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From: Alissa Cooper <alissa@cooperw.in>
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Subject: [rtcweb] AD evaluation: draft-ietf-rtcweb-transports-14
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I have reviewed this document in preparation for IETF last call. It is =
in good shape and I have requested the last call.

While the last call is ongoing, I would like to double-check that this =
document (particularly Section 4) raises no new security considerations =
that should be documented in Section 6. I'm wondering if there are =
scenarios where a malicious application could try to game the priority =
scheme to some ill effect? draft-ietf-tsvwg-rtcweb-qos talks a bit about =
this, so I'm wondering if anything needs to be said here.

And I found a couple of nits to be resolved together with any LC =
comments:

- Sec 3.2 - Please update the citation to point to =
draft-ietf-mmusic-ice-dualstack-fairness-02.

- Sec 3.4 -

"Third, using TCP only between the endpoint and its relay may result
   in less issues with TCP in regards to real-time constraints, e.g. due
   to head of line blocking."
  =20
This is awkwardly phrased. I would suggest something like "Third, using =
TCP between the client's TURN server and its peer may result in more =
performance problems (due to head-of-line blocking) than using UDP."

Alissa=


From nobody Thu Jul  7 14:53:17 2016
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Subject: [rtcweb] Last Call: <draft-ietf-rtcweb-transports-14.txt> (Transports for 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:
- 'Transports for WebRTC'
  <draft-ietf-rtcweb-transports-14.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 2016-07-21. 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


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




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

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


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



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From: Alissa Cooper <alissa@cooperw.in>
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And one bit I left out =E2=80=94 if text is to be added to reference =
draft-ietf-rtcweb-return, that should be done together with addressing =
nits and LC comments.

> On Jul 7, 2016, at 2:33 PM, Alissa Cooper <alissa@cooperw.in> wrote:
>=20
> I have reviewed this document in preparation for IETF last call. It is =
in good shape and I have requested the last call.
>=20
> While the last call is ongoing, I would like to double-check that this =
document (particularly Section 4) raises no new security considerations =
that should be documented in Section 6. I'm wondering if there are =
scenarios where a malicious application could try to game the priority =
scheme to some ill effect? draft-ietf-tsvwg-rtcweb-qos talks a bit about =
this, so I'm wondering if anything needs to be said here.
>=20
> And I found a couple of nits to be resolved together with any LC =
comments:
>=20
> - Sec 3.2 - Please update the citation to point to =
draft-ietf-mmusic-ice-dualstack-fairness-02.
>=20
> - Sec 3.4 -
>=20
> "Third, using TCP only between the endpoint and its relay may result
>   in less issues with TCP in regards to real-time constraints, e.g. =
due
>   to head of line blocking."
>=20
> This is awkwardly phrased. I would suggest something like "Third, =
using TCP between the client's TURN server and its peer may result in =
more performance problems (due to head-of-line blocking) than using =
UDP."
>=20
> Alissa
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb


From nobody Thu Jul  7 19:10:23 2016
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Subject: [rtcweb] I-D Action: draft-ietf-rtcweb-sdp-02.txt
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A New Internet-Draft is available from the on-line Internet-Drafts directories.
This draft is a work item of the Real-Time Communication in WEB-browsers of the IETF.

        Title           : SDP for the WebRTC
        Authors         : Suhas Nandakumar
                          Cullen Jennings
	Filename        : draft-ietf-rtcweb-sdp-02.txt
	Pages           : 107
	Date            : 2016-07-07

Abstract:
   The Web Real-Time Communication [WebRTC] working group is charged to
   provide protocol support for direct interactive rich communication
   using audio, video and data between two peers' web browsers.  With in
   the WebRTC framework, Session Description protocol (SDP) [RFC4566] is
   used for negotiating session capabilities between the peers.  Such a
   negotiation happens based on the SDP Offer/Answer exchange mechanism
   described in [RFC3264].

   This document provides an informational reference in describing the
   role of SDP and the Offer/Answer exchange mechanism for the most
   common WebRTC use-cases.

   This SDP examples provided in this document is still a work in
   progress, but it aims to align closest to the evolving standards
   work.


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

There's also a htmlized version available at:
https://tools.ietf.org/html/draft-ietf-rtcweb-sdp-02

A diff from the previous version is available at:
https://www.ietf.org/rfcdiff?url2=draft-ietf-rtcweb-sdp-02


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

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


From nobody Thu Jul  7 19:26:22 2016
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From: Suhas Nandakumar <suhasietf@gmail.com>
Date: Fri, 8 Jul 2016 07:56:17 +0530
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Subject: [rtcweb] draft-ietf-rtcweb-sdp-02 submitted
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Hello All

   We have submitted the -02 version of RTCWEB SDP examples draft which
went through quite a bit of content re-shuffling and updates to match
(attempt) the latest versions of BUNDLE, JSEP and related drafts.

Many thanks to Paul for starting the review of the draft that triggered few
formatting changes, changes related to consistency of attribute values and
their ordering, to enhance the readability throughout the draft.

Please refer to the following HTML version while reviewing/reading the
draft:

https://tools.ietf.org/id/draft-ietf-rtcweb-sdp-02.html

[[ It might take a short while for the data-tracker to render the above
HTML version though ]]

Here is the high-level summary of changes
1. As far as possible, the examples reflect latest BUNDLE and JSEP
requirements.
2. SCTP examples are updated
3. Visual markers, separators are added in the SDP table
4. SSRC values, IP Address , Port numbers, CNAMEs are consistent to a great
extent now
5. Attributes ordering is consistent across all the examples now
6. RID, Simulcast examples are matched against their latest draft versions.

Please give it a read and let us know your comments/feedback.

Cheers
Suhas/Cullen

--94eb2c0b7ce8079b26053716872d
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<div dir=3D"ltr">Hello All<div><br></div><div>=C2=A0 =C2=A0We have submitte=
d the -02 version of RTCWEB SDP examples draft which went through quite a b=
it of content re-shuffling and updates to match (attempt) the latest versio=
ns of BUNDLE, JSEP and related drafts.=C2=A0</div><div><br></div><div>Many =
thanks to Paul for starting the review of the draft that triggered few form=
atting changes, changes related to consistency of attribute values and thei=
r ordering, to enhance the readability throughout the draft.</div><div><br>=
</div><div>Please refer to the following HTML version while reviewing/readi=
ng the draft:=C2=A0</div><div><br></div><div><a href=3D"https://tools.ietf.=
org/id/draft-ietf-rtcweb-sdp-02.html">https://tools.ietf.org/id/draft-ietf-=
rtcweb-sdp-02.html</a></div><div><br></div><div>[[ It might take a short wh=
ile for the data-tracker to render the above HTML version though ]]</div><d=
iv><br></div><div>Here is the high-level summary of changes</div><div>1. As=
 far as possible, the examples reflect latest BUNDLE and JSEP requirements.=
</div><div>2. SCTP examples are updated=C2=A0</div><div>3. Visual markers, =
separators are added in the SDP table</div><div>4. SSRC values, IP Address =
, Port numbers, CNAMEs are consistent to a great extent now</div><div>5. At=
tributes ordering is consistent across all the examples now</div><div>6. RI=
D, Simulcast examples are matched against their latest draft versions.</div=
><div><br></div><div>Please give it a read and let us know your comments/fe=
edback.</div><div><br></div><div>Cheers</div><div>Suhas/Cullen</div><div><b=
r></div></div>

--94eb2c0b7ce8079b26053716872d--


From nobody Thu Jul  7 19:48:05 2016
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Subject: [rtcweb] I-D Action: draft-ietf-rtcweb-jsep-15.txt
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A New Internet-Draft is available from the on-line Internet-Drafts directories.
This draft is a work item of the Real-Time Communication in WEB-browsers of the IETF.

        Title           : Javascript Session Establishment Protocol
        Authors         : Justin Uberti
                          Cullen Jennings
                          Eric Rescorla
	Filename        : draft-ietf-rtcweb-jsep-15.txt
	Pages           : 88
	Date            : 2016-07-07

Abstract:
   This document describes the mechanisms for allowing a Javascript
   application to control the signaling plane of a multimedia session
   via the interface specified in the W3C RTCPeerConnection API, and
   discusses how this relates to existing signaling protocols.


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

There's also a htmlized version available at:
https://tools.ietf.org/html/draft-ietf-rtcweb-jsep-15

A diff from the previous version is available at:
https://www.ietf.org/rfcdiff?url2=draft-ietf-rtcweb-jsep-15


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

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


From nobody Sat Jul  9 14:27:48 2016
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To: Alissa Cooper <alissa@cooperw.in>, rtcweb@ietf.org
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From: Harald Alvestrand <harald@alvestrand.no>
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Subject: Re: [rtcweb] AD evaluation: draft-ietf-rtcweb-transports-14
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Thank you - I have filed this review as
https://github.com/rtcweb-wg/rtcweb-transport/issues/21 and will address
it after the Last Call ends.


Den 07. juli 2016 23:33, skrev Alissa Cooper:
> I have reviewed this document in preparation for IETF last call. It is in good shape and I have requested the last call.
> 
> While the last call is ongoing, I would like to double-check that this document (particularly Section 4) raises no new security considerations that should be documented in Section 6. I'm wondering if there are scenarios where a malicious application could try to game the priority scheme to some ill effect? draft-ietf-tsvwg-rtcweb-qos talks a bit about this, so I'm wondering if anything needs to be said here.
> 
> And I found a couple of nits to be resolved together with any LC comments:
> 
> - Sec 3.2 - Please update the citation to point to draft-ietf-mmusic-ice-dualstack-fairness-02.
> 
> - Sec 3.4 -
> 
> "Third, using TCP only between the endpoint and its relay may result
>    in less issues with TCP in regards to real-time constraints, e.g. due
>    to head of line blocking."
>    
> This is awkwardly phrased. I would suggest something like "Third, using TCP between the client's TURN server and its peer may result in more performance problems (due to head-of-line blocking) than using UDP."
> 
> Alissa
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb
> 


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Quick questions for folks =E2=80=A6  is an idea that we should explore a =
bit deeper before making decisions about it? or should we not proceed? =
or should we proceed? Just looking for feedback on where this should go =
next=20

Thanks, Cullen


> On Apr 18, 2016, at 7:59 AM, Roman Shpount <roman@telurix.com =
<mailto:roman@telurix.com>> wrote:
>=20
> On Mon, Apr 4, 2016 at 9:10 AM, Eric Rescorla <ekr@rtfm.com =
<mailto:ekr@rtfm.com>> wrote:
> 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
>=20
> 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.
>=20
> Regards,
> _____________
> Roman Shpount
> =20
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org <mailto:rtcweb@ietf.org>
> https://www.ietf.org/mailman/listinfo/rtcweb


--Apple-Mail=_4DF3B869-1E94-4A20-8850-E85B182EF845
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<html><head><meta http-equiv=3D"Content-Type" content=3D"text/html =
charset=3Dutf-8"><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""><div class=3D""><br class=3D""></div>Quick questions for =
folks =E2=80=A6 &nbsp;is an idea that we should explore a bit deeper =
before making decisions about it? or should we not proceed? or should we =
proceed? Just looking for feedback on where this should go =
next&nbsp;<div class=3D""><br class=3D""></div><div class=3D"">Thanks, =
Cullen</div><div class=3D""><br class=3D""><div class=3D""><br =
class=3D""><div class=3D""><blockquote type=3D"cite" class=3D""><div =
class=3D"">On Apr 18, 2016, at 7:59 AM, 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"gmail_extra"><div class=3D""><div =
class=3D"gmail_signature">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""></div></div><div class=3D"gmail_quote"><blockquote =
class=3D"gmail_quote" style=3D"margin:0px 0px 0px =
0.8ex;border-left-width:1px;border-left-style:solid;border-left-color:rgb(=
204,204,204);padding-left:1ex"><div dir=3D"ltr" class=3D""><div =
class=3D"">I wanted to call your attention to a draft I just published =
with a possibly stupid<br class=3D""></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></blockquote><div class=3D""><br =
class=3D""></div><div class=3D"">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.</div><div class=3D""><br class=3D""></div><div =
class=3D"">Regards,</div><div class=3D""><div =
class=3D"gmail_signature">_____________<br class=3D"">Roman =
Shpount</div></div><div class=3D"">&nbsp;</div></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""><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></div></body></html>=

--Apple-Mail=_4DF3B869-1E94-4A20-8850-E85B182EF845--


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From: "Black, David" <david.black@emc.com>
To: Michael Welzl <michawe@ifi.uio.no>, Brian E Carpenter <brian.e.carpenter@gmail.com>
Thread-Topic: [tsvwg] Fall-back to DSCP 0 in draft-ietf-tsvwg-rtcweb-qos ?
Thread-Index: AQHRzhJbsZyuYvv6mE+PQ2kNBpP9KZ/5U/cAgAAHhwCAGfiowA==
Date: Mon, 11 Jul 2016 13:55:53 +0000
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References: <CB087237-108A-44B1-8293-3498F24A2303@ifi.uio.no> <e3ebbf6c-58ab-1f70-3a5c-36a660dc73e7@gmail.com> <4D9967BC-D23D-4DB9-9ABE-9DA6B15B33A8@ifi.uio.no>
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Archived-At: <https://mailarchive.ietf.org/arch/msg/rtcweb/aXHLeMDYbDh1w1ux6OtHQ60Cgx8>
Cc: "Cullen Jennings \(fluffy\)" <fluffy@cisco.com>, "rtcweb@ietf.org" <rtcweb@ietf.org>, "tsvwg@ietf.org" <tsvwg@ietf.org>
Subject: Re: [rtcweb] [tsvwg] Fall-back to DSCP 0 in draft-ietf-tsvwg-rtcweb-qos ?
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To: <david.black@emc.com>, <michawe@ifi.uio.no>, <brian.e.carpenter@gmail.com>
Thread-Topic: [tsvwg] Fall-back to DSCP 0 in draft-ietf-tsvwg-rtcweb-qos ?
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Subject: Re: [rtcweb] [tsvwg] Fall-back to DSCP 0 in draft-ietf-tsvwg-rtcweb-qos ?
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From: Roman Shpount <roman@telurix.com>
Date: Mon, 11 Jul 2016 12:19:10 -0400
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Subject: Re: [rtcweb] [MMUSIC]  Tunnelling DTLS in SDP
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--001a114370be2dff9405375e8335
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I would definitely like to proceed with this. There are certain issues
related to forking and new associations being setup by the answering party,
but I think they are resolvable. On the other hand, the benefits of reduced
call setup up time definitely make this worthwhile.

_____________
Roman Shpount

On Mon, Jul 11, 2016 at 8:22 AM, Cullen Jennings <fluffy@cisco.com> wrote:

>
> Quick questions for folks =E2=80=A6  is an idea that we should explore a =
bit
> deeper before making decisions about it? or should we not proceed? or
> should we proceed? Just looking for feedback on where this should go next
>
> Thanks, Cullen
>
>
> On Apr 18, 2016, at 7:59 AM, Roman Shpount <roman@telurix.com> wrote:
>
> 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 message=
s
>> 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 wil=
l
> 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
>
> _______________________________________________
> 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
>
>

--001a114370be2dff9405375e8335
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<div dir=3D"ltr">I would definitely like to proceed with this. There are ce=
rtain issues related to forking and new associations being setup by the ans=
wering party, but I think they are resolvable. On the other hand, the benef=
its of reduced call setup up time definitely make this worthwhile.</div><di=
v class=3D"gmail_extra"><br clear=3D"all"><div><div class=3D"gmail_signatur=
e" data-smartmail=3D"gmail_signature">_____________<br>Roman Shpount</div><=
/div>
<br><div class=3D"gmail_quote">On Mon, Jul 11, 2016 at 8:22 AM, Cullen Jenn=
ings <span dir=3D"ltr">&lt;<a href=3D"mailto:fluffy@cisco.com" target=3D"_b=
lank">fluffy@cisco.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:1=
ex"><div style=3D"word-wrap:break-word"><div><br></div>Quick questions for =
folks =E2=80=A6 =C2=A0is an idea that we should explore a bit deeper before=
 making decisions about it? or should we not proceed? or should we proceed?=
 Just looking for feedback on where this should go next=C2=A0<div><br></div=
><div>Thanks, Cullen</div><div><br><div><br><div><blockquote type=3D"cite">=
<div><div class=3D"h5"><div>On Apr 18, 2016, at 7:59 AM, Roman Shpount &lt;=
<a href=3D"mailto:roman@telurix.com" target=3D"_blank">roman@telurix.com</a=
>&gt; wrote:</div><br></div></div><div><div><div class=3D"h5"><div dir=3D"l=
tr"><div class=3D"gmail_extra"><div><div>On Mon, Apr 4, 2016 at 9:10 AM, Er=
ic 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"gm=
ail_quote"><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"><div>I wanted to call your att=
ention to a draft I just published with a possibly stupid<br></div><div>ide=
a.</div><div><br></div><div><a href=3D"https://tools.ietf.org/html/draft-re=
scorla-dtls-in-sdp-00" target=3D"_blank">https://tools.ietf.org/html/draft-=
rescorla-dtls-in-sdp-00</a><br></div><div><br></div><div>A nontrivial fract=
ion of call setup time in WebRTC is the DTLS handshake.</div><div>This docu=
ment describes how to piggyback the first few handshake messages</div><div>=
in the SDP offer/answer exchange, thus reducing latency.</div><div><br></di=
v><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 offere=
r 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 solv=
ed with some sort of fallback mechanism to either regular DTLS setup or to =
sending a ClientHello in the answer.</div><div><br></div><div>Regards,</div=
><div><div>_____________<br>Roman Shpount</div></div><div>=C2=A0</div></div=
></div></div></div></div><span class=3D"">
_______________________________________________<br>rtcweb mailing list<br><=
a href=3D"mailto:rtcweb@ietf.org" target=3D"_blank">rtcweb@ietf.org</a><br>=
<a href=3D"https://www.ietf.org/mailman/listinfo/rtcweb" target=3D"_blank">=
https://www.ietf.org/mailman/listinfo/rtcweb</a><br></span></div></blockquo=
te></div><br></div></div></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>

--001a114370be2dff9405375e8335--


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Cc: Cullen Jennings <fluffy@cisco.com>, RTCWeb IETF <rtcweb@ietf.org>, mmusic WG <mmusic@ietf.org>
Subject: Re: [rtcweb] [MMUSIC]  Tunnelling DTLS in SDP
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> On 11 Jul 2016, at 17:19, Roman Shpount <roman@telurix.com> wrote:
>=20
> I would definitely like to proceed with this. There are certain issues =
related to forking and new associations being setup by the answering =
party, but I think they are resolvable. On the other hand, the benefits =
of reduced call setup up time definitely make this worthwhile.

As I said (remotely) at the last meeting, =20
I am distinctly _unkeen_ on the aesthetics of this - it ties us to ever =
uglier and larger SDP, it breaks software layer separation in the =
stacks,=20
it will no doubt presume a single DTLS implementation, it will be _fun_ =
to test, and I anticipate it will open some ugly attack vectors
to the javascript kiddies.=20

Last I heard EKR was going to experiment with it and come back with a =
view on how (in)accurate my doom-saying was.=20
I haven't heard the reply...

- If folks want a corridor conversation about this next week in Berlin, =
I'll be there.

T.

