font-variant: normal; font-weight: normal; =
letter-spacing: normal; line-height: normal; orphans: 2; text-align: =
auto; text-indent: 0px; text-transform: none; white-space: normal; =
widows: 2; word-spacing: 0px; -webkit-border-horizontal-spacing: 0px; =
-webkit-border-vertical-spacing: 0px; =
-webkit-text-decorations-in-effect: none; -webkit-text-size-adjust: =
auto; -webkit-text-stroke-width: 0; "><div class=3D"">Tim Panton - =
Web/VoIP consultant and implementor</div><div class=3D""><a =
href=3D"http://www.westhawk.co.uk" =
class=3D"">www.westhawk.co.uk</a></div><div class=3D""><br =
class=3D""></div></span><br class=3D"Apple-interchange-newline">

</div>
<br class=3D""></body></html>=

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To: Tim Panton <tim@phonefromhere.com>
References: <CABcZeBOM1KoXpXFhvjS753EVpsMENWVen3CCdFj8ry36vPH0dg@mail.gmail.com> <57038B35.8040102@alvestrand.no> <BA5D10EA-D10D-4458-9FCB-272ECB7BFCB6@phonefromhere.com>
From: Harald Alvestrand <harald@alvestrand.no>
Message-ID: <570390ED.8080109@alvestrand.no>
Date: Tue, 5 Apr 2016 12:18:21 +0200
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Subject: Re: [rtcweb] Tunnelling DTLS in SDP
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On 04/05/2016 12:04 PM, Tim Panton wrote:
>
>> On 5 Apr 2016, at 10:53, Harald Alvestrand <harald@alvestrand.no
>> <mailto:harald@alvestrand.no>> wrote:
>>
>> On first read, this makes sense to me.
>>
>> I wonder if it could/should be made into a general concept, to fit
>> with the tendency in WebRTC:NG to separate signalling format even
>> more from operation?
>>
>> We could call it "out of band DTLS setup", say that in general, a
>> DTLS session can be started in one medium (SDP signalling, in this
>> case), and continued in another medium (the DTLS-protected media
>> channel), and have a section describing the details of carrying
>> DTLS-over-SDP.
>>
>> When viewing it in this way, using the same technique with Jabber or
>> proprietary signalling becomes a reasonably obvious exercise. There
>> are some other twists that seem obvious too - for instance, one could
>> continue the setup over the SDP channel in subsequent offer/answers
>> if the first exchange failed to set up a media channel. I'm not sure
>> that makes sense, though.
>>
>> One SDP twist: If forking happens, it could be treated like any other
>> attempt to generate multiple answers to a ClientHello, I think. I'm
>> sure it's well defined how to respond to that - it's an obvious
>> attack. Only one leg of the fork would ever succeed, I assume.
>
> Doesn’t passing the DTLS Hellos through javascript open us to a whole
> pile of bid-down attacks where
> the lists of supported crypto suites and extensions are manipulated in
> the js ? E.G It  might allow the javascript to insert 
> the magic extension that says "this isn’t being recorded”  without the
> browser actually being in that mode. 
> Or is there a way that DTLS can detect this ?

It certainly reduces the setup security from one where only network
on-path attackers can do MITM attacks on the handshake to one where
everyone with access to the Javascript can do MITM attacks on the handshake.

Given that DTLS is supposed to be a reasonable option when you have
on-path attackers on the network path, I certainly hope the defenses are
adequate in this case too, but I don't know DTLS well enough to be able
to say what those defenses are.

>
> T.
>
>
>
>>
>>
>> On 04/04/2016 03:10 PM, Eric Rescorla wrote:
>>> Hi folks,
>>>
>>> I wanted to call your attention to a draft I just published with a
>>> possibly stupid
>>> idea.
>>>
>>> https://tools.ietf.org/html/draft-rescorla-dtls-in-sdp-00
>>>
>>> A nontrivial fraction of call setup time in WebRTC is the DTLS
>>> handshake.
>>> This document describes how to piggyback the first few handshake
>>> messages
>>> in the SDP offer/answer exchange, thus reducing latency.
>>>
>>> Comments welcome.
>>>
>>> -Ekr
>>>
>>>
>>>
>>>
>>>
>>> _______________________________________________
>>> rtcweb mailing list
>>> rtcweb@ietf.org
>>> https://www.ietf.org/mailman/listinfo/rtcweb
>>
>>
>> -- 
>> Surveillance is pervasive. Go Dark.
>> _______________________________________________
>> rtcweb mailing list
>> rtcweb@ietf.org <mailto:rtcweb@ietf.org>
>> https://www.ietf.org/mailman/listinfo/rtcweb
>
> Tim Panton - Web/VoIP consultant and implementor
> www.westhawk.co.uk <http://www.westhawk.co.uk>
>
>
>


-- 
Surveillance is pervasive. Go Dark.


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<html>
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  </head>
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    <div class="moz-cite-prefix">On 04/05/2016 12:04 PM, Tim Panton
      wrote:<br>
    </div>
    <blockquote
      cite="mid:BA5D10EA-D10D-4458-9FCB-272ECB7BFCB6@phonefromhere.com"
      type="cite">
      <meta http-equiv="Content-Type" content="text/html;
        charset=windows-1252">
      <meta http-equiv="Content-Type" content="text/html;
        charset=windows-1252">
      <br class="">
      <div class="">
        <blockquote type="cite" class="">
          <div class="">On 5 Apr 2016, at 10:53, Harald Alvestrand &lt;<a
              moz-do-not-send="true" href="mailto:harald@alvestrand.no"
              class=""><a class="moz-txt-link-abbreviated" href="mailto:harald@alvestrand.no">harald@alvestrand.no</a></a>&gt; wrote:</div>
          <br class="Apple-interchange-newline">
          <div class="">
            <meta content="text/html; charset=windows-1252"
              http-equiv="Content-Type" class="">
            <div text="#000000" bgcolor="#FFFFFF" class="">
              <div class="moz-cite-prefix">On first read, this makes
                sense to me.<br class="">
                <br class="">
                I wonder if it could/should be made into a general
                concept, to fit with the tendency in WebRTC:NG to
                separate signalling format even more from operation?<br
                  class="">
                <br class="">
                We could call it "out of band DTLS setup", say that in
                general, a DTLS session can be started in one medium
                (SDP signalling, in this case), and continued in another
                medium (the DTLS-protected media channel), and have a
                section describing the details of carrying
                DTLS-over-SDP.<br class="">
                <br class="">
                When viewing it in this way, using the same technique
                with Jabber or proprietary signalling becomes a
                reasonably obvious exercise. There are some other twists
                that seem obvious too - for instance, one could continue
                the setup over the SDP channel in subsequent
                offer/answers if the first exchange failed to set up a
                media channel. I'm not sure that makes sense, though.<br
                  class="">
                <br class="">
                One SDP twist: If forking happens, it could be treated
                like any other attempt to generate multiple answers to a
                ClientHello, I think. I'm sure it's well defined how to
                respond to that - it's an obvious attack. Only one leg
                of the fork would ever succeed, I assume.<br class="">
              </div>
            </div>
          </div>
        </blockquote>
        <div class=""><br class="">
        </div>
        <div class="">Doesn’t passing the DTLS Hellos through javascript
          open us to a whole pile of bid-down attacks where</div>
        <div class="">the lists of supported crypto suites and
          extensions are manipulated in the js ? E.G It  might allow the
          javascript to insert </div>
        <div class="">the magic extension that says "this isn’t being
          recorded”  without the browser actually being in that mode. </div>
        <div class="">Or is there a way that DTLS can detect this ?</div>
      </div>
    </blockquote>
    <br>
    It certainly reduces the setup security from one where only network
    on-path attackers can do MITM attacks on the handshake to one where
    everyone with access to the Javascript can do MITM attacks on the
    handshake.<br>
    <br>
    Given that DTLS is supposed to be a reasonable option when you have
    on-path attackers on the network path, I certainly hope the defenses
    are adequate in this case too, but I don't know DTLS well enough to
    be able to say what those defenses are.<br>
    <br>
    <blockquote
      cite="mid:BA5D10EA-D10D-4458-9FCB-272ECB7BFCB6@phonefromhere.com"
      type="cite">
      <div class="">
        <div class=""><br class="">
        </div>
        <div class="">T.</div>
        <div class=""><br class="">
        </div>
        <div class=""><br class="">
        </div>
        <br class="">
        <blockquote type="cite" class="">
          <div class="">
            <div text="#000000" bgcolor="#FFFFFF" class="">
              <div class="moz-cite-prefix"> <br class="">
                <br class="">
                On 04/04/2016 03:10 PM, Eric Rescorla wrote:<br class="">
              </div>
              <blockquote
cite="mid:CABcZeBOM1KoXpXFhvjS753EVpsMENWVen3CCdFj8ry36vPH0dg@mail.gmail.com"
                type="cite" class="">
                <div dir="ltr" class="">Hi folks,
                  <div class=""><br class="">
                  </div>
                  <div class="">I wanted to call your attention to a
                    draft I just published with a possibly stupid</div>
                  <div class="">idea.</div>
                  <div class=""><br class="">
                  </div>
                  <div class=""><a moz-do-not-send="true"
                      href="https://tools.ietf.org/html/draft-rescorla-dtls-in-sdp-00"
                      class="">https://tools.ietf.org/html/draft-rescorla-dtls-in-sdp-00</a><br
                      class="">
                  </div>
                  <div class=""><br class="">
                  </div>
                  <div class="">A nontrivial fraction of call setup time
                    in WebRTC is the DTLS handshake.</div>
                  <div class="">This document describes how to piggyback
                    the first few handshake messages</div>
                  <div class="">in the SDP offer/answer exchange, thus
                    reducing latency.</div>
                  <div class=""><br class="">
                  </div>
                  <div class="">Comments welcome.</div>
                  <div class=""><br class="">
                  </div>
                  <div class="">-Ekr</div>
                  <div class=""><br class="">
                  </div>
                  <div class=""><br class="">
                  </div>
                  <div class=""><br class="">
                  </div>
                </div>
                <br class="">
                <fieldset class="mimeAttachmentHeader"></fieldset>
                <br class="">
                <pre class="" wrap="">_______________________________________________
rtcweb mailing list
<a moz-do-not-send="true" class="moz-txt-link-abbreviated" href="mailto:rtcweb@ietf.org">rtcweb@ietf.org</a>
<a moz-do-not-send="true" class="moz-txt-link-freetext" href="https://www.ietf.org/mailman/listinfo/rtcweb">https://www.ietf.org/mailman/listinfo/rtcweb</a>
</pre>
              </blockquote>
              <br class="">
              <br class="">
              <pre class="moz-signature" cols="72">-- 
Surveillance is pervasive. Go Dark.
</pre>
            </div>
            _______________________________________________<br class="">
            rtcweb mailing list<br class="">
            <a moz-do-not-send="true" href="mailto:rtcweb@ietf.org"
              class="">rtcweb@ietf.org</a><br class="">
            <a moz-do-not-send="true"
              href="https://www.ietf.org/mailman/listinfo/rtcweb"
              class="">https://www.ietf.org/mailman/listinfo/rtcweb</a><br
              class="">
          </div>
        </blockquote>
      </div>
      <br class="">
      <div apple-content-edited="true" class="">
        <span class="Apple-style-span" style="border-collapse: separate;
          color: rgb(0, 0, 0); font-family: Helvetica; font-size: 12px;
          font-style: normal; font-variant: normal; font-weight: normal;
          letter-spacing: normal; line-height: normal; orphans: 2;
          text-align: auto; text-indent: 0px; text-transform: none;
          white-space: normal; widows: 2; word-spacing: 0px;
          -webkit-border-horizontal-spacing: 0px;
          -webkit-border-vertical-spacing: 0px;
          -webkit-text-decorations-in-effect: none;
          -webkit-text-size-adjust: auto; -webkit-text-stroke-width: 0;
          ">
          <div class="">Tim Panton - Web/VoIP consultant and implementor</div>
          <div class=""><a moz-do-not-send="true"
              href="http://www.westhawk.co.uk" class="">www.westhawk.co.uk</a></div>
          <div class=""><br class="">
          </div>
        </span><br class="Apple-interchange-newline">
      </div>
      <br class="">
    </blockquote>
    <br>
    <br>
    <pre class="moz-signature" cols="72">-- 
Surveillance is pervasive. Go Dark.
</pre>
  </body>
</html>

--------------030401020809000303050907--


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From: Victor Pascual Avila <victor.pascual.avila@gmail.com>
To: Justin Uberti <juberti@google.com>
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Subject: Re: [rtcweb] Positioning draft-ietf-rtcweb-ip-address as a standards-track document
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+1

On Tue, Apr 5, 2016 at 4:08 AM, Justin Uberti <juberti@google.com> wrote:
> Thanks for the concrete feedback. I agree with the proposed direction.
>
> On Sat, Apr 2, 2016 at 3:51 PM, Harald Alvestrand <harald@alvestrand.no>
> wrote:
>>
>> I've read through draft-ietf-rtcweb-ip-address again, with an eye to
>> seeing how it would read as a standards-track document.
>>
>> As such, it seems to me to provide rather tentative language in a few
>> places, which could be strengthened by use of the CAPITALIZED WORDS.
>> I think this is a Good Thing; if this is standards-track, with
>> imperative, normative language, an implementation can claim support for
>> RFC XXXX and expect the statement to be understood.
>> (Remember - there is no protocol police. Nothing forces people to
>> support RFC XXXX.)
>>
>> Changes needed:
>>
>> - Add the RFC 2119 incantation, including Stefan's addition ("lowercase
>> words mean what they usually do")
>>
>> - Replace all occurences of "We recommend that..." with "an
>> implementation MUST".
>>
>> - In section 3, end of first paragraph: "   Specifically, WebRTC
>> should:" -> "Implementations of this specification will:"
>>
>> - In section 4: "We recommend Mode 1 ... " -> "Mode 1 and mode 2 MUST be
>> supported. Mode 1 SHOULD be the default when the user has not granted
>> access to camera / microphone. Mode 2 SHOULD be the default when the
>> user has granted access to camera / microphone. The implementation MAY
>> provide means of letting the user or administrator select between modes.
>> These means MUST NOT be placed under the control of WebRTC applications.=
"
>>
>> I think this would make the spec a more useful tool for specifying what
>> an application does.
>>
>>
>>
>> --
>> Surveillance is pervasive. Go Dark.
>>
>>
>> _______________________________________________
>> rtcweb mailing list
>> rtcweb@ietf.org
>> https://www.ietf.org/mailman/listinfo/rtcweb
>
>
>
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb
>



--=20
Victor Pascual =C3=81vila


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From: Eric Rescorla <ekr@rtfm.com>
Date: Tue, 5 Apr 2016 09:39:36 -0300
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To: Harald Alvestrand <harald@alvestrand.no>
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Subject: Re: [rtcweb] Tunnelling DTLS in SDP
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--94eb2c0b14bcb2bd59052fbc2553
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On Tue, Apr 5, 2016 at 7:18 AM, Harald Alvestrand <harald@alvestrand.no>
wrote:

> On 04/05/2016 12:04 PM, Tim Panton wrote:
>
>
> On 5 Apr 2016, at 10:53, Harald Alvestrand < <harald@alvestrand.no>
> harald@alvestrand.no> wrote:
>
> On first read, this makes sense to me.
>
> I wonder if it could/should be made into a general concept, to fit with
> the tendency in WebRTC:NG to separate signalling format even more from
> operation?
>
> We could call it "out of band DTLS setup", say that in general, a DTLS
> session can be started in one medium (SDP signalling, in this case), and
> continued in another medium (the DTLS-protected media channel), and have =
a
> section describing the details of carrying DTLS-over-SDP.
>
> When viewing it in this way, using the same technique with Jabber or
> proprietary signalling becomes a reasonably obvious exercise. There are
> some other twists that seem obvious too - for instance, one could continu=
e
> the setup over the SDP channel in subsequent offer/answers if the first
> exchange failed to set up a media channel. I'm not sure that makes sense,
> though.
>
> One SDP twist: If forking happens, it could be treated like any other
> attempt to generate multiple answers to a ClientHello, I think. I'm sure
> it's well defined how to respond to that - it's an obvious attack. Only o=
ne
> leg of the fork would ever succeed, I assume.
>
>
> Doesn=E2=80=99t passing the DTLS Hellos through javascript open us to a w=
hole pile
> of bid-down attacks where
> the lists of supported crypto suites and extensions are manipulated in th=
e
> js ? E.G It  might allow the javascript to insert
> the magic extension that says "this isn=E2=80=99t being recorded=E2=80=9D=
  without the
> browser actually being in that mode.
> Or is there a way that DTLS can detect this ?
>
>
> It certainly reduces the setup security from one where only network
> on-path attackers can do MITM attacks on the handshake to one where
> everyone with access to the Javascript can do MITM attacks on the handsha=
ke.
>

Well, attackers who control the signaling can put themselves on the network
by manipulating the
ICE parameters.


>
> Given that DTLS is supposed to be a reasonable option when you have
> on-path attackers on the network path, I certainly hope the defenses are
> adequate in this case too, but I don't know DTLS well enough to be able t=
o
> say what those defenses are.
>

However the messages are sent, the handshake is tied to the endpoint keys.

-Ekr


>
> T.
>
>
>
>
>
> On 04/04/2016 03:10 PM, Eric Rescorla wrote:
>
> Hi folks,
>
> I wanted to call your attention to a draft I just published with a
> possibly stupid
> idea.
>
> https://tools.ietf.org/html/draft-rescorla-dtls-in-sdp-00
>
> A nontrivial fraction of call setup time in WebRTC is the DTLS handshake.
> This document describes how to piggyback the first few handshake messages
> in the SDP offer/answer exchange, thus reducing latency.
>
> Comments welcome.
>
> -Ekr
>
>
>
>
>
> _______________________________________________
> rtcweb mailing listrtcweb@ietf.orghttps://www.ietf.org/mailman/listinfo/r=
tcweb
>
>
>
> --
> Surveillance is pervasive. Go Dark.
>
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb
>
>
> Tim Panton - Web/VoIP consultant and implementor
> www.westhawk.co.uk
>
>
>
>
>
> --
> Surveillance is pervasive. Go Dark.
>
>
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb
>
>

--94eb2c0b14bcb2bd59052fbc2553
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Content-Transfer-Encoding: quoted-printable

<div dir=3D"ltr"><br><div class=3D"gmail_extra"><br><div class=3D"gmail_quo=
te">On Tue, Apr 5, 2016 at 7:18 AM, Harald Alvestrand <span dir=3D"ltr">&lt=
;<a href=3D"mailto:harald@alvestrand.no" target=3D"_blank">harald@alvestran=
d.no</a>&gt;</span> wrote:<br><blockquote class=3D"gmail_quote" style=3D"ma=
rgin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">
 =20
   =20
 =20
  <div text=3D"#000000" bgcolor=3D"#FFFFFF"><span class=3D"">
    <div>On 04/05/2016 12:04 PM, Tim Panton
      wrote:<br>
    </div>
    <blockquote type=3D"cite">
     =20
     =20
      <br>
      <div>
        <blockquote type=3D"cite">
          <div>On 5 Apr 2016, at 10:53, Harald Alvestrand &lt;<a href=3D"ma=
ilto:harald@alvestrand.no" target=3D"_blank"></a><a href=3D"mailto:harald@a=
lvestrand.no" target=3D"_blank">harald@alvestrand.no</a>&gt; wrote:</div>
          <br>
          <div>
           =20
            <div text=3D"#000000" bgcolor=3D"#FFFFFF">
              <div>On first read, this makes
                sense to me.<br>
                <br>
                I wonder if it could/should be made into a general
                concept, to fit with the tendency in WebRTC:NG to
                separate signalling format even more from operation?<br>
                <br>
                We could call it &quot;out of band DTLS setup&quot;, say th=
at in
                general, a DTLS session can be started in one medium
                (SDP signalling, in this case), and continued in another
                medium (the DTLS-protected media channel), and have a
                section describing the details of carrying
                DTLS-over-SDP.<br>
                <br>
                When viewing it in this way, using the same technique
                with Jabber or proprietary signalling becomes a
                reasonably obvious exercise. There are some other twists
                that seem obvious too - for instance, one could continue
                the setup over the SDP channel in subsequent
                offer/answers if the first exchange failed to set up a
                media channel. I&#39;m not sure that makes sense, though.<b=
r>
                <br>
                One SDP twist: If forking happens, it could be treated
                like any other attempt to generate multiple answers to a
                ClientHello, I think. I&#39;m sure it&#39;s well defined ho=
w to
                respond to that - it&#39;s an obvious attack. Only one leg
                of the fork would ever succeed, I assume.<br>
              </div>
            </div>
          </div>
        </blockquote>
        <div><br>
        </div>
        <div>Doesn=E2=80=99t passing the DTLS Hellos through javascript
          open us to a whole pile of bid-down attacks where</div>
        <div>the lists of supported crypto suites and
          extensions are manipulated in the js ? E.G It =C2=A0might allow t=
he
          javascript to insert=C2=A0</div>
        <div>the magic extension that says &quot;this isn=E2=80=99t being
          recorded=E2=80=9D =C2=A0without the browser actually being in tha=
t mode.=C2=A0</div>
        <div>Or is there a way that DTLS can detect this ?</div>
      </div>
    </blockquote>
    <br></span>
    It certainly reduces the setup security from one where only network
    on-path attackers can do MITM attacks on the handshake to one where
    everyone with access to the Javascript can do MITM attacks on the
    handshake.<br></div></blockquote><div><br></div><div>Well, attackers wh=
o control the signaling can put themselves on the network by manipulating t=
he</div><div>ICE parameters.</div><div>=C2=A0</div><blockquote class=3D"gma=
il_quote" style=3D"margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-lef=
t:1ex"><div text=3D"#000000" bgcolor=3D"#FFFFFF">
    <br>
    Given that DTLS is supposed to be a reasonable option when you have
    on-path attackers on the network path, I certainly hope the defenses
    are adequate in this case too, but I don&#39;t know DTLS well enough to
    be able to say what those defenses are.</div></blockquote><div><br></di=
v><div>However the messages are sent, the handshake is tied to the endpoint=
 keys.</div><div><br></div><div>-Ekr</div><div><br></div><blockquote class=
=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1px #ccc solid;padd=
ing-left:1ex"><div text=3D"#000000" bgcolor=3D"#FFFFFF"><div><div class=3D"=
h5">
    <br>
    <blockquote type=3D"cite">
      <div>
        <div><br>
        </div>
        <div>T.</div>
        <div><br>
        </div>
        <div><br>
        </div>
        <br>
        <blockquote type=3D"cite">
          <div>
            <div text=3D"#000000" bgcolor=3D"#FFFFFF">
              <div> <br>
                <br>
                On 04/04/2016 03:10 PM, Eric Rescorla wrote:<br>
              </div>
              <blockquote type=3D"cite">
                <div dir=3D"ltr">Hi folks,
                  <div><br>
                  </div>
                  <div>I wanted to call your attention to a
                    draft I just published with a possibly stupid</div>
                  <div>idea.</div>
                  <div><br>
                  </div>
                  <div><a href=3D"https://tools.ietf.org/html/draft-rescorl=
a-dtls-in-sdp-00" target=3D"_blank">https://tools.ietf.org/html/draft-resco=
rla-dtls-in-sdp-00</a><br>
                  </div>
                  <div><br>
                  </div>
                  <div>A nontrivial fraction of call setup time
                    in WebRTC is the DTLS handshake.</div>
                  <div>This document describes how to piggyback
                    the first few handshake messages</div>
                  <div>in the SDP offer/answer exchange, thus
                    reducing latency.</div>
                  <div><br>
                  </div>
                  <div>Comments welcome.</div>
                  <div><br>
                  </div>
                  <div>-Ekr</div>
                  <div><br>
                  </div>
                  <div><br>
                  </div>
                  <div><br>
                  </div>
                </div>
                <br>
                <fieldset></fieldset>
                <br>
                <pre>_______________________________________________
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>
              <br>
              <br>
              <pre cols=3D"72">--=20
Surveillance is pervasive. Go Dark.
</pre>
            </div>
            _______________________________________________<br>
            rtcweb mailing list<br>
            <a href=3D"mailto:rtcweb@ietf.org" target=3D"_blank">rtcweb@iet=
f.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>
        </blockquote>
      </div>
      <br>
      <div>
        <span style=3D"border-collapse:separate;color:rgb(0,0,0);font-famil=
y:Helvetica;font-size:12px;font-style:normal;font-variant:normal;font-weigh=
t:normal;letter-spacing:normal;line-height:normal;text-align:auto;text-inde=
nt:0px;text-transform:none;white-space:normal;word-spacing:0px">
          <div>Tim Panton - Web/VoIP consultant and implementor</div>
          <div><a href=3D"http://www.westhawk.co.uk" target=3D"_blank">www.=
westhawk.co.uk</a></div>
          <div><br>
          </div>
        </span><br>
      </div>
      <br>
    </blockquote>
    <br>
    <br>
    <pre cols=3D"72">--=20
Surveillance is pervasive. Go Dark.
</pre>
  </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" rel=3D"noreferrer"=
 target=3D"_blank">https://www.ietf.org/mailman/listinfo/rtcweb</a><br>
<br></blockquote></div><br></div></div>

--94eb2c0b14bcb2bd59052fbc2553--


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Subject: Re: [rtcweb] Tunnelling DTLS in SDP
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On Tue, Apr 5, 2016 at 5:53 AM, Harald Alvestrand <harald@alvestrand.no>
wrote:

>
> One SDP twist: If forking happens, it could be treated like any other
> attempt to generate multiple answers to a ClientHello, I think. I'm sure
> it's well defined how to respond to that - it's an obvious attack. Only one
> leg of the fork would ever succeed, I assume.
>

The biggest issue with reusing ClientHello in case of forking would be
reusing client random across multiple connections. Unless I am missing
something, this should be generally safe.