>=20
> _____________
> Roman Shpount
>=20
> On Mon, Jul 11, 2016 at 8:22 AM, Cullen Jennings <fluffy@cisco.com =
<mailto:fluffy@cisco.com>> wrote:
>=20
> Quick questions for folks =E2=80=A6  is an idea that we should explore =
a bit deeper before making decisions about it? or should we not proceed? =
or should we proceed? Just looking for feedback on where this should go =
next=20
>=20
> Thanks, Cullen
>=20
>=20
>> On Apr 18, 2016, at 7:59 AM, Roman Shpount <roman@telurix.com =
<mailto:roman@telurix.com>> wrote:
>>=20
>> On Mon, Apr 4, 2016 at 9:10 AM, Eric Rescorla <ekr@rtfm.com =
<mailto:ekr@rtfm.com>> wrote:
>> 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
>>=20
>> 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.
>>=20
>> Regards,
>> _____________
>> Roman Shpount
>> =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
> _______________________________________________
> mmusic mailing list
> mmusic@ietf.org <mailto:mmusic@ietf.org>
> https://www.ietf.org/mailman/listinfo/mmusic =
<https://www.ietf.org/mailman/listinfo/mmusic>
>=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>

--Apple-Mail=_4D84090A-9399-4B02-8FA3-F1FE227809B0
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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 11 Jul 2016, at 17:19, 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" style=3D"font-family: Helvetica; font-size: 12px; =
font-style: normal; font-variant-caps: normal; font-weight: normal; =
letter-spacing: normal; orphans: auto; text-align: start; text-indent: =
0px; text-transform: none; white-space: normal; widows: auto; =
word-spacing: 0px; -webkit-text-stroke-width: 0px;" class=3D"">I would =
definitely like to proceed with this. There are certain issues related =
to forking and new associations being setup by the answering party, but =
I think they are resolvable. On the other hand, the benefits of reduced =
call setup up time definitely make this =
worthwhile.</div></div></blockquote><div><br class=3D""></div><div>As I =
said (remotely) at the last meeting, &nbsp;</div><div>I am distinctly =
_unkeen_ on the aesthetics of this - it ties us to ever uglier and =
larger SDP, it breaks software layer separation in the =
stacks,&nbsp;</div><div>it will no doubt presume a single DTLS =
implementation, it will be _fun_ to test, and I anticipate it will open =
some ugly attack vectors</div><div>to the javascript =
kiddies.&nbsp;</div><div><br class=3D""></div>Last I heard EKR was going =
to experiment with it and come back with a view on how (in)accurate my =
doom-saying was.&nbsp;</div><div>I haven't heard the =
reply...</div><div><br class=3D""></div><div>- If folks want a corridor =
conversation about this next week in Berlin, I'll be =
there.</div><div><br class=3D""></div><div>T.</div><div><br =
class=3D""><blockquote type=3D"cite" class=3D""><div class=3D""><div =
class=3D"gmail_extra" style=3D"font-family: Helvetica; font-size: 12px; =
font-style: normal; font-variant-caps: normal; font-weight: normal; =
letter-spacing: normal; orphans: auto; text-align: start; text-indent: =
0px; text-transform: none; white-space: normal; widows: auto; =
word-spacing: 0px; -webkit-text-stroke-width: 0px;"><br clear=3D"all" =
class=3D""><div class=3D""><div class=3D"gmail_signature" =
data-smartmail=3D"gmail_signature">_____________<br class=3D"">Roman =
Shpount</div></div><br class=3D""><div class=3D"gmail_quote">On Mon, Jul =
11, 2016 at 8:22 AM, Cullen Jennings<span =
class=3D"Apple-converted-space">&nbsp;</span><span dir=3D"ltr" =
class=3D"">&lt;<a href=3D"mailto:fluffy@cisco.com" target=3D"_blank" =
class=3D"">fluffy@cisco.com</a>&gt;</span><span =
class=3D"Apple-converted-space">&nbsp;</span>wrote:<br =
class=3D""><blockquote class=3D"gmail_quote" style=3D"margin: 0px 0px =
0px 0.8ex; border-left-width: 1px; border-left-color: rgb(204, 204, =
204); border-left-style: solid; padding-left: 1ex;"><div =
style=3D"word-wrap: break-word;" class=3D""><div class=3D""><br =
class=3D""></div>Quick questions for folks =E2=80=A6 &nbsp;is an idea =
that we should explore a bit deeper before making decisions about it? or =
should we not proceed? or should we proceed? Just looking for feedback =
on where this should go next&nbsp;<div class=3D""><br =
class=3D""></div><div class=3D"">Thanks, Cullen</div><div class=3D""><br =
class=3D""><div class=3D""><br class=3D""><div class=3D""><blockquote =
type=3D"cite" class=3D""><div class=3D""><div class=3D"h5"><div =
class=3D"">On Apr 18, 2016, at 7:59 AM, Roman Shpount &lt;<a =
href=3D"mailto:roman@telurix.com" target=3D"_blank" =
class=3D"">roman@telurix.com</a>&gt; wrote:</div><br =
class=3D""></div></div><div class=3D""><div class=3D""><div =
class=3D"h5"><div dir=3D"ltr" class=3D""><div class=3D"gmail_extra"><div =
class=3D""><div class=3D"">On Mon, Apr 4, 2016 at 9:10 AM, Eric =
Rescorla<span class=3D"Apple-converted-space">&nbsp;</span><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><span =
class=3D"Apple-converted-space">&nbsp;</span>wrote:<br =
class=3D""></div></div><div class=3D"gmail_quote"><blockquote =
class=3D"gmail_quote" style=3D"margin: 0px 0px 0px 0.8ex; =
border-left-width: 1px; border-left-style: solid; border-left-color: =
rgb(204, 204, 204); padding-left: 1ex;"><div dir=3D"ltr" class=3D""><div =
class=3D"">I wanted to call your attention to a draft I just published =
with a possibly stupid<br class=3D""></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></blockquote><div class=3D""><br =
class=3D""></div><div class=3D"">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.</div><div class=3D""><br class=3D""></div><div =
class=3D"">Regards,</div><div class=3D""><div class=3D"">_____________<br =
class=3D"">Roman Shpount</div></div><div =
class=3D"">&nbsp;</div></div></div></div></div></div><span =
class=3D"">_______________________________________________<br =
class=3D"">rtcweb mailing list<br class=3D""><a =
href=3D"mailto:rtcweb@ietf.org" target=3D"_blank" =
class=3D"">rtcweb@ietf.org</a><br class=3D""><a =
href=3D"https://www.ietf.org/mailman/listinfo/rtcweb" target=3D"_blank" =
class=3D"">https://www.ietf.org/mailman/listinfo/rtcweb</a><br =
class=3D""></span></div></blockquote></div><br =
class=3D""></div></div></div><br =
class=3D"">_______________________________________________<br =
class=3D"">mmusic mailing list<br class=3D""><a =
href=3D"mailto:mmusic@ietf.org" class=3D"">mmusic@ietf.org</a><br =
class=3D""><a href=3D"https://www.ietf.org/mailman/listinfo/mmusic" =
rel=3D"noreferrer" target=3D"_blank" =
class=3D"">https://www.ietf.org/mailman/listinfo/mmusic</a><br =
class=3D""><br class=3D""></blockquote></div><br class=3D""></div><span =
style=3D"font-family: Helvetica; font-size: 12px; font-style: normal; =
font-variant-caps: normal; font-weight: normal; letter-spacing: normal; =
orphans: auto; text-align: start; text-indent: 0px; text-transform: =
none; white-space: normal; widows: auto; word-spacing: 0px; =
-webkit-text-stroke-width: 0px; float: none; display: inline =
!important;" =
class=3D"">_______________________________________________</span><br =
style=3D"font-family: Helvetica; font-size: 12px; font-style: normal; =
font-variant-caps: normal; font-weight: normal; letter-spacing: normal; =
orphans: auto; text-align: start; text-indent: 0px; text-transform: =
none; white-space: normal; widows: auto; word-spacing: 0px; =
-webkit-text-stroke-width: 0px;" class=3D""><span style=3D"font-family: =
Helvetica; font-size: 12px; font-style: normal; font-variant-caps: =
normal; font-weight: normal; letter-spacing: normal; orphans: auto; =
text-align: start; text-indent: 0px; text-transform: none; white-space: =
normal; widows: auto; word-spacing: 0px; -webkit-text-stroke-width: 0px; =
float: none; display: inline !important;" class=3D"">rtcweb mailing =
list</span><br style=3D"font-family: Helvetica; font-size: 12px; =
font-style: normal; font-variant-caps: normal; font-weight: normal; =
letter-spacing: normal; orphans: auto; text-align: start; text-indent: =
0px; text-transform: none; white-space: normal; widows: auto; =
word-spacing: 0px; -webkit-text-stroke-width: 0px;" class=3D""><a =
href=3D"mailto:rtcweb@ietf.org" style=3D"font-family: Helvetica; =
font-size: 12px; font-style: normal; font-variant-caps: normal; =
font-weight: normal; letter-spacing: normal; orphans: auto; text-align: =
start; text-indent: 0px; text-transform: none; white-space: normal; =
widows: auto; word-spacing: 0px; -webkit-text-stroke-width: 0px;" =
class=3D"">rtcweb@ietf.org</a><br style=3D"font-family: Helvetica; =
font-size: 12px; font-style: normal; font-variant-caps: normal; =
font-weight: normal; letter-spacing: normal; orphans: auto; text-align: =
start; text-indent: 0px; text-transform: none; white-space: normal; =
widows: auto; word-spacing: 0px; -webkit-text-stroke-width: 0px;" =
class=3D""><a href=3D"https://www.ietf.org/mailman/listinfo/rtcweb" =
style=3D"font-family: Helvetica; font-size: 12px; font-style: normal; =
font-variant-caps: normal; font-weight: normal; letter-spacing: normal; =
orphans: auto; text-align: start; text-indent: 0px; text-transform: =
none; white-space: normal; widows: auto; word-spacing: 0px; =
-webkit-text-stroke-width: 0px;" =
class=3D"">https://www.ietf.org/mailman/listinfo/rtcweb</a></div></blockqu=
ote></div><br class=3D""></body></html>=

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From nobody Mon Jul 11 09:59:03 2016
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From: Christer Holmberg <christer.holmberg@ericsson.com>
To: T H Panton <tim@phonefromhere.com>, Roman Shpount <roman@telurix.com>
Thread-Topic: [MMUSIC] [rtcweb]   Tunnelling DTLS in SDP
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Date: Mon, 11 Jul 2016 16:58:54 +0000
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References: <CABcZeBOM1KoXpXFhvjS753EVpsMENWVen3CCdFj8ry36vPH0dg@mail.gmail.com> <CAD5OKxt2+YAJfR1q4qyhv_0D7AqMmrsTENTGNw-_vo9Dzr-ejA@mail.gmail.com> <16EBD211-7FAD-43F1-A855-639EC17A17FE@cisco.com> <CAD5OKxsA8UOE83GP67h0NB9H3Lqmw0eC6nhjB=A=7GwdeZf3fw@mail.gmail.com>, <7DF2166A-3AF4-4591-AB25-8C5415875E8D@phonefromhere.com>
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Archived-At: <https://mailarchive.ietf.org/arch/msg/rtcweb/uUREAX7E7E_f_wOTp4NdsEMvtqE>
Cc: Cullen Jennings <fluffy@cisco.com>, RTCWeb IETF <rtcweb@ietf.org>, mmusic WG <mmusic@ietf.org>
Subject: Re: [rtcweb] [MMUSIC]    Tunnelling DTLS in SDP
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Hi,

All I'll say at this point is that you cannot assume that forking entities =
will remove the attribute. You either need to have a fallback procedure in =
case of forking, or say that the mechanism MUST NOT be used in environments=
 where forking can occur (which makes the mechanism more or less useless in=
 virtually all SIP environments).

Regards,

Christer


Sent from my Windows Phone
________________________________
From: T H Panton<mailto:tim@phonefromhere.com>
Sent: =FD11/=FD07/=FD2016 19:33
To: Roman Shpount<mailto:roman@telurix.com>
Cc: Cullen Jennings<mailto:fluffy@cisco.com>; RTCWeb IETF<mailto:rtcweb@iet=
f.org>; mmusic WG<mailto:mmusic@ietf.org>
Subject: Re: [MMUSIC] [rtcweb]   Tunnelling DTLS in SDP


On 11 Jul 2016, at 17:19, Roman Shpount <roman@telurix.com<mailto:roman@tel=
urix.com>> wrote:

I would definitely like to proceed with this. There are certain issues rela=
ted to forking and new associations being setup by the answering party, but=
 I think they are resolvable. On the other hand, the benefits of reduced ca=
ll setup up time definitely make this worthwhile.

As I said (remotely) at the last meeting,
I am distinctly _unkeen_ on the aesthetics of this - it ties us to ever ugl=
ier and larger SDP, it breaks software layer separation in the stacks,
it will no doubt presume a single DTLS implementation, it will be _fun_ to =
test, and I anticipate it will open some ugly attack vectors
to the javascript kiddies.

Last I heard EKR was going to experiment with it and come back with a view =
on how (in)accurate my doom-saying was.
I haven't heard the reply...

- If folks want a corridor conversation about this next week in Berlin, I'l=
l be there.

T.


_____________
Roman Shpount

On Mon, Jul 11, 2016 at 8:22 AM, Cullen Jennings <fluffy@cisco.com<mailto:f=
luffy@cisco.com>> wrote:

Quick questions for folks =85  is an idea that we should explore a bit deep=
er before making decisions about it? or should we not proceed? or should we=
 proceed? Just looking for feedback on where this should go next

Thanks, Cullen


On Apr 18, 2016, at 7:59 AM, Roman Shpount <roman@telurix.com<mailto:roman@=
telurix.com>> wrote:

On Mon, Apr 4, 2016 at 9:10 AM, Eric Rescorla <ekr@rtfm.com<mailto: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 answ=
erer can start a new DTLS association. When answerer starts a new session i=
t will have exactly the same problems as forking, since this will create mu=
ltiple DTLS sessions with the same client random. This can be solved with s=
ome sort of fallback mechanism to either regular DTLS setup or to sending a=
 ClientHello in the answer.

Regards,
_____________
Roman Shpount

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


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


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


--_000_7594FB04B1934943A5C02806D1A2204B4768FEBBESESSMB208erics_
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<html>
<head>
<meta http-equiv=3D"Content-Type" content=3D"text/html; charset=3Dwindows-1=
256">
</head>
<body class=3D"" style=3D"word-wrap:break-word">
<div>
<div style=3D"font-family:Calibri,sans-serif; font-size:11pt">Hi,<br>
<br>
All I'll say at this point is that you cannot assume that forking entities =
will remove the attribute. You either need to have a fallback procedure in =
case of forking, or say that the mechanism MUST NOT be used in environments=
 where forking can occur (which
 makes the mechanism more or less useless in virtually all SIP environments=
).<br>
<br>
Regards,<br>
<br>
Christer<br>
<br>
<br>
Sent from my Windows Phone</div>
</div>
<div dir=3D"ltr">
<hr>
<span style=3D"font-family:Calibri,sans-serif; font-size:11pt; font-weight:=
bold">From:
</span><span style=3D"font-family:Calibri,sans-serif; font-size:11pt"><a hr=
ef=3D"mailto:tim@phonefromhere.com">T H Panton</a></span><br>
<span style=3D"font-family:Calibri,sans-serif; font-size:11pt; font-weight:=
bold">Sent:
</span><span style=3D"font-family:Calibri,sans-serif; font-size:11pt">=FD11=
/=FD07/=FD2016 19:33</span><br>
<span style=3D"font-family:Calibri,sans-serif; font-size:11pt; font-weight:=
bold">To:
</span><span style=3D"font-family:Calibri,sans-serif; font-size:11pt"><a hr=
ef=3D"mailto:roman@telurix.com">Roman Shpount</a></span><br>
<span style=3D"font-family:Calibri,sans-serif; font-size:11pt; font-weight:=
bold">Cc:
</span><span style=3D"font-family:Calibri,sans-serif; font-size:11pt"><a hr=
ef=3D"mailto:fluffy@cisco.com">Cullen Jennings</a>;
<a href=3D"mailto:rtcweb@ietf.org">RTCWeb IETF</a>; <a href=3D"mailto:mmusi=
c@ietf.org">
mmusic WG</a></span><br>
<span style=3D"font-family:Calibri,sans-serif; font-size:11pt; font-weight:=
bold">Subject:
</span><span style=3D"font-family:Calibri,sans-serif; font-size:11pt">Re: [=
MMUSIC] [rtcweb]&nbsp;&nbsp; Tunnelling DTLS in SDP</span><br>
<br>
</div>
<div><br class=3D"">
<div>
<blockquote type=3D"cite" class=3D"">
<div class=3D"">On 11 Jul 2016, at 17:19, Roman Shpount &lt;<a href=3D"mail=
to: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"" style=3D"font-family:Helvetica; font-size:12px;=
 font-style:normal; font-weight:normal; letter-spacing:normal; orphans:auto=
; text-align:start; text-indent:0px; text-transform:none; white-space:norma=
l; widows:auto; word-spacing:0px">
I would definitely like to proceed with this. There are certain issues rela=
ted to forking and new associations being setup by the answering party, but=
 I think they are resolvable. On the other hand, the benefits of reduced ca=
ll setup up time definitely make
 this worthwhile.</div>
</div>
</blockquote>
<div><br class=3D"">
</div>
<div>As I said (remotely) at the last meeting, &nbsp;</div>
<div>I am distinctly _unkeen_ on the aesthetics of this - it ties us to eve=
r uglier and larger SDP, it breaks software layer separation in the stacks,=
&nbsp;</div>
<div>it will no doubt presume a single DTLS implementation, it will be _fun=
_ to test, and I anticipate it will open some ugly attack vectors</div>
<div>to the javascript kiddies.&nbsp;</div>
<div><br class=3D"">
</div>
Last I heard EKR was going to experiment with it and come back with a view =
on how (in)accurate my doom-saying was.&nbsp;</div>
<div>I haven't heard the reply...</div>
<div><br class=3D"">
</div>
<div>- If folks want a corridor conversation about this next week in Berlin=
, I'll be there.</div>
<div><br class=3D"">
</div>
<div>T.</div>
<div><br class=3D"">
<blockquote type=3D"cite" class=3D"">
<div class=3D"">
<div class=3D"gmail_extra" style=3D"font-family:Helvetica; font-size:12px; =
font-style:normal; font-weight:normal; letter-spacing:normal; orphans:auto;=
 text-align:start; text-indent:0px; text-transform:none; white-space:normal=
; widows:auto; word-spacing:0px">
<br clear=3D"all" class=3D"">
<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, Jul 11, 2016 at 8:22 AM, Cullen Jennings=
<span class=3D"Apple-converted-space">&nbsp;</span><span dir=3D"ltr" class=
=3D"">&lt;<a href=3D"mailto:fluffy@cisco.com" target=3D"_blank" class=3D"">=
fluffy@cisco.com</a>&gt;</span><span class=3D"Apple-converted-space">&nbsp;=
</span>wrote:<br class=3D"">
<blockquote class=3D"gmail_quote" style=3D"margin:0px 0px 0px 0.8ex; border=
-left-width:1px; border-left-color:rgb(204,204,204); border-left-style:soli=
d; padding-left:1ex">
<div class=3D"" style=3D"word-wrap:break-word">
<div class=3D""><br class=3D"">
</div>
Quick questions for folks =85 &nbsp;is an idea that we should explore a bit=
 deeper before making decisions about it? or should we not proceed? or shou=
ld we proceed? Just looking for feedback on where this should go next&nbsp;
<div class=3D""><br class=3D"">
</div>
<div class=3D"">Thanks, Cullen</div>
<div class=3D""><br class=3D"">
<div class=3D""><br class=3D"">
<div class=3D"">
<blockquote type=3D"cite" class=3D"">
<div class=3D"">
<div class=3D"h5">
<div class=3D"">On Apr 18, 2016, at 7:59 AM, Roman Shpount &lt;<a href=3D"m=
ailto:roman@telurix.com" target=3D"_blank" class=3D"">roman@telurix.com</a>=
&gt; wrote:</div>
<br class=3D"">
</div>
</div>
<div class=3D"">
<div class=3D"">
<div class=3D"h5">
<div dir=3D"ltr" class=3D"">
<div class=3D"gmail_extra">
<div class=3D"">
<div class=3D"">On Mon, Apr 4, 2016 at 9:10 AM, Eric Rescorla<span class=3D=
"Apple-converted-space">&nbsp;</span><span dir=3D"ltr" class=3D"">&lt;<a hr=
ef=3D"mailto:ekr@rtfm.com" target=3D"_blank" class=3D"">ekr@rtfm.com</a>&gt=
;</span><span class=3D"Apple-converted-space">&nbsp;</span>wrote:<br class=
=3D"">
</div>
</div>
<div class=3D"gmail_quote">
<blockquote class=3D"gmail_quote" style=3D"margin:0px 0px 0px 0.8ex; border=
-left-width:1px; border-left-style:solid; border-left-color:rgb(204,204,204=
); padding-left:1ex">
<div dir=3D"ltr" class=3D"">
<div class=3D"">I wanted to call your attention to a draft I just published=
 with a possibly stupid<br class=3D"">
</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-r=
escorla-dtls-in-sdp-00</a><br class=3D"">
</div>
<div class=3D""><br class=3D"">
</div>
<div class=3D"">A nontrivial fraction of call setup time in WebRTC is the D=
TLS handshake.</div>
<div class=3D"">This document describes how to piggyback the first few hand=
shake messages</div>
<div class=3D"">in the SDP offer/answer exchange, thus reducing latency.</d=
iv>
<div class=3D""><br class=3D"">
</div>
<div class=3D"">Comments welcome.</div>
<div class=3D""><br class=3D"">
</div>
</div>
</blockquote>
<div class=3D""><br class=3D"">
</div>
<div class=3D"">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 thi=
s 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.</div>
<div class=3D""><br class=3D"">
</div>
<div class=3D"">Regards,</div>
<div class=3D"">
<div class=3D"">_____________<br class=3D"">
Roman Shpount</div>
</div>
<div class=3D"">&nbsp;</div>
</div>
</div>
</div>
</div>
</div>
<span class=3D"">_______________________________________________<br class=
=3D"">
rtcweb mailing list<br class=3D"">
<a href=3D"mailto:rtcweb@ietf.org" target=3D"_blank" class=3D"">rtcweb@ietf=
.org</a><br class=3D"">
<a href=3D"https://www.ietf.org/mailman/listinfo/rtcweb" target=3D"_blank" =
class=3D"">https://www.ietf.org/mailman/listinfo/rtcweb</a><br class=3D"">
</span></div>
</blockquote>
</div>
<br class=3D"">
</div>
</div>
</div>
<br class=3D"">
_______________________________________________<br class=3D"">
mmusic mailing list<br class=3D"">
<a href=3D"mailto:mmusic@ietf.org" class=3D"">mmusic@ietf.org</a><br class=
=3D"">
<a href=3D"https://www.ietf.org/mailman/listinfo/mmusic" rel=3D"noreferrer"=
 target=3D"_blank" class=3D"">https://www.ietf.org/mailman/listinfo/mmusic<=
/a><br class=3D"">
<br class=3D"">
</blockquote>
</div>
<br class=3D"">
</div>
<span class=3D"" style=3D"font-family:Helvetica; font-size:12px; font-style=
:normal; font-weight:normal; letter-spacing:normal; orphans:auto; text-alig=
n:start; text-indent:0px; text-transform:none; white-space:normal; widows:a=
uto; word-spacing:0px; float:none; display:inline!important">______________=
_________________________________</span><br class=3D"" style=3D"font-family=
:Helvetica; font-size:12px; font-style:normal; font-weight:normal; letter-s=
pacing:normal; orphans:auto; text-align:start; text-indent:0px; text-transf=
orm:none; white-space:normal; widows:auto; word-spacing:0px">
<span class=3D"" style=3D"font-family:Helvetica; font-size:12px; font-style=
:normal; font-weight:normal; letter-spacing:normal; orphans:auto; text-alig=
n:start; text-indent:0px; text-transform:none; white-space:normal; widows:a=
uto; word-spacing:0px; float:none; display:inline!important">rtcweb
 mailing list</span><br class=3D"" style=3D"font-family:Helvetica; font-siz=
e:12px; font-style:normal; font-weight:normal; letter-spacing:normal; orpha=
ns:auto; text-align:start; text-indent:0px; text-transform:none; white-spac=
e:normal; widows:auto; word-spacing:0px">
<a href=3D"mailto:rtcweb@ietf.org" class=3D"" style=3D"font-family:Helvetic=
a; font-size:12px; font-style:normal; font-weight:normal; letter-spacing:no=
rmal; orphans:auto; text-align:start; text-indent:0px; text-transform:none;=
 white-space:normal; widows:auto; word-spacing:0px">rtcweb@ietf.org</a><br =
class=3D"" style=3D"font-family:Helvetica; font-size:12px; font-style:norma=
l; font-weight:normal; letter-spacing:normal; orphans:auto; text-align:star=
t; text-indent:0px; text-transform:none; white-space:normal; widows:auto; w=
ord-spacing:0px">
<a href=3D"https://www.ietf.org/mailman/listinfo/rtcweb" class=3D"" style=
=3D"font-family:Helvetica; font-size:12px; font-style:normal; font-weight:n=
ormal; letter-spacing:normal; orphans:auto; text-align:start; text-indent:0=
px; text-transform:none; white-space:normal; widows:auto; word-spacing:0px"=
>https://www.ietf.org/mailman/listinfo/rtcweb</a></div>
</blockquote>
</div>
<br class=3D"">
</div>
</body>
</html>

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From: =?UTF-8?Q?I=C3=B1aki_Baz_Castillo?= <ibc@aliax.net>
Date: Mon, 11 Jul 2016 19:11:34 +0200
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To: Christer Holmberg <christer.holmberg@ericsson.com>
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Archived-At: <https://mailarchive.ietf.org/arch/msg/rtcweb/_d-rrua_gSQSI74gyhfYdLrbd_s>
Cc: Cullen Jennings <fluffy@cisco.com>, RTCWeb IETF <rtcweb@ietf.org>, mmusic WG <mmusic@ietf.org>
Subject: Re: [rtcweb] [MMUSIC]  Tunnelling DTLS in SDP
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2016-07-11 18:58 GMT+02:00 Christer Holmberg <christer.holmberg@ericsson.co=
m>:
> All I'll say at this point is that you cannot assume that forking entitie=
s
> will remove the attribute. You either need to have a fallback procedure i=
n
> case of forking, or say that the mechanism MUST NOT be used in environmen=
ts
> where forking can occur (which makes the mechanism more or less useless i=
n
> virtually all SIP environments).

SIP devices can just ignore this spec like if it was Outbound or GRUU.


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


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From: "Black, David" <david.black@emc.com>
To: "Ruediger.Geib@telekom.de" <Ruediger.Geib@telekom.de>, "michawe@ifi.uio.no" <michawe@ifi.uio.no>, "brian.e.carpenter@gmail.com" <brian.e.carpenter@gmail.com>
Thread-Topic: [tsvwg] Fall-back to DSCP 0 in draft-ietf-tsvwg-rtcweb-qos ?
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Date: Mon, 11 Jul 2016 18:22:23 +0000
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Subject: Re: [rtcweb] [tsvwg] Fall-back to DSCP 0 in draft-ietf-tsvwg-rtcweb-qos ?
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From: Eric Rescorla <ekr@rtfm.com>
Date: Mon, 11 Jul 2016 15:40:40 -0700
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To: T H Panton <tim@phonefromhere.com>
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Cc: Cullen Jennings <fluffy@cisco.com>, RTCWeb IETF <rtcweb@ietf.org>, mmusic WG <mmusic@ietf.org>
Subject: Re: [rtcweb] [MMUSIC]  Tunnelling DTLS in SDP
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On Mon, Jul 11, 2016 at 9:33 AM, T H Panton <tim@phonefromhere.com> wrote:

>
> On 11 Jul 2016, at 17:19, Roman Shpount <roman@telurix.com> wrote:
>
> I would definitely like to proceed with this. There are certain issues
> related to forking and new associations being setup by the answering part=
y,
> but I think they are resolvable. On the other hand, the benefits of reduc=
ed
> call setup up time definitely make this worthwhile.
>
>
> As I said (remotely) at the last meeting,
> I am distinctly _unkeen_ on the aesthetics of this - it ties us to ever
> uglier and larger SDP, it breaks software layer separation in the stacks,
> it will no doubt presume a single DTLS implementation, it will be _fun_ t=
o
> test, and I anticipate it will open some ugly attack vectors
> to the javascript kiddies.
>

> Last I heard EKR was going to experiment with it and come back with a vie=
w
> on how (in)accurate my doom-saying was.
> I haven't heard the reply...
>

We're working on an implementation now but it took a while to divert the
staff.

-Ekr


>
> - If folks want a corridor conversation about this next week in Berlin,
> I'll be there.
>
> T.
>
>
> _____________
> Roman Shpount
>
> On Mon, Jul 11, 2016 at 8:22 AM, Cullen Jennings <fluffy@cisco.com> wrote=
:
>
>>
>> Quick questions for folks =E2=80=A6  is an idea that we should explore a=
 bit
>> deeper before making decisions about it? or should we not proceed? or
>> should we proceed? Just looking for feedback on where this should go nex=
t
>>
>> Thanks, Cullen
>>
>>
>> On Apr 18, 2016, at 7:59 AM, Roman Shpount <roman@telurix.com> wrote:
>>
>> 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 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.
>>>
>>>
>> 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 wi=
ll
>> 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
>>
>> _______________________________________________
>> 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
>>
>>
> _______________________________________________
> 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
>
>

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<div dir=3D"ltr"><br><div class=3D"gmail_extra"><br><div class=3D"gmail_quo=
te">On Mon, Jul 11, 2016 at 9:33 AM, T H Panton <span dir=3D"ltr">&lt;<a hr=
ef=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 class=3D""><blockquote type=3D"cite"><div>=
On 11 Jul 2016, at 17:19, Roman Shpount &lt;<a href=3D"mailto:roman@telurix=
.com" target=3D"_blank">roman@telurix.com</a>&gt; wrote:</div><br><div><div=
 dir=3D"ltr" style=3D"font-family:Helvetica;font-size:12px;font-style:norma=
l;font-weight:normal;letter-spacing:normal;text-align:start;text-indent:0px=
;text-transform:none;white-space:normal;word-spacing:0px">I would definitel=
y like to proceed with this. There are certain issues related to forking an=
d new associations being setup by the answering party, but I think they are=
 resolvable. On the other hand, the benefits of reduced call setup up time =
definitely make this worthwhile.</div></div></blockquote><div><br></div></s=
pan><div>As I said (remotely) at the last meeting, =C2=A0</div><div>I am di=
stinctly _unkeen_ on the aesthetics of this - it ties us to ever uglier and=
 larger SDP, it breaks software layer separation in the stacks,=C2=A0</div>=
<div>it will no doubt presume a single DTLS implementation, it will be _fun=
_ to test, and I anticipate it will open some ugly attack vectors</div><div=
>to the javascript kiddies.=C2=A0</div></div></div></blockquote><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=
>Last I heard EKR was going to experiment with it and come back with a view=
 on how (in)accurate my doom-saying was.=C2=A0</div><div>I haven&#39;t hear=
d the reply...</div></div></blockquote><div><br></div><div>We&#39;re workin=
g on an implementation now but it took a while to divert the staff.</div><d=
iv><br></div><div>-Ekr</div><div>=C2=A0</div><blockquote class=3D"gmail_quo=
te" style=3D"margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex"=
><div style=3D"word-wrap:break-word"><div><br></div><div>- If folks want a =
corridor conversation about this next week in Berlin, I&#39;ll be there.</d=
iv><div><br></div><div>T.</div><div><div class=3D"h5"><div><br><blockquote =
type=3D"cite"><div><div class=3D"gmail_extra" style=3D"font-family:Helvetic=
a;font-size:12px;font-style:normal;font-weight:normal;letter-spacing:normal=
;text-align:start;text-indent:0px;text-transform:none;white-space:normal;wo=
rd-spacing:0px"><br clear=3D"all"><div><div data-smartmail=3D"gmail_signatu=
re">_____________<br>Roman Shpount</div></div><br><div class=3D"gmail_quote=
">On Mon, Jul 11, 2016 at 8:22 AM, Cullen Jennings<span>=C2=A0</span><span =
dir=3D"ltr">&lt;<a href=3D"mailto:fluffy@cisco.com" target=3D"_blank">fluff=
y@cisco.com</a>&gt;</span><span>=C2=A0</span>wrote:<br><blockquote class=3D=
"gmail_quote" style=3D"margin:0px 0px 0px 0.8ex;border-left-width:1px;borde=
r-left-color:rgb(204,204,204);border-left-style:solid;padding-left:1ex"><di=
v style=3D"word-wrap:break-word"><div><br></div>Quick questions for folks =
=E2=80=A6 =C2=A0is an idea that we should explore a bit deeper before makin=
g decisions about it? or should we not proceed? or should we proceed? Just =
looking for feedback on where this should go next=C2=A0<div><br></div><div>=
Thanks, Cullen</div><div><br><div><br><div><blockquote type=3D"cite"><div><=
div><div>On Apr 18, 2016, at 7:59 AM, Roman Shpount &lt;<a href=3D"mailto:r=
oman@telurix.com" target=3D"_blank">roman@telurix.com</a>&gt; wrote:</div><=
br></div></div><div><div><div><div dir=3D"ltr"><div class=3D"gmail_extra"><=
div><div>On Mon, Apr 4, 2016 at 9:10 AM, Eric Rescorla<span>=C2=A0</span><s=
pan dir=3D"ltr">&lt;<a href=3D"mailto:ekr@rtfm.com" target=3D"_blank">ekr@r=
tfm.com</a>&gt;</span><span>=C2=A0</span>wrote:<br></div></div><div class=
=3D"gmail_quote"><blockquote class=3D"gmail_quote" style=3D"margin:0px 0px =
0px 0.8ex;border-left-width:1px;border-left-style:solid;border-left-color:r=
gb(204,204,204);padding-left:1ex"><div dir=3D"ltr"><div>I wanted to call yo=
ur attention to a draft I just published with a possibly stupid<br></div><d=
iv>idea.</div><div><br></div><div><a href=3D"https://tools.ietf.org/html/dr=
aft-rescorla-dtls-in-sdp-00" target=3D"_blank">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>Thi=
s document describes how to piggyback the first few handshake messages</div=
><div>in the SDP offer/answer exchange, thus reducing latency.</div><div><b=
r></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 same problems as forking, since this=
 will create multiple DTLS sessions with the same client random. This can b=
e solved with some sort of fallback mechanism to either regular DTLS setup =
or to sending a ClientHello in the answer.</div><div><br></div><div>Regards=
,</div><div><div>_____________<br>Roman Shpount</div></div><div>=C2=A0</div=
></div></div></div></div></div><span>______________________________________=
_________<br>rtcweb mailing list<br><a href=3D"mailto:rtcweb@ietf.org" targ=
et=3D"_blank">rtcweb@ietf.org</a><br><a href=3D"https://www.ietf.org/mailma=
n/listinfo/rtcweb" target=3D"_blank">https://www.ietf.org/mailman/listinfo/=
rtcweb</a><br></span></div></blockquote></div><br></div></div></div><br>___=
____________________________________________<br>mmusic mailing list<br><a h=
ref=3D"mailto:mmusic@ietf.org" target=3D"_blank">mmusic@ietf.org</a><br><a =
href=3D"https://www.ietf.org/mailman/listinfo/mmusic" rel=3D"noreferrer" ta=
rget=3D"_blank">https://www.ietf.org/mailman/listinfo/mmusic</a><br><br></b=
lockquote></div><br></div><span style=3D"font-family:Helvetica;font-size:12=
px;font-style:normal;font-weight:normal;letter-spacing:normal;text-align:st=
art;text-indent:0px;text-transform:none;white-space:normal;word-spacing:0px=
;float:none;display:inline!important">_____________________________________=
__________</span><br style=3D"font-family:Helvetica;font-size:12px;font-sty=
le:normal;font-weight:normal;letter-spacing:normal;text-align:start;text-in=
dent:0px;text-transform:none;white-space:normal;word-spacing:0px"><span sty=
le=3D"font-family:Helvetica;font-size:12px;font-style:normal;font-weight:no=
rmal;letter-spacing:normal;text-align:start;text-indent:0px;text-transform:=
none;white-space:normal;word-spacing:0px;float:none;display:inline!importan=
t">rtcweb mailing list</span><br style=3D"font-family:Helvetica;font-size:1=
2px;font-style:normal;font-weight:normal;letter-spacing:normal;text-align:s=
tart;text-indent:0px;text-transform:none;white-space:normal;word-spacing:0p=
x"><a href=3D"mailto:rtcweb@ietf.org" style=3D"font-family:Helvetica;font-s=
ize:12px;font-style:normal;font-weight:normal;letter-spacing:normal;text-al=
ign:start;text-indent:0px;text-transform:none;white-space:normal;word-spaci=
ng:0px" target=3D"_blank">rtcweb@ietf.org</a><br style=3D"font-family:Helve=
tica;font-size:12px;font-style:normal;font-weight:normal;letter-spacing:nor=
mal;text-align:start;text-indent:0px;text-transform:none;white-space:normal=
;word-spacing:0px"><a href=3D"https://www.ietf.org/mailman/listinfo/rtcweb"=
 style=3D"font-family:Helvetica;font-size:12px;font-style:normal;font-weigh=
t:normal;letter-spacing:normal;text-align:start;text-indent:0px;text-transf=
orm:none;white-space:normal;word-spacing:0px" target=3D"_blank">https://www=
.ietf.org/mailman/listinfo/rtcweb</a></div></blockquote></div><br></div></d=
iv></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>

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From: Ted Hardie <ted.ietf@gmail.com>
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Subject: [rtcweb] Agenda change: Friday meeting returned to pool
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Howdy,

Fewer of the drafts were updated and fewer reviews received than the chairs
expected.  Given the very tight timing for this IETF and the overlaps in
ART, we have agreed with the ADs to return the Friday slot to the pool for
rescheduling.

Our current agenda proposal is this:

Agenda
Administrivia (5 mins)
JSEP  (90 mins)
draft-ietf-rtcweb-sdp  (15 mins)
Matters from other WGs (10 mins)

Comments, edits, and suggestions welcome,

Ted, Cullen, Sean

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<div dir=3D"ltr"><div><div>Howdy,<br><br></div>Fewer of the drafts were upd=
ated and fewer reviews received than the chairs expected.=C2=A0 Given the v=
ery tight timing for this IETF and the overlaps in ART, we have agreed with=
 the ADs to return the Friday slot to the pool for rescheduling.<br><br></d=
iv>Our current agenda proposal is this:<br><br><pre>Agenda=20
Administrivia (5 mins)
JSEP  (90 mins)
draft-ietf-rtcweb-sdp  (15 mins)
Matters from other WGs (10 mins)<br><br></pre><pre>Comments, edits, and sug=
gestions welcome,<br><br></pre><pre>Ted, Cullen, Sean <br></pre></div>

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To: "Black, David" <david.black@emc.com>
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Cc: "fluffy@cisco.com" <fluffy@cisco.com>, "Ruediger.Geib@telekom.de" <Ruediger.Geib@telekom.de>, "tsvwg@ietf.org" <tsvwg@ietf.org>, "rtcweb@ietf.org" <rtcweb@ietf.org>, "brian.e.carpenter@gmail.com" <brian.e.carpenter@gmail.com>
Subject: Re: [rtcweb] [tsvwg] Fall-back to DSCP 0 in draft-ietf-tsvwg-rtcweb-qos ?
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The DART work warns of the pitfalls.=20

My suggestion is that for protocols using ICE that STUN packets could be use=
d for this path check - if this is desired. I am not sure we need to hold th=
e current draft for this sort of addition.

Gorry

On 11 Jul 2016, at 19:22, Black, David <david.black@emc.com> wrote:

>> are you asking, whether the treatment of unexpected DSCPs
>=20
> No, I'm asking which draft should document what a WebRTC application or a b=
rowser
> hosting a WebRTC application should do in order to be robust to black-hole=

> (unreachability) behavior caused by use of a DSCP other than 000000.
>=20
>> Didn't read the DART work before replying, but I assume the situation isn=
't
>> covered in sufficient detail there.
>=20
> Nope, it isn't ...
>=20
> Thanks, --David
>=20
>> -----Original Message-----
>> From: Ruediger.Geib@telekom.de [mailto:Ruediger.Geib@telekom.de]
>> Sent: Monday, July 11, 2016 10:34 AM
>> To: Black, David; michawe@ifi.uio.no; brian.e.carpenter@gmail.com
>> Cc: fluffy@cisco.com; rtcweb@ietf.org; tsvwg@ietf.org
>> Subject: AW: [tsvwg] Fall-back to DSCP 0 in draft-ietf-tsvwg-rtcweb-qos ?=

>>=20
>> Hi David,
>>=20
>> are you asking, whether the treatment of unexpected DSCPs
>> - should be documented within draft-ietf-tsvwg-rtcweb-qos or
>> - should be documented within draft-jennings-ice-rtcweb-timing-01 or
>> - should be documented within a to be written draft-unexpected-dscp-
>> treatment?
>>=20
>> We've discussed bleaching vs transport of unexpected DSCPs while working o=
n
>> diffserv-intercon as a possible measure (but never discussed dropping). I=
 favor a
>> separate and generic RFC dealing with the issue, if that is what you ask f=
or. I think
>> the issue of unexpected DSCP treatment is not rtcweb/STUN/... specific. T=
he
>> same holds for the avoidance of blackholing due to DSCP marks. Hints to a=
void it
>> should be part of that draft.
>>=20
>> Didn't read the DART work before replying, but I assume the situation isn=
't
>> covered in sufficient detail there.
>>=20
>> Regards,
>>=20
>> Ruediger
>>=20
>>=20
>> -----Urspr=C3=BCngliche Nachricht-----
>> Von: tsvwg [mailto:tsvwg-bounces@ietf.org] Im Auftrag von Black, David
>> Gesendet: Montag, 11. Juli 2016 15:56
>> An: Michael Welzl; Brian E Carpenter
>> Cc: Cullen Jennings (fluffy); rtcweb@ietf.org; tsvwg@ietf.org
>> Betreff: Re: [tsvwg] Fall-back to DSCP 0 in draft-ietf-tsvwg-rtcweb-qos ?=

>>=20
>> [+rtcweb]
>>=20
>>>>> We believe that draft-ietf-tsvwg-rtcweb-qos should say: if setting
>>>>> the DSCP
>>> values as proposed here consistently produces packet drops, senders
>>> should fall back to DSCP 0. WebRTC shouldn't "fail" because of the DSCP.=

>>>>=20
>>>> This is an interesting and worrying result indeed, but it's in no
>>>> way specific to WebRTC. I think a generic RFC ("Happy Eyeballs with
>>>> Differentiated
>>> Services"?)
>>>> would be the way to go.
>>>=20
>>> Well, if you use the DSCP within your own administrative domain for
>>> DiffServ, you should be safe=E2=80=A6. so I guess this problem mostly ma=
tters
>>> for usage across domains, or even end-to-end.
>>> =3D> It seems to me that it=E2=80=99s broader than WebRTC in exactly the=
 same
>>> way as
>>> RFC7657 is broader than WebRTC.
>>=20
>> We have an immediate issue about what to add to the rtcweb-qos draft, whi=
ch is
>> IESG approved, and will go to the RFC Editor once this concern is resolve=
d.
>> While I greatly appreciate Michael's suggestion of text, I wonder whether=
 this is
>> part of a larger problem ...
>>=20
>> Cullen recently sent a message to TSVWG about initial use of bandwidth by=
 ICE
>> for STUN:
>>=20
>> https://www.ietf.org/mail-archive/web/tsvwg/current/msg14497.html
>>=20
>> That  message begins with:
>>=20
>>> When WebRTC connections start, they send a bunch of STUN packets as part=
 of
>> ICE.
>>> Typically most of these STUN packets do not actually reach a valid
>>> destination so they have 100% packet loss from the point of view of
>>> the sender. This is discussed more in
>>> draft-jennings-ice-rtcweb-timing-01
>>=20
>> This feels like part of the same problem - black-holing caused by DSCP se=
lection
>> feels like it could be dealt with as part of STUN probing, although some c=
aution is
>> in order  (e.g., if EF is used to set up audio).
>>=20
>> What's the "right way" to deal with this possibility that non-default DSC=
P usage by
>> Web RTC may cause black-holing, and which draft should specify the black-=
hole-
>> avoidance behavior?
>>=20
>> Thanks, --David
>>=20
>>> -----Original Message-----
>>> From: tsvwg [mailto:tsvwg-bounces@ietf.org] On Behalf Of Michael Welzl
>>> Sent: Friday, June 24, 2016 4:52 PM
>>> To: Brian E Carpenter
>>> Cc: tsvwg@ietf.org
>>> Subject: Re: [tsvwg] Fall-back to DSCP 0 in draft-ietf-tsvwg-rtcweb-qos ?=

>>>=20
>>> Hi,
>>>=20
>>>=20
>>>> On 24. jun. 2016, at 22.25, Brian E Carpenter
>>>> <brian.e.carpenter@gmail.com>
>>> wrote:
>>>>=20
>>>>> We believe that draft-ietf-tsvwg-rtcweb-qos should say: if setting
>>>>> the DSCP
>>> values as proposed here consistently produces packet drops, senders
>>> should fall back to DSCP 0. WebRTC shouldn't "fail" because of the DSCP.=