Another issue is what will happen if two DTLS association are established
for a single m= line (i.e. RTCP bundle is not used). Does this mean the
same ClientHello is used for both connection?

Regards,
_____________
Roman Shpount

--089e01494ff2575a7c052fbcb88d
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<div dir=3D"ltr"><div class=3D"gmail_extra"><div><div class=3D"gmail_signat=
ure">On Tue, Apr 5, 2016 at 5:53 AM, Harald Alvestrand <span dir=3D"ltr">&l=
t;<a href=3D"mailto:harald@alvestrand.no" target=3D"_blank">harald@alvestra=
nd.no</a>&gt;</span> wrote:<br></div></div><div class=3D"gmail_quote"><bloc=
kquote class=3D"gmail_quote" style=3D"margin:0px 0px 0px 0.8ex;border-left-=
width:1px;border-left-style:solid;border-left-color:rgb(204,204,204);paddin=
g-left:1ex">
 =20
   =20
 =20
  <div text=3D"#000000" bgcolor=3D"#FFFFFF">
    <div><br>
      One SDP twist: If forking happens, it could be treated like any
      other attempt to generate multiple answers to a ClientHello, I
      think. I&#39;m sure it&#39;s well defined how to respond to that - it=
&#39;s an
      obvious attack. Only one leg of the fork would ever succeed, I
      assume.</div></div></blockquote><div><br></div><div>The biggest issue=
 with reusing ClientHello in case of forking would be reusing client random=
 across multiple connections. Unless I am missing something, this should be=
 generally safe.</div><div><br></div><div>Another issue is what will happen=
 if two DTLS association are established for a single m=3D line (i.e. RTCP =
bundle is not used). Does this mean the same ClientHello is used for both c=
onnection?=C2=A0</div><div><br></div><div>Regards,</div><div><div><div clas=
s=3D"gmail_signature">_____________<br>Roman Shpount</div></div></div><div>=
<br></div></div></div></div>

--089e01494ff2575a7c052fbcb88d--


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From: "Drage, Keith (Nokia - GB)" <keith.drage@nokia.com>
To: EXT Eric Rescorla <ekr@rtfm.com>, Harald Alvestrand <harald@alvestrand.no>
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--_000_949EF20990823C4C85C18D59AA11AD8BADEB9E7DFR712WXCHMBA11z_--


From nobody Tue Apr  5 07:20:09 2016
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From: "Drage, Keith (Nokia - GB)" <keith.drage@nokia.com>
To: "rtcweb@ietf.org" <rtcweb@ietf.org>, Justin Uberti <juberti@google.com>
Thread-Topic: [rtcweb] Positioning draft-ietf-rtcweb-ip-address as a standards-track document
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From nobody Tue Apr  5 13:19:14 2016
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From: "Drage, Keith (Nokia - GB)" <keith.drage@nokia.com>
To: "rtcweb@ietf.org" <rtcweb@ietf.org>
Thread-Topic: IP gathering permission
Thread-Index: AdGPeGd58V/HGoaQSbeQrBprO/9n+Q==
Date: Tue, 5 Apr 2016 20:19:07 +0000
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Subject: [rtcweb] IP gathering permission
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The discussion in the WG (Tuesday p.m. session) that I have just listened t=
o causes me concern.

Any system that gathers personal information needs to meet the local regula=
tory framework. For example, throughout most of Europe that is the privacy =
directive and its local implementation in various European countries. While=
 I don't expect people to analyse the protocol in terms of each local regul=
atory framework, I believe we do need to provide the tools to allow these t=
o be met.=20

Ultimately, it is not just a matter of whether the user can comprehend what=
 he has been asked, but whether the regulator is also happy that the needs =
of the local regulation have been met. Many times that will come down to th=
e view of the local regulator as to whether the situation has been made suf=
ficiently clear to the end user.

Certainly we need to avoid any assumption that might be provided by the dra=
ft that because information A has been released, that information B is also=
 OK from a privacy consideration.

Maybe we need a statement that says any selection needs to meet the privacy=
 framework imposed on the system, by regulation, by contract, etc.

Regards

Keith


From nobody Tue Apr  5 13:25:57 2016
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Subject: [rtcweb] Comments on draft-ietf-rtcweb-fec-03
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Section 4.2:

Support for receiving packets with FEC in them is mandatory for Opus. 
The useinbandfec=1 parameter indicates that the receiver is actually 
prepared to *use* that data (and thus that the bandwidth overhead is not 
wasted).

The reference to the RTP draft should probably also be updated to point 
to the final RFC.

I'd propose the following text:

'For Opus, a receiver can indicate that it is prepared to use incoming 
FEC data with the "useinbandfec=1" parameter, as specified in [RFC7587], 
Section 6.1.'

Section 7:

I think it's a little unclear what MUST be supported here, particularly 
for in-band FEC. I believe both Opus and AMR require that all receivers 
can handle packets with in-band FEC. Are receivers that are WebRTC 
endpoints required to actually do something with the data they receive 
(and, e.g., to always set useinbandfec=1 for Opus)? Are senders required 
to include in-band FEC whenever the receiver asks for it (and network 
conditions/application requests support it, as indicated in Section 8)? 
I.e., is this mandatory to support (which in the case of the explicitly 
named codecs, is already mandatory), or mandatory to use? It might be 
good to clarify.

Whether or not these comments are addressed, I think this document is 
ready for WGLC.


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To: "rtcweb@ietf.org" <rtcweb@ietf.org>, Justin Uberti <juberti@google.com>
From: Magnus Westerlund <magnus.westerlund@ericsson.com>
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Subject: [rtcweb] AMR and max-red in draft-ietf-rtcweb-fec
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Hi,

As raised in the WG session there are where an open issue if there need 
to be any specification of the AMR/AMR-WB max-red parameter.

So RFC4867 do have the following on offer/answer usage of the parameter:

       -  The parameter "max-red" is a stream property parameter.  For
          send-only or send-recv unicast media streams, the parameter
          declares the limitation on redundancy that the stream sender
          will use.  For recvonly streams, it indicates the desired value
          for the stream sent to the receiver.  The answerer MAY change
          the value, but is RECOMMENDED to use the same limitation as the
          offer declares.  In the case of multicast, the offerer MAY
          declare a limitation; this SHALL be answered using the same
          value.  A media sender using this payload format is RECOMMENDED
          to always include the "max-red" parameter.  This information is
          likely to simplify the media stream handling in the receiver.
          This is especially true if no redundancy will be used, in which
          case "max-red" is set to 0.  As this parameter was not defined
          originally, some senders will not declare this parameter even
          if it will limit or not send redundancy at all.

Most notable is that media senders are RECOMMENDED to include it. Also 
note that it is the sender that indicates the limitation it intends to 
adhere to for the streams using this RTP payload type.

I also like to note that 3GPP in its specification for its Multimedia 
Telephony, has specified that one shall not use a value larger than 220 
ms, and the value should be a multiple of 20 ms.  Please see 3GPP TS 
26.114 version 13.3.0
http://www.3gpp.org/ftp/Specs/archive/26_series/26.114/26114-d30.zip

So from my perspective I think a WebRTC endpoint simply should follow 
the AMR RTP Payload specification when supported and include the 
parameter to indicate the actual upper range of the redundancy it 
intended to use. It is good to know about the limitation that 3GPP has 
defined as one likely reason to support AMR/AMR-WB is to be able to 
interoperate with 3GPP Services.

Thus I would suggest that a small addition to the Section 4.3 text and 
corrections in the first sentence:

OLD:
    For AMR/AMR-WB, support for redundant encoding, and the maximum
    supported depth, are controlled by the 'max-red' parameter, as
    specified in [RFC4867], Section 8.1.

NEW:
    For AMR/AMR-WB, support for redundant encoding, and the maximum
    used depth, is indicated by the 'max-red' parameter, as
    specified in [RFC4867], Section 8.1 and 8.2. The 3GPP Multimedia
    Telephony services specifies a maximum value of 220 ms for
    'max-red', see [3GPP.26.114].


New Informational Reference:

[3GPP.26.114]

<reference anchor="3GPP.26.114"><front><title>IP Multimedia Subsystem 
(IMS); Multimedia telephony; Media handling and 
interaction</title><author><organization>3GPP</organization></author><date 
day="17" month="March" year="2016"/></front><seriesInfo name="3GPP TS" 
value="26.114 13.3.0"/><format type="HTML" 
target="http://www.3gpp.org/ftp/Specs/html-info/26114.htm"/></reference>

Opinions?

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] AMR and max-red in draft-ietf-rtcweb-fec
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Hello Magnus

I support your proposal. I would even suggest to relate this to the draft-i=
etf-rtcweb-audio-codecs-for-interop (where more information about use of AM=
R and AMR-WB is provided ) and specify more explicitly to include it in the=
 offer/answer with an integer value multiple of 20ms not exceeding 220ms.
However, we can stay with your initial proposal if you or anyone else think=
s it is preferable.

Kind regards

St=E9phane

OLD:
    For AMR/AMR-WB, support for redundant encoding, and the maximum
    supported depth, are controlled by the 'max-red' parameter, as
    specified in [RFC4867], Section 8.1.

NEW:
    For AMR/AMR-WB, support for redundant encoding, and the maximum
    used depth, is indicated by the 'max-red' parameter, as
    specified in [RFC4867], Section 8.1 and 8.2. The 3GPP Multimedia
    Telephony services specifies a maximum value of 220 ms for
    'max-red', see [3GPP.26.114]. If AMR or AMR-WB are offered for interope=
rability purpose,
   see [draft-ietf-rtcweb-audio-codecs-for-interop], the max-red parameter =
must be included
  in the offer and answer and set to an integer value multiple of 20ms.

-----Message d'origine-----
De=A0: rtcweb [mailto:rtcweb-bounces@ietf.org] De la part de Magnus Westerl=
und
Envoy=E9=A0: mardi 5 avril 2016 23:02
=C0=A0: rtcweb@ietf.org; Justin Uberti
Objet=A0: [rtcweb] AMR and max-red in draft-ietf-rtcweb-fec

Hi,

As raised in the WG session there are where an open issue if there need to =
be any specification of the AMR/AMR-WB max-red parameter.

So RFC4867 do have the following on offer/answer usage of the parameter:

       -  The parameter "max-red" is a stream property parameter.  For
          send-only or send-recv unicast media streams, the parameter
          declares the limitation on redundancy that the stream sender
          will use.  For recvonly streams, it indicates the desired value
          for the stream sent to the receiver.  The answerer MAY change
          the value, but is RECOMMENDED to use the same limitation as the
          offer declares.  In the case of multicast, the offerer MAY
          declare a limitation; this SHALL be answered using the same
          value.  A media sender using this payload format is RECOMMENDED
          to always include the "max-red" parameter.  This information is
          likely to simplify the media stream handling in the receiver.
          This is especially true if no redundancy will be used, in which
          case "max-red" is set to 0.  As this parameter was not defined
          originally, some senders will not declare this parameter even
          if it will limit or not send redundancy at all.

Most notable is that media senders are RECOMMENDED to include it. Also note=
 that it is the sender that indicates the limitation it intends to adhere t=
o for the streams using this RTP payload type.

I also like to note that 3GPP in its specification for its Multimedia Telep=
hony, has specified that one shall not use a value larger than 220 ms, and =
the value should be a multiple of 20 ms.  Please see 3GPP TS
26.114 version 13.3.0
http://www.3gpp.org/ftp/Specs/archive/26_series/26.114/26114-d30.zip

So from my perspective I think a WebRTC endpoint simply should follow the A=
MR RTP Payload specification when supported and include the parameter to in=
dicate the actual upper range of the redundancy it intended to use. It is g=
ood to know about the limitation that 3GPP has defined as one likely reason=
 to support AMR/AMR-WB is to be able to interoperate with 3GPP Services.

Thus I would suggest that a small addition to the Section 4.3 text and corr=
ections in the first sentence:

OLD:
    For AMR/AMR-WB, support for redundant encoding, and the maximum
    supported depth, are controlled by the 'max-red' parameter, as
    specified in [RFC4867], Section 8.1.

NEW:
    For AMR/AMR-WB, support for redundant encoding, and the maximum
    used depth, is indicated by the 'max-red' parameter, as
    specified in [RFC4867], Section 8.1 and 8.2. The 3GPP Multimedia
    Telephony services specifies a maximum value of 220 ms for
    'max-red', see [3GPP.26.114].


New Informational Reference:

[3GPP.26.114]

<reference anchor=3D"3GPP.26.114"><front><title>IP Multimedia Subsystem (IM=
S); Multimedia telephony; Media handling and interaction</title><author><or=
ganization>3GPP</organization></author><date
day=3D"17" month=3D"March" year=3D"2016"/></front><seriesInfo name=3D"3GPP =
TS"=20
value=3D"26.114 13.3.0"/><format type=3D"HTML"=20
target=3D"http://www.3gpp.org/ftp/Specs/html-info/26114.htm"/></reference>

Opinions?

Cheers

Magnus Westerlund

----------------------------------------------------------------------
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
----------------------------------------------------------------------

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

___________________________________________________________________________=
______________________________________________

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Orange decline toute responsabilite si ce message a ete altere, deforme ou =
falsifie. Merci.

This message and its attachments may contain confidential or privileged inf=
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they should not be distributed, used or copied without authorisation.
If you have received this email in error, please notify the sender and dele=
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As emails may be altered, Orange is not liable for messages that have been =
modified, changed or falsified.
Thank you.


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From: <stephane.proust@orange.com>
To: Magnus Westerlund <magnus.westerlund@ericsson.com>, "rtcweb@ietf.org" <rtcweb@ietf.org>, Justin Uberti <juberti@google.com>
Thread-Topic: [rtcweb] AMR and max-red in draft-ietf-rtcweb-fec
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Sorry, the last part of the sentence in the proposal was missing  :  the ma=
x-red parameter must be included
  in the offer and answer and set to an integer value multiple of 20ms not =
exceeding 220ms.

OLD:
    For AMR/AMR-WB, support for redundant encoding, and the maximum
    supported depth, are controlled by the 'max-red' parameter, as
    specified in [RFC4867], Section 8.1.

NEW:
    For AMR/AMR-WB, support for redundant encoding, and the maximum
    used depth, is indicated by the 'max-red' parameter, as
    specified in [RFC4867], Section 8.1 and 8.2. The 3GPP Multimedia
    Telephony services specifies a maximum value of 220 ms for
    'max-red', see [3GPP.26.114]. If AMR or AMR-WB are offered for interope=
rability purpose,
   see [draft-ietf-rtcweb-audio-codecs-for-interop], the max-red parameter =
must be included
  in the offer and answer and set to an integer value multiple of 20ms not =
exceeding 220ms.

-----Message d'origine-----
De=A0: rtcweb [mailto:rtcweb-bounces@ietf.org] De la part de stephane.prous=
t@orange.com
Envoy=E9=A0: mercredi 6 avril 2016 10:18
=C0=A0: Magnus Westerlund; rtcweb@ietf.org; Justin Uberti
Objet=A0: Re: [rtcweb] AMR and max-red in draft-ietf-rtcweb-fec

Hello Magnus

I support your proposal. I would even suggest to relate this to the draft-i=
etf-rtcweb-audio-codecs-for-interop (where more information about use of AM=
R and AMR-WB is provided ) and specify more explicitly to include it in the=
 offer/answer with an integer value multiple of 20ms not exceeding 220ms.
However, we can stay with your initial proposal if you or anyone else think=
s it is preferable.

Kind regards

St=E9phane

OLD:
    For AMR/AMR-WB, support for redundant encoding, and the maximum
    supported depth, are controlled by the 'max-red' parameter, as
    specified in [RFC4867], Section 8.1.

NEW:
    For AMR/AMR-WB, support for redundant encoding, and the maximum
    used depth, is indicated by the 'max-red' parameter, as
    specified in [RFC4867], Section 8.1 and 8.2. The 3GPP Multimedia
    Telephony services specifies a maximum value of 220 ms for
    'max-red', see [3GPP.26.114]. If AMR or AMR-WB are offered for interope=
rability purpose,
   see [draft-ietf-rtcweb-audio-codecs-for-interop], the max-red parameter =
must be included
  in the offer and answer and set to an integer value multiple of 20ms.

-----Message d'origine-----
De=A0: rtcweb [mailto:rtcweb-bounces@ietf.org] De la part de Magnus Westerl=
und Envoy=E9=A0: mardi 5 avril 2016 23:02 =C0=A0: rtcweb@ietf.org; Justin U=
berti Objet=A0: [rtcweb] AMR and max-red in draft-ietf-rtcweb-fec

Hi,

As raised in the WG session there are where an open issue if there need to =
be any specification of the AMR/AMR-WB max-red parameter.

So RFC4867 do have the following on offer/answer usage of the parameter:

       -  The parameter "max-red" is a stream property parameter.  For
          send-only or send-recv unicast media streams, the parameter
          declares the limitation on redundancy that the stream sender
          will use.  For recvonly streams, it indicates the desired value
          for the stream sent to the receiver.  The answerer MAY change
          the value, but is RECOMMENDED to use the same limitation as the
          offer declares.  In the case of multicast, the offerer MAY
          declare a limitation; this SHALL be answered using the same
          value.  A media sender using this payload format is RECOMMENDED
          to always include the "max-red" parameter.  This information is
          likely to simplify the media stream handling in the receiver.
          This is especially true if no redundancy will be used, in which
          case "max-red" is set to 0.  As this parameter was not defined
          originally, some senders will not declare this parameter even
          if it will limit or not send redundancy at all.

Most notable is that media senders are RECOMMENDED to include it. Also note=
 that it is the sender that indicates the limitation it intends to adhere t=
o for the streams using this RTP payload type.

I also like to note that 3GPP in its specification for its Multimedia Telep=
hony, has specified that one shall not use a value larger than 220 ms, and =
the value should be a multiple of 20 ms.  Please see 3GPP TS
26.114 version 13.3.0
http://www.3gpp.org/ftp/Specs/archive/26_series/26.114/26114-d30.zip

So from my perspective I think a WebRTC endpoint simply should follow the A=
MR RTP Payload specification when supported and include the parameter to in=
dicate the actual upper range of the redundancy it intended to use. It is g=
ood to know about the limitation that 3GPP has defined as one likely reason=
 to support AMR/AMR-WB is to be able to interoperate with 3GPP Services.

Thus I would suggest that a small addition to the Section 4.3 text and corr=
ections in the first sentence:

OLD:
    For AMR/AMR-WB, support for redundant encoding, and the maximum
    supported depth, are controlled by the 'max-red' parameter, as
    specified in [RFC4867], Section 8.1.

NEW:
    For AMR/AMR-WB, support for redundant encoding, and the maximum
    used depth, is indicated by the 'max-red' parameter, as
    specified in [RFC4867], Section 8.1 and 8.2. The 3GPP Multimedia
    Telephony services specifies a maximum value of 220 ms for
    'max-red', see [3GPP.26.114].


New Informational Reference:

[3GPP.26.114]

<reference anchor=3D"3GPP.26.114"><front><title>IP Multimedia Subsystem (IM=
S); Multimedia telephony; Media handling and interaction</title><author><or=
ganization>3GPP</organization></author><date
day=3D"17" month=3D"March" year=3D"2016"/></front><seriesInfo name=3D"3GPP =
TS"=20
value=3D"26.114 13.3.0"/><format type=3D"HTML"=20
target=3D"http://www.3gpp.org/ftp/Specs/html-info/26114.htm"/></reference>

Opinions?

Cheers

Magnus Westerlund

----------------------------------------------------------------------
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
----------------------------------------------------------------------

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

___________________________________________________________________________=
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This message and its attachments may contain confidential or privileged inf=
ormation that may be protected by law; they should not be distributed, used=
 or copied without authorisation.
If you have received this email in error, please notify the sender and dele=
te this message and its attachments.
As emails may be altered, Orange is not liable for messages that have been =
modified, changed or falsified.
Thank you.

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

___________________________________________________________________________=
______________________________________________

Ce message et ses pieces jointes peuvent contenir des informations confiden=
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pas etre diffuses, exploites ou copies sans autorisation. Si vous avez recu=
 ce message par erreur, veuillez le signaler
a l'expediteur et le detruire ainsi que les pieces jointes. Les messages el=
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Orange decline toute responsabilite si ce message a ete altere, deforme ou =
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This message and its attachments may contain confidential or privileged inf=
ormation that may be protected by law;
they should not be distributed, used or copied without authorisation.
If you have received this email in error, please notify the sender and dele=
te this message and its attachments.
As emails may be altered, Orange is not liable for messages that have been =
modified, changed or falsified.
Thank you.


From nobody Wed Apr  6 06:18:48 2016
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A New Internet-Draft is available from the on-line Internet-Drafts directories.
This draft is a work item of the Real-Time Communication in WEB-browsers of the IETF.

        Title           : Application Layer Protocol Negotiation for Web Real-Time Communications (WebRTC) 
        Author          : Martin Thomson
	Filename        : draft-ietf-rtcweb-alpn-03.txt
	Pages           : 7
	Date            : 2016-04-05

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 from web applications.


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

There's also a htmlized version available at:
https://tools.ietf.org/html/draft-ietf-rtcweb-alpn-03

A diff from the previous version is available at:
https://www.ietf.org/rfcdiff?url2=draft-ietf-rtcweb-alpn-03


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

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


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--001a1141dca2c96fe4052fd87563
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Section 3.3

   Note that this mechanism can only carry redundancy information for the
   immediately preceding packet; as such the decoder cannot fully
   recover multiple consecutive lost packets.


[BA] In our experience, this has turned out to be a significant problem

on wireless networks where burst loss is common.


Section 4.1


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


   Because of the lower packet rate of audio encodings, usually a single
   packet per frame, use of a separate FEC stream comes with a higher
   overhead than other mechanisms, and therefore is NOT RECOMMENDED.


[BA] In our implementation, we have found that using external FEC

supporting burst loss resilience provides audio quality superior

to RFC 2198 redundancy with less overhead, iff the level of

protection can be dynamically adjusted (e.g. it is only used if

needed). Since that is recommended in Section 8, NOT RECOMMENDED

seems too harsh. Maybe optional?


Section 5.1


   For video content, use of a separate FEC stream with the RTP payload
   format described in [I-D.ietf-payload-flexible-fec-scheme
<https://tools.ietf.org/html/draft-ietf-rtcweb-fec-03#ref-I-D.ietf-payload-flexible-fec-scheme>]
is
   RECOMMENDED.


[BA] The above RECOMMENDED seems to be contradicted by a MUST in

Section 7.


Section 7


 To support the functionality recommended above, implementations MUST
   support the relevant mechanisms for their supported audio codecs, as
   described in Section 4
<https://tools.ietf.org/html/draft-ietf-rtcweb-fec-03#section-4>, and
the general FEC mechanism described in
   [I-D.ietf-payload-flexible-fec-scheme
<https://tools.ietf.org/html/draft-ietf-rtcweb-fec-03#ref-I-D.ietf-payload-flexible-fec-scheme>].

   Implementations MAY support additional FEC mechanisms if desired,
   e.g.  [RFC5109 <https://tools.ietf.org/html/rfc5109>].


[BA] The above paragraph seems to indicate that flex fec is a MUST,
rather than merely RECOMMENDED. Personally, I think a MUST for
flex fec is too much, and that since existing implementations support
ulpfec, this should be upgraded to RECOMMENDED.

RECOMMENDED seems more suitable for flexible fec since we don't have much

implementation experience yet.  My recommendation would be to wait to

publish the flex fec document until we had some implementation

experience, since if the flex fec goes to RFC before that we could

have some surprises (including late IPR declarations) that might lead

us to regret publishing early.


Section 8


   Given this, WebRTC
   implementations SHOULD only transmit the amount of FEC needed to
   protect against the observed packet loss (which can be determined,
   e.g., by monitoring transmit packet loss data from RTCP Receiver
   Reports [RFC3550 <https://tools.ietf.org/html/rfc3550>]), or the
application indicates it is willing to pay
   a quality penalty to proactively avoid losses.


[BA] Very much in agreement with this - particularly since RTX can
be a quite viable robustness mechanism, particularly when combined
with temporal scalability. Where RTX is not sufficient (e.g. RTT
is too high), it might be worth mentioning selective FEC protection
(e.g. of the base layer only). Since VP8, VP9 and future video
codecs can all benefit from this, it's not that esoteric.

--001a1141dca2c96fe4052fd87563
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<div dir=3D"ltr">Section 3.3<div><br><div><pre class=3D"" style=3D"font-siz=
e:13.3333px;margin-top:0px;margin-bottom:0px;color:rgb(0,0,0)">   Note that=
 this mechanism can only carry redundancy information for the
   immediately preceding packet; as such the decoder cannot fully
   recover multiple consecutive lost packets.</pre><pre class=3D"" style=3D=
"font-size:13.3333px;margin-top:0px;margin-bottom:0px;color:rgb(0,0,0)"><br=
></pre><pre class=3D"" style=3D"font-size:13.3333px;margin-top:0px;margin-b=
ottom:0px;color:rgb(0,0,0)">[BA] In our experience, this has turned out to =
be a significant problem</pre><pre class=3D"" style=3D"font-size:13.3333px;=
margin-top:0px;margin-bottom:0px;color:rgb(0,0,0)">on wireless networks whe=
re burst loss is common. </pre><pre class=3D"" style=3D"font-size:13.3333px=
;margin-top:0px;margin-bottom:0px;color:rgb(0,0,0)"><br></pre><pre class=3D=
"" style=3D"font-size:13.3333px;margin-top:0px;margin-bottom:0px;color:rgb(=
0,0,0)">Section 4.1</pre><pre class=3D"" style=3D"font-size:13.3333px;margi=
n-top:0px;margin-bottom:0px;color:rgb(0,0,0)"><br></pre><pre class=3D"" sty=
le=3D"font-size:13.3333px;margin-top:0px;margin-bottom:0px;color:rgb(0,0,0)=
"><pre class=3D"" style=3D"font-size:13.3333px;margin-top:0px;margin-bottom=
:0px">   When using the Opus codec, use of the built-in Opus FEC mechanism =
is
   RECOMMENDED.  This provides reasonable protection of the audio stream
   against typical losses, with modest overhead.  Note that as indicated
   above the built-in Opus FEC only provides single-frame redundancy; if
   multi-packet protection is needed, the built-in FEC should be
   combined with [<a href=3D"https://tools.ietf.org/html/rfc2198" title=3D"=
&quot;RTP Payload for Redundant Audio Data&quot;">RFC2198</a>] redundancy t=
o protect the N-2th, N-3rd, etc.
   packets.</pre><pre class=3D"" style=3D"font-size:13.3333px;margin-top:0p=
x;margin-bottom:0px"><br></pre><pre class=3D"" style=3D"font-size:13.3333px=
;margin-top:0px;margin-bottom:0px"><pre class=3D"" style=3D"font-size:13.33=
33px;margin-top:0px;margin-bottom:0px">   Because of the lower packet rate =
of audio encodings, usually a single
   packet per frame, use of a separate FEC stream comes with a higher
   overhead than other mechanisms, and therefore is NOT RECOMMENDED.</pre><=
/pre><pre class=3D"" style=3D"font-size:13.3333px;margin-top:0px;margin-bot=
tom:0px"><br></pre><pre class=3D"" style=3D"font-size:13.3333px;margin-top:=
0px;margin-bottom:0px">[BA] In our implementation, we have found that using=
 external FEC </pre><pre class=3D"" style=3D"font-size:13.3333px;margin-top=
:0px;margin-bottom:0px">supporting burst loss resilience provides audio qua=
lity superior</pre><pre class=3D"" style=3D"font-size:13.3333px;margin-top:=
0px;margin-bottom:0px">to RFC 2198 redundancy with less overhead,=C2=A0iff =
the level of</pre><pre class=3D"" style=3D"font-size:13.3333px;margin-top:0=
px;margin-bottom:0px">protection can be dynamically adjusted (e.g. it is on=
ly used if</pre><pre class=3D"" style=3D"font-size:13.3333px;margin-top:0px=
;margin-bottom:0px">needed). Since that is recommended in Section 8, NOT RE=
COMMENDED=C2=A0</pre><pre class=3D"" style=3D"font-size:13.3333px;margin-to=
p:0px;margin-bottom:0px">seems too harsh. Maybe optional?</pre><pre class=
=3D"" style=3D"font-size:13.3333px;margin-top:0px;margin-bottom:0px"><br></=
pre><pre class=3D"" style=3D"font-size:13.3333px;margin-top:0px;margin-bott=
om:0px">Section 5.1</pre><pre class=3D"" style=3D"font-size:13.3333px;margi=
n-top:0px;margin-bottom:0px"><br></pre><pre class=3D"" style=3D"font-size:1=
3.3333px;margin-top:0px;margin-bottom:0px"><pre class=3D"" style=3D"font-si=
ze:13.3333px;margin-top:0px;margin-bottom:0px"><pre class=3D"" style=3D"fon=
t-size:13.3333px;margin-top:0px;margin-bottom:0px">   For video content, us=
e of a separate FEC stream with the RTP payload
   format described in [<a href=3D"https://tools.ietf.org/html/draft-ietf-r=
tcweb-fec-03#ref-I-D.ietf-payload-flexible-fec-scheme">I-D.ietf-payload-fle=
xible-fec-scheme</a>] is
   RECOMMENDED. </pre></pre><pre class=3D"" style=3D"font-size:13.3333px;ma=
rgin-top:0px;margin-bottom:0px"><br></pre><pre class=3D"" style=3D"font-siz=
e:13.3333px;margin-top:0px;margin-bottom:0px">[BA] The above RECOMMENDED se=
ems to be contradicted by a MUST in</pre><pre class=3D"" style=3D"font-size=
:13.3333px;margin-top:0px;margin-bottom:0px">Section 7. </pre></pre><pre cl=
ass=3D"" style=3D"font-size:13.3333px;margin-top:0px;margin-bottom:0px"><br=
></pre><pre class=3D"" style=3D"font-size:13.3333px;margin-top:0px;margin-b=
ottom:0px"><font face=3D"arial, sans-serif">Section 7</font></pre><pre clas=
s=3D"" style=3D"font-size:13.3333px;margin-top:0px;margin-bottom:0px"><font=
 face=3D"arial, sans-serif"><br></font></pre><pre class=3D"" style=3D"font-=
size:13.3333px;margin-top:0px;margin-bottom:0px"><pre class=3D"" style=3D"f=
ont-size:13.3333px;margin-top:0px;margin-bottom:0px"> To support the functi=
onality recommended above, implementations MUST
   support the relevant mechanisms for their supported audio codecs, as
   described in <a href=3D"https://tools.ietf.org/html/draft-ietf-rtcweb-fe=
c-03#section-4">Section 4</a>, and the general FEC mechanism described in
   [<a href=3D"https://tools.ietf.org/html/draft-ietf-rtcweb-fec-03#ref-I-D=
.ietf-payload-flexible-fec-scheme">I-D.ietf-payload-flexible-fec-scheme</a>=
].

   Implementations MAY support additional FEC mechanisms if desired,
   e.g.  [<a href=3D"https://tools.ietf.org/html/rfc5109" title=3D"&quot;RT=
P Payload Format for Generic Forward Error Correction&quot;">RFC5109</a>].
</pre><div><br></div><div>[BA] The above paragraph seems to indicate that f=
lex fec is a MUST,</div><div>rather than merely RECOMMENDED.  Personally, I=
 think a MUST for</div><div>flex fec is too much, and that since existing i=
mplementations support</div><div>ulpfec, this should be upgraded to RECOMME=
NDED.=C2=A0</div><div><br></div><pre class=3D"" style=3D"font-size:13.3333p=
x;margin-top:0px;margin-bottom:0px"><pre class=3D"" style=3D"font-size:13.3=
333px;margin-top:0px;margin-bottom:0px">RECOMMENDED seems more suitable for=
 flexible fec since we don&#39;t have much</pre><pre class=3D"" style=3D"fo=
nt-size:13.3333px;margin-top:0px;margin-bottom:0px">implementation experien=
ce yet.  My recommendation would be to wait to</pre><pre class=3D"" style=
=3D"font-size:13.3333px;margin-top:0px;margin-bottom:0px">publish the flex =
fec document until we had some implementation</pre><pre class=3D"" style=3D=
"font-size:13.3333px;margin-top:0px;margin-bottom:0px">experience, since if=
 the flex fec goes to RFC before that we could</pre><pre class=3D"" style=
=3D"font-size:13.3333px;margin-top:0px;margin-bottom:0px">have some surpris=
es (including late IPR declarations) that might lead</pre><pre class=3D"" s=
tyle=3D"font-size:13.3333px;margin-top:0px;margin-bottom:0px">us to regret =
publishing early. </pre><pre class=3D"" style=3D"font-size:13.3333px;margin=
-top:0px;margin-bottom:0px"><br></pre><pre class=3D"" style=3D"font-size:13=
.3333px;margin-top:0px;margin-bottom:0px">Section 8</pre><pre class=3D"" st=
yle=3D"font-size:13.3333px;margin-top:0px;margin-bottom:0px"><br></pre><pre=
 class=3D"" style=3D"font-size:13.3333px;margin-top:0px;margin-bottom:0px">=
<pre class=3D"" style=3D"font-size:13.3333px;margin-top:0px;margin-bottom:0=
px">   Given this, WebRTC
   implementations SHOULD only transmit the amount of FEC needed to
   protect against the observed packet loss (which can be determined,
   e.g., by monitoring transmit packet loss data from RTCP Receiver
   Reports [<a href=3D"https://tools.ietf.org/html/rfc3550" title=3D"&quot;=
RTP: A Transport Protocol for Real-Time Applications&quot;">RFC3550</a>]), =
or the application indicates it is willing to pay
   a quality penalty to proactively avoid losses.</pre></pre></pre><div><br=
></div><div>[BA] Very much in agreement with this - particularly since RTX =
can</div><div>be a quite viable robustness mechanism, particularly when com=
bined</div><div>with temporal scalability.  Where RTX is not sufficient (e.=
g. RTT</div><div>is too high), it might be worth mentioning selective FEC p=
rotection</div><div>(e.g. of the base layer only).   Since VP8, VP9 and fut=
ure video</div><div>codecs can all benefit from this, it&#39;s not that eso=
teric.</div></pre><pre class=3D"" style=3D"font-size:13.3333px;margin-top:0=
px;margin-bottom:0px"><font face=3D"arial, sans-serif"><br></font></pre><pr=
e class=3D"" style=3D"font-size:13.3333px;margin-top:0px;margin-bottom:0px"=
> </pre><pre class=3D"" style=3D"font-size:13.3333px;margin-top:0px;margin-=
bottom:0px"><br></pre><pre class=3D"" style=3D"font-size:13.3333px;margin-t=
op:0px;margin-bottom:0px"><br></pre></pre></div></div></div>

--001a1141dca2c96fe4052fd87563--


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Subject: [rtcweb] Last Call: <draft-ietf-rtcweb-alpn-03.txt> (Application Layer Protocol Negotiation for Web Real-Time Communications (WebRTC)) to Proposed Standard
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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:
- 'Application Layer Protocol Negotiation for Web Real-Time
   Communications (WebRTC)'
  <draft-ietf-rtcweb-alpn-03.txt> as Proposed Standard

The IESG plans to make a decision in the next few weeks, and solicits
final comments on this action. Please send substantive comments to the
ietf@ietf.org mailing lists by 2016-04-21. 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


   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 from web applications.




The file can be obtained via
https://datatracker.ietf.org/doc/draft-ietf-rtcweb-alpn/

IESG discussion can be tracked via
https://datatracker.ietf.org/doc/draft-ietf-rtcweb-alpn/ballot/


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



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To: <stephane.proust@orange.com>, "rtcweb@ietf.org" <rtcweb@ietf.org>, Justin Uberti <juberti@google.com>
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Subject: Re: [rtcweb] AMR and max-red in draft-ietf-rtcweb-fec
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Hi,

I am generally fine with Stephane's proposal. I do note that the "must" 
in the last sentence probably need to be a "MUST", otherwise if the 
intention is just to point to requirements that exists, then please 
reformulate this to not use RFC2119 terms, even in lower cases.

Cheers

Magnus

Den 2016-04-06 kl. 06:28, skrev stephane.proust@orange.com:
> Sorry, the last part of the sentence in the proposal was missing  :  the max-red parameter must be included
>    in the offer and answer and set to an integer value multiple of 20ms not exceeding 220ms.
>
> OLD:
>      For AMR/AMR-WB, support for redundant encoding, and the maximum
>      supported depth, are controlled by the 'max-red' parameter, as
>      specified in [RFC4867], Section 8.1.
>
> NEW:
>      For AMR/AMR-WB, support for redundant encoding, and the maximum
>      used depth, is indicated by the 'max-red' parameter, as
>      specified in [RFC4867], Section 8.1 and 8.2. The 3GPP Multimedia
>      Telephony services specifies a maximum value of 220 ms for
>      'max-red', see [3GPP.26.114]. If AMR or AMR-WB are offered for interoperability purpose,
>     see [draft-ietf-rtcweb-audio-codecs-for-interop], the max-red parameter must be included
>    in the offer and answer and set to an integer value multiple of 20ms not exceeding 220ms.
>
> -----Message d'origine-----
> De : rtcweb [mailto:rtcweb-bounces@ietf.org] De la part de stephane.proust@orange.com
> Envoyé : mercredi 6 avril 2016 10:18
> Ŕ : Magnus Westerlund; rtcweb@ietf.org; Justin Uberti
> Objet : Re: [rtcweb] AMR and max-red in draft-ietf-rtcweb-fec
>
> Hello Magnus
>
> I support your proposal. I would even suggest to relate this to the draft-ietf-rtcweb-audio-codecs-for-interop (where more information about use of AMR and AMR-WB is provided ) and specify more explicitly to include it in the offer/answer with an integer value multiple of 20ms not exceeding 220ms.
> However, we can stay with your initial proposal if you or anyone else thinks it is preferable.
>
> Kind regards
>
> Stéphane
>
> OLD:
>      For AMR/AMR-WB, support for redundant encoding, and the maximum
>      supported depth, are controlled by the 'max-red' parameter, as
>      specified in [RFC4867], Section 8.1.
>
> NEW:
>      For AMR/AMR-WB, support for redundant encoding, and the maximum
>      used depth, is indicated by the 'max-red' parameter, as
>      specified in [RFC4867], Section 8.1 and 8.2. The 3GPP Multimedia
>      Telephony services specifies a maximum value of 220 ms for
>      'max-red', see [3GPP.26.114]. If AMR or AMR-WB are offered for interoperability purpose,
>     see [draft-ietf-rtcweb-audio-codecs-for-interop], the max-red parameter must be included
>    in the offer and answer and set to an integer value multiple of 20ms.
>
> -----Message d'origine-----
> De : rtcweb [mailto:rtcweb-bounces@ietf.org] De la part de Magnus Westerlund Envoyé : mardi 5 avril 2016 23:02 Ŕ : rtcweb@ietf.org; Justin Uberti Objet : [rtcweb] AMR and max-red in draft-ietf-rtcweb-fec
>
> Hi,
>
> As raised in the WG session there are where an open issue if there need to be any specification of the AMR/AMR-WB max-red parameter.
>
> So RFC4867 do have the following on offer/answer usage of the parameter:
>
>         -  The parameter "max-red" is a stream property parameter.  For
>            send-only or send-recv unicast media streams, the parameter
>            declares the limitation on redundancy that the stream sender
>            will use.  For recvonly streams, it indicates the desired value
>            for the stream sent to the receiver.  The answerer MAY change
>            the value, but is RECOMMENDED to use the same limitation as the
>            offer declares.  In the case of multicast, the offerer MAY
>            declare a limitation; this SHALL be answered using the same
>            value.  A media sender using this payload format is RECOMMENDED
>            to always include the "max-red" parameter.  This information is
>            likely to simplify the media stream handling in the receiver.
>            This is especially true if no redundancy will be used, in which
>            case "max-red" is set to 0.  As this parameter was not defined
>            originally, some senders will not declare this parameter even
>            if it will limit or not send redundancy at all.
>
> Most notable is that media senders are RECOMMENDED to include it. Also note that it is the sender that indicates the limitation it intends to adhere to for the streams using this RTP payload type.
>
> I also like to note that 3GPP in its specification for its Multimedia Telephony, has specified that one shall not use a value larger than 220 ms, and the value should be a multiple of 20 ms.  Please see 3GPP TS
> 26.114 version 13.3.0
> http://www.3gpp.org/ftp/Specs/archive/26_series/26.114/26114-d30.zip
>
> So from my perspective I think a WebRTC endpoint simply should follow the AMR RTP Payload specification when supported and include the parameter to indicate the actual upper range of the redundancy it intended to use. It is good to know about the limitation that 3GPP has defined as one likely reason to support AMR/AMR-WB is to be able to interoperate with 3GPP Services.
>
> Thus I would suggest that a small addition to the Section 4.3 text and corrections in the first sentence:
>
> OLD:
>      For AMR/AMR-WB, support for redundant encoding, and the maximum
>      supported depth, are controlled by the 'max-red' parameter, as
>      specified in [RFC4867], Section 8.1.
>
> NEW:
>      For AMR/AMR-WB, support for redundant encoding, and the maximum
>      used depth, is indicated by the 'max-red' parameter, as
>      specified in [RFC4867], Section 8.1 and 8.2. The 3GPP Multimedia
>      Telephony services specifies a maximum value of 220 ms for
>      'max-red', see [3GPP.26.114].
>
>
> New Informational Reference:
>
> [3GPP.26.114]
>
> <reference anchor="3GPP.26.114"><front><title>IP Multimedia Subsystem (IMS); Multimedia telephony; Media handling and interaction</title><author><organization>3GPP</organization></author><date
> day="17" month="March" year="2016"/></front><seriesInfo name="3GPP TS"
> value="26.114 13.3.0"/><format type="HTML"
> target="http://www.3gpp.org/ftp/Specs/html-info/26114.htm"/></reference>
>
> Opinions?
>
> 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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> https://www.ietf.org/mailman/listinfo/rtcweb
>
> _________________________________________________________________________________________________________________________
>
> Ce message et ses pieces jointes peuvent contenir des informations confidentielles ou privilegiees et ne doivent donc
> pas etre diffuses, exploites ou copies sans autorisation. Si vous avez recu ce message par erreur, veuillez le signaler
> a l'expediteur et le detruire ainsi que les pieces jointes. Les messages electroniques etant susceptibles d'alteration,
> Orange decline toute responsabilite si ce message a ete altere, deforme ou falsifie. Merci.
>
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> Thank you.
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>


-- 

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: Peter Thatcher <pthatcher@google.com>
Date: Thu, 7 Apr 2016 08:03:02 -0700
Message-ID: <CAJrXDUHyxd=n5XUkBjQaDTKYnALkg9C-da0rx0z0Q988DA6=Sg@mail.gmail.com>
To: Tim Panton <tim@phonefromhere.com>
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Cc: "rtcweb@ietf.org" <rtcweb@ietf.org>, mmusic WG <mmusic@ietf.org>
Subject: Re: [rtcweb] [MMUSIC]  Tunnelling DTLS in SDP
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On Tue, Apr 5, 2016 at 2:52 AM, Tim Panton <tim@phonefromhere.com> wrote:

>
> On 5 Apr 2016, at 03:06, Justin Uberti <juberti@google.com> wrote:
>
>
>
> On Mon, Apr 4, 2016 at 8:39 AM, pfh <tim@phonefromhere.com> wrote:
>
>>
>> On 4 Apr 2016, at 14:10, Eric Rescorla <ekr@rtfm.com> wrote:
>>
>> Hi folks,
>>
>> I wanted to call your attention to a draft I just published with a
>> possibly stupid
>> idea.
>>
>> https://tools.ietf.org/html/draft-rescorla-dtls-in-sdp-00
>>
>> A nontrivial fraction of call setup time in WebRTC is the DTLS handshake=
.
>> This document describes how to piggyback the first few handshake message=
s
>> in the SDP offer/answer exchange, thus reducing latency.
>>
>>
>> It strikes me we could get the same reduced latency benefits by
>> piggybacking on ICE
>> rather than SDP, e.g. embedding the DTLS packet as data in a new STUN
>> attribute type.
>>
>> The downside of piggybacking on the SDP is that you are increasing the
>> trust you have to place in the
>> signalling server undermining the elegant decoupling we have at the
>> moment between signalling and
>> media. (The SDES issues of logging keys in the web servers apply to the
>> public certs as well).
>>
>
> I don't think the logging issue applies here. Only the public key would b=
e
> logged, and it's already transmitted in cleartext during a normal DTLS
> handshake (it's a public key, after all).
>
>
> True, but over a different path.
>
>
>> It also significantly clutters the SDP (even more) !
>>
>
> Please see Jonathan Lennox's comment about garbage piles. I think the onl=
y
> truly unattractive part here is the fact that the messages will need to b=
e
> base64ed, resulting in minor blowup. But that seems like a reasonable
> downside for 1+ RTT upside.
>
>
> Just when the WebRTC NG was freeing us from Offer/Answer and SDP=E2=80=A6=
.
>
> Sigh.
>
>
=E2=80=8BThe general idea here is "put the beginning of the crypto handshak=
e in the
signaling"=E2=80=8B.  That's not tied to SDP.  WebRTC NG could easily do th=
e same
thing without SDP.