>>>>=20
>>>> This is an interesting and worrying result indeed, but it's in no
>>>> way specific to WebRTC. I think a generic RFC ("Happy Eyeballs with
>>>> Differentiated
>>> Services"?)
>>>> would be the way to go.
>>>=20
>>> Well, if you use the DSCP within your own administrative domain for
>>> DiffServ, you should be safe=E2=80=A6. so I guess this problem mostly ma=
tters
>>> for usage across domains, or even end-to-end.
>>> =3D> It seems to me that it=E2=80=99s broader than WebRTC in exactly the=
 same
>>> way as
>>> RFC7657 is broader than WebRTC.
>>>=20
>>> As a side note, I think the term =E2=80=9CHappy Eyeballs=E2=80=9D is mis=
leading here.
>>> This term, to me, indicates parallel usage of multiple packet types to
>>> see which one(s)
>>> succeed(s) and then try to make the best choice.
>>> This isn=E2=80=99t what you would do here - we=E2=80=99re talking about a=
 rare failure
>>> situation. It=E2=80=99s really only something I=E2=80=99d suggest as a f=
all-back, upon
>>> seeing communication failing entirely, over a longer period (perhaps in l=
ine with
>> a circuit-breaker).
>>>=20
>>>=20
>>>> But (as with a lot of recent discussion about IPv6 extension headers
>>>> causing packet drops) we really need to understand why
>>> this
>>>> happens. Is it traffic policing that fails to reclassify and re-mark
>>>> packets, misguided firewalls, or what?
>>>=20
>>> I have no clue=E2=80=A6 but as my previous email said, we have indicatio=
ns
>>> that this behavior occurred close to the senders (most of them being
>>> in people=E2=80=99s homes), hence it=E2=80=99s more likely that this was=
 done by a
>>> middlebox or an edge router than by a core router.
>>> That=E2=80=99s as much as we know...
>>>=20
>>> Cheers,
>>> Michael
>>>=20
>>>=20
>>>>=20
>>>> Regards
>>>>  Brian
>>>>=20
>>>>> On 25/06/2016 00:17, Michael Welzl wrote:
>>>>> Dear all,
>>>>>=20
>>>>> We recently did DSCP related measurements that lead us to propose a
>>>>> change
>>> to draft-ietf-tsvwg-rtcweb-qos.
>>>>> These measurements are reported in this paper:
>>>>> http://heim.ifi.uio.no/~michawe/research/publications/anrw16-
>>> measurement.pdf
>>>>> ... which will be presented at the ANRW workshop, the Saturday
>>>>> before the
>>> IETF meeting.
>>>>>=20
>>>>> The paper reports on tests between people's homes and 3 servers
>>>>> that we
>>> ran - one in Oregon (amazon cloud) and two in Norway. This gave us 185
>>> distinct paths (IP address pairs). The test was a TCP handshake, done
>>> with raw sockets, and we played around with values in the IP header.
>>> Meanwhile, Runa (main
>>> author) did some more tests from various public places during a trip
>>> from Norway to India, essentially confirming the finding we discuss
>>> here; we'll ask for a slot to report about both the data in the ANRW
>>> paper and the newer data in the MAPRG session.
>>>>>=20
>>>>> The number of data points isn't exactly huge in both cases, but
>>>>> still relevant,
>>> we believe: it is about a persistent failure that we saw on different
>>> paths, and so we think it's probably reasonable to take it into
>>> account (given that a fix shouldn't be very complicated either).
>>>>>=20
>>>>> We used, as input, DSCP values from the table in draft-ietf-tsvwg-rtcw=
eb-
>> qos.
>>>>> Here are some sentences copy + pasted from our paper:
>>>>>=20
>>>>> ***
>>>>> "To minimize the chance that congestion-based drops make us believe
>>>>> in a
>>> failure
>>>>> to communicate using certain values in the IP header, we re-tried
>>>>> failed
>>> packet
>>>>> exchanges up to three times, and we sent an ICMP packet just ahead
>>>>> of every measurement packet. We only assumed a communication
>>>>> failure when the
>>> test
>>>>> failed three times and the ICMP packet succeeded."
>>>>>=20
>>>>> "Considering table 1 in [6], we assumed the flow type =E2=80=9CInterac=
tive
>>>>> Video with
>>> or
>>>>> without Audio=E2=80=9D with application priorities =E2=80=9CVery low=E2=
=80=9D (DSCP value
>>>>> CS1 (8)),
>>> =E2=80=9CLow=E2=80=9D
>>>>> (DF (0)) or =E2=80=9CMedium=E2=80=9D (AF42 (36)), and flow type =E2=80=
=9CAudio=E2=80=9D with
>>>>> application
>>> prioritiy
>>>>> =E2=80=9CHigh=E2=80=9D (EF PHB (46))."
>>>>>=20
>>>>> "We did see consistent packet drops too: on 23 paths for DSCP value
>>>>> 8, 21
>>> paths
>>>>> for 36, 19 for 46."
>>>>> ***
>>>>>=20
>>>>> We also saw DSCP value changes, but nothing very alarming there -
>>>>> in one
>>> case, AF42 was changed to AF41, which is even nice  :)     however, this=

>> consistent
>>> drop behavior that appears only when using a DSCP value !=3D 0 isn't goo=
d at all.
>>> Now these things could in principle have happened close to our 3
>>> servers, meaning there were only few devices that did it, but in fact
>>> this measurement was a part of a larger measurement campaign in which
>>> we also checked where such packet drops typically happened. The result
>>> was: most such drops seem to be close to the client, which may
>>> indicate that there were in fact multiple such boxes.
>>>>>=20
>>>>> We believe that draft-ietf-tsvwg-rtcweb-qos should say: if setting
>>>>> the DSCP
>>> values as proposed here consistently produces packet drops, senders
>>> should fall back to DSCP 0. WebRTC shouldn't "fail" because of the DSCP.=

>>>>>=20
>>>>> Cheers,
>>>>> Michael
>=20


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Hi!

Some notes from my first reading of this document:

* Abstract
"This SDP examples provided in this document is still a work in
   progress, but it aims to align closest to the evolving standards
   work.
=E2=80=9C
s/This/The/
s/closest/closely/   (maybe :-) )

* 1. Introduction
"For this purpose, SDP is used to
   construct [RFC3264 <https://tools.ietf.org/html/rfc3264>] =
Offers/Answers for describing (media and non-
   media) streams as appropriate for the recipients of the session
   description to participate in the session.
=E2=80=9C
I am hesitant to separate =E2=80=9Cmedia and non-media=E2=80=9D so early =
in this document.
If we refer to the definitions in RFC 3264 there=E2=80=99s only =E2=80=9Cm=
edia stream=E2=80=9D and 4566 seems=20
to include everything - not just multimedia - in =E2=80=9Cmedia =
stream=E2=80=9D.

Do we have a reference for a non-media stream?

* Section 3
"[WebRTC <>] proposes JavaScript application to fully specify and =
control
   the signaling plane of a multimedia session as described in the JSEP
   specification [I-D.ietf-rtcweb-jsep =
<https://tools.ietf.org/html/draft-ietf-rtcweb-sdp-02#ref-I-D.ietf-rtcweb-=
jsep>]. =E2=80=9C

s/application/applications/

Maybe add a third bullet describing SDP=E2=80=99s function to bind the =
SDP
to the actual media - the ice-pwd=E2=80=99s. There=E2=80=99s a =
transitive trust
there which we could mention in the bullets.

* Section 4
Do we have a definition of a "WebRTC session=E2=80=9D somewhere?

"The Offer/Answer [RFC3264 <https://tools.ietf.org/html/rfc3264>] model =
specifies rule for the bilateral
   exchange of Session Description Protocol (SDP) messages for creation
   of multimedia streams.
=E2=80=9C
s/rule/rules/
I would add =E2=80=9Cbidirectional=E2=80=9D in the front of =
=E2=80=9Cmultimedia streams=E2=80=9D

"It defines protocol with involved
   participants exchanging desired session characteristics from each
   others perspective constructed as SDP to negotiate the session
   between them.=E2=80=9D
This sentence is hard to parse and need a rewrite. Maybe something along

=E2=80=9CIt defines a protocol which enables participants of a desired =
session
to exchange session characteristics, based on a SDP constructed
from their own perspective=E2=80=9D

I would NOT use the word negotiate here. It=E2=80=99s not really =
negotation,
it=E2=80=99s an offer/answer exchange=E2=80=A6 It=E2=80=99s really hard =
to describe when I=E2=80=99m
trying to teach people SDP, but I found that the word =E2=80=9Cnegotiate=E2=
=80=9D=20
puts them in the wrong place. Multiple offer/answer exchanges can be
seen as a negotiation though.

"If the session is accepted the Offer/Answer model
   guarantees a common view of the multimedia session between the
   participants.=E2=80=9D
After many years of working with SIP and SDP offer/exchanges I=20
would avoid the word =E2=80=9Cguarantees=E2=80=9D in this very sentence. =
;-)

"At any time, either participant MAY generate a new SDP offer that
   updates the session in progress.=E2=80=9D
Since this is the brief overview aimed to educate SDP offer/answer,
I would rephrase this to:
=E2=80=9CAt any time either participant MAY initiate a new offer/answer
exchange to update a session in progress by generating a new SDP =
offer.=E2=80=9D

"With in the context of WebRTC, the Offer/Answer model defines the
   state-machinery for WebRTC peers to negotiate session descriptions
   between them during the initial setup stages as well as for eventual
   session updates.=E2=80=9D
There=E2=80=99s the word =E2=80=9Cnegotiate=E2=80=9D again. And you =
don=E2=80=99t negotiate SDPs,
you negotiate "webrtc sessions=E2=80=9D or =E2=80=9Csessions=E2=80=9D.=20=


5. WebRTC SDP examples
The term =E2=80=9Cnon-media=E2=80=9D is used here again.

"Supports NAT traversal using ICE mechanism=E2=80=9D
I don=E2=80=99t agree that ICE is there to support NAT traversal. It=E2=80=
=99s a=20
way to describe it that leads to a lot of misunderstandings. The TURN
server supports NAT traversal, ICE is a discovery protocol to find the
optimal stream path. ICE is also the way to handle dual stack networks.
Maybe rephrasing it would be a good idea to avoid
future misunderstandings:
=E2=80=9CSupports optimal network path discovery and NAT traversal using =
ICE and TURN mechanisms=E2=80=9D

Section 5.1:
s/Eventhough/Even though/
s/an WebRTC/a WebRTC/


I=E2=80=99ve only glanced over the actual examples, it will need much =
more time. Here=E2=80=99s
a few quick comments:

- a=3Dinactive in 5.2.4 has a reference to RFC 3264. I think a proper =
reference
  would be RFC 4566 SDP
- 5.2.5: DTMF is no longer RFC 2833 - it is 4733 now.


General:
I would like to see some examples using the current IP protocol, not the =
legacy IPv4,
as well as dual stack examples.

- An IPv6 only web browser offering an IPv4 through a relay ice =
candidate
- A dual stack client
- A full IPv6 exchange - maybe switch one of the examples using IPv4 to =
IPv6
  just to show the syntax of address and port number

Regards,
/Olle

=20


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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"">Hi!<div class=3D""><br class=3D""></div><div class=3D"">Some =
notes from my first reading of this document:</div><div class=3D""><br =
class=3D""></div><div class=3D"">* Abstract</div><div class=3D"">"<span =
style=3D"font-size: 13.310184478759766px;" class=3D"">This SDP examples =
provided in this document is still a work in</span></div><pre =
style=3D"font-size: 13.310184478759766px; margin-top: 0px; =
margin-bottom: 0px;" class=3D"">   progress, but it aims to align =
closest to the evolving standards
   work.</pre><div class=3D"">=E2=80=9C</div><div =
class=3D"">s/This/The/</div><div class=3D"">s/closest/closely/ &nbsp; =
(maybe :-) )</div><div class=3D""><br class=3D""></div><div class=3D"">* =
1. Introduction</div><div class=3D"">"<span style=3D"font-size: =
13.310184478759766px;" class=3D"">For this purpose, SDP is used =
to</span></div><pre class=3D"newpage" style=3D"font-size: =
13.310184478759766px; margin-top: 0px; margin-bottom: 0px; =
page-break-before: always;">   construct [<a =
href=3D"https://tools.ietf.org/html/rfc3264" title=3D"&quot;An =
Offer/Answer Model with Session Description Protocol (SDP)&quot;" =
class=3D"">RFC3264</a>] Offers/Answers for describing (media and non-
   media) streams as appropriate for the recipients of the session
   description to participate in the session.</pre><div =
class=3D"">=E2=80=9C</div><div class=3D"">I am hesitant to separate =
=E2=80=9Cmedia and non-media=E2=80=9D so early in this =
document.</div><div class=3D"">If we refer to the definitions in RFC =
3264 there=E2=80=99s only =E2=80=9Cmedia stream=E2=80=9D and 4566 =
seems&nbsp;</div><div class=3D"">to include everything - not just =
multimedia - in =E2=80=9Cmedia stream=E2=80=9D.</div><div class=3D""><br =
class=3D""></div><div class=3D"">Do we have a reference for a non-media =
stream?</div><div class=3D""><br class=3D""></div><div class=3D"">* =
Section 3</div><div class=3D"">"<span style=3D"font-size: =
13.310184478759766px;" class=3D"">[</span><a name=3D"ref-WebRTC" =
id=3D"ref-WebRTC" style=3D"font-size: 13.310184478759766px;" =
class=3D"">WebRTC</a><span style=3D"font-size: 13.310184478759766px;" =
class=3D"">] proposes JavaScript application to fully specify and =
control</span></div><pre class=3D"newpage" style=3D"font-size: =
13.310184478759766px; margin-top: 0px; margin-bottom: 0px; =
page-break-before: always;">   the signaling plane of a multimedia =
session as described in the JSEP
   specification [<a =
href=3D"https://tools.ietf.org/html/draft-ietf-rtcweb-sdp-02#ref-I-D.ietf-=
rtcweb-jsep" class=3D"">I-D.ietf-rtcweb-jsep</a>]. =E2=80=9C</pre><pre =
class=3D"newpage" style=3D"font-size: 13.310184478759766px; margin-top: =
0px; margin-bottom: 0px; page-break-before: always;"><br =
class=3D""></pre><pre class=3D"newpage" style=3D"font-size: =
13.310184478759766px; margin-top: 0px; margin-bottom: 0px; =
page-break-before: always;">s/application/applications/</pre><pre =
class=3D"newpage" style=3D"font-size: 13.310184478759766px; margin-top: =
0px; margin-bottom: 0px; page-break-before: always;"><br =
class=3D""></pre><pre class=3D"newpage" style=3D"font-size: =
13.310184478759766px; margin-top: 0px; margin-bottom: 0px; =
page-break-before: always;">Maybe add a third bullet describing SDP=E2=80=99=
s function to bind the SDP</pre><pre class=3D"newpage" style=3D"font-size:=
 13.310184478759766px; margin-top: 0px; margin-bottom: 0px; =
page-break-before: always;">to the actual media - the ice-pwd=E2=80=99s. =
There=E2=80=99s a transitive trust</pre><pre class=3D"newpage" =
style=3D"font-size: 13.310184478759766px; margin-top: 0px; =
margin-bottom: 0px; page-break-before: always;">there which we could =
mention in the bullets.</pre><pre class=3D"newpage" style=3D"font-size: =
13.310184478759766px; margin-top: 0px; margin-bottom: 0px; =
page-break-before: always;"><br class=3D""></pre><pre class=3D"newpage" =
style=3D"font-size: 13.310184478759766px; margin-top: 0px; =
margin-bottom: 0px; page-break-before: always;">* Section 4</pre><pre =
class=3D"newpage" style=3D"margin-top: 0px; margin-bottom: 0px; =
page-break-before: always;"><font size=3D"3" class=3D"">Do we have a =
definition of a "WebRTC session=E2=80=9D somewhere?</font></pre><pre =
class=3D"newpage" style=3D"margin-top: 0px; margin-bottom: 0px; =
page-break-before: always;"><font size=3D"3" class=3D""><br =
class=3D""></font></pre><pre class=3D"newpage" style=3D"margin-top: 0px; =
margin-bottom: 0px; page-break-before: always;"><font size=3D"3" =
class=3D"">"</font><span style=3D"font-size: 13.310184478759766px;" =
class=3D"">The Offer/Answer [</span><a =
href=3D"https://tools.ietf.org/html/rfc3264" title=3D"&quot;An =
Offer/Answer Model with Session Description Protocol (SDP)&quot;" =
style=3D"font-size: 13.310184478759766px;" class=3D"">RFC3264</a><span =
style=3D"font-size: 13.310184478759766px;" class=3D"">] model specifies =
rule for the bilateral</span></pre><pre class=3D"newpage" =
style=3D"font-size: 13.310184478759766px; margin-top: 0px; =
margin-bottom: 0px; page-break-before: always;">   exchange of Session =
Description Protocol (SDP) messages for creation
   of multimedia streams.</pre><div class=3D"">=E2=80=9C</div><div =
class=3D"">s/rule/rules/</div><div class=3D"">I would add =
=E2=80=9Cbidirectional=E2=80=9D in the front of =E2=80=9Cmultimedia =
streams=E2=80=9D</div><div class=3D""><br class=3D""></div><div =
class=3D"">"<span style=3D"font-size: 13.310184478759766px;" class=3D"">It=
 defines protocol with involved</span></div><pre class=3D"newpage" =
style=3D"font-size: 13.310184478759766px; margin-top: 0px; =
margin-bottom: 0px; page-break-before: always;">   participants =
exchanging desired session characteristics from each
   others perspective constructed as SDP to negotiate the session
   between them.=E2=80=9D</pre><pre class=3D"newpage" style=3D"font-size: =
13.310184478759766px; margin-top: 0px; margin-bottom: 0px; =
page-break-before: always;">This sentence is hard to parse and need a =
rewrite. Maybe something along</pre><pre class=3D"newpage" =
style=3D"font-size: 13.310184478759766px; margin-top: 0px; =
margin-bottom: 0px; page-break-before: always;"><br class=3D""></pre><pre =
class=3D"newpage" style=3D"font-size: 13.310184478759766px; margin-top: =
0px; margin-bottom: 0px; page-break-before: always;">=E2=80=9CIt defines =
a protocol which enables participants of a desired session</pre><pre =
class=3D"newpage" style=3D"font-size: 13.310184478759766px; margin-top: =
0px; margin-bottom: 0px; page-break-before: always;">to exchange session =
characteristics, based on a SDP constructed</pre><pre class=3D"newpage" =
style=3D"font-size: 13.310184478759766px; margin-top: 0px; =
margin-bottom: 0px; page-break-before: always;">from their own =
perspective=E2=80=9D</pre><pre class=3D"newpage" style=3D"font-size: =
13.310184478759766px; margin-top: 0px; margin-bottom: 0px; =
page-break-before: always;"><br class=3D""></pre><pre class=3D"newpage" =
style=3D"font-size: 13.310184478759766px; margin-top: 0px; =
margin-bottom: 0px; page-break-before: always;">I would NOT use the word =
negotiate here. It=E2=80=99s not really negotation,</pre><pre =
class=3D"newpage" style=3D"font-size: 13.310184478759766px; margin-top: =
0px; margin-bottom: 0px; page-break-before: always;">it=E2=80=99s an =
offer/answer exchange=E2=80=A6 It=E2=80=99s really hard to describe when =
I=E2=80=99m</pre><pre class=3D"newpage" style=3D"font-size: =
13.310184478759766px; margin-top: 0px; margin-bottom: 0px; =
page-break-before: always;">trying to teach people SDP, but I found that =
the word =E2=80=9Cnegotiate=E2=80=9D </pre><pre class=3D"newpage" =
style=3D"font-size: 13.310184478759766px; margin-top: 0px; =
margin-bottom: 0px; page-break-before: always;">puts them in the wrong =
place. Multiple offer/answer exchanges can be</pre><pre class=3D"newpage" =
style=3D"font-size: 13.310184478759766px; margin-top: 0px; =
margin-bottom: 0px; page-break-before: always;">seen as a negotiation =
though.</pre><pre class=3D"newpage" style=3D"font-size: =
13.310184478759766px; margin-top: 0px; margin-bottom: 0px; =
page-break-before: always;"><br class=3D""></pre><pre class=3D"newpage" =
style=3D"font-size: 13.310184478759766px; margin-top: 0px; =
margin-bottom: 0px; page-break-before: always;">"<span style=3D"font-size:=
 13.310184478759766px;" class=3D"">If the session is accepted the =
Offer/Answer model</span></pre><pre class=3D"newpage" style=3D"font-size: =
13.310184478759766px; margin-top: 0px; margin-bottom: 0px; =
page-break-before: always;">   guarantees a common view of the =
multimedia session between the
   participants.=E2=80=9D</pre><pre class=3D"newpage" style=3D"font-size: =
13.310184478759766px; margin-top: 0px; margin-bottom: 0px; =
page-break-before: always;">After many years of working with SIP and SDP =
offer/exchanges I </pre><pre class=3D"newpage" style=3D"font-size: =
13.310184478759766px; margin-top: 0px; margin-bottom: 0px; =
page-break-before: always;">would avoid the word =E2=80=9Cguarantees=E2=80=
=9D in this very sentence. ;-)</pre><pre class=3D"newpage" =
style=3D"font-size: 13.310184478759766px; margin-top: 0px; =
margin-bottom: 0px; page-break-before: always;"><br class=3D""></pre><pre =
class=3D"newpage" style=3D"font-size: 13.310184478759766px; margin-top: =
0px; margin-bottom: 0px; page-break-before: always;">"<span =
style=3D"font-size: 13.310184478759766px;" class=3D"">At any time, =
either participant MAY generate a new SDP offer that</span></pre><pre =
class=3D"newpage" style=3D"font-size: 13.310184478759766px; margin-top: =
0px; margin-bottom: 0px; page-break-before: always;">   updates the =
session in progress.=E2=80=9D</pre><pre class=3D"newpage" =
style=3D"font-size: 13.310184478759766px; margin-top: 0px; =
margin-bottom: 0px; page-break-before: always;">Since this is the brief =
overview aimed to educate SDP offer/answer,</pre><pre class=3D"newpage" =
style=3D"font-size: 13.310184478759766px; margin-top: 0px; =
margin-bottom: 0px; page-break-before: always;">I would rephrase this =
to:</pre><pre class=3D"newpage" style=3D"font-size: =
13.310184478759766px; margin-top: 0px; margin-bottom: 0px; =
page-break-before: always;">=E2=80=9CAt any time either participant MAY =
initiate a new offer/answer</pre><pre class=3D"newpage" =
style=3D"font-size: 13.310184478759766px; margin-top: 0px; =
margin-bottom: 0px; page-break-before: always;">exchange to update a =
session in progress by generating a new SDP offer.=E2=80=9D</pre><pre =
class=3D"newpage" style=3D"font-size: 13.310184478759766px; margin-top: =
0px; margin-bottom: 0px; page-break-before: always;"><br =
class=3D""></pre><pre class=3D"newpage" style=3D"font-size: =
13.310184478759766px; margin-top: 0px; margin-bottom: 0px; =
page-break-before: always;">"<span style=3D"font-size: =
13.310184478759766px;" class=3D"">With in the context of WebRTC, the =
Offer/Answer model defines the</span></pre><pre class=3D"newpage" =
style=3D"font-size: 13.310184478759766px; margin-top: 0px; =
margin-bottom: 0px; page-break-before: always;">   state-machinery for =
WebRTC peers to negotiate session descriptions
   between them during the initial setup stages as well as for eventual
   session updates.=E2=80=9D</pre><pre class=3D"newpage" =
style=3D"font-size: 13.310184478759766px; margin-top: 0px; =
margin-bottom: 0px; page-break-before: always;">There=E2=80=99s the word =
=E2=80=9Cnegotiate=E2=80=9D again. And you don=E2=80=99t negotiate =
SDPs,</pre><pre class=3D"newpage" style=3D"font-size: =
13.310184478759766px; margin-top: 0px; margin-bottom: 0px; =
page-break-before: always;">you negotiate "webrtc sessions=E2=80=9D or =
=E2=80=9Csessions=E2=80=9D. </pre><pre class=3D"newpage" =
style=3D"font-size: 13.310184478759766px; margin-top: 0px; =
margin-bottom: 0px; page-break-before: always;"><br class=3D""></pre><pre =
class=3D"newpage" style=3D"font-size: 13.310184478759766px; margin-top: =
0px; margin-bottom: 0px; page-break-before: always;">5. WebRTC SDP =
examples</pre><pre class=3D"newpage" style=3D"font-size: =
13.310184478759766px; margin-top: 0px; margin-bottom: 0px; =
page-break-before: always;">The term =E2=80=9Cnon-media=E2=80=9D is used =
here again.</pre><pre class=3D"newpage" style=3D"font-size: =
13.310184478759766px; margin-top: 0px; margin-bottom: 0px; =
page-break-before: always;"><br class=3D""></pre><pre class=3D"newpage" =
style=3D"margin-top: 0px; margin-bottom: 0px; page-break-before: =
always;"><font size=3D"3" class=3D"">"</font><span style=3D"font-size: =
13.310184478759766px;" class=3D"">Supports NAT traversal using ICE =
mechanism</span><font size=3D"3" class=3D"">=E2=80=9D</font></pre><pre =
class=3D"newpage" style=3D"margin-top: 0px; margin-bottom: 0px; =
page-break-before: always;"><font size=3D"3" class=3D"">I don=E2=80=99t =
agree that ICE is there to support NAT traversal. It=E2=80=99s a =
</font></pre><pre class=3D"newpage" style=3D"margin-top: 0px; =
margin-bottom: 0px; page-break-before: always;"><font size=3D"3" =
class=3D"">way to describe it that leads to a lot of misunderstandings. =
The TURN</font></pre><pre class=3D"newpage" style=3D"margin-top: 0px; =
margin-bottom: 0px; page-break-before: always;"><font size=3D"3" =
class=3D"">server supports NAT traversal, ICE is a discovery protocol to =
find the</font></pre><pre class=3D"newpage" style=3D"margin-top: 0px; =
margin-bottom: 0px; page-break-before: always;"><font size=3D"3" =
class=3D"">optimal stream path. ICE is also the way to handle dual stack =
networks.</font></pre><pre class=3D"newpage" style=3D"margin-top: 0px; =
margin-bottom: 0px; page-break-before: always;"><font size=3D"3" =
class=3D"">Maybe rephrasing it would be a good idea to =
avoid</font></pre><pre class=3D"newpage" style=3D"margin-top: 0px; =
margin-bottom: 0px; page-break-before: always;"><font size=3D"3" =
class=3D"">future misunderstandings:</font></pre><pre class=3D"newpage" =
style=3D"margin-top: 0px; margin-bottom: 0px; page-break-before: =
always;"><font size=3D"3" class=3D"">=E2=80=9CSupports optimal network =
path discovery and NAT traversal using ICE and TURN =
mechanisms=E2=80=9D</font></pre><pre class=3D"newpage" =
style=3D"margin-top: 0px; margin-bottom: 0px; page-break-before: =
always;"><font size=3D"3" class=3D""><br class=3D""></font></pre><pre =
class=3D"newpage" style=3D"margin-top: 0px; margin-bottom: 0px; =
page-break-before: always;"><font size=3D"3" class=3D"">Section =
5.1:</font></pre><pre class=3D"newpage" style=3D"margin-top: 0px; =
margin-bottom: 0px; page-break-before: always;"><font size=3D"3" =
class=3D"">s/</font><span style=3D"font-size: 13.310184478759766px;" =
class=3D"">Eventhough/Even though/</span></pre><pre class=3D"newpage" =
style=3D"margin-top: 0px; margin-bottom: 0px; page-break-before: =
always;"><font size=3D"3" class=3D"">s/</font><span style=3D"font-size: =
13.310184478759766px;" class=3D"">an WebRTC/a WebRTC/</span></pre><pre =
class=3D"newpage" style=3D"margin-top: 0px; margin-bottom: 0px; =
page-break-before: always;"><span style=3D"font-size: =
13.310184478759766px;" class=3D""><br class=3D""></span></pre><pre =
class=3D"newpage" style=3D"margin-top: 0px; margin-bottom: 0px; =
page-break-before: always;"><span style=3D"font-size: =
13.310184478759766px;" class=3D""><br class=3D""></span></pre><pre =
class=3D"newpage" style=3D"margin-top: 0px; margin-bottom: 0px; =
page-break-before: always;"><font size=3D"3" class=3D"">I=E2=80=99ve =
only glanced over the actual examples, it will need much more time. =
Here=E2=80=99s</font></pre><pre class=3D"newpage" style=3D"margin-top: =
0px; margin-bottom: 0px; page-break-before: always;"><font size=3D"3" =
class=3D"">a few quick comments:</font></pre><pre class=3D"newpage" =
style=3D"margin-top: 0px; margin-bottom: 0px; page-break-before: =
always;"><font size=3D"3" class=3D""><br class=3D""></font></pre><pre =
class=3D"newpage" style=3D"margin-top: 0px; margin-bottom: 0px; =
page-break-before: always;"><font size=3D"3" class=3D"">- a=3Dinactive =
in 5.2.4 has a reference to RFC 3264. I think a proper =
reference</font></pre><pre class=3D"newpage" style=3D"margin-top: 0px; =
margin-bottom: 0px; page-break-before: always;"><font size=3D"3" =
class=3D"">  would be RFC 4566 SDP</font></pre><pre class=3D"newpage" =
style=3D"margin-top: 0px; margin-bottom: 0px; page-break-before: =
always;"><font size=3D"3" class=3D"">- 5.2.5: DTMF is no longer RFC 2833 =
- it is 4733 now.</font></pre><pre class=3D"newpage" style=3D"margin-top: =
0px; margin-bottom: 0px; page-break-before: always;"><font size=3D"3" =
class=3D""><br class=3D""></font></pre><pre class=3D"newpage" =
style=3D"margin-top: 0px; margin-bottom: 0px; page-break-before: =
always;"><font size=3D"3" class=3D""><br class=3D""></font></pre><pre =
class=3D"newpage" style=3D"margin-top: 0px; margin-bottom: 0px; =
page-break-before: always;"><font size=3D"3" =
class=3D"">General:</font></pre><pre class=3D"newpage" =
style=3D"margin-top: 0px; margin-bottom: 0px; page-break-before: =
always;"><font size=3D"3" class=3D"">I would like to see some examples =
using the current IP protocol, not the legacy IPv4,</font></pre><pre =
class=3D"newpage" style=3D"margin-top: 0px; margin-bottom: 0px; =
page-break-before: always;"><font size=3D"3" class=3D"">as well as dual =
stack examples.</font></pre><pre class=3D"newpage" style=3D"margin-top: =
0px; margin-bottom: 0px; page-break-before: always;"><font size=3D"3" =
class=3D""><br class=3D""></font></pre><pre class=3D"newpage" =
style=3D"margin-top: 0px; margin-bottom: 0px; page-break-before: =
always;"><font size=3D"3" class=3D"">- An IPv6 only web browser offering =
an IPv4 through a relay ice candidate</font></pre><pre class=3D"newpage" =
style=3D"margin-top: 0px; margin-bottom: 0px; page-break-before: =
always;"><font size=3D"3" class=3D"">- A dual stack =
client</font></pre><pre class=3D"newpage" style=3D"margin-top: 0px; =
margin-bottom: 0px; page-break-before: always;">- A full IPv6 exchange - =
maybe switch one of the examples using IPv4 to IPv6</pre><pre =
class=3D"newpage" style=3D"margin-top: 0px; margin-bottom: 0px; =
page-break-before: always;">  just to show the syntax of address and =
port number</pre><pre class=3D"newpage" style=3D"margin-top: 0px; =
margin-bottom: 0px; page-break-before: always;"><br class=3D""></pre><pre =
class=3D"newpage" style=3D"margin-top: 0px; margin-bottom: 0px; =
page-break-before: always;">Regards,</pre><pre class=3D"newpage" =
style=3D"margin-top: 0px; margin-bottom: 0px; page-break-before: =
always;">/Olle</pre><pre class=3D"newpage" style=3D"margin-top: 0px; =
margin-bottom: 0px; page-break-before: always;"><br class=3D""></pre><pre =
class=3D"newpage" style=3D"margin-top: 0px; margin-bottom: 0px; =
page-break-before: always;"><font size=3D"3" class=3D""> =
</font></pre><div class=3D""><br class=3D""></div></body></html>=