>
>
>
>> As you point out, it weakens the usefulness of longer term certs, which
>> would be a major nuisance IMHO.
>>
>> Tim.
>>
>>
>> _______________________________________________
>> rtcweb mailing list
>> rtcweb@ietf.org
>> https://www.ietf.org/mailman/listinfo/rtcweb
>>
>>
>
>
> _______________________________________________
> mmusic mailing list
> mmusic@ietf.org
> https://www.ietf.org/mailman/listinfo/mmusic
>
>

--001a114551525aa568052fe662ec
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<div dir=3D"ltr"><div class=3D"gmail_default" style=3D"font-family:arial,he=
lvetica,sans-serif"><br></div><div class=3D"gmail_extra"><br><div class=3D"=
gmail_quote">On Tue, Apr 5, 2016 at 2:52 AM, Tim Panton <span dir=3D"ltr">&=
lt;<a href=3D"mailto:tim@phonefromhere.com" target=3D"_blank">tim@phonefrom=
here.com</a>&gt;</span> wrote:<br><blockquote class=3D"gmail_quote" style=
=3D"margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex"><div sty=
le=3D"word-wrap:break-word"><br><div><span class=3D""><blockquote type=3D"c=
ite"><div>On 5 Apr 2016, at 03:06, Justin Uberti &lt;<a href=3D"mailto:jube=
rti@google.com" target=3D"_blank">juberti@google.com</a>&gt; wrote:</div><b=
r><div><div dir=3D"ltr"><br><div class=3D"gmail_extra"><br><div class=3D"gm=
ail_quote">On Mon, Apr 4, 2016 at 8:39 AM, pfh <span dir=3D"ltr">&lt;<a hre=
f=3D"mailto:tim@phonefromhere.com" target=3D"_blank">tim@phonefromhere.com<=
/a>&gt;</span> wrote:<br><blockquote class=3D"gmail_quote" style=3D"margin:=
0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex"><div style=3D"word-=
wrap:break-word"><br><div><span><blockquote type=3D"cite"><div>On 4 Apr 201=
6, at 14:10, Eric Rescorla &lt;<a href=3D"mailto:ekr@rtfm.com" target=3D"_b=
lank">ekr@rtfm.com</a>&gt; wrote:</div><br><div><div dir=3D"ltr">Hi folks,<=
div><br></div><div>I wanted to call your attention to a draft I just publis=
hed with a possibly stupid</div><div>idea.</div><div><br></div><div><a href=
=3D"https://tools.ietf.org/html/draft-rescorla-dtls-in-sdp-00" target=3D"_b=
lank">https://tools.ietf.org/html/draft-rescorla-dtls-in-sdp-00</a><br></di=
v><div><br></div><div>A nontrivial fraction of call setup time in WebRTC is=
 the DTLS handshake.</div><div>This document describes how to piggyback the=
 first few handshake messages</div><div>in the SDP offer/answer exchange, t=
hus reducing latency.</div></div></div></blockquote><div><br></div></span><=
div>It strikes me we could get the same reduced latency benefits by piggyba=
cking on ICE</div><div>rather than SDP, e.g. embedding the DTLS packet as d=
ata in a new STUN attribute type.</div><div><br></div><div>The downside of =
piggybacking on the SDP is that you are increasing the trust you have to pl=
ace in the</div><div>signalling server undermining the elegant decoupling w=
e have at the moment between signalling and=C2=A0</div><div>media. (The SDE=
S issues of logging keys in the web servers apply to the public certs as we=
ll).</div></div></div></blockquote><div><br></div><div>I don&#39;t think th=
e logging issue applies here. Only the public key would be logged, and it&#=
39;s already transmitted in cleartext during a normal DTLS handshake (it&#3=
9;s a public key, after all).=C2=A0</div></div></div></div></div></blockquo=
te><div><br></div></span><div>True, but over a different path.=C2=A0</div><=
span class=3D""><br><blockquote type=3D"cite"><div><div dir=3D"ltr"><div cl=
ass=3D"gmail_extra"><div class=3D"gmail_quote"><blockquote class=3D"gmail_q=
uote" style=3D"margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1e=
x"><div style=3D"word-wrap:break-word"><div><div><br></div><div>It also sig=
nificantly clutters the SDP (even more) !</div></div></div></blockquote><di=
v><br></div><div>Please see Jonathan Lennox&#39;s comment about garbage pil=
es. I think the only truly unattractive part here is the fact that the mess=
ages will need to be base64ed, resulting in minor blowup. But that seems li=
ke a reasonable downside for 1+ RTT upside.</div></div></div></div></div></=
blockquote><div><br></div></span>Just when the WebRTC NG was freeing us fro=
m Offer/Answer and SDP=E2=80=A6.</div><div><br></div><div>Sigh.</div><span =
class=3D""><div><br></div></span></div></blockquote><div><br></div><div><di=
v class=3D"gmail_default" style=3D"font-family:arial,helvetica,sans-serif">=
=E2=80=8BThe general idea here is &quot;put the beginning of the crypto han=
dshake in the signaling&quot;=E2=80=8B.=C2=A0 That&#39;s not tied to SDP.=
=C2=A0 WebRTC NG could easily do the same thing without SDP.=C2=A0</div><br=
></div><div>=C2=A0</div><blockquote class=3D"gmail_quote" style=3D"margin:0=
 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex"><div style=3D"word-w=
rap:break-word"><span class=3D""><div><br><blockquote type=3D"cite"><div><d=
iv dir=3D"ltr"><div class=3D"gmail_extra"><div class=3D"gmail_quote"><div><=
br></div><blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;borde=
r-left:1px #ccc solid;padding-left:1ex"><div style=3D"word-wrap:break-word"=
><div><div><br></div><div>As you point out, it weakens the usefulness of lo=
nger term certs, which</div><div>would be a major nuisance IMHO.</div><div>=
<br></div><div>Tim.</div></div><br></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" rel=3D"noreferrer"=
 target=3D"_blank">https://www.ietf.org/mailman/listinfo/rtcweb</a><br>
<br></blockquote></div><br></div></div>
</div></blockquote></div><br></span></div><br>_____________________________=
__________________<br>
mmusic mailing list<br>
<a href=3D"mailto:mmusic@ietf.org">mmusic@ietf.org</a><br>
<a href=3D"https://www.ietf.org/mailman/listinfo/mmusic" rel=3D"noreferrer"=
 target=3D"_blank">https://www.ietf.org/mailman/listinfo/mmusic</a><br>
<br></blockquote></div><br></div></div>

--001a114551525aa568052fe662ec--


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From: Peter Thatcher <pthatcher@google.com>
Date: Thu, 7 Apr 2016 08:07:10 -0700
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To: Harald Alvestrand <harald@alvestrand.no>
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Subject: Re: [rtcweb] Tunnelling DTLS in SDP
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On Tue, Apr 5, 2016 at 2:53 AM, Harald Alvestrand <harald@alvestrand.no>
wrote:

> On first read, this makes sense to me.
>
> I wonder if it could/should be made into a general concept, to fit with
> the tendency in WebRTC:NG to separate signalling format even more from
> operation?
>
> We could call it "out of band DTLS setup", say that in general, a DTLS
> session can be started in one medium (SDP signalling, in this case), and
> continued in another medium (the DTLS-protected media channel), and have a
> section describing the details of carrying DTLS-over-SDP.
>

> When viewing it in this way, using the same technique with Jabber or
> proprietary signalling becomes a reasonably obvious exercise. There are
> some other twists that seem obvious too - for instance, one could continue
> the setup over the SDP channel in subsequent offer/answers if the first
> exchange failed to set up a media channel. I'm not sure that makes sense,
> though.
>

In ORTC language, we could simply have an event on the DtlsTransport that
is something like .onpacket and another that's .addpacket, and when the
transport wants to send a packet, it lets the JS know so it can do so over
the signalling path, and then the JS can take care of optimizing it (or
not).


>
> One SDP twist: If forking happens, it could be treated like any other
> attempt to generate multiple answers to a ClientHello, I think. I'm sure
> it's well defined how to respond to that - it's an obvious attack. Only one
> leg of the fork would ever succeed, I assume.
>
>
>
> On 04/04/2016 03:10 PM, Eric Rescorla wrote:
>
> Hi folks,
>
> I wanted to call your attention to a draft I just published with a
> possibly stupid
> idea.
>
> https://tools.ietf.org/html/draft-rescorla-dtls-in-sdp-00
>
> A nontrivial fraction of call setup time in WebRTC is the DTLS handshake.
> This document describes how to piggyback the first few handshake messages
> in the SDP offer/answer exchange, thus reducing latency.
>
> Comments welcome.
>
> -Ekr
>
>
>
>
>
> _______________________________________________
> 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
>
>

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<div dir=3D"ltr"><div class=3D"gmail_default" style=3D"font-family:arial,he=
lvetica,sans-serif"><br></div><div class=3D"gmail_extra"><br><div class=3D"=
gmail_quote">On Tue, Apr 5, 2016 at 2:53 AM, Harald Alvestrand <span dir=3D=
"ltr">&lt;<a href=3D"mailto:harald@alvestrand.no" target=3D"_blank">harald@=
alvestrand.no</a>&gt;</span> wrote:<br><blockquote class=3D"gmail_quote" st=
yle=3D"margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">
 =20
   =20
 =20
  <div text=3D"#000000" bgcolor=3D"#FFFFFF">
    <div>On first read, this makes sense to me.<br>
      <br>
      I wonder if it could/should be made into a general concept, to fit
      with the tendency in WebRTC:NG to separate signalling format even
      more from operation?<br>
      <br>
      We could call it &quot;out of band DTLS setup&quot;, say that in gene=
ral, a
      DTLS session can be started in one medium (SDP signalling, in this
      case), and continued in another medium (the DTLS-protected media
      channel), and have a section describing the details of carrying
      DTLS-over-SDP.=C2=A0</div></div></blockquote><blockquote class=3D"gma=
il_quote" style=3D"margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-lef=
t:1ex"><div text=3D"#000000" bgcolor=3D"#FFFFFF"><div>
      <br>
      When viewing it in this way, using the same technique with Jabber
      or proprietary signalling becomes a reasonably obvious exercise.
      There are some other twists that seem obvious too - for instance,
      one could continue the setup over the SDP channel in subsequent
      offer/answers if the first exchange failed to set up a media
      channel. I&#39;m not sure that makes sense, though.<br></div></div></=
blockquote><div><div class=3D"gmail_default" style=3D"font-family:arial,hel=
vetica,sans-serif;display:inline"><br></div></div><div><div class=3D"gmail_=
default" style=3D"font-family:arial,helvetica,sans-serif;display:inline">In=
 ORTC language, we could simply have an event on the DtlsTransport that is =
something like .onpacket and another that&#39;s .addpacket, and when the tr=
ansport wants to send a packet, it lets the JS know so it can do so over th=
e signalling path, and then the JS can take care of optimizing it (or not).=
 =C2=A0</div></div><div>=C2=A0</div><blockquote class=3D"gmail_quote" style=
=3D"margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex"><div tex=
t=3D"#000000" bgcolor=3D"#FFFFFF"><div>
      <br>
      One SDP twist: If forking happens, it could be treated like any
      other attempt to generate multiple answers to a ClientHello, I
      think. I&#39;m sure it&#39;s well defined how to respond to that - it=
&#39;s an
      obvious attack. Only one leg of the fork would ever succeed, I
      assume.<div><div class=3D"h5"><br>
      <br>
      <br>
      On 04/04/2016 03:10 PM, Eric Rescorla wrote:<br>
    </div></div></div>
    <blockquote type=3D"cite"><div><div class=3D"h5">
      <div dir=3D"ltr">Hi folks,
        <div><br>
        </div>
        <div>I wanted to call your attention to a draft I just published
          with a possibly stupid</div>
        <div>idea.</div>
        <div><br>
        </div>
        <div><a href=3D"https://tools.ietf.org/html/draft-rescorla-dtls-in-=
sdp-00" target=3D"_blank">https://tools.ietf.org/html/draft-rescorla-dtls-i=
n-sdp-00</a><br>
        </div>
        <div><br>
        </div>
        <div>A nontrivial fraction of call setup time in WebRTC is the
          DTLS handshake.</div>
        <div>This document describes how to piggyback the first few
          handshake messages</div>
        <div>in the SDP offer/answer exchange, thus reducing latency.</div>
        <div><br>
        </div>
        <div>Comments welcome.</div>
        <div><br>
        </div>
        <div>-Ekr</div>
        <div><br>
        </div>
        <div><br>
        </div>
        <div><br>
        </div>
      </div>
      <br>
      <fieldset></fieldset>
      <br>
      </div></div><span class=3D""><pre>___________________________________=
____________
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>
    </span></blockquote><span class=3D"HOEnZb"><font color=3D"#888888">
    <br>
    <br>
    <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" rel=3D"noreferrer"=
 target=3D"_blank">https://www.ietf.org/mailman/listinfo/rtcweb</a><br>
<br></blockquote></div><br></div></div>

--001a114661dc1b5c01052fe6710d--


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From: Martin Thomson <martin.thomson@gmail.com>
To: Jonathan Lennox <jonathan@vidyo.com>, Eric Rescorla <ekr@rtfm.com>, Alissa Cooper <alissa@cooperw.in>,  Cullen Jennings <fluffy@cisco.com>, "rtcweb@ietf.org" <rtcweb@ietf.org>
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Subject: [rtcweb] friendly amendments to text on DTMF resolution
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grammar of the first sentence (Jonathan)
"default duration" for gaps should be "default gap" (ekr)
strike "such as browsers" from second paragraph (Alissa)
change "will not" to "are not require to" (Martin/Fluffy)


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Sorry that we were not able to get to the slides on this during the WG
meeting.  Here's the proposal from Justin, taken for the slides he prepared.


Original text:

WebRTC incorporates an explicit permission grant for access to local audio
and video, which are typically much more sensitive than the aforementioned
IP address information. If the user has consented to media access, this
should also allow WebRTC to gather all possible candidates and determine
the absolute best route for media traffic.

Proposed replacement:

Gathering all possible candidates SHOULD only be performed upon user
consent, which thwarts the typical drive-by enumeration attacks. The
details of this consent are left to the implementation; one potential
mechanism is to key this off getUserMedia consent.

The getUserMedia suggestion takes into account that the user has provided
some consent to the application already; that when doing so the user
typically wants to engage in a conversational session, which benefits most
from an optimal network path, and lastly, the fact that the underlying
issue is complex and difficult to explain, making explicit consent for
enumeration troublesome.

--001a113ce79e3e422d052ffce334
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<div dir=3D"ltr"><div>Sorry that we were not able to get to the slides on t=
his during the WG meeting.=C2=A0 Here&#39;s the proposal from Justin, taken=
 for the slides he prepared.<br><br><br></div>Original text:<br><div><br>We=
bRTC incorporates an explicit permission grant for access to local audio an=
d video, which are typically much more sensitive than the aforementioned IP=
 address information. If the user has consented to media access, this shoul=
d also allow WebRTC to gather all possible candidates and determine the abs=
olute best route for media traffic.<br><br></div><div>Proposed replacement:=
<br></div><div><br>Gathering all possible candidates SHOULD only be perform=
ed upon user consent, which thwarts the typical drive-by enumeration attack=
s. The details of this consent are left to the implementation; one potentia=
l mechanism is to key this off getUserMedia consent.<br><br>The getUserMedi=
a suggestion takes into account that the user has provided some consent to =
the application already; that when doing so the user typically wants to eng=
age in a conversational session, which benefits most from an optimal networ=
k path, and lastly, the fact that the underlying issue is complex and diffi=
cult to explain, making explicit consent for enumeration troublesome.<br></=
div></div>

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From: Paul Kyzivat <pkyzivat@alum.mit.edu>
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Subject: [rtcweb] Followup to discussion of O/A and codecs from Friday session
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During the rtcweb session today, while discussing (I think) JSEP issue 
#247 (Document what should happen when there are no matching codecs in 
answer), there was a question about happens in a particular case. I 
think it was:

- X offers codecs A and B (sendrecv)
- Y answers codec B (sendrecv)

The question was whether Y may use codec A when sending to X. Jonathan 
gave an answer I disagreed with and we had a chat about it. He got an 
action item to follow up afterwords, then he asked if I would do it.

As I read RFC3264, the answer is:

Yes, in the above case Y may send A to X.

Then a subsequent scenario came up: what would happen if Y has a need to 
send a (re)offer? Wouldn't this result in an inability to send using A? 
(Let's assume that Y is unable/unwilling to *receive* A but can send it, 
even though this is a weird case.)

- Y offers codec B (sendrecv)
- X answers codec B and A (sendrecv)

This *is* permitted by 3264. And in this case Y will be allowed to 
continue sending using codec A.

Relevant pieces of 3264 supporting this interpretation:

- section 5.1 (Generating the Initial Offer/Unicast Streams):

    ... If multiple formats are listed, it
    means that the offerer is capable of making use of any of those
    formats during the session.  In other words, the answerer MAY change
    formats in the middle of the session, making use of any of the
    formats listed, without sending a new offer.

- section 6.1 (Generating the Answer/Unicast Streams):

    ... For streams marked as sendrecv in the answer,
    the "m=" line MUST contain at least one codec the answerer is willing
    to both send and receive, from amongst those listed in the offer.
    The stream MAY indicate additional media formats, not listed in the
    corresponding stream in the offer, that the answerer is willing to
    send or receive (of course, it [the answerer] will not be able to
    send them at this time, since it was not listed in the offer).

- section 8.3.2 (Changing the Set of Media Formats):

    The list of media formats used in the session MAY be changed.  To do
    this, the offerer creates a new media description, with the list of
    media formats in the "m=" line different from the corresponding media
    stream in the previous SDP.  This list MAY include new formats, and
    MAY remove formats present from the previous SDP.  ...

    The corresponding media stream in the answer is formulated as
    described in Section 6, and may result in a change in media formats
    as well.  Similarly, as described in Section 6, as soon as it sends
    its answer, the answerer MUST begin sending media using any formats
    in the offer that were also present in the answer, and SHOULD use the
    most preferred format in the offer that was also listed in the answer
    (assuming the stream allows for sending), and MUST NOT send using any
    formats that are not in the offer, even if they were present in a
    previous SDP from the peer.  Similarly, when the offerer receives the
    answer, it MUST begin sending media using any formats in the answer,
    and SHOULD use the most preferred one (assuming the stream allows for
    sending), and MUST NOT send using any formats that are not in the
    answer, even if they were present in a previous SDP from the peer.

I'm not too clear on how this relates to issue #247, but at least I hope 
we can agree on what 3264 says.

	Thanks,
	Paul


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From: Harald Alvestrand <harald@alvestrand.no>
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Subject: Re: [rtcweb] Followup to discussion of O/A and codecs from Friday session
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This indicates that the answer's omission of a codec does not allow the
offerer to deallocate resources that it has reserved for decoding that
type of stream (such as hardware).

Under this interpretation, if I reserve hardware for decoding VP8 and
H.264, and I get back an answer that indicates VP8 only, I cannot
deallocate my H.264 resources, since the answerer may choose to send me
H.264 anyway.

This contradicts something I think I've heard Cullen claim multiple
times, but I may have misunderstood Cullen.


On 04/08/2016 11:41 PM, Paul Kyzivat wrote:
> During the rtcweb session today, while discussing (I think) JSEP issue
> #247 (Document what should happen when there are no matching codecs in
> answer), there was a question about happens in a particular case. I
> think it was:
>
> - X offers codecs A and B (sendrecv)
> - Y answers codec B (sendrecv)
>
> The question was whether Y may use codec A when sending to X. Jonathan
> gave an answer I disagreed with and we had a chat about it. He got an
> action item to follow up afterwords, then he asked if I would do it.
>
> As I read RFC3264, the answer is:
>
> Yes, in the above case Y may send A to X.
>
> Then a subsequent scenario came up: what would happen if Y has a need
> to send a (re)offer? Wouldn't this result in an inability to send
> using A? (Let's assume that Y is unable/unwilling to *receive* A but
> can send it, even though this is a weird case.)
>
> - Y offers codec B (sendrecv)
> - X answers codec B and A (sendrecv)
>
> This *is* permitted by 3264. And in this case Y will be allowed to
> continue sending using codec A.
>
> Relevant pieces of 3264 supporting this interpretation:
>
> - section 5.1 (Generating the Initial Offer/Unicast Streams):
>
>    ... If multiple formats are listed, it
>    means that the offerer is capable of making use of any of those
>    formats during the session.  In other words, the answerer MAY change
>    formats in the middle of the session, making use of any of the
>    formats listed, without sending a new offer.
>
> - section 6.1 (Generating the Answer/Unicast Streams):
>
>    ... For streams marked as sendrecv in the answer,
>    the "m=" line MUST contain at least one codec the answerer is willing
>    to both send and receive, from amongst those listed in the offer.
>    The stream MAY indicate additional media formats, not listed in the
>    corresponding stream in the offer, that the answerer is willing to
>    send or receive (of course, it [the answerer] will not be able to
>    send them at this time, since it was not listed in the offer).
>
> - section 8.3.2 (Changing the Set of Media Formats):
>
>    The list of media formats used in the session MAY be changed.  To do
>    this, the offerer creates a new media description, with the list of
>    media formats in the "m=" line different from the corresponding media
>    stream in the previous SDP.  This list MAY include new formats, and
>    MAY remove formats present from the previous SDP.  ...
>
>    The corresponding media stream in the answer is formulated as
>    described in Section 6, and may result in a change in media formats
>    as well.  Similarly, as described in Section 6, as soon as it sends
>    its answer, the answerer MUST begin sending media using any formats
>    in the offer that were also present in the answer, and SHOULD use the
>    most preferred format in the offer that was also listed in the answer
>    (assuming the stream allows for sending), and MUST NOT send using any
>    formats that are not in the offer, even if they were present in a
>    previous SDP from the peer.  Similarly, when the offerer receives the
>    answer, it MUST begin sending media using any formats in the answer,
>    and SHOULD use the most preferred one (assuming the stream allows for
>    sending), and MUST NOT send using any formats that are not in the
>    answer, even if they were present in a previous SDP from the peer.
>
> I'm not too clear on how this relates to issue #247, but at least I
> hope we can agree on what 3264 says.
>
>     Thanks,
>     Paul
>
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb


-- 
Surveillance is pervasive. Go Dark.


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Subject: Re: [rtcweb] Followup to discussion of O/A and codecs from Friday session
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In Section 6 of RFC 3264:

> The answerer MUST send using a media format in the offer that is also listed in the answer
>
> So, if X offers A and B, and Y answers B, how would Y be able to send A?
It's not in both the offer and the answer.