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> On 12 Jul 2016, at 09:55, Olle E. Johansson <oej@edvina.net> wrote:
>=20
>=20
> General:
> I would like to see some examples using the current IP protocol, not =
the legacy IPv4,
> as well as dual stack examples.
>=20
> - An IPv6 only web browser offering an IPv4 through a relay ice =
candidate
> - A dual stack client
> - A full IPv6 exchange - maybe switch one of the examples using IPv4 =
to IPv6
>   just to show the syntax of address and port number
>=20
In addition I think an offer/answer with different packetization times, =
like
alaw/30 ms, Opus/60 ms or something similar. Yes, I=E2=80=99m going down =
a rat hole,
but it will come up and we=E2=80=99d better document a proposal for how =
to handle this.

If one wants to be messy you can add DTMF or CNG to an offer/answer =
using
alaw at 8000 hz and Opus at 48000 making it necessary to have multiple =
DTMF
and CNG for that media stream. But that may be out of scope for this =
particular
document, even though it may add clarity.

/O=

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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 12 Jul 2016, at 09:55, Olle E. Johansson &lt;<a =
href=3D"mailto:oej@edvina.net" class=3D"">oej@edvina.net</a>&gt; =
wrote:</div><br class=3D"Apple-interchange-newline"><div class=3D""><pre =
class=3D"newpage" style=3D"font-size: 14px; font-style: normal; =
font-variant-caps: normal; font-weight: normal; letter-spacing: normal; =
orphans: auto; text-align: start; text-indent: 0px; text-transform: =
none; widows: auto; word-spacing: 0px; -webkit-text-stroke-width: 0px; =
margin-top: 0px; margin-bottom: 0px; page-break-before: always;"><font =
size=3D"3" class=3D""><br =
class=3D"Apple-interchange-newline">General:</font></pre><pre =
class=3D"newpage" style=3D"font-size: 14px; font-style: normal; =
font-variant-caps: normal; font-weight: normal; letter-spacing: normal; =
orphans: auto; text-align: start; text-indent: 0px; text-transform: =
none; widows: auto; word-spacing: 0px; -webkit-text-stroke-width: 0px; =
margin-top: 0px; margin-bottom: 0px; page-break-before: always;"><font =
size=3D"3" class=3D"">I would like to see some examples using the =
current IP protocol, not the legacy IPv4,</font></pre><pre =
class=3D"newpage" style=3D"font-size: 14px; font-style: normal; =
font-variant-caps: normal; font-weight: normal; letter-spacing: normal; =
orphans: auto; text-align: start; text-indent: 0px; text-transform: =
none; widows: auto; word-spacing: 0px; -webkit-text-stroke-width: 0px; =
margin-top: 0px; margin-bottom: 0px; page-break-before: always;"><font =
size=3D"3" class=3D"">as well as dual stack examples.</font></pre><pre =
class=3D"newpage" style=3D"font-size: 14px; font-style: normal; =
font-variant-caps: normal; font-weight: normal; letter-spacing: normal; =
orphans: auto; text-align: start; text-indent: 0px; text-transform: =
none; widows: auto; word-spacing: 0px; -webkit-text-stroke-width: 0px; =
margin-top: 0px; margin-bottom: 0px; page-break-before: always;"><font =
size=3D"3" class=3D""><br class=3D""></font></pre><pre class=3D"newpage" =
style=3D"font-size: 14px; font-style: normal; font-variant-caps: normal; =
font-weight: normal; letter-spacing: normal; orphans: auto; text-align: =
start; text-indent: 0px; text-transform: none; widows: auto; =
word-spacing: 0px; -webkit-text-stroke-width: 0px; margin-top: 0px; =
margin-bottom: 0px; page-break-before: always;"><font size=3D"3" =
class=3D"">- An IPv6 only web browser offering an IPv4 through a relay =
ice candidate</font></pre><pre class=3D"newpage" style=3D"font-size: =
14px; font-style: normal; font-variant-caps: normal; font-weight: =
normal; letter-spacing: normal; orphans: auto; text-align: start; =
text-indent: 0px; text-transform: none; widows: auto; word-spacing: 0px; =
-webkit-text-stroke-width: 0px; margin-top: 0px; margin-bottom: 0px; =
page-break-before: always;"><font size=3D"3" class=3D"">- A dual stack =
client</font></pre><pre class=3D"newpage" style=3D"font-size: 14px; =
font-style: normal; font-variant-caps: normal; font-weight: normal; =
letter-spacing: normal; orphans: auto; text-align: start; text-indent: =
0px; text-transform: none; widows: auto; word-spacing: 0px; =
-webkit-text-stroke-width: 0px; margin-top: 0px; margin-bottom: 0px; =
page-break-before: always;">- A full IPv6 exchange - maybe switch one of =
the examples using IPv4 to IPv6</pre><pre class=3D"newpage" =
style=3D"font-size: 14px; font-style: normal; font-variant-caps: normal; =
font-weight: normal; letter-spacing: normal; orphans: auto; text-align: =
start; text-indent: 0px; text-transform: none; widows: auto; =
word-spacing: 0px; -webkit-text-stroke-width: 0px; margin-top: 0px; =
margin-bottom: 0px; page-break-before: always;">  just to show the =
syntax of address and port number</pre><br =
class=3D"Apple-interchange-newline"></div></blockquote></div>In addition =
I think an offer/answer with different packetization times, like<div =
class=3D"">alaw/30 ms, Opus/60 ms or something similar. Yes, I=E2=80=99m =
going down a rat hole,</div><div class=3D"">but it will come up and =
we=E2=80=99d better document a proposal for how to handle =
this.</div><div class=3D""><br class=3D""></div><div class=3D"">If one =
wants to be messy you can add DTMF or CNG to an offer/answer =
using</div><div class=3D"">alaw at 8000 hz and Opus at 48000 making it =
necessary to have multiple DTMF</div><div class=3D"">and CNG for that =
media stream. But that may be out of scope for this particular</div><div =
class=3D"">document, even though it may add clarity.</div><div =
class=3D""><br class=3D""></div><div class=3D"">/O</div></body></html>=

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From: Suhas Nandakumar <suhasietf@gmail.com>
Date: Tue, 12 Jul 2016 18:15:06 +0530
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Subject: Re: [rtcweb] draft-ietf-rtcweb-sdp-02 submitted
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Hello Olle

  Thanks for the review. I will go through the other comments in a while
but wanted to share one thought on the ptime feedback

JSEP-15 says

   - If this m=3D section is for media with configurable frame sizes, e.g.
   audio, an "a=3Dmaxptime" line, indicating the smallest of the maximum
   supported frame sizes out of all codecs included above, as specified in
   [RFC4566]
   <https://tools.ietf.org/id/draft-ietf-rtcweb-jsep-15.html#RFC4566>,
   Section 6."


The current set of examples try to follow JSEP specification to the closest
in how the SDP is constructed barring few open issues on handing certain
attributes

Please let us know your thoughts  on various ptimes across codecs is
something that should go in JSEP spec SDP rules as well ?

Thanks
Suhas

On Tue, Jul 12, 2016 at 5:40 PM, Olle E. Johansson <oej@edvina.net> wrote:

>
> On 12 Jul 2016, at 09:55, Olle E. Johansson <oej@edvina.net> wrote:
>
>
> General:
>
> I would like to see some examples using the current IP protocol, not the =
legacy IPv4,
>
> as well as dual stack examples.
>
>
> - An IPv6 only web browser offering an IPv4 through a relay ice candidate
>
> - A dual stack client
>
> - A full IPv6 exchange - maybe switch one of the examples using IPv4 to I=
Pv6
>
>   just to show the syntax of address and port number
>
>
> In addition I think an offer/answer with different packetization times,
> like
> alaw/30 ms, Opus/60 ms or something similar. Yes, I=E2=80=99m going down =
a rat
> hole,
> but it will come up and we=E2=80=99d better document a proposal for how t=
o handle
> this.
>
> If one wants to be messy you can add DTMF or CNG to an offer/answer using
> alaw at 8000 hz and Opus at 48000 making it necessary to have multiple DT=
MF
> and CNG for that media stream. But that may be out of scope for this
> particular
> document, even though it may add clarity.
>
> /O
>

--001a114d6e0c742e2c05376fa3ba
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<div dir=3D"ltr">Hello Olle<div><br></div><div>=C2=A0 Thanks for the review=
. I will go through the other comments in a while but wanted to share one t=
hought on the ptime feedback</div><div><br></div><div>JSEP-15 says</div><ul=
 style=3D"color:rgb(0,0,0);font-family:verdana,helvetica,arial,sans-serif;f=
ont-size:13.3333px"><li style=3D"margin-left:2em;margin-right:2em">If this =
m=3D section is for media with configurable frame sizes, e.g. audio, an &qu=
ot;a=3Dmaxptime&quot; line, indicating the smallest of the maximum supporte=
d frame sizes out of all codecs included above, as specified in=C2=A0<a hre=
f=3D"https://tools.ietf.org/id/draft-ietf-rtcweb-jsep-15.html#RFC4566" styl=
e=3D"text-decoration:none">[RFC4566]</a>, Section 6.&quot;</li></ul><div><b=
r></div><div><font color=3D"#000000" face=3D"verdana, helvetica, arial, san=
s-serif"><span style=3D"font-size:13.3333px">The current set of examples tr=
y to follow JSEP specification to the closest in how the SDP is constructed=
 barring few open issues on handing certain attributes</span></font></div><=
div><font color=3D"#000000" face=3D"verdana, helvetica, arial, sans-serif">=
<span style=3D"font-size:13.3333px"><br></span></font></div><div><font colo=
r=3D"#000000" face=3D"verdana, helvetica, arial, sans-serif"><span style=3D=
"font-size:13.3333px">Please let us know your thoughts =C2=A0on various pti=
mes across codecs is something that should go in JSEP spec SDP rules as wel=
l ?</span></font></div><div><font color=3D"#000000" face=3D"verdana, helvet=
ica, arial, sans-serif"><span style=3D"font-size:13.3333px"><br></span></fo=
nt></div><div><font color=3D"#000000" face=3D"verdana, helvetica, arial, sa=
ns-serif"><span style=3D"font-size:13.3333px">Thanks</span></font></div><di=
v><font color=3D"#000000" face=3D"verdana, helvetica, arial, sans-serif"><s=
pan style=3D"font-size:13.3333px">Suhas</span></font></div></div><div class=
=3D"gmail_extra"><br><div class=3D"gmail_quote">On Tue, Jul 12, 2016 at 5:4=
0 PM, Olle E. Johansson <span dir=3D"ltr">&lt;<a href=3D"mailto:oej@edvina.=
net" target=3D"_blank">oej@edvina.net</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"><span class=3D""><br=
><div><blockquote type=3D"cite"><div>On 12 Jul 2016, at 09:55, Olle E. Joha=
nsson &lt;<a href=3D"mailto:oej@edvina.net" target=3D"_blank">oej@edvina.ne=
t</a>&gt; wrote:</div><br><div><pre style=3D"font-size:14px;font-style:norm=
al;font-weight:normal;letter-spacing:normal;text-align:start;text-indent:0p=
x;text-transform:none;word-spacing:0px;margin-top:0px;margin-bottom:0px"><f=
ont size=3D"3"><br>General:</font></pre><pre style=3D"font-size:14px;font-s=
tyle:normal;font-weight:normal;letter-spacing:normal;text-align:start;text-=
indent:0px;text-transform:none;word-spacing:0px;margin-top:0px;margin-botto=
m:0px"><font size=3D"3">I would like to see some examples using the current=
 IP protocol, not the legacy IPv4,</font></pre><pre style=3D"font-size:14px=
;font-style:normal;font-weight:normal;letter-spacing:normal;text-align:star=
t;text-indent:0px;text-transform:none;word-spacing:0px;margin-top:0px;margi=
n-bottom:0px"><font size=3D"3">as well as dual stack examples.</font></pre>=
<pre style=3D"font-size:14px;font-style:normal;font-weight:normal;letter-sp=
acing:normal;text-align:start;text-indent:0px;text-transform:none;word-spac=
ing:0px;margin-top:0px;margin-bottom:0px"><font size=3D"3"><br></font></pre=
><pre style=3D"font-size:14px;font-style:normal;font-weight:normal;letter-s=
pacing:normal;text-align:start;text-indent:0px;text-transform:none;word-spa=
cing:0px;margin-top:0px;margin-bottom:0px"><font size=3D"3">- An IPv6 only =
web browser offering an IPv4 through a relay ice candidate</font></pre><pre=
 style=3D"font-size:14px;font-style:normal;font-weight:normal;letter-spacin=
g:normal;text-align:start;text-indent:0px;text-transform:none;word-spacing:=
0px;margin-top:0px;margin-bottom:0px"><font size=3D"3">- A dual stack clien=
t</font></pre><pre style=3D"font-size:14px;font-style:normal;font-weight:no=
rmal;letter-spacing:normal;text-align:start;text-indent:0px;text-transform:=
none;word-spacing:0px;margin-top:0px;margin-bottom:0px">- A full IPv6 excha=
nge - maybe switch one of the examples using IPv4 to IPv6</pre><pre style=
=3D"font-size:14px;font-style:normal;font-weight:normal;letter-spacing:norm=
al;text-align:start;text-indent:0px;text-transform:none;word-spacing:0px;ma=
rgin-top:0px;margin-bottom:0px">  just to show the syntax of address and po=
rt number</pre><br></div></blockquote></div></span>In addition I think an o=
ffer/answer with different packetization times, like<div>alaw/30 ms, Opus/6=
0 ms or something similar. Yes, I=E2=80=99m going down a rat hole,</div><di=
v>but it will come up and we=E2=80=99d better document a proposal for how t=
o handle this.</div><div><br></div><div>If one wants to be messy you can ad=
d DTMF or CNG to an offer/answer using</div><div>alaw at 8000 hz and Opus a=
t 48000 making it necessary to have multiple DTMF</div><div>and CNG for tha=
t media stream. But that may be out of scope for this particular</div><div>=
document, even though it may add clarity.</div><span class=3D"HOEnZb"><font=
 color=3D"#888888"><div><br></div><div>/O</div></font></span></div></blockq=
uote></div><br></div>

--001a114d6e0c742e2c05376fa3ba--


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From: "Black, David" <david.black@emc.com>
To: "Gorry (erg)" <gorry@erg.abdn.ac.uk>
Thread-Topic: [tsvwg] Fall-back to DSCP 0 in draft-ietf-tsvwg-rtcweb-qos ?
Thread-Index: AQHRzhJbsZyuYvv6mE+PQ2kNBpP9KZ/5U/cAgAAHhwCAGfiowIAAC+UAgABFHwCAASQSAIAAEh2g
Date: Tue, 12 Jul 2016 12:55:49 +0000
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References: <CB087237-108A-44B1-8293-3498F24A2303@ifi.uio.no> <e3ebbf6c-58ab-1f70-3a5c-36a660dc73e7@gmail.com> <4D9967BC-D23D-4DB9-9ABE-9DA6B15B33A8@ifi.uio.no> <CE03DB3D7B45C245BCA0D243277949362F5D6D94@MX307CL04.corp.emc.com> <f06ecef448a84c6787acd3fda02b56ec@HE101653.emea1.cds.t-internal.com> <CE03DB3D7B45C245BCA0D243277949362F5D8376@MX307CL04.corp.emc.com> <648D8616-0AB5-4420-A829-1481B30237F4@erg.abdn.ac.uk>
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Cc: "fluffy@cisco.com" <fluffy@cisco.com>, "Ruediger.Geib@telekom.de" <Ruediger.Geib@telekom.de>, "tsvwg@ietf.org" <tsvwg@ietf.org>, "Black, David" <david.black@emc.com>, "rtcweb@ietf.org" <rtcweb@ietf.org>, "brian.e.carpenter@gmail.com" <brian.e.carpenter@gmail.com>
Subject: Re: [rtcweb] [tsvwg] Fall-back to DSCP 0 in draft-ietf-tsvwg-rtcweb-qos ?
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> On 12 Jul 2016, at 14:45, Suhas Nandakumar <suhasietf@gmail.com> =
wrote:
>=20
> Hello Olle
>=20
>   Thanks for the review. I will go through the other comments in a =
while but wanted to share one thought on the ptime feedback
>=20
> JSEP-15 says
> If this m=3D section is for media with configurable frame sizes, e.g. =
audio, an "a=3Dmaxptime" line, indicating the smallest of the maximum =
supported frame sizes out of all codecs included above, as specified in =
[RFC4566] =
<https://tools.ietf.org/id/draft-ietf-rtcweb-jsep-15.html#RFC4566>, =
Section 6.=E2=80=9D
That=E2=80=99s a good setting, but not the same. If you have codec Y =
with ptimes settable from 10 to 130 and codec Z with ptimes settable =
from 20 to 250,
the maxptime would be 130 if I understand you right.