On Fri, Apr 8, 2016 at 3:01 PM, Harald Alvestrand <harald@alvestrand.no>
wrote:

> This indicates that the answer's omission of a codec does not allow the
> offerer to deallocate resources that it has reserved for decoding that
> type of stream (such as hardware).
>
> Under this interpretation, if I reserve hardware for decoding VP8 and
> H.264, and I get back an answer that indicates VP8 only, I cannot
> deallocate my H.264 resources, since the answerer may choose to send me
> H.264 anyway.
>
> This contradicts something I think I've heard Cullen claim multiple
> times, but I may have misunderstood Cullen.
>
>
> On 04/08/2016 11:41 PM, Paul Kyzivat wrote:
> > During the rtcweb session today, while discussing (I think) JSEP issue
> > #247 (Document what should happen when there are no matching codecs in
> > answer), there was a question about happens in a particular case. I
> > think it was:
> >
> > - X offers codecs A and B (sendrecv)
> > - Y answers codec B (sendrecv)
> >
> > The question was whether Y may use codec A when sending to X. Jonathan
> > gave an answer I disagreed with and we had a chat about it. He got an
> > action item to follow up afterwords, then he asked if I would do it.
> >
> > As I read RFC3264, the answer is:
> >
> > Yes, in the above case Y may send A to X.
> >
> > Then a subsequent scenario came up: what would happen if Y has a need
> > to send a (re)offer? Wouldn't this result in an inability to send
> > using A? (Let's assume that Y is unable/unwilling to *receive* A but
> > can send it, even though this is a weird case.)
> >
> > - Y offers codec B (sendrecv)
> > - X answers codec B and A (sendrecv)
> >
> > This *is* permitted by 3264. And in this case Y will be allowed to
> > continue sending using codec A.
> >
> > Relevant pieces of 3264 supporting this interpretation:
> >
> > - section 5.1 (Generating the Initial Offer/Unicast Streams):
> >
> >    ... If multiple formats are listed, it
> >    means that the offerer is capable of making use of any of those
> >    formats during the session.  In other words, the answerer MAY change
> >    formats in the middle of the session, making use of any of the
> >    formats listed, without sending a new offer.
> >
> > - section 6.1 (Generating the Answer/Unicast Streams):
> >
> >    ... For streams marked as sendrecv in the answer,
> >    the "m=" line MUST contain at least one codec the answerer is willing
> >    to both send and receive, from amongst those listed in the offer.
> >    The stream MAY indicate additional media formats, not listed in the
> >    corresponding stream in the offer, that the answerer is willing to
> >    send or receive (of course, it [the answerer] will not be able to
> >    send them at this time, since it was not listed in the offer).
> >
> > - section 8.3.2 (Changing the Set of Media Formats):
> >
> >    The list of media formats used in the session MAY be changed.  To do
> >    this, the offerer creates a new media description, with the list of
> >    media formats in the "m=" line different from the corresponding media
> >    stream in the previous SDP.  This list MAY include new formats, and
> >    MAY remove formats present from the previous SDP.  ...
> >
> >    The corresponding media stream in the answer is formulated as
> >    described in Section 6, and may result in a change in media formats
> >    as well.  Similarly, as described in Section 6, as soon as it sends
> >    its answer, the answerer MUST begin sending media using any formats
> >    in the offer that were also present in the answer, and SHOULD use the
> >    most preferred format in the offer that was also listed in the answer
> >    (assuming the stream allows for sending), and MUST NOT send using any
> >    formats that are not in the offer, even if they were present in a
> >    previous SDP from the peer.  Similarly, when the offerer receives the
> >    answer, it MUST begin sending media using any formats in the answer,
> >    and SHOULD use the most preferred one (assuming the stream allows for
> >    sending), and MUST NOT send using any formats that are not in the
> >    answer, even if they were present in a previous SDP from the peer.
> >
> > I'm not too clear on how this relates to issue #247, but at least I
> > hope we can agree on what 3264 says.
> >
> >     Thanks,
> >     Paul
> >
> > _______________________________________________
> > rtcweb mailing list
> > rtcweb@ietf.org
> > https://www.ietf.org/mailman/listinfo/rtcweb
>
>
> --
> Surveillance is pervasive. Go Dark.
>
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb
>

--001a114268885d90940530009131
Content-Type: text/html; charset=UTF-8
Content-Transfer-Encoding: quoted-printable

<div dir=3D"ltr">In Section 6 of RFC 3264:<div><blockquote class=3D"gmail_q=
uote" style=3D"margin:0px 0px 0px 0.8ex;border-left-width:1px;border-left-c=
olor:rgb(204,204,204);border-left-style:solid;padding-left:1ex"><pre style=
=3D"color:rgb(0,0,0);word-wrap:break-word;white-space:pre-wrap">The answere=
r MUST send using a media format in the offer that is also listed in the an=
swer</pre></blockquote><div><span style=3D"font-size:12.8px">So, if X offer=
s A and B, and Y answers B, how would Y be able to send A? It&#39;s not in =
both the offer and the answer.</span></div></div></div><div class=3D"gmail_=
extra"><br><div class=3D"gmail_quote">On Fri, Apr 8, 2016 at 3:01 PM, Haral=
d Alvestrand <span dir=3D"ltr">&lt;<a href=3D"mailto:harald@alvestrand.no" =
target=3D"_blank">harald@alvestrand.no</a>&gt;</span> wrote:<br><blockquote=
 class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1px #ccc soli=
d;padding-left:1ex">This indicates that the answer&#39;s omission of a code=
c does not allow the<br>
offerer to deallocate resources that it has reserved for decoding that<br>
type of stream (such as hardware).<br>
<br>
Under this interpretation, if I reserve hardware for decoding VP8 and<br>
H.264, and I get back an answer that indicates VP8 only, I cannot<br>
deallocate my H.264 resources, since the answerer may choose to send me<br>
H.264 anyway.<br>
<br>
This contradicts something I think I&#39;ve heard Cullen claim multiple<br>
times, but I may have misunderstood Cullen.<br>
<div class=3D"HOEnZb"><div class=3D"h5"><br>
<br>
On 04/08/2016 11:41 PM, Paul Kyzivat wrote:<br>
&gt; During the rtcweb session today, while discussing (I think) JSEP issue=
<br>
&gt; #247 (Document what should happen when there are no matching codecs in=
<br>
&gt; answer), there was a question about happens in a particular case. I<br=
>
&gt; think it was:<br>
&gt;<br>
&gt; - X offers codecs A and B (sendrecv)<br>
&gt; - Y answers codec B (sendrecv)<br>
&gt;<br>
&gt; The question was whether Y may use codec A when sending to X. Jonathan=
<br>
&gt; gave an answer I disagreed with and we had a chat about it. He got an<=
br>
&gt; action item to follow up afterwords, then he asked if I would do it.<b=
r>
&gt;<br>
&gt; As I read RFC3264, the answer is:<br>
&gt;<br>
&gt; Yes, in the above case Y may send A to X.<br>
&gt;<br>
&gt; Then a subsequent scenario came up: what would happen if Y has a need<=
br>
&gt; to send a (re)offer? Wouldn&#39;t this result in an inability to send<=
br>
&gt; using A? (Let&#39;s assume that Y is unable/unwilling to *receive* A b=
ut<br>
&gt; can send it, even though this is a weird case.)<br>
&gt;<br>
&gt; - Y offers codec B (sendrecv)<br>
&gt; - X answers codec B and A (sendrecv)<br>
&gt;<br>
&gt; This *is* permitted by 3264. And in this case Y will be allowed to<br>
&gt; continue sending using codec A.<br>
&gt;<br>
&gt; Relevant pieces of 3264 supporting this interpretation:<br>
&gt;<br>
&gt; - section 5.1 (Generating the Initial Offer/Unicast Streams):<br>
&gt;<br>
&gt;=C2=A0 =C2=A0 ... If multiple formats are listed, it<br>
&gt;=C2=A0 =C2=A0 means that the offerer is capable of making use of any of=
 those<br>
&gt;=C2=A0 =C2=A0 formats during the session.=C2=A0 In other words, the ans=
werer MAY change<br>
&gt;=C2=A0 =C2=A0 formats in the middle of the session, making use of any o=
f the<br>
&gt;=C2=A0 =C2=A0 formats listed, without sending a new offer.<br>
&gt;<br>
&gt; - section 6.1 (Generating the Answer/Unicast Streams):<br>
&gt;<br>
&gt;=C2=A0 =C2=A0 ... For streams marked as sendrecv in the answer,<br>
&gt;=C2=A0 =C2=A0 the &quot;m=3D&quot; line MUST contain at least one codec=
 the answerer is willing<br>
&gt;=C2=A0 =C2=A0 to both send and receive, from amongst those listed in th=
e offer.<br>
&gt;=C2=A0 =C2=A0 The stream MAY indicate additional media formats, not lis=
ted in the<br>
&gt;=C2=A0 =C2=A0 corresponding stream in the offer, that the answerer is w=
illing to<br>
&gt;=C2=A0 =C2=A0 send or receive (of course, it [the answerer] will not be=
 able to<br>
&gt;=C2=A0 =C2=A0 send them at this time, since it was not listed in the of=
fer).<br>
&gt;<br>
&gt; - section 8.3.2 (Changing the Set of Media Formats):<br>
&gt;<br>
&gt;=C2=A0 =C2=A0 The list of media formats used in the session MAY be chan=
ged.=C2=A0 To do<br>
&gt;=C2=A0 =C2=A0 this, the offerer creates a new media description, with t=
he list of<br>
&gt;=C2=A0 =C2=A0 media formats in the &quot;m=3D&quot; line different from=
 the corresponding media<br>
&gt;=C2=A0 =C2=A0 stream in the previous SDP.=C2=A0 This list MAY include n=
ew formats, and<br>
&gt;=C2=A0 =C2=A0 MAY remove formats present from the previous SDP.=C2=A0 .=
..<br>
&gt;<br>
&gt;=C2=A0 =C2=A0 The corresponding media stream in the answer is formulate=
d as<br>
&gt;=C2=A0 =C2=A0 described in Section 6, and may result in a change in med=
ia formats<br>
&gt;=C2=A0 =C2=A0 as well.=C2=A0 Similarly, as described in Section 6, as s=
oon as it sends<br>
&gt;=C2=A0 =C2=A0 its answer, the answerer MUST begin sending media using a=
ny formats<br>
&gt;=C2=A0 =C2=A0 in the offer that were also present in the answer, and SH=
OULD use the<br>
&gt;=C2=A0 =C2=A0 most preferred format in the offer that was also listed i=
n the answer<br>
&gt;=C2=A0 =C2=A0 (assuming the stream allows for sending), and MUST NOT se=
nd using any<br>
&gt;=C2=A0 =C2=A0 formats that are not in the offer, even if they were pres=
ent in a<br>
&gt;=C2=A0 =C2=A0 previous SDP from the peer.=C2=A0 Similarly, when the off=
erer receives the<br>
&gt;=C2=A0 =C2=A0 answer, it MUST begin sending media using any formats in =
the answer,<br>
&gt;=C2=A0 =C2=A0 and SHOULD use the most preferred one (assuming the strea=
m allows for<br>
&gt;=C2=A0 =C2=A0 sending), and MUST NOT send using any formats that are no=
t in the<br>
&gt;=C2=A0 =C2=A0 answer, even if they were present in a previous SDP from =
the peer.<br>
&gt;<br>
&gt; I&#39;m not too clear on how this relates to issue #247, but at least =
I<br>
&gt; hope we can agree on what 3264 says.<br>
&gt;<br>
&gt;=C2=A0 =C2=A0 =C2=A0Thanks,<br>
&gt;=C2=A0 =C2=A0 =C2=A0Paul<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" rel=3D"norefe=
rrer" target=3D"_blank">https://www.ietf.org/mailman/listinfo/rtcweb</a><br=
>
<br>
<br>
</div></div><span class=3D"HOEnZb"><font color=3D"#888888">--<br>
Surveillance is pervasive. Go Dark.<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" rel=3D"noreferrer"=
 target=3D"_blank">https://www.ietf.org/mailman/listinfo/rtcweb</a><br>
</div></div></blockquote></div><br></div>

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Date: Fri, 8 Apr 2016 18:21:23 -0400
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On Fri, Apr 8, 2016 at 6:01 PM, Harald Alvestrand <harald@alvestrand.no>
wrote:

> This indicates that the answer's omission of a codec does not allow the
> offerer to deallocate resources that it has reserved for decoding that
> type of stream (such as hardware).
>
> Under this interpretation, if I reserve hardware for decoding VP8 and
> H.264, and I get back an answer that indicates VP8 only, I cannot
> deallocate my H.264 resources, since the answerer may choose to send me
> H.264 anyway.
>
>
Based on what I heard from multiple people (including Jonathan Rosenberg
when this document was originally discussed), Offer/Answer was always
intended to be asymmetrical. Each side can declare all the codecs it is
willing to receive and can send any codec remote party supports. I saw
multiple examples of this being implemented and used in exactly this manner.

In your example, if you need to free resources for H.264, you need to do
another Offer/Answer exchange in which you will send an offer with VP8 only.

Regards,
_____________
Roman Shpount

--001a1140b4727a580c0530009d58
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<div dir=3D"ltr"><div class=3D"gmail_extra"><div><div class=3D"gmail_signat=
ure">On Fri, Apr 8, 2016 at 6:01 PM, Harald Alvestrand <span dir=3D"ltr">&l=
t;<a href=3D"mailto:harald@alvestrand.no" target=3D"_blank">harald@alvestra=
nd.no</a>&gt;</span> wrote:<br></div></div><div class=3D"gmail_quote"><bloc=
kquote class=3D"gmail_quote" style=3D"margin:0px 0px 0px 0.8ex;border-left-=
width:1px;border-left-style:solid;border-left-color:rgb(204,204,204);paddin=
g-left:1ex">This indicates that the answer&#39;s omission of a codec does n=
ot allow the<br>
offerer to deallocate resources that it has reserved for decoding that<br>
type of stream (such as hardware).<br>
<br>
Under this interpretation, if I reserve hardware for decoding VP8 and<br>
H.264, and I get back an answer that indicates VP8 only, I cannot<br>
deallocate my H.264 resources, since the answerer may choose to send me<br>
H.264 anyway.<br>
<br></blockquote><div><br></div>Based on what I heard from multiple people =
(including Jonathan Rosenberg when this document was originally discussed),=
 Offer/Answer was always intended to be asymmetrical. Each side can declare=
 all the codecs it is willing to receive and can send any codec remote part=
y supports. I saw multiple examples of this being implemented and used in e=
xactly this manner.</div><div class=3D"gmail_quote"><br></div><div class=3D=
"gmail_quote">In your example, if you need to free resources for H.264, you=
 need to do another Offer/Answer exchange in which you will send an offer w=
ith VP8 only.</div><div class=3D"gmail_quote"><br></div><div class=3D"gmail=
_quote">Regards,<br><div><div class=3D"gmail_signature">_____________<br>Ro=
man Shpount</div></div><div>=C2=A0</div></div></div></div>

--001a1140b4727a580c0530009d58--


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From: Roman Shpount <roman@telurix.com>
To: Taylor Brandstetter <deadbeef@google.com>
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Subject: Re: [rtcweb] Followup to discussion of O/A and codecs from Friday session
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On Fri, Apr 8, 2016 at 6:18 PM, Taylor Brandstetter <deadbeef@google.com>
wrote:

> In Section 6 of RFC 3264:
>
>> The answerer MUST send using a media format in the offer that is also listed in the answer
>>
>> So, if X offers A and B, and Y answers B, how would Y be able to send A?
> It's not in both the offer and the answer.
>
>
A does not have to be in both offer and answer. Offer/Answer is declarative
capability exchange with each side declaring what it is willing to receive.
It is not intended to pick a single codec. Negotiation is considered
successful if there is at least one common codec, but there are no other
requirements. Each party can send any codec remote party is willing to
receive and can switch between the codecs remote party supports at any
time. So in this example, Y can not only send A but can switch between A
and B at any time.

Also, answer can include codecs which were not present in the offer, so the
whole situation can be reversed.
_____________
Roman Shpount

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<div dir=3D"ltr"><div class=3D"gmail_extra"><div><div class=3D"gmail_signat=
ure"><br></div></div><div class=3D"gmail_quote">On Fri, Apr 8, 2016 at 6:18=
 PM, Taylor Brandstetter <span dir=3D"ltr">&lt;<a href=3D"mailto:deadbeef@g=
oogle.com" target=3D"_blank">deadbeef@google.com</a>&gt;</span> wrote:<br><=
blockquote class=3D"gmail_quote" style=3D"margin:0px 0px 0px 0.8ex;border-l=
eft-width:1px;border-left-style:solid;border-left-color:rgb(204,204,204);pa=
dding-left:1ex"><div dir=3D"ltr">In Section 6 of RFC 3264:<div><blockquote =
class=3D"gmail_quote" style=3D"margin:0px 0px 0px 0.8ex;border-left-width:1=
px;border-left-color:rgb(204,204,204);border-left-style:solid;padding-left:=
1ex"><pre style=3D"color:rgb(0,0,0);word-wrap:break-word;white-space:pre-wr=
ap">The answerer MUST send using a media format in the offer that is also l=
isted in the answer</pre></blockquote><div><span style=3D"font-size:12.8px"=
>So, if X offers A and B, and Y answers B, how would Y be able to send A? I=
t&#39;s not in both the offer and the answer.</span></div></div></div><div =
class=3D""><div class=3D"h5"><div class=3D"gmail_extra"><br></div></div></d=
iv></blockquote><div><br></div><div>A does not have to be in both offer and=
 answer. Offer/Answer is declarative capability exchange with each side dec=
laring what it is willing to receive. It is not intended to pick a single c=
odec. Negotiation is considered successful if there is at least one common =
codec, but there are no other requirements. Each party can send any codec r=
emote party is willing to receive and can switch between the codecs remote =
party supports at any time. So in this example, Y can not only send A but c=
an switch between A and B at any time.</div><div><br></div><div>Also, answe=
r can include codecs which were not present in the offer, so the whole situ=
ation can be reversed.</div><div><div class=3D"gmail_signature">___________=
__<br>Roman Shpount</div></div><div>=C2=A0</div></div></div></div>

--bcaec502d300528d2e053000b3e2--


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From: Jean-Marc Valin <jmvalin@mozilla.com>
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-----BEGIN PGP SIGNED MESSAGE-----
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On 04/08/2016 10:40 AM, Martin Thomson wrote:
> grammar of the first sentence (Jonathan) "default duration" for
> gaps should be "default gap" (ekr) strike "such as browsers" from
> second paragraph (Alissa) change "will not" to "are not require to"
> (Martin/Fluffy)

Can someone point me to the latest text to which the above applies? I
did not attend Buenos Aires and could not find anything in the meeting
slides.

Cheers,

	Jean-Marc
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From: Taylor Brandstetter <deadbeef@google.com>
To: Roman Shpount <roman@telurix.com>
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Subject: Re: [rtcweb] Followup to discussion of O/A and codecs from Friday session
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I understand if that was the intent, but that's not what RFC 3264 seems to
say. Both Section 6 and Section 8.3.2 state that the answer MUST send a
codec listed in both the offer and the answer.

So is RFC 3264 wrong? Or is my interpretation of the rule wrong?

On Fri, Apr 8, 2016 at 3:27 PM, Roman Shpount <roman@telurix.com> wrote:

>
> On Fri, Apr 8, 2016 at 6:18 PM, Taylor Brandstetter <deadbeef@google.com>
> wrote:
>
>> In Section 6 of RFC 3264:
>>
>>> The answerer MUST send using a media format in the offer that is also listed in the answer
>>>
>>> So, if X offers A and B, and Y answers B, how would Y be able to send A?
>> It's not in both the offer and the answer.
>>
>>
> A does not have to be in both offer and answer. Offer/Answer is
> declarative capability exchange with each side declaring what it is willing
> to receive. It is not intended to pick a single codec. Negotiation is
> considered successful if there is at least one common codec, but there are
> no other requirements. Each party can send any codec remote party is
> willing to receive and can switch between the codecs remote party supports
> at any time. So in this example, Y can not only send A but can switch
> between A and B at any time.
>
> Also, answer can include codecs which were not present in the offer, so
> the whole situation can be reversed.
> _____________
> Roman Shpount
>
>

--001a11c00a305e0f9a053000d228
Content-Type: text/html; charset=UTF-8
Content-Transfer-Encoding: quoted-printable

<div dir=3D"ltr">I understand if that was the intent, but that&#39;s not wh=
at RFC 3264 seems to say. Both Section 6 and Section 8.3.2 state that the a=
nswer MUST send a codec listed in both the offer and the answer.<div><br></=
div><div>So is RFC 3264 wrong? Or is my interpretation of the rule wrong?</=
div></div><div class=3D"gmail_extra"><br><div class=3D"gmail_quote">On Fri,=
 Apr 8, 2016 at 3:27 PM, Roman Shpount <span dir=3D"ltr">&lt;<a href=3D"mai=
lto:roman@telurix.com" target=3D"_blank">roman@telurix.com</a>&gt;</span> w=
rote:<br><blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;borde=
r-left:1px #ccc solid;padding-left:1ex"><div dir=3D"ltr"><div class=3D"gmai=
l_extra"><div><div><br></div></div><div class=3D"gmail_quote"><span class=
=3D"">On Fri, Apr 8, 2016 at 6:18 PM, Taylor Brandstetter <span dir=3D"ltr"=
>&lt;<a href=3D"mailto:deadbeef@google.com" target=3D"_blank">deadbeef@goog=
le.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-style:solid;bord=
er-left-color:rgb(204,204,204);padding-left:1ex"><div dir=3D"ltr">In Sectio=
n 6 of RFC 3264:<div><blockquote class=3D"gmail_quote" style=3D"margin:0px =
0px 0px 0.8ex;border-left-width:1px;border-left-color:rgb(204,204,204);bord=
er-left-style:solid;padding-left:1ex"><pre style=3D"color:rgb(0,0,0);word-w=
rap:break-word;white-space:pre-wrap">The answerer MUST send using a media f=
ormat in the offer that is also listed in the answer</pre></blockquote><div=
><span style=3D"font-size:12.8px">So, if X offers A and B, and Y answers B,=
 how would Y be able to send A? It&#39;s not in both the offer and the answ=
er.</span></div></div></div><div><div><div class=3D"gmail_extra"><br></div>=
</div></div></blockquote><div><br></div></span><div>A does not have to be i=
n both offer and answer. Offer/Answer is declarative capability exchange wi=
th each side declaring what it is willing to receive. It is not intended to=
 pick a single codec. Negotiation is considered successful if there is at l=
east one common codec, but there are no other requirements. Each party can =
send any codec remote party is willing to receive and can switch between th=
e codecs remote party supports at any time. So in this example, Y can not o=
nly send A but can switch between A and B at any time.</div><div><br></div>=
<div>Also, answer can include codecs which were not present in the offer, s=
o the whole situation can be reversed.</div><div><div>_____________<span cl=
ass=3D"HOEnZb"><font color=3D"#888888"><br>Roman Shpount</font></span></div=
></div><div>=C2=A0</div></div></div></div>
</blockquote></div><br></div>

--001a11c00a305e0f9a053000d228--


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From: Taylor Brandstetter <deadbeef@google.com>
To: Roman Shpount <roman@telurix.com>
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Subject: Re: [rtcweb] Followup to discussion of O/A and codecs from Friday session
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--001a113e490a1748d9053000f902
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Actually, I see that Section 5.1 states:
>
> [T]he answerer MAY change formats in the middle of the session, making use of any of the formats listed [in the offer], without sending a new offer.
>
> So, does the combination of these statements imply that the answerer MUST
start by sending a format common to both the offer and the answer, but
later MAY switch to a format specific to the offer? That seems odd, but I
can't think of any other conclusion.

On Fri, Apr 8, 2016 at 3:36 PM, Taylor Brandstetter <deadbeef@google.com>
wrote:

> I understand if that was the intent, but that's not what RFC 3264 seems to
> say. Both Section 6 and Section 8.3.2 state that the answer MUST send a
> codec listed in both the offer and the answer.
>
> So is RFC 3264 wrong? Or is my interpretation of the rule wrong?
>
> On Fri, Apr 8, 2016 at 3:27 PM, Roman Shpount <roman@telurix.com> wrote:
>
>>
>> On Fri, Apr 8, 2016 at 6:18 PM, Taylor Brandstetter <deadbeef@google.com>
>> wrote:
>>
>>> In Section 6 of RFC 3264:
>>>
>>>> The answerer MUST send using a media format in the offer that is also listed in the answer
>>>>
>>>> So, if X offers A and B, and Y answers B, how would Y be able to send
>>> A? It's not in both the offer and the answer.
>>>
>>>
>> A does not have to be in both offer and answer. Offer/Answer is
>> declarative capability exchange with each side declaring what it is willing
>> to receive. It is not intended to pick a single codec. Negotiation is
>> considered successful if there is at least one common codec, but there are
>> no other requirements. Each party can send any codec remote party is
>> willing to receive and can switch between the codecs remote party supports
>> at any time. So in this example, Y can not only send A but can switch
>> between A and B at any time.
>>
>> Also, answer can include codecs which were not present in the offer, so
>> the whole situation can be reversed.
>> _____________
>> Roman Shpount
>>
>>
>
>

--001a113e490a1748d9053000f902
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<div dir=3D"ltr">Actually, I see that Section 5.1 states:<blockquote class=
=3D"gmail_quote" style=3D"margin:0px 0px 0px 0.8ex;border-left-width:1px;bo=
rder-left-color:rgb(204,204,204);border-left-style:solid;padding-left:1ex">=
<pre style=3D"color:rgb(0,0,0);word-wrap:break-word;white-space:pre-wrap">[=
T]he answerer MAY change formats in the middle of the session, making use o=
f any of the formats listed [in the offer], without sending a new offer.</p=
re></blockquote><div>So, does the combination of these statements imply tha=
t the answerer MUST start by sending a format common to both the offer and =
the answer, but later MAY switch to a format specific to the offer? That se=
ems odd, but I can&#39;t think of any other conclusion.<br></div></div><div=
 class=3D"gmail_extra"><br><div class=3D"gmail_quote">On Fri, Apr 8, 2016 a=
t 3:36 PM, Taylor Brandstetter <span dir=3D"ltr">&lt;<a href=3D"mailto:dead=
beef@google.com" target=3D"_blank">deadbeef@google.com</a>&gt;</span> wrote=
:<br><blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-le=
ft:1px #ccc solid;padding-left:1ex"><div dir=3D"ltr">I understand if that w=
as the intent, but that&#39;s not what RFC 3264 seems to say. Both Section =
6 and Section 8.3.2 state that the answer MUST send a codec listed in both =
the offer and the answer.<div><br></div><div>So is RFC 3264 wrong? Or is my=
 interpretation of the rule wrong?</div></div><div class=3D"HOEnZb"><div cl=
ass=3D"h5"><div class=3D"gmail_extra"><br><div class=3D"gmail_quote">On Fri=
, Apr 8, 2016 at 3:27 PM, Roman Shpount <span dir=3D"ltr">&lt;<a href=3D"ma=
ilto:roman@telurix.com" target=3D"_blank">roman@telurix.com</a>&gt;</span> =
wrote:<br><blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;bord=
er-left:1px #ccc solid;padding-left:1ex"><div dir=3D"ltr"><div class=3D"gma=
il_extra"><div><div><br></div></div><div class=3D"gmail_quote"><span>On Fri=
, Apr 8, 2016 at 6:18 PM, Taylor Brandstetter <span dir=3D"ltr">&lt;<a href=
=3D"mailto:deadbeef@google.com" target=3D"_blank">deadbeef@google.com</a>&g=
t;</span> wrote:<br><blockquote class=3D"gmail_quote" style=3D"margin:0px 0=
px 0px 0.8ex;border-left-width:1px;border-left-style:solid;border-left-colo=
r:rgb(204,204,204);padding-left:1ex"><div dir=3D"ltr">In Section 6 of RFC 3=
264:<div><blockquote class=3D"gmail_quote" style=3D"margin:0px 0px 0px 0.8e=
x;border-left-width:1px;border-left-color:rgb(204,204,204);border-left-styl=
e:solid;padding-left:1ex"><pre style=3D"color:rgb(0,0,0);word-wrap:break-wo=
rd;white-space:pre-wrap">The answerer MUST send using a media format in the=
 offer that is also listed in the answer</pre></blockquote><div><span style=
=3D"font-size:12.8px">So, if X offers A and B, and Y answers B, how would Y=
 be able to send A? It&#39;s not in both the offer and the answer.</span></=
div></div></div><div><div><div class=3D"gmail_extra"><br></div></div></div>=
</blockquote><div><br></div></span><div>A does not have to be in both offer=
 and answer. Offer/Answer is declarative capability exchange with each side=
 declaring what it is willing to receive. It is not intended to pick a sing=
le codec. Negotiation is considered successful if there is at least one com=
mon codec, but there are no other requirements. Each party can send any cod=
ec remote party is willing to receive and can switch between the codecs rem=
ote party supports at any time. So in this example, Y can not only send A b=
ut can switch between A and B at any time.</div><div><br></div><div>Also, a=
nswer can include codecs which were not present in the offer, so the whole =
situation can be reversed.</div><div><div>_____________<span><font color=3D=
"#888888"><br>Roman Shpount</font></span></div></div><div>=C2=A0</div></div=
></div></div>
</blockquote></div><br></div>
</div></div></blockquote></div><br></div>

--001a113e490a1748d9053000f902--


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From: Roman Shpount <roman@telurix.com>
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Subject: Re: [rtcweb] Followup to discussion of O/A and codecs from Friday session
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Taylor,

After rereading RFC3264, I can see that you are correct: If the codec is
not present in the answer, offerer MAY stop listening for this payload type
(https://tools.ietf.org/html/rfc3264#section-7).

Sorry for confusion,

_____________
Roman Shpount

On Fri, Apr 8, 2016 at 6:36 PM, Taylor Brandstetter <deadbeef@google.com>
wrote:

> I understand if that was the intent, but that's not what RFC 3264 seems to
> say. Both Section 6 and Section 8.3.2 state that the answer MUST send a
> codec listed in both the offer and the answer.
>
> So is RFC 3264 wrong? Or is my interpretation of the rule wrong?
>
> On Fri, Apr 8, 2016 at 3:27 PM, Roman Shpount <roman@telurix.com> wrote:
>
>>
>> On Fri, Apr 8, 2016 at 6:18 PM, Taylor Brandstetter <deadbeef@google.com>
>> wrote:
>>
>>> In Section 6 of RFC 3264:
>>>
>>>> The answerer MUST send using a media format in the offer that is also listed in the answer
>>>>
>>>> So, if X offers A and B, and Y answers B, how would Y be able to send
>>> A? It's not in both the offer and the answer.
>>>
>>>
>> A does not have to be in both offer and answer. Offer/Answer is
>> declarative capability exchange with each side declaring what it is willing
>> to receive. It is not intended to pick a single codec. Negotiation is
>> considered successful if there is at least one common codec, but there are
>> no other requirements. Each party can send any codec remote party is
>> willing to receive and can switch between the codecs remote party supports
>> at any time. So in this example, Y can not only send A but can switch
>> between A and B at any time.
>>
>> Also, answer can include codecs which were not present in the offer, so
>> the whole situation can be reversed.
>> _____________
>> Roman Shpount
>>
>>
>
>

--001a113f91ae478d3a053000f938
Content-Type: text/html; charset=UTF-8
Content-Transfer-Encoding: quoted-printable

<div dir=3D"ltr">Taylor,<div><br></div><div>After rereading RFC3264, I can =
see that you are correct: If the codec is not present in the answer, offere=
r MAY stop listening for this payload type (<a href=3D"https://tools.ietf.o=
rg/html/rfc3264#section-7">https://tools.ietf.org/html/rfc3264#section-7</a=
>).</div><div><br></div><div>Sorry for confusion,</div></div><div class=3D"=
gmail_extra"><br clear=3D"all"><div><div class=3D"gmail_signature">________=
_____<br>Roman Shpount</div></div>
<br><div class=3D"gmail_quote">On Fri, Apr 8, 2016 at 6:36 PM, Taylor Brand=
stetter <span dir=3D"ltr">&lt;<a href=3D"mailto:deadbeef@google.com" target=
=3D"_blank">deadbeef@google.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">I understand if that was the intent, but tha=
t&#39;s not what RFC 3264 seems to say. Both Section 6 and Section 8.3.2 st=
ate that the answer MUST send a codec listed in both the offer and the answ=
er.<div><br></div><div>So is RFC 3264 wrong? Or is my interpretation of the=
 rule wrong?</div></div><div class=3D"HOEnZb"><div class=3D"h5"><div class=
=3D"gmail_extra"><br><div class=3D"gmail_quote">On Fri, Apr 8, 2016 at 3:27=
 PM, Roman Shpount <span dir=3D"ltr">&lt;<a href=3D"mailto:roman@telurix.co=
m" target=3D"_blank">roman@telurix.com</a>&gt;</span> wrote:<br><blockquote=
 class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1px #ccc soli=
d;padding-left:1ex"><div dir=3D"ltr"><div class=3D"gmail_extra"><div><div><=
br></div></div><div class=3D"gmail_quote"><span>On Fri, Apr 8, 2016 at 6:18=
 PM, Taylor Brandstetter <span dir=3D"ltr">&lt;<a href=3D"mailto:deadbeef@g=
oogle.com" target=3D"_blank">deadbeef@google.com</a>&gt;</span> wrote:<br><=
blockquote class=3D"gmail_quote" style=3D"margin:0px 0px 0px 0.8ex;border-l=
eft-width:1px;border-left-style:solid;border-left-color:rgb(204,204,204);pa=
dding-left:1ex"><div dir=3D"ltr">In Section 6 of RFC 3264:<div><blockquote =
class=3D"gmail_quote" style=3D"margin:0px 0px 0px 0.8ex;border-left-width:1=
px;border-left-color:rgb(204,204,204);border-left-style:solid;padding-left:=
1ex"><pre style=3D"color:rgb(0,0,0);word-wrap:break-word;white-space:pre-wr=
ap">The answerer MUST send using a media format in the offer that is also l=
isted in the answer</pre></blockquote><div><span style=3D"font-size:12.8px"=
>So, if X offers A and B, and Y answers B, how would Y be able to send A? I=
t&#39;s not in both the offer and the answer.</span></div></div></div><div>=
<div><div class=3D"gmail_extra"><br></div></div></div></blockquote><div><br=
></div></span><div>A does not have to be in both offer and answer. Offer/An=
swer is declarative capability exchange with each side declaring what it is=
 willing to receive. It is not intended to pick a single codec. Negotiation=
 is considered successful if there is at least one common codec, but there =
are no other requirements. Each party can send any codec remote party is wi=
lling to receive and can switch between the codecs remote party supports at=
 any time. So in this example, Y can not only send A but can switch between=
 A and B at any time.</div><div><br></div><div>Also, answer can include cod=
ecs which were not present in the offer, so the whole situation can be reve=
rsed.</div><div><div>_____________<span><font color=3D"#888888"><br>Roman S=
hpount</font></span></div></div><div>=C2=A0</div></div></div></div>
</blockquote></div><br></div>
</div></div></blockquote></div><br></div>

--001a113f91ae478d3a053000f938--


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From: Roman Shpount <roman@telurix.com>
To: "rtcweb@ietf.org" <rtcweb@ietf.org>, Justin Uberti <juberti@google.com>,  Christer Holmberg <christer.holmberg@ericsson.com>
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Subject: [rtcweb] DTLS role during SDP renegotiation
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--089e01537a4079562805300194b9
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Hi All,

I've listened through the IEFT95 session recording and I wanted to clarify
one thing. DTLS setup attribute MUST be set to the currently negotiated
role if not accompanied by ICE restart and SHOULD be actpass in case of ICE
restart. The reason for this is that answerer is allowed to start a new
DTLS association in response to the offer with ICE restart. When answerer
starts a new DTLS association, it might be beneficial to change the DTLS
setup role. Answerer is not allowed to start new DTLS association if
offerer did not initiate an ICE restart since new transport cannot be
created for the DTLS association, so offerer MUST preserve the setup role
in this case.

Regards,
_____________
Roman Shpount

--089e01537a4079562805300194b9
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Content-Transfer-Encoding: quoted-printable

<div dir=3D"ltr"><div>Hi All,</div><div><br></div>I&#39;ve listened through=
 the IEFT95 session recording and I wanted to clarify one thing. DTLS setup=
 attribute MUST be set to the currently negotiated role if not accompanied =
by ICE restart and SHOULD be actpass in case of ICE restart. The reason for=
 this is that answerer is allowed to start a new DTLS association in respon=
se to the offer with ICE restart. When answerer starts a new DTLS associati=
on, it might be beneficial to change the DTLS setup role. Answerer is not a=
llowed to start new DTLS association if offerer did not initiate an ICE res=
tart since new transport cannot be created for the DTLS association, so off=
erer MUST preserve the setup role in this case.<div><br></div><div>Regards,=
<br clear=3D"all"><div><div class=3D"gmail_signature">_____________<br>Roma=
n Shpount</div></div>
</div></div>

--089e01537a4079562805300194b9--


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From: Paul Kyzivat <pkyzivat@alum.mit.edu>
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Subject: Re: [rtcweb] Followup to discussion of O/A and codecs from Friday session
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On 4/8/16 6:47 PM, Roman Shpount wrote:
> Taylor,
>
> After rereading RFC3264, I can see that you are correct: If the codec is
> not present in the answer, offerer MAY stop listening for this payload
> type (https://tools.ietf.org/html/rfc3264#section-7).
>
> Sorry for confusion,

Hmm.

Also, the end of section 6.1 says:

    The answerer MUST send using a media format in the offer
    that is also listed in the answer,

Together, those contradict what I said above.

So I guess Harald gets his wish to be able to discard resources for 
codecs that weren't mentioned in the answer.

Unfortunately this also means it isn't possible to have asymmetric 
behavior. For instance, it means you can't have telephone-events work in 
one direction and not the other.

	Sorry for missing the above,
	Paul


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From: "Cullen Jennings (fluffy)" <fluffy@cisco.com>
To: Jean-Marc Valin <jmvalin@mozilla.com>
Thread-Topic: friendly amendments to text on DTMF resolution
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Cc: Jonathan Lennox <jonathan@vidyo.com>, Ted Hardie <hardie@google.com>, "rtcweb@ietf.org" <rtcweb@ietf.org>
Subject: Re: [rtcweb] friendly amendments to text on DTMF resolution
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> On Apr 8, 2016, at 8:40 AM, Martin Thomson <martin.thomson@gmail.com> wro=
te:
>=20
> grammar of the first sentence (Jonathan)
> "default duration" for gaps should be "default gap" (ekr)
> strike "such as browsers" from second paragraph (Alissa)
> change "will not" to "are not require to" (Martin/Fluffy)

Just wanted to +1 that Martin got all the points I recall on this. =


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From: Ted Hardie <ted.ietf@gmail.com>
Date: Wed, 13 Apr 2016 13:31:41 -0700
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Subject: [rtcweb] Confirmation of DTMF issue
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--001a113e3966870a6c053063abee
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With Martin's friendly amendments, the room at IETF 95 agreed to the follow
as the resolution of the DTMF discussion.  The chairs would like anyone who
has new issues with this to let us know as soon as possible and no later
than Wednesday, April 20th, noon UTC,


"DTMF events generated by a WebRTC endpoint MUST have a duration of no more
than 8000 ms and no less than 40 ms.  The recommended default duration is
100 ms for each tone. The gap between events MUST be no less than 30 ms;
 the recommended default gap duration is 70 ms.

WebRTC endpoints are not required to do anything with RFC 4733 tones sent
to them, except gracefully drop them. There is currently no API to inform
JavaScript about the received DTMF or other RFC 4733 tones."

regards,

Ted, Cullen, and Sean

--001a113e3966870a6c053063abee
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<div dir=3D"ltr"><div><div>With Martin&#39;s friendly amendments, the room =
at IETF 95 agreed to the follow as the resolution of the DTMF discussion.=
=C2=A0 The chairs would like anyone who has new issues with this to let us =
know as soon as possible and no later than Wednesday, April 20th, noon UTC,=
 <br><br><br>&quot;<span style=3D"font-family:arial,helvetica,sans-serif"><=
span style=3D"font-size:10.6667px;color:rgb(0,0,0);background-color:transpa=
rent;font-weight:400;font-style:normal;font-variant:normal;text-decoration:=
none;vertical-align:baseline">DTMF events generated by a WebRTC endpoint MU=
ST have a duration of no more than 8000 ms and no less than 40 ms.=C2=A0 Th=
e recommended default duration is 100 ms for each tone. The gap between eve=
nts MUST be no less than 30 ms; =C2=A0the recommended default gap duration =
is 70 ms.</span><br><br><span style=3D"font-size:10.6667px;color:rgb(0,0,0)=
;background-color:transparent;font-weight:400;font-style:normal;font-varian=
t:normal;text-decoration:none;vertical-align:baseline">WebRTC endpoints are=
 not required to do anything with RFC 4733 tones sent to them, except grace=
fully drop them. There is currently no API to inform JavaScript about the r=
eceived DTMF or other RFC 4733 tones.&quot;<br><br></span></span></div><spa=
n style=3D"font-family:arial,helvetica,sans-serif"><span style=3D"font-size=
:10.6667px;color:rgb(0,0,0);background-color:transparent;font-weight:400;fo=
nt-style:normal;font-variant:normal;text-decoration:none;vertical-align:bas=
eline">regards,<br><br></span></span></div><span style=3D"font-family:arial=
,helvetica,sans-serif"><span style=3D"font-size:10.6667px;color:rgb(0,0,0);=
background-color:transparent;font-weight:400;font-style:normal;font-variant=
:normal;text-decoration:none;vertical-align:baseline">Ted, Cullen, and Sean=
<br></span></span></div>

--001a113e3966870a6c053063abee--


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From: Roman Shpount <roman@telurix.com>
To: Ted Hardie <ted.ietf@gmail.com>
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Subject: Re: [rtcweb] Confirmation of DTMF issue
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--047d7bdc08a66b0078053063f178
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I support this language.

_____________
Roman Shpount

On Wed, Apr 13, 2016 at 4:31 PM, Ted Hardie <ted.ietf@gmail.com> wrote:

> With Martin's friendly amendments, the room at IETF 95 agreed to the
> follow as the resolution of the DTMF discussion.  The chairs would like
> anyone who has new issues with this to let us know as soon as possible and
> no later than Wednesday, April 20th, noon UTC,
>
>
> "DTMF events generated by a WebRTC endpoint MUST have a duration of no
> more than 8000 ms and no less than 40 ms.  The recommended default duration
> is 100 ms for each tone. The gap between events MUST be no less than 30 ms;
>  the recommended default gap duration is 70 ms.
>
> WebRTC endpoints are not required to do anything with RFC 4733 tones sent
> to them, except gracefully drop them. There is currently no API to inform
> JavaScript about the received DTMF or other RFC 4733 tones."
>
> regards,
>
> Ted, Cullen, and Sean
>
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb
>
>

--047d7bdc08a66b0078053063f178
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<div dir=3D"ltr">I support this language.</div><div class=3D"gmail_extra"><=
br clear=3D"all"><div><div class=3D"gmail_signature">_____________<br>Roman=
 Shpount</div></div>
<br><div class=3D"gmail_quote">On Wed, Apr 13, 2016 at 4:31 PM, Ted Hardie =
<span dir=3D"ltr">&lt;<a href=3D"mailto:ted.ietf@gmail.com" target=3D"_blan=
k">ted.ietf@gmail.com</a>&gt;</span> wrote:<br><blockquote class=3D"gmail_q=
uote" style=3D"margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1e=
x"><div dir=3D"ltr"><div><div>With Martin&#39;s friendly amendments, the ro=
om at IETF 95 agreed to the follow as the resolution of the DTMF discussion=
.=C2=A0 The chairs would like anyone who has new issues with this to let us=
 know as soon as possible and no later than Wednesday, April 20th, noon UTC=
, <br><br><br>&quot;<span style=3D"font-family:arial,helvetica,sans-serif">=
<span style=3D"font-size:10.6667px;color:rgb(0,0,0);background-color:transp=
arent;font-weight:400;font-style:normal;font-variant:normal;text-decoration=
:none;vertical-align:baseline">DTMF events generated by a WebRTC endpoint M=
UST have a duration of no more than 8000 ms and no less than 40 ms.=C2=A0 T=
he recommended default duration is 100 ms for each tone. The gap between ev=
ents MUST be no less than 30 ms; =C2=A0the recommended default gap duration=
 is 70 ms.</span><br><br><span style=3D"font-size:10.6667px;color:rgb(0,0,0=
);background-color:transparent;font-weight:400;font-style:normal;font-varia=
nt:normal;text-decoration:none;vertical-align:baseline">WebRTC endpoints ar=
e not required to do anything with RFC 4733 tones sent to them, except grac=
efully drop them. There is currently no API to inform JavaScript about the =
received DTMF or other RFC 4733 tones.&quot;<br><br></span></span></div><sp=
an style=3D"font-family:arial,helvetica,sans-serif"><span style=3D"font-siz=
e:10.6667px;color:rgb(0,0,0);background-color:transparent;font-weight:400;f=
ont-style:normal;font-variant:normal;text-decoration:none;vertical-align:ba=
seline">regards,<br><br></span></span></div><span style=3D"font-family:aria=
l,helvetica,sans-serif"><span style=3D"font-size:10.6667px;color:rgb(0,0,0)=
;background-color:transparent;font-weight:400;font-style:normal;font-varian=
t:normal;text-decoration:none;vertical-align:baseline">Ted, Cullen, and Sea=
n<br></span></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" rel=3D"noreferrer"=
 target=3D"_blank">https://www.ietf.org/mailman/listinfo/rtcweb</a><br>
<br></blockquote></div><br></div>

--047d7bdc08a66b0078053063f178--


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From: Martin Thomson <martin.thomson@gmail.com>
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Subject: Re: [rtcweb] Confirmation of DTMF issue
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+1

A bigger +1 to finishing this.  I never realized that immersion
coating was possible for something as large as a bike shed.

On 13 April 2016 at 17:31, Ted Hardie <ted.ietf@gmail.com> wrote:
> With Martin's friendly amendments, the room at IETF 95 agreed to the follow
> as the resolution of the DTMF discussion.  The chairs would like anyone who
> has new issues with this to let us know as soon as possible and no later
> than Wednesday, April 20th, noon UTC,
>
>
> "DTMF events generated by a WebRTC endpoint MUST have a duration of no more
> than 8000 ms and no less than 40 ms.  The recommended default duration is
> 100 ms for each tone. The gap between events MUST be no less than 30 ms;
> the recommended default gap duration is 70 ms.
>
> WebRTC endpoints are not required to do anything with RFC 4733 tones sent to
> them, except gracefully drop them. There is currently no API to inform
> JavaScript about the received DTMF or other RFC 4733 tones."
>
> regards,
>
> Ted, Cullen, and Sean
>
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb
>


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From: "Asveren, Tolga" <tasveren@sonusnet.com>
To: Martin Thomson <martin.thomson@gmail.com>, Ted Hardie <ted.ietf@gmail.com>
Thread-Topic: [rtcweb] Confirmation of DTMF issue
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Cc: Cullen Jennings <fluffy@cisco.com>, "rtcweb@ietf.org" <rtcweb@ietf.org>
Subject: Re: [rtcweb] Confirmation of DTMF issue
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The text looks good to me too.