What I am looking for is the weird situation where I want to offer codec =
Y with the default ptime of 40 and codec Z with the default ptime of 50.
This is a wormhole, since the a=3Dptime setting is kind of a mystery =
that a few brave people have been trying to solve - does it apply
to ALL codecs in a offered stream and if so - how to we handle an offer =
like above. Codec Y and codec Z may have incompatible
ptimes too, so one default ptime for the stream may not work at all.=20
>=20
> The current set of examples try to follow JSEP specification to the =
closest in how the SDP is constructed barring few open issues on handing =
certain attributes
Ok, then this issue, which is generic SDP may be out of scope for this =
document - but it will pop up at some unspecified time
in the future so I wanted to suggest that the time is now.=20
>=20
> Please let us know your thoughts  on various ptimes across codecs is =
something that should go in JSEP spec SDP rules as well ?
I need to look a bit more into JSEP for that. While waiting for me to do =
that, others may offer their opinions :-)

/O
>=20
> Thanks
> Suhas
>=20
> On Tue, Jul 12, 2016 at 5:40 PM, Olle E. Johansson <oej@edvina.net =
<mailto:oej@edvina.net>> wrote:
>=20
>> On 12 Jul 2016, at 09:55, Olle E. Johansson <oej@edvina.net =
<mailto:oej@edvina.net>> wrote:
>>=20
>>=20
>> General:
>> I would like to see some examples using the current IP protocol, not =
the legacy IPv4,
>> as well as dual stack examples.
>>=20
>> - An IPv6 only web browser offering an IPv4 through a relay ice =
candidate
>> - A dual stack client
>> - A full IPv6 exchange - maybe switch one of the examples using IPv4 =
to IPv6
>>   just to show the syntax of address and port number
>>=20
> In addition I think an offer/answer with different packetization =
times, like
> alaw/30 ms, Opus/60 ms or something similar. Yes, I=E2=80=99m going =
down a rat hole,
> but it will come up and we=E2=80=99d better document a proposal for =
how to handle this.
>=20
> If one wants to be messy you can add DTMF or CNG to an offer/answer =
using
> alaw at 8000 hz and Opus at 48000 making it necessary to have multiple =
DTMF
> and CNG for that media stream. But that may be out of scope for this =
particular
> document, even though it may add clarity.
>=20
> /O
>=20


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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 12 Jul 2016, at 14:45, Suhas Nandakumar &lt;<a =
href=3D"mailto:suhasietf@gmail.com" class=3D"">suhasietf@gmail.com</a>&gt;=
 wrote:</div><br class=3D"Apple-interchange-newline"><div class=3D""><div =
dir=3D"ltr" class=3D"">Hello Olle<div class=3D""><br class=3D""></div><div=
 class=3D"">&nbsp; Thanks for the review. I will go through the other =
comments in a while but wanted to share one thought on the ptime =
feedback</div><div class=3D""><br class=3D""></div><div class=3D"">JSEP-15=
 says</div><ul style=3D"font-family: verdana, helvetica, arial, =
sans-serif; font-size: 13.3333px;" class=3D""><li =
style=3D"margin-left:2em;margin-right:2em" class=3D"">If this m=3D =
section is for media with configurable frame sizes, e.g. audio, an =
"a=3Dmaxptime" line, indicating the smallest of the maximum supported =
frame sizes out of all codecs included above, as specified in&nbsp;<a =
href=3D"https://tools.ietf.org/id/draft-ietf-rtcweb-jsep-15.html#RFC4566" =
style=3D"text-decoration:none" class=3D"">[RFC4566]</a>, Section =
6.=E2=80=9D</li></ul></div></div></blockquote>That=E2=80=99s a good =
setting, but not the same. If you have codec Y with ptimes settable from =
10 to 130 and codec Z with ptimes settable from 20 to 250,</div><div>the =
maxptime would be 130 if I understand you right.</div><div><br =
class=3D""></div><div>What I am looking for is the weird situation where =
I want to offer codec Y with the default ptime of 40 and codec Z with =
the default ptime of 50.</div><div>This is a wormhole, since the a=3Dptime=
 setting is kind of a mystery that a few brave people have been trying =
to solve - does it apply</div><div>to ALL codecs in a offered stream and =
if so - how to we handle an offer like above. Codec Y and codec Z may =
have incompatible</div><div>ptimes too, so one default ptime for the =
stream may not work at all.&nbsp;<br class=3D""><blockquote type=3D"cite" =
class=3D""><div class=3D""><div dir=3D"ltr" class=3D""><div class=3D""><br=
 class=3D""></div><div class=3D""><font face=3D"verdana, helvetica, =
arial, sans-serif" class=3D""><span style=3D"font-size:13.3333px" =
class=3D"">The current set of examples try to follow JSEP specification =
to the closest in how the SDP is constructed barring few open issues on =
handing certain =
attributes</span></font></div></div></div></blockquote>Ok, then this =
issue, which is generic SDP may be out of scope for this document - but =
it will pop up at some unspecified time</div><div>in the future so I =
wanted to suggest that the time is now.&nbsp;<br class=3D""><blockquote =
type=3D"cite" class=3D""><div class=3D""><div dir=3D"ltr" class=3D""><div =
class=3D""><font face=3D"verdana, helvetica, arial, sans-serif" =
class=3D""><span style=3D"font-size:13.3333px" class=3D""><br =
class=3D""></span></font></div><div class=3D""><font face=3D"verdana, =
helvetica, arial, sans-serif" class=3D""><span =
style=3D"font-size:13.3333px" class=3D"">Please let us know your =
thoughts &nbsp;on various ptimes across codecs is something that should =
go in JSEP spec SDP rules as well =
?</span></font></div></div></div></blockquote>I need to look a bit more =
into JSEP for that. While waiting for me to do that, others may offer =
their opinions :-)</div><div><br class=3D""></div><div>/O<br =
class=3D""><blockquote type=3D"cite" class=3D""><div class=3D""><div =
dir=3D"ltr" class=3D""><div class=3D""><font face=3D"verdana, helvetica, =
arial, sans-serif" class=3D""><span style=3D"font-size:13.3333px" =
class=3D""><br class=3D""></span></font></div><div class=3D""><font =
face=3D"verdana, helvetica, arial, sans-serif" class=3D""><span =
style=3D"font-size:13.3333px" class=3D"">Thanks</span></font></div><div =
class=3D""><font face=3D"verdana, helvetica, arial, sans-serif" =
class=3D""><span style=3D"font-size:13.3333px" =
class=3D"">Suhas</span></font></div></div><div class=3D"gmail_extra"><br =
class=3D""><div class=3D"gmail_quote">On Tue, Jul 12, 2016 at 5:40 PM, =
Olle E. Johansson <span dir=3D"ltr" class=3D"">&lt;<a =
href=3D"mailto:oej@edvina.net" target=3D"_blank" =
class=3D"">oej@edvina.net</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""><span class=3D""><br class=3D""><div class=3D""><blockquote =
type=3D"cite" class=3D""><div class=3D"">On 12 Jul 2016, at 09:55, Olle =
E. Johansson &lt;<a href=3D"mailto:oej@edvina.net" target=3D"_blank" =
class=3D"">oej@edvina.net</a>&gt; wrote:</div><br class=3D""><div =
class=3D""><pre =
style=3D"font-size:14px;font-style:normal;font-weight:normal;letter-spacin=
g:normal;text-align:start;text-indent:0px;text-transform:none;word-spacing=
:0px;margin-top:0px;margin-bottom:0px" class=3D""><font size=3D"3" =
class=3D""><br class=3D"">General:</font></pre><pre =
style=3D"font-size:14px;font-style:normal;font-weight:normal;letter-spacin=
g:normal;text-align:start;text-indent:0px;text-transform:none;word-spacing=
:0px;margin-top:0px;margin-bottom:0px" class=3D""><font size=3D"3" =
class=3D"">I would like to see some examples using the current IP =
protocol, not the legacy IPv4,</font></pre><pre =
style=3D"font-size:14px;font-style:normal;font-weight:normal;letter-spacin=
g:normal;text-align:start;text-indent:0px;text-transform:none;word-spacing=
:0px;margin-top:0px;margin-bottom:0px" class=3D""><font size=3D"3" =
class=3D"">as well as dual stack examples.</font></pre><pre =
style=3D"font-size:14px;font-style:normal;font-weight:normal;letter-spacin=
g:normal;text-align:start;text-indent:0px;text-transform:none;word-spacing=
:0px;margin-top:0px;margin-bottom:0px" class=3D""><font size=3D"3" =
class=3D""><br class=3D""></font></pre><pre =
style=3D"font-size:14px;font-style:normal;font-weight:normal;letter-spacin=
g:normal;text-align:start;text-indent:0px;text-transform:none;word-spacing=
:0px;margin-top:0px;margin-bottom:0px" class=3D""><font size=3D"3" =
class=3D"">- An IPv6 only web browser offering an IPv4 through a relay =
ice candidate</font></pre><pre =
style=3D"font-size:14px;font-style:normal;font-weight:normal;letter-spacin=
g:normal;text-align:start;text-indent:0px;text-transform:none;word-spacing=
:0px;margin-top:0px;margin-bottom:0px" class=3D""><font size=3D"3" =
class=3D"">- A dual stack client</font></pre><pre =
style=3D"font-size:14px;font-style:normal;font-weight:normal;letter-spacin=
g:normal;text-align:start;text-indent:0px;text-transform:none;word-spacing=
:0px;margin-top:0px;margin-bottom:0px" class=3D"">- A full IPv6 exchange =
- maybe switch one of the examples using IPv4 to IPv6</pre><pre =
style=3D"font-size:14px;font-style:normal;font-weight:normal;letter-spacin=
g:normal;text-align:start;text-indent:0px;text-transform:none;word-spacing=
:0px;margin-top:0px;margin-bottom:0px" class=3D"">  just to show the =
syntax of address and port number</pre><br =
class=3D""></div></blockquote></div></span>In addition I think an =
offer/answer with different packetization times, like<div =
class=3D"">alaw/30 ms, Opus/60 ms or something similar. Yes, I=E2=80=99m =
going down a rat hole,</div><div class=3D"">but it will come up and =
we=E2=80=99d better document a proposal for how to handle =
this.</div><div class=3D""><br class=3D""></div><div class=3D"">If one =
wants to be messy you can add DTMF or CNG to an offer/answer =
using</div><div class=3D"">alaw at 8000 hz and Opus at 48000 making it =
necessary to have multiple DTMF</div><div class=3D"">and CNG for that =
media stream. But that may be out of scope for this particular</div><div =
class=3D"">document, even though it may add clarity.</div><span =
class=3D"HOEnZb"><font color=3D"#888888" class=3D""><div class=3D""><br =
class=3D""></div><div =
class=3D"">/O</div></font></span></div></blockquote></div><br =
class=3D""></div>
</div></blockquote></div><br class=3D""></body></html>=

--Apple-Mail=_7513E0C8-5BE4-44FD-9CE4-F6406A843CD1--


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From: "Cullen Jennings (fluffy)" <fluffy@cisco.com>
To: "Black, David" <david.black@emc.com>
Thread-Topic: [rtcweb] [tsvwg] Fall-back to DSCP 0 in draft-ietf-tsvwg-rtcweb-qos ?
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Subject: Re: [rtcweb] [tsvwg] Fall-back to DSCP 0 in draft-ietf-tsvwg-rtcweb-qos ?
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From: "Cullen Jennings (fluffy)" <fluffy@cisco.com>
To: "Black, David" <david.black@emc.com>
Thread-Topic: [tsvwg] Fall-back to DSCP 0 in draft-ietf-tsvwg-rtcweb-qos ?
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Date: Tue, 12 Jul 2016 16:20:29 +0000
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References: <CB087237-108A-44B1-8293-3498F24A2303@ifi.uio.no> <e3ebbf6c-58ab-1f70-3a5c-36a660dc73e7@gmail.com> <4D9967BC-D23D-4DB9-9ABE-9DA6B15B33A8@ifi.uio.no> <CE03DB3D7B45C245BCA0D243277949362F5D6D94@MX307CL04.corp.emc.com> <f06ecef448a84c6787acd3fda02b56ec@HE101653.emea1.cds.t-internal.com> <CE03DB3D7B45C245BCA0D243277949362F5D8376@MX307CL04.corp.emc.com> <648D8616-0AB5-4420-A829-1481B30237F4@erg.abdn.ac.uk> <CE03DB3D7B45C245BCA0D243277949362F5DA0F5@MX307CL04.corp.emc.com>
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Cc: "Gorry \(erg\)" <gorry@erg.abdn.ac.uk>, "Ruediger.Geib@telekom.de" <Ruediger.Geib@telekom.de>, "tsvwg@ietf.org" <tsvwg@ietf.org>, RTCWeb IETF <rtcweb@ietf.org>, Brian Carpenter <brian.e.carpenter@gmail.com>
Subject: Re: [rtcweb] [tsvwg] Fall-back to DSCP 0 in draft-ietf-tsvwg-rtcweb-qos ?
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On Jul 12, 2016, at 8:55 AM, Black, David <david.black@emc.com<mailto:david=
.black@emc.com>> wrote:

The DART work warns of the pitfalls.

My suggestion is that for protocols using ICE that STUN packets could be us=
ed for
this path check - if this is desired. I am not sure we need to hold the cur=
rent draft
for this sort of addition.

I'd add a sentence to say that ICE uses STUN for this sort of DSCP-specific=
 path
check and cite the draft that specifies how that path checking is performed=
.
Which draft would that be?

I concur that this draft on DSCP usage should not specify how path checking=
 is done.

+1


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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-lin=
e-break: after-white-space;" class=3D"">
<br class=3D"">
<div>
<blockquote type=3D"cite" class=3D"">
<div class=3D"">On Jul 12, 2016, at 8:55 AM, Black, David &lt;<a href=3D"ma=
ilto:david.black@emc.com" class=3D"">david.black@emc.com</a>&gt; wrote:</di=
v>
<br class=3D"Apple-interchange-newline">
<div class=3D"">
<div class=3D"">
<blockquote type=3D"cite" style=3D"font-family: Helvetica; font-size: 14px;=
 font-style: normal; font-variant-caps: normal; font-weight: normal; letter=
-spacing: normal; orphans: auto; text-align: start; text-indent: 0px; text-=
transform: none; white-space: normal; widows: auto; word-spacing: 0px; -web=
kit-text-stroke-width: 0px;" class=3D"">
The DART work warns of the pitfalls.<br class=3D"">
<br class=3D"">
My suggestion is that for protocols using ICE that STUN packets could be us=
ed for<br class=3D"">
this path check - if this is desired. I am not sure we need to hold the cur=
rent draft<br class=3D"">
for this sort of addition.<br class=3D"">
</blockquote>
<br style=3D"font-family: Helvetica; font-size: 14px; font-style: normal; f=
ont-variant-caps: normal; font-weight: normal; letter-spacing: normal; orph=
ans: auto; text-align: start; text-indent: 0px; text-transform: none; white=
-space: normal; widows: auto; word-spacing: 0px; -webkit-text-stroke-width:=
 0px;" class=3D"">
<span style=3D"font-family: Helvetica; font-size: 14px; font-style: normal;=
 font-variant-caps: normal; font-weight: normal; letter-spacing: normal; or=
phans: auto; text-align: start; text-indent: 0px; text-transform: none; whi=
te-space: normal; widows: auto; word-spacing: 0px; -webkit-text-stroke-widt=
h: 0px; float: none; display: inline !important;" class=3D"">I'd
 add a sentence to say that ICE uses STUN for this sort of DSCP-specific pa=
th</span><br style=3D"font-family: Helvetica; font-size: 14px; font-style: =
normal; font-variant-caps: normal; font-weight: normal; letter-spacing: nor=
mal; orphans: auto; text-align: start; text-indent: 0px; text-transform: no=
ne; white-space: normal; widows: auto; word-spacing: 0px; -webkit-text-stro=
ke-width: 0px;" class=3D"">
<span style=3D"font-family: Helvetica; font-size: 14px; font-style: normal;=
 font-variant-caps: normal; font-weight: normal; letter-spacing: normal; or=
phans: auto; text-align: start; text-indent: 0px; text-transform: none; whi=
te-space: normal; widows: auto; word-spacing: 0px; -webkit-text-stroke-widt=
h: 0px; float: none; display: inline !important;" class=3D"">check
 and cite the draft that specifies how that path checking is performed.</sp=
an><br style=3D"font-family: Helvetica; font-size: 14px; font-style: normal=
; font-variant-caps: normal; font-weight: normal; letter-spacing: normal; o=
rphans: auto; text-align: start; text-indent: 0px; text-transform: none; wh=
ite-space: normal; widows: auto; word-spacing: 0px; -webkit-text-stroke-wid=
th: 0px;" class=3D"">
<span style=3D"font-family: Helvetica; font-size: 14px; font-style: normal;=
 font-variant-caps: normal; font-weight: normal; letter-spacing: normal; or=
phans: auto; text-align: start; text-indent: 0px; text-transform: none; whi=
te-space: normal; widows: auto; word-spacing: 0px; -webkit-text-stroke-widt=
h: 0px; float: none; display: inline !important;" class=3D"">Which
 draft would that be?</span><br style=3D"font-family: Helvetica; font-size:=
 14px; font-style: normal; font-variant-caps: normal; font-weight: normal; =
letter-spacing: normal; orphans: auto; text-align: start; text-indent: 0px;=
 text-transform: none; white-space: normal; widows: auto; word-spacing: 0px=
; -webkit-text-stroke-width: 0px;" class=3D"">
<br style=3D"font-family: Helvetica; font-size: 14px; font-style: normal; f=
ont-variant-caps: normal; font-weight: normal; letter-spacing: normal; orph=
ans: auto; text-align: start; text-indent: 0px; text-transform: none; white=
-space: normal; widows: auto; word-spacing: 0px; -webkit-text-stroke-width:=
 0px;" class=3D"">
<span style=3D"font-family: Helvetica; font-size: 14px; font-style: normal;=
 font-variant-caps: normal; font-weight: normal; letter-spacing: normal; or=
phans: auto; text-align: start; text-indent: 0px; text-transform: none; whi=
te-space: normal; widows: auto; word-spacing: 0px; -webkit-text-stroke-widt=
h: 0px; float: none; display: inline !important;" class=3D"">I
 concur that this draft on DSCP usage should not specify how path checking =
is done.</span></div>
</div>
</blockquote>
</div>
<br class=3D"">
<div class=3D"">&#43;1&nbsp;</div>
<div class=3D""><br class=3D"">
</div>
</body>
</html>

--_000_FF3823F7D4744AE3AF0F3935ED8C5381ciscocom_--


From nobody Tue Jul 12 10:12:13 2016
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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:
- 'Transports for WebRTC'
  <draft-ietf-rtcweb-transports-14.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 2016-08-01. Exceptionally, comments may be
sent to iesg@ietf.org instead. In either case, please retain the
beginning of the Subject line to allow automated sorting.

Abstract


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




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

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


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



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From: Magnus Westerlund <magnus.westerlund@ericsson.com>
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Hi,

I have reviewed this document in IETF last call, sorry I missed the last 
WG last call, thus in this stage instead. I do support the publication 
of this document. However, I think the below comments do need to be 
considered before publication approval.

1. Section 3.3:


    When a client gathers all IPv6 addresses on a host, and both
    temporary addresses and permanent addresses of the same scope are
    present, the client SHOULD discard the permanent addresses before
    exposing addresses to the application or using them in ICE.  This is
    consistent with the default policy described in [RFC6724].

    If some of the temporary IPv6 addresses, but not all, are marked
    deprecated, the client SHOULD discard the deprecated addresses.

I haven't noticed this before, but if you have both permanent and 
temporary addresses, can all the temporary be marked deprecated? If that 
can occur, does this specification need to say something in this case 
which should be used, a deprecated temporary or a permanent one?