Thanks,
Tolga

> -----Original Message-----
> From: rtcweb [mailto:rtcweb-bounces@ietf.org] On Behalf Of Martin
> Thomson
> Sent: Wednesday, April 13, 2016 9:45 PM
> To: Ted Hardie <ted.ietf@gmail.com>
> Cc: Cullen Jennings <fluffy@cisco.com>; rtcweb@ietf.org
> Subject: Re: [rtcweb] Confirmation of DTMF issue
>=20
> +1
>=20
> A bigger +1 to finishing this.  I never realized that immersion coating w=
as
> possible for something as large as a bike shed.
>=20
> On 13 April 2016 at 17:31, Ted Hardie <ted.ietf@gmail.com> wrote:
> > With Martin's friendly amendments, the room at IETF 95 agreed to the
> > follow as the resolution of the DTMF discussion.  The chairs would
> > like anyone who has new issues with this to let us know as soon as
> > possible and no later than Wednesday, April 20th, noon UTC,
> >
> >
> > "DTMF events generated by a WebRTC endpoint MUST have a duration of
> no
> > more than 8000 ms and no less than 40 ms.  The recommended default
> > duration is
> > 100 ms for each tone. The gap between events MUST be no less than 30
> > ms; the recommended default gap duration is 70 ms.
> >
> > WebRTC endpoints are not required to do anything with RFC 4733 tones
> > sent to them, except gracefully drop them. There is currently no API
> > to inform JavaScript about the received DTMF or other RFC 4733 tones."
> >
> > regards,
> >
> > Ted, Cullen, and Sean
> >
> > _______________________________________________
> > 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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To: Ted Hardie <ted.ietf@gmail.com>, "rtcweb@ietf.org" <rtcweb@ietf.org>, Cullen Jennings <fluffy@cisco.com>, Sean Turner <sean@sn3rd.com>
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-----BEGIN PGP SIGNED MESSAGE-----
Hash: SHA256

+1

On 04/13/2016 04:31 PM, Ted Hardie wrote:
> With Martin's friendly amendments, the room at IETF 95 agreed to
> the follow as the resolution of the DTMF discussion.  The chairs
> would like anyone who has new issues with this to let us know as
> soon as possible and no later than Wednesday, April 20th, noon
> UTC,
> 
> 
> "DTMF events generated by a WebRTC endpoint MUST have a duration of
> no more than 8000 ms and no less than 40 ms.  The recommended
> default duration is 100 ms for each tone. The gap between events
> MUST be no less than 30 ms;  the recommended default gap duration
> is 70 ms.
> 
> WebRTC endpoints are not required to do anything with RFC 4733
> tones sent to them, except gracefully drop them. There is currently
> no API to inform JavaScript about the received DTMF or other RFC
> 4733 tones."
> 
> regards,
> 
> Ted, Cullen, and Sean
> 
> 
> _______________________________________________ rtcweb mailing
> list rtcweb@ietf.org https://www.ietf.org/mailman/listinfo/rtcweb
> 
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Subject: Re: [rtcweb] Last call redux: draft-ietf-rtcweb-rtp-usage-26
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No comments were received, so this document will revert to the RFC =
Editor.

> On Mar 23, 2016, at 1:44 PM, Alissa Cooper <alissa@cooperw.in> wrote:
>=20
> draft-ietf-rtcweb-rtp-usage-25 was approved for publication in July =
2015 and is in the RFC Editor's queue. Subsequently, the RTCWEB WG =
identified a change that needed to be made in Section 4.5 related to =
multiplexing RTP and RTCP. The change is reflected in =
draft-ietf-rtcweb-rtp-usage-26 and copied below.
>=20
> The IESG will not revisit its decision to approve this document for =
publication unless it receives feedback from the community indicating a =
need to do so. Please send substantive comments to the ietf@ietf.org =
mailing list by 2016-04-13. Exceptionally, comments may be sent to =
iesg@ietf.org instead.
>=20
> The change is:
>=20
> OLD:
>=20
> If SDP is used for signalling, this
> negotiation MUST use the attributes defined in [RFC5761].  For
> backwards compatibility, implementations are also REQUIRED to support
> RTP and RTCP sent on separate transport-layer flows.
>=20
> NEW:
>=20
> If SDP is used for signalling, this
> negotiation MUST use the mechanism defined in [RFC5761].
> Implementations can also support sending RTP and RTCP on separate
> transport-layer flows, but this is OPTIONAL to implement.  If an
> implementation does not support RTP and RTCP sent on separate
> transport-layer flows, it MUST indicate that using the mechanism
> defined in [I-D.ietf-mmusic-mux-exclusive].
>=20
> The document is available at =
https://datatracker.ietf.org/doc/draft-ietf-rtcweb-rtp-usage/
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb


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----------------------------------------------------------------------
COMMENT:
----------------------------------------------------------------------

The text below is due to be added in Section 3 of the document, pending
list confirmation of consensus in the room from IETF 95, which is due to
wrap up on April 20.

"DTMF events generated by a WebRTC endpoint MUST have a duration of no
more than 8000 ms and no less than 40 ms.  The recommended default
duration is 100 ms for each tone. The gap between events MUST be no less
than 30 ms; the recommended default gap duration is 70 ms.

WebRTC endpoints are not required to do anything with RFC 4733 tones sent
to them, except gracefully drop them. There is currently no API to inform
JavaScript about the received DTMF or other RFC 4733 tones."



From nobody Fri Apr 15 05:56:28 2016
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----------------------------------------------------------------------
COMMENT:
----------------------------------------------------------------------

In Section 5: AEC is a SHOULD implement with no reference. Is this well
understood to be implementable without any reference?

In section 8: acronym VBR is used for the first time without explanation.



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Subject: Re: [rtcweb] Last call redux: draft-ietf-rtcweb-rtp-usage-26
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No comments were received, so this document will revert to the RFC =
Editor.

> On Mar 23, 2016, at 1:44 PM, Alissa Cooper <alissa@cooperw.in> wrote:
>=20
> draft-ietf-rtcweb-rtp-usage-25 was approved for publication in July =
2015 and is in the RFC Editor's queue. Subsequently, the RTCWEB WG =
identified a change that needed to be made in Section 4.5 related to =
multiplexing RTP and RTCP. The change is reflected in =
draft-ietf-rtcweb-rtp-usage-26 and copied below.
>=20
> The IESG will not revisit its decision to approve this document for =
publication unless it receives feedback from the community indicating a =
need to do so. Please send substantive comments to the ietf@ietf.org =
mailing list by 2016-04-13. Exceptionally, comments may be sent to =
iesg@ietf.org instead.
>=20
> The change is:
>=20
> OLD:
>=20
> If SDP is used for signalling, this
> negotiation MUST use the attributes defined in [RFC5761].  For
> backwards compatibility, implementations are also REQUIRED to support
> RTP and RTCP sent on separate transport-layer flows.
>=20
> NEW:
>=20
> If SDP is used for signalling, this
> negotiation MUST use the mechanism defined in [RFC5761].
> Implementations can also support sending RTP and RTCP on separate
> transport-layer flows, but this is OPTIONAL to implement.  If an
> implementation does not support RTP and RTCP sent on separate
> transport-layer flows, it MUST indicate that using the mechanism
> defined in [I-D.ietf-mmusic-mux-exclusive].
>=20
> The document is available at =
https://datatracker.ietf.org/doc/draft-ietf-rtcweb-rtp-usage/
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb



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From nobody Sun Apr 17 22:44:28 2016
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To: Alexey Melnikov <aamelnikov@fastmail.fm>, The IESG <iesg@ietf.org>
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From: Harald Alvestrand <harald@alvestrand.no>
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Subject: Re: [rtcweb] Alexey Melnikov's No Objection on draft-ietf-rtcweb-audio-10: (with COMMENT)
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On 04/15/2016 02:56 PM, Alexey Melnikov wrote:
>
> ----------------------------------------------------------------------
> COMMENT:
> ----------------------------------------------------------------------
>
> In Section 5: AEC is a SHOULD implement with no reference. Is this well
> understood to be implementable without any reference?

I believe the concept of AEC (echo cancellation) is sufficiently well
understood that it can be implemented without a reference.

Specific echo cancellation algorithms are typically quite fluid and/or
proprietary; I do not believe that the Internet would benefit from
referencing any single AEC algorithm.
Two endpoints do not need to agree on an AEC algorithm to function.

>
> In section 8: acronym VBR is used for the first time without explanation.
>
>
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb


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From: Alexey Melnikov <aamelnikov@fastmail.fm>
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Subject: Re: [rtcweb] Alexey Melnikov's No Objection on draft-ietf-rtcweb-audio-10: (with COMMENT)
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Hi Harald,

> On 18 Apr 2016, at 06:44, Harald Alvestrand <harald@alvestrand.no> wrote:
> 
>> On 04/15/2016 02:56 PM, Alexey Melnikov wrote:
>> 
>> ----------------------------------------------------------------------
>> COMMENT:
>> ----------------------------------------------------------------------
>> 
>> In Section 5: AEC is a SHOULD implement with no reference. Is this well
>> understood to be implementable without any reference?
> 
> I believe the concept of AEC (echo cancellation) is sufficiently well
> understood that it can be implemented without a reference.
> 
> Specific echo cancellation algorithms are typically quite fluid and/or
> proprietary; I do not believe that the Internet would benefit from
> referencing any single AEC algorithm.
> Two endpoints do not need to agree on an AEC algorithm to function.

Ok, fair enough.

It would be good to expand the acronym for ignorant people like me though.
> 
>> 
>> In section 8: acronym VBR is used for the first time without explanation.
>> 
>> 
>> _______________________________________________
>> rtcweb mailing list
>> rtcweb@ietf.org
>> https://www.ietf.org/mailman/listinfo/rtcweb
> 


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Subject: Re: [rtcweb] Alexey Melnikov's No Objection on draft-ietf-rtcweb-audio-10: (with COMMENT)
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On 2016/04/18 16:30, Alexey Melnikov wrote:

>> On 18 Apr 2016, at 06:44, Harald Alvestrand <harald@alvestrand.no> wrote:

>>> In Section 5: AEC is a SHOULD implement with no reference. Is this well
>>> understood to be implementable without any reference?
>>
>> I believe the concept of AEC (echo cancellation) is sufficiently well
>> understood that it can be implemented without a reference.
>>
>> Specific echo cancellation algorithms are typically quite fluid and/or
>> proprietary; I do not believe that the Internet would benefit from
>> referencing any single AEC algorithm.
>> Two endpoints do not need to agree on an AEC algorithm to function.
>
> Ok, fair enough.
>
> It would be good to expand the acronym for ignorant people like me though.

I wanted to make exactly the same comment, but then I checked, and the 
expansion is given in the TOC and the title of section 5 (the only place 
where the acronym is used without expansion), so it might be good enough.

Regards,   Martin.


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From: Alexey Melnikov <aamelnikov@fastmail.fm>
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Subject: Re: [rtcweb] Alexey Melnikov's No Objection on draft-ietf-rtcweb-audio-10: (with COMMENT)
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> On 18 Apr 2016, at 08:46, Martin J. D=C3=BCrst <duerst@it.aoyama.ac.jp> wr=
ote:
>=20
> On 2016/04/18 16:30, Alexey Melnikov wrote:
>=20
>>> On 18 Apr 2016, at 06:44, Harald Alvestrand <harald@alvestrand.no> wrote=
:
>=20
>>>> In Section 5: AEC is a SHOULD implement with no reference. Is this well=

>>>> understood to be implementable without any reference?
>>>=20
>>> I believe the concept of AEC (echo cancellation) is sufficiently well
>>> understood that it can be implemented without a reference.
>>>=20
>>> Specific echo cancellation algorithms are typically quite fluid and/or
>>> proprietary; I do not believe that the Internet would benefit from
>>> referencing any single AEC algorithm.
>>> Two endpoints do not need to agree on an AEC algorithm to function.
>>=20
>> Ok, fair enough.
>>=20
>> It would be good to expand the acronym for ignorant people like me though=
.
>=20
> I wanted to make exactly the same comment, but then I checked, and the exp=
ansion is given in the TOC and the title of section 5 (the only place where t=
he acronym is used without expansion), so it might be good enough.

Ok. RFC Editor is quite good at fixing these anyway.



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From: Roman Shpount <roman@telurix.com>
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Subject: Re: [rtcweb] Tunnelling DTLS in SDP
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--001a1140a2b621b1a00530c39d18
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On Mon, Apr 4, 2016 at 9:10 AM, Eric Rescorla <ekr@rtfm.com> wrote:

> I wanted to call your attention to a draft I just published with a
> possibly stupid
> idea.
>
> https://tools.ietf.org/html/draft-rescorla-dtls-in-sdp-00
>
> A nontrivial fraction of call setup time in WebRTC is the DTLS handshake.
> This document describes how to piggyback the first few handshake messages
> in the SDP offer/answer exchange, thus reducing latency.
>
> Comments welcome.
>
>
One other issue I thought off was that with DTLS SDP either offerer or
answerer can start a new DTLS association. When answerer starts a new
session it will have exactly the same problems as forking, since this will
create multiple DTLS sessions with the same client random. This can be
solved with some sort of fallback mechanism to either regular DTLS setup or
to sending a ClientHello in the answer.

Regards,
_____________
Roman Shpount

--001a1140a2b621b1a00530c39d18
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<div dir=3D"ltr"><div class=3D"gmail_extra"><div><div class=3D"gmail_signat=
ure">On Mon, Apr 4, 2016 at 9:10 AM, Eric Rescorla <span dir=3D"ltr">&lt;<a=
 href=3D"mailto:ekr@rtfm.com" target=3D"_blank">ekr@rtfm.com</a>&gt;</span>=
 wrote:<br></div></div><div class=3D"gmail_quote"><blockquote class=3D"gmai=
l_quote" style=3D"margin:0px 0px 0px 0.8ex;border-left-width:1px;border-lef=
t-style:solid;border-left-color:rgb(204,204,204);padding-left:1ex"><div dir=
=3D"ltr"><div>I wanted to call your attention to a draft I just published w=
ith a possibly stupid<br></div><div>idea.</div><div><br></div><div><a href=
=3D"https://tools.ietf.org/html/draft-rescorla-dtls-in-sdp-00" target=3D"_b=
lank">https://tools.ietf.org/html/draft-rescorla-dtls-in-sdp-00</a><br></di=
v><div><br></div><div>A nontrivial fraction of call setup time in WebRTC is=
 the DTLS handshake.</div><div>This document describes how to piggyback the=
 first few handshake messages</div><div>in the SDP offer/answer exchange, t=
hus reducing latency.</div><div><br></div><div>Comments welcome.</div><div>=
<br></div></div></blockquote><div><br></div><div>One other issue I thought =
off was that with DTLS SDP either offerer or answerer can start a new DTLS =
association. When answerer starts a new session it will have exactly the sa=
me problems as forking, since this will create multiple DTLS sessions with =
the same client random. This can be solved with some sort of fallback mecha=
nism to either regular DTLS setup or to sending a ClientHello in the answer=
.</div><div><br></div><div>Regards,</div><div><div class=3D"gmail_signature=
">_____________<br>Roman Shpount</div></div><div>=C2=A0</div></div></div></=
div>

--001a1140a2b621b1a00530c39d18--


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----------------------------------------------------------------------
COMMENT:
----------------------------------------------------------------------

Shucheng LIU's OPS DIR review:

**** Editorial ****

 

* Section 2, page 3:

> 

>    o  Legacy networks: In this document, legacy networks encompass the

>       conversational networks that are already deployed like the PSTN,

>       the PLMN, the IP/IMS networks offering VoIP services, including

>       3GPP "4G" Evolved Packet System[TS23.002]

 

Missing space in "Evolved Packet System[TS23.002]"

 

 

* Section 2, page 3:

>  o  PSTN:Public Switched Telephone Network

 

Missing space.

 

 

* Section 3, page 4:

>  Consequently,

>    a significant number of calls are likely to occur between terminals

>    supporting WebRTC endpoints and other terminals like mobile
handsets,

>    fixed VoIP terminals, DECT terminals that do not support WebRTC

>    endpoints nor implement OPUS. 

 

Seems should  s/terminals, DECT terminals/terminals, and DECT terminals/

 

 

* Section 3: each of the bullets is separated by two blank lines rather
than a single one.

 

 

* Section 4.1.1, page 5:

> especially

 

s/especially/specially/

 

 

* Section 4.1.3, page 5:

>    The payload format to be used for AMR-WB is described in [RFC4867]

>    with bandwidth efficient format and one speech frame encapsulated
in

>    each RTP packets

 

s/packets/packet/

 

 

* Section 4.2.1, page 6:

>  This include both mobile phone calls using GSM and 3G

 

s/include/includes/

 

 

* Section 4.2.1, page 6:

> such as, GSMA voice IMS profile for VoLTE in [IR.92].

 

Please remove the comma.

 

 

* Section 4.2.1, page 6:

>    degrading the high efficiency over mobile radio access.References

> for

 

Missing space.

 

 

* Section 4.2.3, page 7:

>    The payload format to be used for AMR is described in [RFC4867]
with

>    bandwidth efficient format and one speech frame encapsulated in
each

>    RTP packets.

 

s/packets/packet/



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----------------------------------------------------------------------
COMMENT:
----------------------------------------------------------------------


section 3: MOS could do with a reference



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Hello Magnus,

Sorry for the late answer

In order to ensure proper interworking it is worth having a real normative =
"MUST" here.

> OLD:
>      For AMR/AMR-WB, support for redundant encoding, and the maximum
>      supported depth, are controlled by the 'max-red' parameter, as
>      specified in [RFC4867], Section 8.1.
>
> NEW:
>      For AMR/AMR-WB, support for redundant encoding, and the maximum
>      used depth, is indicated by the 'max-red' parameter, as
>      specified in [RFC4867], Section 8.1 and 8.2. The 3GPP Multimedia
>      Telephony services specifies a maximum value of 220 ms for
>      'max-red', see [3GPP.26.114]. If AMR or AMR-WB are offered for inter=
operability purpose,
>     see [draft-ietf-rtcweb-audio-codecs-for-interop], the max-red paramet=
er MUST be included
>    in the offer and answer and set to an integer value multiple of 20ms n=
ot exceeding 220ms.

St=E9phane

-----Message d'origine-----
De=A0: Magnus Westerlund [mailto:magnus.westerlund@ericsson.com]=20
Envoy=E9=A0: jeudi 7 avril 2016 16:30
=C0=A0: PROUST St=E9phane IMT/OLPS; rtcweb@ietf.org; Justin Uberti
Objet=A0: Re: [rtcweb] AMR and max-red in draft-ietf-rtcweb-fec

Hi,

I am generally fine with Stephane's proposal. I do note that the "must"=20
in the last sentence probably need to be a "MUST", otherwise if the intenti=
on is just to point to requirements that exists, then please reformulate th=
is to not use RFC2119 terms, even in lower cases.