2. Section 3.4:

    If TCP connections are used, RTP framing according to [RFC4571] MUST
    be used, both for the RTP packets and for the DTLS packets used to
    carry data channels.

I think the last part of this sentence, unintentionally excludes the 
DTLS handshake packets. The ICE TCP spec also specifies that the STUN 
connectivity checks are using RFC4571 framing. Thus, I think some 
reformulation of this sentence is in order.

3. Section 3.5:

    WebRTC implementations MUST support multiplexing of DTLS and RTP over
    the same port pair, as described in the DTLS-SRTP specification
    [RFC5764], section 5.1.2.  All application layer protocol payloads
    over this DTLS connection are SCTP packets.

This text should reference also the update to RFC5764:
https://datatracker.ietf.org/doc/draft-ietf-avtcore-rfc5764-mux-fixes/

That document is publication requested so even as normative reference it 
will hopefully have minimal impact on the publication progress of this 
document.

4. Section 4.2:

    The implementation MAY turn off use of DSCP markings if it detects
    symptoms of unexpected behaviour like priority inversion or blocking
    of packets with certain DSCP markings.  The detection of these
    conditions is implementation dependent.

As raised on the TSVWG and RTCWEB mailing list recently in regards to 
draft-ietf-tsvwg-rtcweb-qos there have been a recent study 
(https://irtf.org/anrw/2016/anrw16-final17.pdf) showing that non zero 
DSCP values resulted in consistent failures in 10-13% of the tested 
paths. So I think it worth considering if this text needs to be sharpen, 
and possibly actually point to a solution that needs to be specified?

5. Section 8.1:

    [I-D.martinsen-mmusic-ice-dualstack-fairness]
               Martinsen, P., Reddy, T., and P. Patil, "ICE IPv4/IPv6
               Dual Stack Fairness", draft-martinsen-mmusic-ice-
               dualstack-fairness-02 (work in progress), February 2015.

Can be updated as this is now an WG item in ICE.

Cheers

Magnus Westerlund

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Services, Media and Network features, Ericsson Research EAB/TXM
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Färögatan 6                 | Mobile +46 73 0949079
SE-164 80 Stockholm, Sweden | mailto: magnus.westerlund@ericsson.com
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To: "Cullen Jennings (fluffy)" <fluffy@cisco.com>, "Black, David" <david.black@emc.com>
References: <CB087237-108A-44B1-8293-3498F24A2303@ifi.uio.no> <e3ebbf6c-58ab-1f70-3a5c-36a660dc73e7@gmail.com> <4D9967BC-D23D-4DB9-9ABE-9DA6B15B33A8@ifi.uio.no> <CE03DB3D7B45C245BCA0D243277949362F5D6D94@MX307CL04.corp.emc.com> <09203BA7-ED00-405C-AA66-C31D411A2B11@cisco.com>
From: Magnus Westerlund <magnus.westerlund@ericsson.com>
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Subject: Re: [rtcweb] [tsvwg] Fall-back to DSCP 0 in draft-ietf-tsvwg-rtcweb-qos ?
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Den 2016-07-12 kl. 18:19, skrev Cullen Jennings (fluffy):
>
> short answer here but as David suggested â€¦  some implementation use
> the STUN packets in ICE  or just  in WebRTC style liveness tests to
> do the tests of if a given DSCP works or not. In general ICE is a
> good tool to take a bunch of possible paths, test which work, and
> select the best.

I do agree that how you do the path checks when setting DSCP values != 0 
is dependent on the context. For the WebRTC I do agree doing checks 
using ICE is quite reasonable.

We already have similar path testing usages of ICE in the ECN for RTP 
specification (RFC6679), see Section 7.2.1. I will note that taking this 
as blueprint for DSCP testing, what is needed clearly requires a new 
separate specification. The components needs are: 1) A new STUN 
parameter to request the ICE peer to echo the DSCP field value received. 
2) A ICE capability parameter to be used in signalling negotiations to 
determine capability for this feature. 3) Behaviour specification on how 
to test values and interpret responses. This include things like if one 
should actually establish multiple candidate pairs one with DSCP testing 
and one without?

So the question here is how to proceed with this issue. So I would 
suggest the following way forward.

1. draft-ietf-tsvwg-rtcweb-qos identifies the issue and recommends the 
user to apply path verification methods but don't specify them.

2. Someone takes on the task to write a DSCP path verification extension 
to ICE.

3. The natural place to indicate the need/recommendation for 
implementing this functionality would be in draft-ietf-rtcweb-transports 
(Currently in IETF LC). However, here I think we need to have a 
discussion if RTCWEB WG wants to only place a suitable warning about the 
need, and indicate future forthcoming specification or if we hold this 
document up until this solution is available?


Cheers

Magnus Westerlund

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


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From: Christer Holmberg <christer.holmberg@ericsson.com>
To: Suhas Nandakumar <suhasietf@gmail.com>, "rtcweb@ietf.org" <rtcweb@ietf.org>
Thread-Topic: [rtcweb] draft-ietf-rtcweb-sdp-02 submitted
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Subject: Re: [rtcweb] draft-ietf-rtcweb-sdp-02 submitted
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--_000_D3AD51DEBF59christerholmbergericssoncom_
Content-Type: text/plain; charset="Windows-1252"
Content-Transfer-Encoding: quoted-printable

Hi,

The examples are not aligned with all of the latest versions of associated =
drafts. See below.

Q1:

Assuming we want to be compliant with draft-ice-5245bis (even though both J=
SEP and your draft still reference RFC 5245), you should use the SDP 'ice-o=
ptions: ice2=92 attribute (draft-ietf-mmusic-ice-sip-sdp).



Q2:

I think the SDP =91dtls-id=92 attribute (draft-ietf-mmusic-sdp-dtls) should=
 be added.


Q3:

According to draft-bundle, you need to include the SDP =91rtcp-mux-only=92 =
attribute in bundle-only m- lines.

Also in general I think it would be good to have some mux-exclusive example=
s, since they support of non-mux is optional (and since some vendors have i=
ndicated that they will not support non-mux long term).


Regards,

Christer




From: rtcweb <rtcweb-bounces@ietf.org<mailto:rtcweb-bounces@ietf.org>> on b=
ehalf of Suhas Nandakumar <suhasietf@gmail.com<mailto:suhasietf@gmail.com>>
Date: Friday 8 July 2016 at 05:26
To: "rtcweb@ietf.org<mailto:rtcweb@ietf.org>" <rtcweb@ietf.org<mailto:rtcwe=
b@ietf.org>>
Subject: [rtcweb] draft-ietf-rtcweb-sdp-02 submitted

Hello All

   We have submitted the -02 version of RTCWEB SDP examples draft which wen=
t through quite a bit of content re-shuffling and updates to match (attempt=
) the latest versions of BUNDLE, JSEP and related drafts.

Many thanks to Paul for starting the review of the draft that triggered few=
 formatting changes, changes related to consistency of attribute values and=
 their ordering, to enhance the readability throughout the draft.

Please refer to the following HTML version while reviewing/reading the draf=
t:

https://tools.ietf.org/id/draft-ietf-rtcweb-sdp-02.html

[[ It might take a short while for the data-tracker to render the above HTM=
L version though ]]

Here is the high-level summary of changes
1. As far as possible, the examples reflect latest BUNDLE and JSEP requirem=
ents.
2. SCTP examples are updated
3. Visual markers, separators are added in the SDP table
4. SSRC values, IP Address , Port numbers, CNAMEs are consistent to a great=
 extent now
5. Attributes ordering is consistent across all the examples now
6. RID, Simulcast examples are matched against their latest draft versions.

Please give it a read and let us know your comments/feedback.

Cheers
Suhas/Cullen


--_000_D3AD51DEBF59christerholmbergericssoncom_
Content-Type: text/html; charset="Windows-1252"
Content-ID: <17739C6C0ED2A942B875CC0A5649348F@ericsson.com>
Content-Transfer-Encoding: quoted-printable

<html>
<head>
<meta http-equiv=3D"Content-Type" content=3D"text/html; charset=3DWindows-1=
252">
</head>
<body style=3D"word-wrap: break-word; -webkit-nbsp-mode: space; -webkit-lin=
e-break: after-white-space; color: rgb(0, 0, 0); font-size: 14px; font-fami=
ly: Calibri, sans-serif;">
<div style=3D"color: rgb(0, 0, 0); font-family: Calibri, sans-serif; font-s=
ize: 14px;">
Hi,</div>
<div style=3D"color: rgb(0, 0, 0); font-family: Calibri, sans-serif; font-s=
ize: 14px;">
<br>
</div>
<div style=3D"color: rgb(0, 0, 0); font-family: Calibri, sans-serif; font-s=
ize: 14px;">
The examples are not aligned with all of the latest versions of associated =
drafts. See below.</div>
<div style=3D"color: rgb(0, 0, 0); font-family: Calibri, sans-serif; font-s=
ize: 14px;">
<br>
</div>
<div style=3D"color: rgb(0, 0, 0); font-family: Calibri, sans-serif; font-s=
ize: 14px;">
Q1:</div>
<div style=3D"color: rgb(0, 0, 0); font-family: Calibri, sans-serif; font-s=
ize: 14px;">
<br>
</div>
<div style=3D"widows: 1;"><font face=3D"Calibri,sans-serif">Assuming we wan=
t to be compliant with draft-ice-5245bis (even though both JSEP and your dr=
aft still reference RFC 5245), you should use the SDP '</font><span style=
=3D"widows: 1;"><font face=3D"Calibri,sans-serif"><span style=3D"white-spac=
e: pre-wrap;">ice-options:
 ice2</span></font></span><font face=3D"Calibri,sans-serif">=92</font><span=
 style=3D"widows: 1;"><font face=3D"Calibri,sans-serif"><span style=3D"whit=
e-space: pre-wrap;"> attribute (draft-ietf-mmusic-ice-sip-sdp).</span></fon=
t></span></div>
<div style=3D"color: rgb(0, 0, 0); font-family: Calibri, sans-serif; font-s=
ize: 14px;">
<br>
</div>
<div style=3D"color: rgb(0, 0, 0); font-family: Calibri, sans-serif; font-s=
ize: 14px;">
<br>
</div>
<div style=3D"color: rgb(0, 0, 0); font-family: Calibri, sans-serif; font-s=
ize: 14px;">
<br>
</div>
<div style=3D"color: rgb(0, 0, 0); font-family: Calibri, sans-serif; font-s=
ize: 14px;">
Q2:</div>
<div style=3D"color: rgb(0, 0, 0); font-family: Calibri, sans-serif; font-s=
ize: 14px;">
<br>
</div>
<div style=3D"color: rgb(0, 0, 0); font-family: Calibri, sans-serif; font-s=
ize: 14px;">
I think the SDP =91dtls-id=92 attribute&nbsp;<span style=3D"white-space: pr=
e-wrap; widows: 1;">(draft-ietf-mmusic-sdp-dtls)
</span>should be added.</div>
<div style=3D"color: rgb(0, 0, 0); font-family: Calibri, sans-serif; font-s=
ize: 14px;">
<br>
</div>
<div style=3D"color: rgb(0, 0, 0); font-family: Calibri, sans-serif; font-s=
ize: 14px;">
<br>
</div>
<div style=3D"color: rgb(0, 0, 0); font-family: Calibri, sans-serif; font-s=
ize: 14px;">
Q3:</div>
<div style=3D"color: rgb(0, 0, 0); font-family: Calibri, sans-serif; font-s=
ize: 14px;">
<br>
</div>
<div style=3D"color: rgb(0, 0, 0); font-family: Calibri, sans-serif; font-s=
ize: 14px;">
According to draft-bundle, you need to include the SDP =91rtcp-mux-only=92 =
attribute in bundle-only m- lines.</div>
<div style=3D"color: rgb(0, 0, 0); font-family: Calibri, sans-serif; font-s=
ize: 14px;">
<br>
</div>
<div style=3D"color: rgb(0, 0, 0); font-family: Calibri, sans-serif; font-s=
ize: 14px;">
Also in general I think it would be good to have some mux-exclusive example=
s, since they support of non-mux is optional (and since some vendors have i=
ndicated that they will not support non-mux long term).</div>
<div style=3D"color: rgb(0, 0, 0); font-family: Calibri, sans-serif; font-s=
ize: 14px;">
<br>
</div>
<div style=3D"color: rgb(0, 0, 0); font-family: Calibri, sans-serif; font-s=
ize: 14px;">
<br>
</div>
<div style=3D"color: rgb(0, 0, 0); font-family: Calibri, sans-serif; font-s=
ize: 14px;">
Regards,</div>
<div style=3D"color: rgb(0, 0, 0); font-family: Calibri, sans-serif; font-s=
ize: 14px;">
<br>
</div>
<div style=3D"color: rgb(0, 0, 0); font-family: Calibri, sans-serif; font-s=
ize: 14px;">
Christer</div>
<div style=3D"color: rgb(0, 0, 0); font-family: Calibri, sans-serif; font-s=
ize: 14px;">
<br>
</div>
<div style=3D"color: rgb(0, 0, 0); font-family: Calibri, sans-serif; font-s=
ize: 14px;">
<br>
</div>
<div style=3D"color: rgb(0, 0, 0); font-family: Calibri, sans-serif; font-s=
ize: 14px;">
&nbsp;</div>
<div style=3D"color: rgb(0, 0, 0); font-family: Calibri, sans-serif; font-s=
ize: 14px;">
<br>
</div>
<span id=3D"OLK_SRC_BODY_SECTION" style=3D"color: rgb(0, 0, 0); font-family=
: Calibri, sans-serif; font-size: 14px;">
<div style=3D"font-family:Calibri; font-size:11pt; text-align:left; color:b=
lack; BORDER-BOTTOM: medium none; BORDER-LEFT: medium none; PADDING-BOTTOM:=
 0in; PADDING-LEFT: 0in; PADDING-RIGHT: 0in; BORDER-TOP: #b5c4df 1pt solid;=
 BORDER-RIGHT: medium none; PADDING-TOP: 3pt">
<span style=3D"font-weight:bold">From: </span>rtcweb &lt;<a href=3D"mailto:=
rtcweb-bounces@ietf.org">rtcweb-bounces@ietf.org</a>&gt; on behalf of Suhas=
 Nandakumar &lt;<a href=3D"mailto:suhasietf@gmail.com">suhasietf@gmail.com<=
/a>&gt;<br>
<span style=3D"font-weight:bold">Date: </span>Friday 8 July 2016 at 05:26<b=
r>
<span style=3D"font-weight:bold">To: </span>&quot;<a href=3D"mailto:rtcweb@=
ietf.org">rtcweb@ietf.org</a>&quot; &lt;<a href=3D"mailto:rtcweb@ietf.org">=
rtcweb@ietf.org</a>&gt;<br>
<span style=3D"font-weight:bold">Subject: </span>[rtcweb] draft-ietf-rtcweb=
-sdp-02 submitted<br>
</div>
<div><br>
</div>
<div>
<div>
<div dir=3D"ltr">Hello All
<div><br>
</div>
<div>&nbsp; &nbsp;We have submitted the -02 version of RTCWEB SDP examples =
draft which went through quite a bit of content re-shuffling and updates to=
 match (attempt) the latest versions of BUNDLE, JSEP and related drafts.&nb=
sp;</div>
<div><br>
</div>
<div>Many thanks to Paul for starting the review of the draft that triggere=
d few formatting changes, changes related to consistency of attribute value=
s and their ordering, to enhance the readability throughout the draft.</div=
>
<div><br>
</div>
<div>Please refer to the following HTML version while reviewing/reading the=
 draft:&nbsp;</div>
<div><br>
</div>
<div><a href=3D"https://tools.ietf.org/id/draft-ietf-rtcweb-sdp-02.html">ht=
tps://tools.ietf.org/id/draft-ietf-rtcweb-sdp-02.html</a></div>
<div><br>
</div>
<div>[[ It might take a short while for the data-tracker to render the abov=
e HTML version though ]]</div>
<div><br>
</div>
<div>Here is the high-level summary of changes</div>
<div>1. As far as possible, the examples reflect latest BUNDLE and JSEP req=
uirements.</div>
<div>2. SCTP examples are updated&nbsp;</div>
<div>3. Visual markers, separators are added in the SDP table</div>
<div>4. SSRC values, IP Address , Port numbers, CNAMEs are consistent to a =
great extent now</div>
<div>5. Attributes ordering is consistent across all the examples now</div>
<div>6. RID, Simulcast examples are matched against their latest draft vers=
ions.</div>
<div><br>
</div>
<div>Please give it a read and let us know your comments/feedback.</div>
<div><br>
</div>
<div>Cheers</div>
<div>Suhas/Cullen</div>
<div><br>
</div>
</div>
</div>
</div>
</span>
</body>
</html>

--_000_D3AD51DEBF59christerholmbergericssoncom_--


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References: <CAMRcRGTnptR4iw-_u=JWOF1HOo6Ej60DSUuYVn9UY88_yqUW4A@mail.gmail.com> <D3AD51DE.BF59%christer.holmberg@ericsson.com>
From: Suhas Nandakumar <suhasietf@gmail.com>
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To: Christer Holmberg <christer.holmberg@ericsson.com>
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Subject: Re: [rtcweb] draft-ietf-rtcweb-sdp-02 submitted
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Hello Christer

 Thanks for reviewing the draft. Here are few general comments on how the
SDP examples are laid out in the current version
  - It aligns closest to JSEP-14 and its dependencies
  - Wherever there is  conflicts between JSEP-14 and BUNDLE-31 , i made a
choice to stick with JSEP-14

In our presentation at IETF-96 we plan to bring up these differences as one
of the open issues

Please see inline for specific feedback items.


On Thu, Jul 14, 2016 at 5:30 PM, Christer Holmberg <
christer.holmberg@ericsson.com> wrote:

> Hi,
>
> The examples are not aligned with all of the latest versions of associate=
d
> drafts. See below.
>
> Q1:
>
> Assuming we want to be compliant with draft-ice-5245bis (even though both
> JSEP and your draft still reference RFC 5245), you should use the SDP 'ic=
e-options:
> ice2=E2=80=99 attribute (draft-ietf-mmusic-ice-sip-sdp).
>
> [suhas] - Agree .. it is one of open issues on draft alignments with JSEP

>
>
> Q2:
>
> I think the SDP =E2=80=98dtls-id=E2=80=99 attribute (draft-ietf-mmusic-sd=
p-dtls) should
> be added.
>
>
[suhas] . same issues as previous one. JSEP-15 doesn't list
draft-ietf-mmusic-sdp-dtls as dependency


>
> Q3:
>
> According to draft-bundle, you need to include the SDP =E2=80=98rtcp-mux-=
only=E2=80=99
> attribute in bundle-only m- lines.
>
> Also in general I think it would be good to have some mux-exclusive
> examples, since they support of non-mux is optional (and since some vendo=
rs
> have indicated that they will not support non-mux long term).
>
>
[suhas] .. There are few offers that include rtcp-mux-only (atleast 5.3.1)
.. I agree we can add concrete examples on this. Also to note JSEP doesn't
list rtcp-mux-only in its SDP construction rules.  We hope to clarify these
open issues at the upcoming meeting.