Cheers

Magnus

Den 2016-04-06 kl. 06:28, skrev stephane.proust@orange.com:
> Sorry, the last part of the sentence in the proposal was missing  :  the =
max-red parameter must be included
>    in the offer and answer and set to an integer value multiple of 20ms n=
ot exceeding 220ms.
>
> OLD:
>      For AMR/AMR-WB, support for redundant encoding, and the maximum
>      supported depth, are controlled by the 'max-red' parameter, as
>      specified in [RFC4867], Section 8.1.
>
> NEW:
>      For AMR/AMR-WB, support for redundant encoding, and the maximum
>      used depth, is indicated by the 'max-red' parameter, as
>      specified in [RFC4867], Section 8.1 and 8.2. The 3GPP Multimedia
>      Telephony services specifies a maximum value of 220 ms for
>      'max-red', see [3GPP.26.114]. If AMR or AMR-WB are offered for inter=
operability purpose,
>     see [draft-ietf-rtcweb-audio-codecs-for-interop], the max-red paramet=
er must be included
>    in the offer and answer and set to an integer value multiple of 20ms n=
ot exceeding 220ms.
>
> -----Message d'origine-----
> De : rtcweb [mailto:rtcweb-bounces@ietf.org] De la part de=20
> stephane.proust@orange.com Envoy=E9 : mercredi 6 avril 2016 10:18 =C0 :=
=20
> Magnus Westerlund; rtcweb@ietf.org; Justin Uberti Objet : Re: [rtcweb]=20
> AMR and max-red in draft-ietf-rtcweb-fec
>
> Hello Magnus
>
> I support your proposal. I would even suggest to relate this to the draft=
-ietf-rtcweb-audio-codecs-for-interop (where more information about use of =
AMR and AMR-WB is provided ) and specify more explicitly to include it in t=
he offer/answer with an integer value multiple of 20ms not exceeding 220ms.
> However, we can stay with your initial proposal if you or anyone else thi=
nks it is preferable.
>
> Kind regards
>
> St=E9phane
>
> OLD:
>      For AMR/AMR-WB, support for redundant encoding, and the maximum
>      supported depth, are controlled by the 'max-red' parameter, as
>      specified in [RFC4867], Section 8.1.
>
> NEW:
>      For AMR/AMR-WB, support for redundant encoding, and the maximum
>      used depth, is indicated by the 'max-red' parameter, as
>      specified in [RFC4867], Section 8.1 and 8.2. The 3GPP Multimedia
>      Telephony services specifies a maximum value of 220 ms for
>      'max-red', see [3GPP.26.114]. If AMR or AMR-WB are offered for inter=
operability purpose,
>     see [draft-ietf-rtcweb-audio-codecs-for-interop], the max-red paramet=
er must be included
>    in the offer and answer and set to an integer value multiple of 20ms.
>
> -----Message d'origine-----
> De : rtcweb [mailto:rtcweb-bounces@ietf.org] De la part de Magnus=20
> Westerlund Envoy=E9 : mardi 5 avril 2016 23:02 =C0 : rtcweb@ietf.org;=20
> Justin Uberti Objet : [rtcweb] AMR and max-red in=20
> draft-ietf-rtcweb-fec
>
> Hi,
>
> As raised in the WG session there are where an open issue if there need t=
o be any specification of the AMR/AMR-WB max-red parameter.
>
> So RFC4867 do have the following on offer/answer usage of the parameter:
>
>         -  The parameter "max-red" is a stream property parameter.  For
>            send-only or send-recv unicast media streams, the parameter
>            declares the limitation on redundancy that the stream sender
>            will use.  For recvonly streams, it indicates the desired value
>            for the stream sent to the receiver.  The answerer MAY change
>            the value, but is RECOMMENDED to use the same limitation as the
>            offer declares.  In the case of multicast, the offerer MAY
>            declare a limitation; this SHALL be answered using the same
>            value.  A media sender using this payload format is RECOMMENDED
>            to always include the "max-red" parameter.  This information is
>            likely to simplify the media stream handling in the receiver.
>            This is especially true if no redundancy will be used, in which
>            case "max-red" is set to 0.  As this parameter was not defined
>            originally, some senders will not declare this parameter even
>            if it will limit or not send redundancy at all.
>
> Most notable is that media senders are RECOMMENDED to include it. Also no=
te that it is the sender that indicates the limitation it intends to adhere=
 to for the streams using this RTP payload type.
>
> I also like to note that 3GPP in its specification for its Multimedia=20
> Telephony, has specified that one shall not use a value larger than=20
> 220 ms, and the value should be a multiple of 20 ms.  Please see 3GPP=20
> TS
> 26.114 version 13.3.0
> http://www.3gpp.org/ftp/Specs/archive/26_series/26.114/26114-d30.zip
>
> So from my perspective I think a WebRTC endpoint simply should follow the=
 AMR RTP Payload specification when supported and include the parameter to =
indicate the actual upper range of the redundancy it intended to use. It is=
 good to know about the limitation that 3GPP has defined as one likely reas=
on to support AMR/AMR-WB is to be able to interoperate with 3GPP Services.
>
> Thus I would suggest that a small addition to the Section 4.3 text and co=
rrections in the first sentence:
>
> OLD:
>      For AMR/AMR-WB, support for redundant encoding, and the maximum
>      supported depth, are controlled by the 'max-red' parameter, as
>      specified in [RFC4867], Section 8.1.
>
> NEW:
>      For AMR/AMR-WB, support for redundant encoding, and the maximum
>      used depth, is indicated by the 'max-red' parameter, as
>      specified in [RFC4867], Section 8.1 and 8.2. The 3GPP Multimedia
>      Telephony services specifies a maximum value of 220 ms for
>      'max-red', see [3GPP.26.114].
>
>
> New Informational Reference:
>
> [3GPP.26.114]
>
> <reference anchor=3D"3GPP.26.114"><front><title>IP Multimedia Subsystem=
=20
> (IMS); Multimedia telephony; Media handling and=20
> interaction</title><author><organization>3GPP</organization></author><
> date day=3D"17" month=3D"March" year=3D"2016"/></front><seriesInfo=20
> name=3D"3GPP TS"
> value=3D"26.114 13.3.0"/><format type=3D"HTML"
> target=3D"http://www.3gpp.org/ftp/Specs/html-info/26114.htm"/></referenc
> e>
>
> Opinions?
>
> Cheers
>
> Magnus Westerlund
>
> ----------------------------------------------------------------------
> 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
> ----------------------------------------------------------------------
>
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb
>
> ______________________________________________________________________
> ___________________________________________________
>
> Ce message et ses pieces jointes peuvent contenir des informations confid=
entielles ou privilegiees et ne doivent donc pas etre diffuses, exploites o=
u copies sans autorisation. Si vous avez recu ce message par erreur, veuill=
ez le signaler a l'expediteur et le detruire ainsi que les pieces jointes. =
Les messages electroniques etant susceptibles d'alteration, Orange decline =
toute responsabilite si ce message a ete altere, deforme ou falsifie. Merci.
>
> This message and its attachments may contain confidential or privileged i=
nformation that may be protected by law; they should not be distributed, us=
ed or copied without authorisation.
> If you have received this email in error, please notify the sender and de=
lete this message and its attachments.
> As emails may be altered, Orange is not liable for messages that have bee=
n modified, changed or falsified.
> Thank you.
>
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb
>
> ______________________________________________________________________
> ___________________________________________________
>
> Ce message et ses pieces jointes peuvent contenir des informations=20
> confidentielles ou privilegiees et ne doivent donc pas etre diffuses,=20
> exploites ou copies sans autorisation. Si vous avez recu ce message=20
> par erreur, veuillez le signaler a l'expediteur et le detruire ainsi que =
les pieces jointes. Les messages electroniques etant susceptibles d'alterat=
ion, Orange decline toute responsabilite si ce message a ete altere, deform=
e ou falsifie. Merci.
>
> This message and its attachments may contain confidential or=20
> privileged information that may be protected by law; they should not be d=
istributed, used or copied without authorisation.
> If you have received this email in error, please notify the sender and de=
lete this message and its attachments.
> As emails may be altered, Orange is not liable for messages that have bee=
n modified, changed or falsified.
> Thank you.
>
>


--=20

Magnus Westerlund

----------------------------------------------------------------------
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
----------------------------------------------------------------------


___________________________________________________________________________=
______________________________________________

Ce message et ses pieces jointes peuvent contenir des informations confiden=
tielles ou privilegiees et ne doivent donc
pas etre diffuses, exploites ou copies sans autorisation. Si vous avez recu=
 ce message par erreur, veuillez le signaler
a l'expediteur et le detruire ainsi que les pieces jointes. Les messages el=
ectroniques etant susceptibles d'alteration,
Orange decline toute responsabilite si ce message a ete altere, deforme ou =
falsifie. Merci.

This message and its attachments may contain confidential or privileged inf=
ormation that may be protected by law;
they should not be distributed, used or copied without authorisation.
If you have received this email in error, please notify the sender and dele=
te this message and its attachments.
As emails may be altered, Orange is not liable for messages that have been =
modified, changed or falsified.
Thank you.


From nobody Wed Apr 20 09:03:24 2016
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From: Ted Hardie <ted.ietf@gmail.com>
Date: Wed, 20 Apr 2016 09:03:02 -0700
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--001a113d7a8ca60be80530ecbbfb
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Having seen confirmation of this approach and no new issues, we will move
forward with this language.
Thanks to everyone working on the issue,

Ted, Cullen, Sean

On Wed, Apr 13, 2016 at 1:31 PM, Ted Hardie <ted.ietf@gmail.com> wrote:

> With Martin's friendly amendments, the room at IETF 95 agreed to the
> follow as the resolution of the DTMF discussion.  The chairs would like
> anyone who has new issues with this to let us know as soon as possible and
> no later than Wednesday, April 20th, noon UTC,
>
>
> "DTMF events generated by a WebRTC endpoint MUST have a duration of no
> more than 8000 ms and no less than 40 ms.  The recommended default duration
> is 100 ms for each tone. The gap between events MUST be no less than 30 ms;
>  the recommended default gap duration is 70 ms.
>
> WebRTC endpoints are not required to do anything with RFC 4733 tones sent
> to them, except gracefully drop them. There is currently no API to inform
> JavaScript about the received DTMF or other RFC 4733 tones."
>
> regards,
>
> Ted, Cullen, and Sean
>

--001a113d7a8ca60be80530ecbbfb
Content-Type: text/html; charset=UTF-8
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<div dir=3D"ltr"><div><div>Having seen confirmation of this approach and no=
 new issues, we will move forward with this language.<br>Thanks to everyone=
 working on the issue,<br></div><br></div>Ted, Cullen, Sean<br><div><div><d=
iv><div class=3D"gmail_extra"><br><div class=3D"gmail_quote">On Wed, Apr 13=
, 2016 at 1:31 PM, Ted Hardie <span dir=3D"ltr">&lt;<a href=3D"mailto:ted.i=
etf@gmail.com" target=3D"_blank">ted.ietf@gmail.com</a>&gt;</span> wrote:<b=
r><blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:=
1px #ccc solid;padding-left:1ex"><div dir=3D"ltr"><div><div>With Martin&#39=
;s friendly amendments, the room at IETF 95 agreed to the follow as the res=
olution of the DTMF discussion.=C2=A0 The chairs would like anyone who has =
new issues with this to let us know as soon as possible and no later than W=
ednesday, April 20th, noon UTC, <br><br><br>&quot;<span style=3D"font-famil=
y:arial,helvetica,sans-serif"><span style=3D"font-size:10.6667px;color:rgb(=
0,0,0);background-color:transparent;font-weight:400;font-style:normal;font-=
variant:normal;text-decoration:none;vertical-align:baseline">DTMF events ge=
nerated by a WebRTC endpoint MUST have a duration of no more than 8000 ms a=
nd no less than 40 ms.=C2=A0 The recommended default duration is 100 ms for=
 each tone. The gap between events MUST be no less than 30 ms; =C2=A0the re=
commended default gap duration is 70 ms.</span><br><br><span style=3D"font-=
size:10.6667px;color:rgb(0,0,0);background-color:transparent;font-weight:40=
0;font-style:normal;font-variant:normal;text-decoration:none;vertical-align=
:baseline">WebRTC endpoints are not required to do anything with RFC 4733 t=
ones sent to them, except gracefully drop them. There is currently no API t=
o inform JavaScript about the received DTMF or other RFC 4733 tones.&quot;<=
br><br></span></span></div><span style=3D"font-family:arial,helvetica,sans-=
serif"><span style=3D"font-size:10.6667px;color:rgb(0,0,0);background-color=
:transparent;font-weight:400;font-style:normal;font-variant:normal;text-dec=
oration:none;vertical-align:baseline">regards,<br><br></span></span></div><=
span style=3D"font-family:arial,helvetica,sans-serif"><span style=3D"font-s=
ize:10.6667px;color:rgb(0,0,0);background-color:transparent;font-weight:400=
;font-style:normal;font-variant:normal;text-decoration:none;vertical-align:=
baseline">Ted, Cullen, and Sean<br></span></span></div>
</blockquote></div><br></div></div></div></div></div>

--001a113d7a8ca60be80530ecbbfb--


From nobody Wed Apr 20 21:40:24 2016
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Subject: [rtcweb] Spencer Dawkins' Yes on draft-ietf-rtcweb-audio-codecs-for-interop-05: (with COMMENT)
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Spencer Dawkins has entered the following ballot position for
draft-ietf-rtcweb-audio-codecs-for-interop-05: Yes

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----------------------------------------------------------------------
COMMENT:
----------------------------------------------------------------------

I remember that the "but what about THIS audio codec?" discussions were
pretty contentious for a while, and wanted to say that this document does
a really good job of handling that question. Thanks for producing it.



From nobody Thu Apr 21 08:31:30 2016
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A New Internet-Draft is available from the on-line Internet-Drafts directories.
This draft is a work item of the Real-Time Communication in WEB-browsers of the IETF.

        Title           : WebRTC Audio Codec and Processing Requirements
        Authors         : Jean-Marc Valin
                          Cary Bran
	Filename        : draft-ietf-rtcweb-audio-11.txt
	Pages           : 7
	Date            : 2016-04-21

Abstract:
   This document outlines the audio codec and processing requirements
   for WebRTC endpoints.


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

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

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


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From nobody Fri Apr 22 03:26:06 2016
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A New Internet-Draft is available from the on-line Internet-Drafts directories.
This draft is a work item of the Real-Time Communication in WEB-browsers of the IETF.

        Title           : Additional WebRTC audio codecs for interoperability.
        Author          : Stephane Proust
	Filename        : draft-ietf-rtcweb-audio-codecs-for-interop-06.txt
	Pages           : 12
	Date            : 2016-04-22

Abstract:
   To ensure a baseline level of interoperability between WebRTC
   endpoints, a minimum set of required codecs is specified.  However,
   to maximize the possibility to establish the session without the need
   for audio transcoding, it is also recommended to include in the offer
   other suitable audio codecs that are available to the browser.

   This document provides some guidelines on the suitable codecs to be
   considered for WebRTC endpoints to address the most relevant
   interoperability use cases.


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From nobody Mon Apr 25 02:23:56 2016
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which bandwidth modifiers for b= lines does a client need to support?

Chrome currently supports b=AS, however the implementation is closer to 
TIAS semantically. Firefox does not implement either currently.

http://w3c.github.io/webrtc-pc/#widl-RTCRtpEncodingParameters-maxBitrate
uses TIAS as a definition.

For parsing, supporting both is fine (for a while at least).
For serialization I would prefer picking one (TIAS) to having to track 
which variant a peer supports.


From nobody Mon Apr 25 10:05:21 2016
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Subject: [rtcweb] Protocol Action: 'WebRTC Audio Codec and Processing Requirements' to Proposed Standard (draft-ietf-rtcweb-audio-11.txt)
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The IESG has approved the following document:
- 'WebRTC Audio Codec and Processing Requirements'
  (draft-ietf-rtcweb-audio-11.txt) as Proposed Standard

This document is the product of the Real-Time Communication in
WEB-browsers Working Group.

The IESG contact persons are Alexey Melnikov, Ben Campbell and Alissa
Cooper.

A URL of this Internet Draft is:
https://datatracker.ietf.org/doc/draft-ietf-rtcweb-audio/





Technical Summary

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

Working Group Summary

 Some members of the WG wished to add additional MTI codecs. Rational for those are provided in draft-ietf-rtcweb-audio-codecs-for-interop which is referenced by this spec. 

Document Quality

  There are several implementations including FireFox and Chrome. 


Personnel

  Who is the Document Shepherd?
Cullen Jennings

 Who is the Responsible Area Director?
Alissa Cooper


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The IESG has approved the following document:
- 'Additional WebRTC audio codecs for interoperability.'
  (draft-ietf-rtcweb-audio-codecs-for-interop-06.txt) as Informational
RFC

This document is the product of the Real-Time Communication in
WEB-browsers Working Group.

The IESG contact persons are Alexey Melnikov, Ben Campbell and Alissa
Cooper.

A URL of this Internet Draft is:
https://datatracker.ietf.org/doc/draft-ietf-rtcweb-audio-codecs-for-interop/





Technical Summary

  The document explains some additional audio codecs that WebRTC devices and browsers may wish to support and some of reasons that an implementation might choose to use that codec.  It does not specify any mandatory to implement codecs for WebRTC. 

Working Group Summary

  Some people in the WG would like these codecs to be mandatory to implement for WebRTC browsers. 

Document Quality

  There are WebRTC devices (but not browsers) that implement theses codecs.


Personnel

  Who is the Document Shepherd? Who is the Responsible Area
  Director?

  Shepherd is Cullen Jennings. AD is Alissa Cooper. 


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Am 20.03.2016 um 23:31 schrieb internet-drafts@ietf.org:
>
> A New Internet-Draft is available from the on-line Internet-Drafts directories.
> This draft is a work item of the Real-Time Communication in WEB-browsers of the IETF.
>
>          Title           : WebRTC IP Address Handling Recommendations
>          Authors         : Justin Uberti
>                            Guo-wei Shieh
> 	Filename        : draft-ietf-rtcweb-ip-handling-01.txt
> 	Pages           : 8
> 	Date            : 2016-03-20

I stumbled upon a (now deleted) stackoverflow posting yesterday in which 
obfuscated code "used webrtc" to hack the users router.
Basically webrtc was used to get the local ip address from which the 
routers ip is inferred. Then a default admin account is tried.

It turns out the idea is not knew, I heard from some former coworkers 
that they used this in penetration testing before. But it has now 
reached stackoverflow which means there will soon be rumors and all 
kinds of "webrtc allows hacking your router" stories.

The principal problem here is default passwords. WebRTC makes it 
slightly easier by removing the need to run a series of http requests to 
find out the routers ip. Should this be mentioned as an example in #1 of 
the problem statement?


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Alexey Melnikov has entered the following ballot position for
draft-ietf-rtcweb-alpn-03: No Objection

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introductory paragraph, however.)


Please refer to https://www.ietf.org/iesg/statement/discuss-criteria.html
for more information about IESG DISCUSS and COMMENT positions.


The document, along with other ballot positions, can be found here:
https://datatracker.ietf.org/doc/draft-ietf-rtcweb-alpn/



----------------------------------------------------------------------
COMMENT:
----------------------------------------------------------------------

Please excuse my ignorance (pointers would be appreciated, if this is
explained elsewhere): do RTP intermediary need to be updated to
understand this spec?
If yes, how can you enforce requirements on "c-webrtc"?



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On 29 April 2016 at 02:01, Alexey Melnikov <aamelnikov@fastmail.fm> wrote:
> Please excuse my ignorance (pointers would be appreciated, if this is
> explained elsewhere): do RTP intermediary need to be updated to
> understand this spec?

No.  Mostly.  RTP intermediaries can choose to implement this spec as
an endpoint, but there is typically no benefit from doing so.  RTP
intermediaries don't usually have a strong distinction between "the
thing that does (S)RTP" and "the thing that controls signaling" and so
don't gain much by isolating media from the latter.


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On Fri, Apr 29, 2016, at 12:51 AM, Martin Thomson wrote:
> On 29 April 2016 at 02:01, Alexey Melnikov <aamelnikov@fastmail.fm>
> wrote:
> > Please excuse my ignorance (pointers would be appreciated, if this is
> > explained elsewhere): do RTP intermediary need to be updated to
> > understand this spec?
> 
> No.  Mostly.  RTP intermediaries can choose to implement this spec as
> an endpoint, but there is typically no benefit from doing so.  RTP
> intermediaries don't usually have a strong distinction between "the
> thing that does (S)RTP" and "the thing that controls signaling" and so
> don't gain much by isolating media from the latter.

Section 3 says:

   RTP middleboxes and entities that forward media or data cannot
   promise to maintain confidentiality.  Any entity that forwards
   content, or records content for later access by entities other than
   the authenticated peer, MUST NOT offer or accept a session with the
   "c-webrtc" identifier.

So, my questions are:

1) Do RTP middleboxes frequently forward content, or records content for
later access by entities other than
   the authenticated peer?
2) If the answer to 1) is "yes", why would RTP middlebox software
developers bother to update software to comply with this specification?

Please help me understand, if I misunderstood the text.


From nobody Fri Apr 29 03:06:08 2016
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Date: Fri, 29 Apr 2016 20:06:03 +1000
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From: Martin Thomson <martin.thomson@gmail.com>
To: Alexey Melnikov <aamelnikov@fastmail.fm>
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Cc: draft-ietf-rtcweb-alpn@ietf.org, Sean Turner <turners@ieca.com>, "rtcweb@ietf.org" <rtcweb@ietf.org>, The IESG <iesg@ietf.org>, rtcweb-chairs@ietf.org
Subject: Re: [rtcweb] Alexey Melnikov's No Objection on draft-ietf-rtcweb-alpn-03: (with COMMENT)
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On 29 April 2016 at 19:30, Alexey Melnikov <aamelnikov@fastmail.fm> wrote:
> 1) Do RTP middleboxes frequently forward content, or records content for
> later access by entities other than
>    the authenticated peer?
> 2) If the answer to 1) is "yes", why would RTP middlebox software
> developers bother to update software to comply with this specification?

(I should really read my own doc before answering.)

Yes, middleboxes do this all the time.  However, they don't need to
update their software.  Endpoints that implement this spec will assume
that the absence of ALPN means that there is no confidentiality.  And
I realize now that I never actually said that.  That was extremely
remiss of me...

https://github.com/martinthomson/drafts/commit/21b84074a671bd


From nobody Fri Apr 29 07:57:34 2016
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From: "Cullen Jennings (fluffy)" <fluffy@cisco.com>
To: Harald Tveit Alvestrand <harald@alvestrand.no>
Thread-Topic: [rtcweb] Clarification in draft-ietf-rtcweb-transports-11
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Date: Fri, 29 Apr 2016 14:57:10 +0000
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Subject: Re: [rtcweb] Clarification in draft-ietf-rtcweb-transports-11
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Github says this was fixed in version -10 - See https://github.com/rtcweb-w=
g/rtcweb-transport/issues/12

But it does not seem to be in the -10 draft or latest draft so I'm wonderin=
g if it got lost along the way somewhere.=20



> On Feb 8, 2016, at 12:46 AM, Harald Alvestrand <harald@alvestrand.no> wro=
te:
>=20
> This seems reasonable.
>=20
> Filed https://github.com/rtcweb-wg/rtcweb-transport/issues/12 so as to no=
t forget.
>=20
>>=20
>>=20
>> On Wed, Feb 3, 2016 at 7:16 AM, Cullen Jennings <fluffy@iii.ca> wrote:
>> I think we need to make the ALPN a bit more explicit. We agreed to inclu=
de the ALPN header so that a proxy knows it might receive video packet rate=
s tunnelled across it. However the text on this is not 100% clear. Right no=
w it says
>>=20
>>    If it does so, it MUST support the "ALPN" header as
>>    specified in [RFC7639],
>>=20
>> But some people have posted out including this header is optional in 763=
9 so there might be some ubiquity here about what is meant. I think we shou=
ld change the word =93support=94 to =93include=94 to make it clear this nee=
ds to be sent to the proxy so that it would read
>>=20
>>     If it does so, it MUST include the "ALPN" header as
>>    specified in [RFC7639],
>>=20
>> I believe that correctly reflects what we intended on this. There is no =
requirement for the proxy to understand or do anything with this header. Ol=
d proxies will just ignore it and cary on as if it was not there.
>>=20
>> I wouldn't have a problem with this if it were restricted to browsers. H=
owever,
>> we don't want to careful about retroactively branding endpoints which ar=
en't
>> browsers but can talk to WebRTC clients as noncompliant. Maybe this
>> is like codecs where they are "WebRTC Compatible"?
>>=20
>> -Ekr
>>=20
>> Cullen
>>=20
>> (with my individual contributor hat on)
>>=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
>>=20
>>=20
>>=20
>> _______________________________________________
>> rtcweb mailing list
>>=20
>> rtcweb@ietf.org
>> https://www.ietf.org/mailman/listinfo/rtcweb
>=20
>=20
> --=20
> Surveillance is pervasive. Go Dark.
>=20
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb


From nobody Fri Apr 29 08:17:16 2016
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To: "Cullen Jennings (fluffy)" <fluffy@cisco.com>
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From: Harald Alvestrand <harald@alvestrand.no>
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Subject: Re: [rtcweb] Clarification in draft-ietf-rtcweb-transports-11
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Den 29. april 2016 16:57, skrev Cullen Jennings (fluffy):
> 
> Github says this was fixed in version -10 - See https://github.com/rtcweb-wg/rtcweb-transport/issues/12
> 
> But it does not seem to be in the -10 draft or latest draft so I'm wondering if it got lost along the way somewhere. 

You're right. I reopened the issue.

> 
> 
> 
>> On Feb 8, 2016, at 12:46 AM, Harald Alvestrand <harald@alvestrand.no> wrote:
>>
>> This seems reasonable.
>>
>> Filed https://github.com/rtcweb-wg/rtcweb-transport/issues/12 so as to not forget.
>>
>>>
>>>
>>> On Wed, Feb 3, 2016 at 7:16 AM, Cullen Jennings <fluffy@iii.ca> wrote:
>>> I think we need to make the ALPN a bit more explicit. We agreed to include the ALPN header so that a proxy knows it might receive video packet rates tunnelled across it. However the text on this is not 100% clear. Right now it says
>>>
>>>    If it does so, it MUST support the "ALPN" header as
>>>    specified in [RFC7639],
>>>
>>> But some people have posted out including this header is optional in 7639 so there might be some ubiquity here about what is meant. I think we should change the word “support” to “include” to make it clear this needs to be sent to the proxy so that it would read
>>>
>>>     If it does so, it MUST include the "ALPN" header as
>>>    specified in [RFC7639],
>>>
>>> I believe that correctly reflects what we intended on this. There is no requirement for the proxy to understand or do anything with this header. Old proxies will just ignore it and cary on as if it was not there.
>>>
>>> I wouldn't have a problem with this if it were restricted to browsers. However,
>>> we don't want to careful about retroactively branding endpoints which aren't
>>> browsers but can talk to WebRTC clients as noncompliant. Maybe this
>>> is like codecs where they are "WebRTC Compatible"?
>>>
>>> -Ekr
>>>
>>> Cullen
>>>
>>> (with my individual contributor hat on)
>>>
>>>
>>> _______________________________________________
>>> 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.
>>
>> _______________________________________________
>> rtcweb mailing list
>> rtcweb@ietf.org
>> https://www.ietf.org/mailman/listinfo/rtcweb
> 