>
> Regards,
>
> Christer
>
>
>
>
> From: rtcweb <rtcweb-bounces@ietf.org> on behalf of Suhas Nandakumar <
> suhasietf@gmail.com>
> Date: Friday 8 July 2016 at 05:26
> To: "rtcweb@ietf.org" <rtcweb@ietf.org>
> Subject: [rtcweb] draft-ietf-rtcweb-sdp-02 submitted
>
> Hello All
>
>    We have submitted the -02 version of RTCWEB SDP examples draft which
> went through quite a bit of content re-shuffling and updates to match
> (attempt) the latest versions of BUNDLE, JSEP and related drafts.
>
> Many thanks to Paul for starting the review of the draft that triggered
> few formatting changes, changes related to consistency of attribute value=
s
> and their ordering, to enhance the readability throughout the draft.
>
> Please refer to the following HTML version while reviewing/reading the
> draft:
>
> https://tools.ietf.org/id/draft-ietf-rtcweb-sdp-02.html
>
> [[ It might take a short while for the data-tracker to render the above
> HTML version though ]]
>
> Here is the high-level summary of changes
> 1. As far as possible, the examples reflect latest BUNDLE and JSEP
> requirements.
> 2. SCTP examples are updated
> 3. Visual markers, separators are added in the SDP table
> 4. SSRC values, IP Address , Port numbers, CNAMEs are consistent to a
> great extent now
> 5. Attributes ordering is consistent across all the examples now
> 6. RID, Simulcast examples are matched against their latest draft version=
s.
>
> Please give it a read and let us know your comments/feedback.
>
> Cheers
> Suhas/Cullen
>
>

--001a113db05ab47b1f05379821da
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<div dir=3D"ltr">Hello Christer<div><br></div><div>=C2=A0Thanks for reviewi=
ng the draft. Here are few general comments on how the SDP examples are lai=
d out in the current version</div><div>=C2=A0 - It aligns closest to JSEP-1=
4 and its dependencies</div><div>=C2=A0 - Wherever there is =C2=A0conflicts=
 between JSEP-14 and BUNDLE-31 , i made a choice to stick with JSEP-14=C2=
=A0</div><div><br></div><div>In our presentation at IETF-96 we plan to brin=
g up these differences as one of the open issues</div><div><br></div><div>P=
lease see inline for specific feedback items.</div><div><br></div><div clas=
s=3D"gmail_extra"><br><div class=3D"gmail_quote">On Thu, Jul 14, 2016 at 5:=
30 PM, Christer Holmberg <span dir=3D"ltr">&lt;<a href=3D"mailto:christer.h=
olmberg@ericsson.com" target=3D"_blank">christer.holmberg@ericsson.com</a>&=
gt;</span> wrote:<br><blockquote class=3D"gmail_quote" style=3D"margin:0px =
0px 0px 0.8ex;border-left-width:1px;border-left-style:solid;border-left-col=
or:rgb(204,204,204);padding-left:1ex">



<div style=3D"word-wrap:break-word;color:rgb(0,0,0);font-size:14px;font-fam=
ily:Calibri,sans-serif">
<div style=3D"color:rgb(0,0,0);font-family:Calibri,sans-serif;font-size:14p=
x">
Hi,</div>
<div style=3D"color:rgb(0,0,0);font-family:Calibri,sans-serif;font-size:14p=
x">
<br>
</div>
<div style=3D"color:rgb(0,0,0);font-family:Calibri,sans-serif;font-size:14p=
x">
The examples are not aligned with all of the latest versions of associated =
drafts. See below.</div>
<div style=3D"color:rgb(0,0,0);font-family:Calibri,sans-serif;font-size:14p=
x">
<br>
</div>
<div style=3D"color:rgb(0,0,0);font-family:Calibri,sans-serif;font-size:14p=
x">
Q1:</div>
<div style=3D"color:rgb(0,0,0);font-family:Calibri,sans-serif;font-size:14p=
x">
<br>
</div>
<div><font face=3D"Calibri,sans-serif">Assuming we want to be compliant wit=
h draft-ice-5245bis (even though both JSEP and your draft still reference R=
FC 5245), you should use the SDP &#39;</font><span><font face=3D"Calibri,sa=
ns-serif"><span style=3D"white-space:pre-wrap">ice-options:
 ice2</span></font></span><font face=3D"Calibri,sans-serif">=E2=80=99</font=
><span><font face=3D"Calibri,sans-serif"><span style=3D"white-space:pre-wra=
p"> attribute (draft-ietf-mmusic-ice-sip-sdp).</span></font></span></div>
<div style=3D"color:rgb(0,0,0);font-family:Calibri,sans-serif;font-size:14p=
x">
<br></div></div></blockquote><div>[suhas] - Agree .. it is one of open issu=
es on draft alignments with JSEP=C2=A0</div><blockquote class=3D"gmail_quot=
e" style=3D"margin:0px 0px 0px 0.8ex;border-left-width:1px;border-left-styl=
e:solid;border-left-color:rgb(204,204,204);padding-left:1ex"><div style=3D"=
word-wrap:break-word;color:rgb(0,0,0);font-size:14px;font-family:Calibri,sa=
ns-serif"><div style=3D"color:rgb(0,0,0);font-family:Calibri,sans-serif;fon=
t-size:14px">
</div>
<div style=3D"color:rgb(0,0,0);font-family:Calibri,sans-serif;font-size:14p=
x">
<br>
</div>
<div style=3D"color:rgb(0,0,0);font-family:Calibri,sans-serif;font-size:14p=
x">
<br>
</div>
<div style=3D"color:rgb(0,0,0);font-family:Calibri,sans-serif;font-size:14p=
x">
Q2:</div>
<div style=3D"color:rgb(0,0,0);font-family:Calibri,sans-serif;font-size:14p=
x">
<br>
</div>
<div style=3D"color:rgb(0,0,0);font-family:Calibri,sans-serif;font-size:14p=
x">
I think the SDP =E2=80=98dtls-id=E2=80=99 attribute=C2=A0<span style=3D"whi=
te-space:pre-wrap">(draft-ietf-mmusic-sdp-dtls)
</span>should be added.</div>
<div style=3D"color:rgb(0,0,0);font-family:Calibri,sans-serif;font-size:14p=
x">
<br></div></div></blockquote><div><br></div><div>[suhas] . same issues as p=
revious one. JSEP-15 doesn&#39;t list=C2=A0<span style=3D"color:rgb(0,0,0);=
font-family:Calibri,sans-serif;font-size:14px;white-space:pre-wrap">draft-i=
etf-mmusic-</span><span style=3D"color:rgb(0,0,0);font-family:Calibri,sans-=
serif;font-size:14px;white-space:pre-wrap">sdp-dtls as dependency</span></d=
iv><div>=C2=A0</div><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 style=3D"word-wrap:break-word;col=
or:rgb(0,0,0);font-size:14px;font-family:Calibri,sans-serif"><div style=3D"=
color:rgb(0,0,0);font-family:Calibri,sans-serif;font-size:14px">
</div>
<div style=3D"color:rgb(0,0,0);font-family:Calibri,sans-serif;font-size:14p=
x">
<br>
</div>
<div style=3D"color:rgb(0,0,0);font-family:Calibri,sans-serif;font-size:14p=
x">
Q3:</div>
<div style=3D"color:rgb(0,0,0);font-family:Calibri,sans-serif;font-size:14p=
x">
<br>
</div>
<div style=3D"color:rgb(0,0,0);font-family:Calibri,sans-serif;font-size:14p=
x">
According to draft-bundle, you need to include the SDP =E2=80=98rtcp-mux-on=
ly=E2=80=99 attribute in bundle-only m- lines.</div>
<div style=3D"color:rgb(0,0,0);font-family:Calibri,sans-serif;font-size:14p=
x">
<br>
</div>
<div style=3D"color:rgb(0,0,0);font-family:Calibri,sans-serif;font-size:14p=
x">
Also in general I think it would be good to have some mux-exclusive example=
s, since they support of non-mux is optional (and since some vendors have i=
ndicated that they will not support non-mux long term).</div>
<div style=3D"color:rgb(0,0,0);font-family:Calibri,sans-serif;font-size:14p=
x">
<br></div></div></blockquote><div><br></div><div>[suhas] .. There are few o=
ffers that include rtcp-mux-only (atleast 5.3.1) .. I agree we can add conc=
rete examples on this. Also to note JSEP doesn&#39;t list rtcp-mux-only in =
its SDP construction rules.=C2=A0 We hope to clarify these open issues at t=
he upcoming meeting.=C2=A0</div><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 style=3D"word-wrap:br=
eak-word;color:rgb(0,0,0);font-size:14px;font-family:Calibri,sans-serif"><d=
iv style=3D"color:rgb(0,0,0);font-family:Calibri,sans-serif;font-size:14px"=
>
</div>
<div style=3D"color:rgb(0,0,0);font-family:Calibri,sans-serif;font-size:14p=
x">
<br>
</div>
<div style=3D"color:rgb(0,0,0);font-family:Calibri,sans-serif;font-size:14p=
x">
Regards,</div>
<div style=3D"color:rgb(0,0,0);font-family:Calibri,sans-serif;font-size:14p=
x">
<br>
</div>
<div style=3D"color:rgb(0,0,0);font-family:Calibri,sans-serif;font-size:14p=
x">
Christer</div>
<div style=3D"color:rgb(0,0,0);font-family:Calibri,sans-serif;font-size:14p=
x">
<br>
</div>
<div style=3D"color:rgb(0,0,0);font-family:Calibri,sans-serif;font-size:14p=
x">
<br>
</div>
<div style=3D"color:rgb(0,0,0);font-family:Calibri,sans-serif;font-size:14p=
x">
=C2=A0</div>
<div style=3D"color:rgb(0,0,0);font-family:Calibri,sans-serif;font-size:14p=
x">
<br>
</div>
<span style=3D"color:rgb(0,0,0);font-family:Calibri,sans-serif;font-size:14=
px">
<div style=3D"font-family:Calibri;font-size:11pt;text-align:left;color:blac=
k;border-width:1pt medium medium;border-style:solid none none;padding:3pt 0=
in 0in;border-top-color:rgb(181,196,223)">
<span style=3D"font-weight:bold">From: </span>rtcweb &lt;<a href=3D"mailto:=
rtcweb-bounces@ietf.org" target=3D"_blank">rtcweb-bounces@ietf.org</a>&gt; =
on behalf of Suhas Nandakumar &lt;<a href=3D"mailto:suhasietf@gmail.com" ta=
rget=3D"_blank">suhasietf@gmail.com</a>&gt;<br>
<span style=3D"font-weight:bold">Date: </span>Friday 8 July 2016 at 05:26<b=
r>
<span style=3D"font-weight:bold">To: </span>&quot;<a href=3D"mailto:rtcweb@=
ietf.org" target=3D"_blank">rtcweb@ietf.org</a>&quot; &lt;<a href=3D"mailto=
:rtcweb@ietf.org" target=3D"_blank">rtcweb@ietf.org</a>&gt;<br>
<span style=3D"font-weight:bold">Subject: </span>[rtcweb] draft-ietf-rtcweb=
-sdp-02 submitted<br>
</div><div><div class=3D"h5">
<div><br>
</div>
<div>
<div>
<div dir=3D"ltr">Hello All
<div><br>
</div>
<div>=C2=A0 =C2=A0We have submitted the -02 version of RTCWEB SDP examples =
draft which went through quite a bit of content re-shuffling and updates to=
 match (attempt) the latest versions of BUNDLE, JSEP and related drafts.=C2=
=A0</div>
<div><br>
</div>
<div>Many thanks to Paul for starting the review of the draft that triggere=
d few formatting changes, changes related to consistency of attribute value=
s and their ordering, to enhance the readability throughout the draft.</div=
>
<div><br>
</div>
<div>Please refer to the following HTML version while reviewing/reading the=
 draft:=C2=A0</div>
<div><br>
</div>
<div><a href=3D"https://tools.ietf.org/id/draft-ietf-rtcweb-sdp-02.html" ta=
rget=3D"_blank">https://tools.ietf.org/id/draft-ietf-rtcweb-sdp-02.html</a>=
</div>
<div><br>
</div>
<div>[[ It might take a short while for the data-tracker to render the abov=
e HTML version though ]]</div>
<div><br>
</div>
<div>Here is the high-level summary of changes</div>
<div>1. As far as possible, the examples reflect latest BUNDLE and JSEP req=
uirements.</div>
<div>2. SCTP examples are updated=C2=A0</div>
<div>3. Visual markers, separators are added in the SDP table</div>
<div>4. SSRC values, IP Address , Port numbers, CNAMEs are consistent to a =
great extent now</div>
<div>5. Attributes ordering is consistent across all the examples now</div>
<div>6. RID, Simulcast examples are matched against their latest draft vers=
ions.</div>
<div><br>
</div>
<div>Please give it a read and let us know your comments/feedback.</div>
<div><br>
</div>
<div>Cheers</div>
<div>Suhas/Cullen</div>
<div><br>
</div>
</div>
</div>
</div>
</div></div></span>
</div>

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

--001a113db05ab47b1f05379821da--


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From: "Black, David" <david.black@emc.com>
To: Magnus Westerlund <magnus.westerlund@ericsson.com>
Thread-Topic: [tsvwg] [rtcweb] Fall-back to DSCP 0 in draft-ietf-tsvwg-rtcweb-qos ?
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Date: Thu, 14 Jul 2016 21:13:16 +0000
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Subject: Re: [rtcweb] [tsvwg] Fall-back to DSCP 0 in draft-ietf-tsvwg-rtcweb-qos ?
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To: Magnus Westerlund <magnus.westerlund@ericsson.com>, ietf@ietf.org
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Subject: Re: [rtcweb] Extended Last Call: <draft-ietf-rtcweb-transports-14.txt> (Transports for WebRTC) to Proposed Standard
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Thank you, I have filed these as issue #22 to #26 at
github.com/rtcweb-wg/rtcweb-transport/issues.

I think issue #25, which I entitled "DSCP bleaching", may be the
difficult one to resolve.


From nobody Mon Jul 18 00:56:02 2016
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To: "Black, David" <david.black@emc.com>, Magnus Westerlund <magnus.westerlund@ericsson.com>
References: <CB087237-108A-44B1-8293-3498F24A2303@ifi.uio.no> <e3ebbf6c-58ab-1f70-3a5c-36a660dc73e7@gmail.com> <4D9967BC-D23D-4DB9-9ABE-9DA6B15B33A8@ifi.uio.no> <CE03DB3D7B45C245BCA0D243277949362F5D6D94@MX307CL04.corp.emc.com> <09203BA7-ED00-405C-AA66-C31D411A2B11@cisco.com> <4f7acb04-6b37-0f95-3613-c127ff8b31ad@ericsson.com> <CE03DB3D7B45C245BCA0D243277949362F5E4761@MX307CL04.corp.emc.com>
From: Harald Alvestrand <harald@alvestrand.no>
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Subject: Re: [rtcweb] [tsvwg] Fall-back to DSCP 0 in draft-ietf-tsvwg-rtcweb-qos ?
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David,

repeating for transparency what I asked in person:

Can you post the links to the measurements that led to the conclusion
that non-arrival of DSCP-marked packets is a big enough problem that we
should develop specific mechanisms to detect it?

I'd like to make sure we're all on the same page wrt what the problem
is, how we can measure it, and in what contexts the problems are likely
to have significant impact.

(I worry in particular about how many false positives a blackhole
detection mechanism is likely to have, and how much this will in turn
increase randomness in performance as percieved by the user, making
performance problems harder to diagnose.)


On 07/14/2016 11:13 PM, Black, David wrote:
> Magnus,
>
> I think that's a fine suggestion.   I think the next step is:
>
>> 3. The natural place to indicate the need/recommendation for
>> implementing this functionality would be in draft-ietf-rtcweb-transports
>> (Currently in IETF LC). However, here I think we need to have a
>> discussion if RTCWEB WG wants to only place a suitable warning about the
>> need, and indicate future forthcoming specification or if we hold this
>> document up until this solution is available?
> I'll attend the Thu RTCWEB session in Berlin to see how this comes out,
> after which it should be straightforward for the draft authors and yours
> truly to write the sentence or two that draft-ietf-tsvwg-rtcweb-qos will
> need.
>
> Thanks, --David
>
>> -----Original Message-----
>> From: Magnus Westerlund [mailto:magnus.westerlund@ericsson.com]
>> Sent: Thursday, July 14, 2016 4:53 AM
>> To: Cullen Jennings (fluffy); Black, David
>> Cc: RTCWeb IETF; Michael Welzl; tsvwg@ietf.org
>> Subject: Re: [tsvwg] [rtcweb] Fall-back to DSCP 0 in draft-ietf-tsvwg-rtcweb-qos ?
>>
>> Den 2016-07-12 kl. 18:19, skrev Cullen Jennings (fluffy):
>>> short answer here but as David suggested â€¦  some implementation use
>>> the STUN packets in ICE  or just  in WebRTC style liveness tests to
>>> do the tests of if a given DSCP works or not. In general ICE is a
>>> good tool to take a bunch of possible paths, test which work, and
>>> select the best.
>> I do agree that how you do the path checks when setting DSCP values != 0
>> is dependent on the context. For the WebRTC I do agree doing checks
>> using ICE is quite reasonable.
>>
>> We already have similar path testing usages of ICE in the ECN for RTP
>> specification (RFC6679), see Section 7.2.1. I will note that taking this
>> as blueprint for DSCP testing, what is needed clearly requires a new
>> separate specification. The components needs are: 1) A new STUN
>> parameter to request the ICE peer to echo the DSCP field value received.
>> 2) A ICE capability parameter to be used in signalling negotiations to
>> determine capability for this feature. 3) Behaviour specification on how
>> to test values and interpret responses. This include things like if one
>> should actually establish multiple candidate pairs one with DSCP testing
>> and one without?
>>
>> So the question here is how to proceed with this issue. So I would
>> suggest the following way forward.
>>
>> 1. draft-ietf-tsvwg-rtcweb-qos identifies the issue and recommends the
>> user to apply path verification methods but don't specify them.
>>
>> 2. Someone takes on the task to write a DSCP path verification extension
>> to ICE.
>>
>> 3. The natural place to indicate the need/recommendation for
>> implementing this functionality would be in draft-ietf-rtcweb-transports
>> (Currently in IETF LC). However, here I think we need to have a
>> discussion if RTCWEB WG wants to only place a suitable warning about the
>> need, and indicate future forthcoming specification or if we hold this
>> document up until this solution is available?
>>
>>
>> Cheers
>>
>> Magnus Westerlund
>>
>> ----------------------------------------------------------------------
>> Services, Media and Network features, Ericsson Research EAB/TXM
>> ----------------------------------------------------------------------
>> Ericsson AB                 | Phone  +46 10 7148287
>> FÃ¤rÃ¶gatan 6                 | Mobile +46 73 0949079
>> SE-164 80 Stockholm, Sweden | mailto: magnus.westerlund@ericsson.com
>> ----------------------------------------------------------------------
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb


-- 
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From: Magnus Westerlund <magnus.westerlund@ericsson.com>
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Subject: Re: [rtcweb] [tsvwg] Fall-back to DSCP 0 in draft-ietf-tsvwg-rtcweb-qos ?
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Den 2016-07-18 kl. 09:55, skrev Harald Alvestrand:
> David,
>
> repeating for transparency what I asked in person:
>
> Can you post the links to the measurements that led to the conclusion
> that non-arrival of DSCP-marked packets is a big enough problem that we
> should develop specific mechanisms to detect it?
>

https://irtf.org/anrw/2016/anrw16-final17.pdf

> I'd like to make sure we're all on the same page wrt what the problem
> is, how we can measure it, and in what contexts the problems are likely
> to have significant impact.
>
> (I worry in particular about how many false positives a blackhole
> detection mechanism is likely to have, and how much this will in turn
> increase randomness in performance as percieved by the user, making
> performance problems harder to diagnose.)
>
>
> On 07/14/2016 11:13 PM, Black, David wrote:
>> Magnus,
>>
>> I think that's a fine suggestion.   I think the next step is:
>>
>>> 3. The natural place to indicate the need/recommendation for
>>> implementing this functionality would be in draft-ietf-rtcweb-transports
>>> (Currently in IETF LC). However, here I think we need to have a
>>> discussion if RTCWEB WG wants to only place a suitable warning about the
>>> need, and indicate future forthcoming specification or if we hold this
>>> document up until this solution is available?
>> I'll attend the Thu RTCWEB session in Berlin to see how this comes out,
>> after which it should be straightforward for the draft authors and yours
>> truly to write the sentence or two that draft-ietf-tsvwg-rtcweb-qos will
>> need.
>>
>> Thanks, --David
>>
>>> -----Original Message-----
>>> From: Magnus Westerlund [mailto:magnus.westerlund@ericsson.com]
>>> Sent: Thursday, July 14, 2016 4:53 AM
>>> To: Cullen Jennings (fluffy); Black, David
>>> Cc: RTCWeb IETF; Michael Welzl; tsvwg@ietf.org
>>> Subject: Re: [tsvwg] [rtcweb] Fall-back to DSCP 0 in draft-ietf-tsvwg-rtcweb-qos ?
>>>
>>> Den 2016-07-12 kl. 18:19, skrev Cullen Jennings (fluffy):
>>>> short answer here but as David suggested â€¦  some implementation use
>>>> the STUN packets in ICE  or just  in WebRTC style liveness tests to
>>>> do the tests of if a given DSCP works or not. In general ICE is a
>>>> good tool to take a bunch of possible paths, test which work, and
>>>> select the best.
>>> I do agree that how you do the path checks when setting DSCP values != 0
>>> is dependent on the context. For the WebRTC I do agree doing checks
>>> using ICE is quite reasonable.
>>>
>>> We already have similar path testing usages of ICE in the ECN for RTP
>>> specification (RFC6679), see Section 7.2.1. I will note that taking this
>>> as blueprint for DSCP testing, what is needed clearly requires a new
>>> separate specification. The components needs are: 1) A new STUN
>>> parameter to request the ICE peer to echo the DSCP field value received.
>>> 2) A ICE capability parameter to be used in signalling negotiations to
>>> determine capability for this feature. 3) Behaviour specification on how
>>> to test values and interpret responses. This include things like if one
>>> should actually establish multiple candidate pairs one with DSCP testing
>>> and one without?
>>>
>>> So the question here is how to proceed with this issue. So I would
>>> suggest the following way forward.
>>>
>>> 1. draft-ietf-tsvwg-rtcweb-qos identifies the issue and recommends the
>>> user to apply path verification methods but don't specify them.
>>>
>>> 2. Someone takes on the task to write a DSCP path verification extension
>>> to ICE.
>>>
>>> 3. The natural place to indicate the need/recommendation for
>>> implementing this functionality would be in draft-ietf-rtcweb-transports
>>> (Currently in IETF LC). However, here I think we need to have a
>>> discussion if RTCWEB WG wants to only place a suitable warning about the
>>> need, and indicate future forthcoming specification or if we hold this
>>> document up until this solution is available?
>>>
>>>
>>> Cheers
>>>
>>> Magnus Westerlund
>>>
>>> ----------------------------------------------------------------------
>>> Services, Media and Network features, Ericsson Research EAB/TXM
>>> ----------------------------------------------------------------------
>>> Ericsson AB                 | Phone  +46 10 7148287
>>> FÃ¤rÃ¶gatan 6                 | Mobile +46 73 0949079
>>> SE-164 80 Stockholm, Sweden | mailto: magnus.westerlund@ericsson.com
>>> ----------------------------------------------------------------------
>> _______________________________________________
>> rtcweb mailing list
>> rtcweb@ietf.org
>> https://www.ietf.org/mailman/listinfo/rtcweb
>
>


-- 

Magnus Westerlund

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


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To: rtcweb@ietf.org, tsvwg@ietf.org
From: Randell Jesup <randell-ietf@jesup.org>
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Subject: Re: [rtcweb] [tsvwg] Fall-back to DSCP 0 in draft-ietf-tsvwg-rtcweb-qos ?
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On 7/18/2016 4:15 AM, Magnus Westerlund wrote:
> Den 2016-07-18 kl. 09:55, skrev Harald Alvestrand:
>> David,
>>
>> repeating for transparency what I asked in person:
>>
>> Can you post the links to the measurements that led to the conclusion
>> that non-arrival of DSCP-marked packets is a big enough problem that we
>> should develop specific mechanisms to detect it?
>>
>
> https://irtf.org/anrw/2016/anrw16-final17.pdf

Interesting (especially the drops; remarking to 0 is what I'd expect).  
10+ % chance of a non-working path definitely requires us to anticipate it.

Using ICE makes some sense, but I'm a little concerned at multiplying 
the combinatorial number of connectivity checks yet again.

Note this would need to be redone on ICE restart or any connection 
change of course.

Intermediate route changes could also cause it to suddenly fail if DSCP 
is in-use.  Fallback to DSCP 0 (or re-probing) on connection loss (or 
when doing media retransmits?) may also be needed, though restarting ICE 
should work, if ICE can re-establish quickly.

-- 
Randell Jesup -- rjesup a t mozilla d o t com
Please please please don't email randell-ietf@jesup.org!  Way too much spam


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<html>
  <head>
    <meta content=3D"text/html; charset=3Dutf-8" http-equiv=3D"Content-Ty=
pe">
  </head>
  <body bgcolor=3D"#FFFFFF" text=3D"#000000">
    <div class=3D"moz-cite-prefix">On 7/18/2016 4:15 AM, Magnus Westerlun=
d
      wrote:<br>
    </div>
    <blockquote
      cite=3D"mid:f0c9b4a4-afaf-c6dc-856c-dd27bd5049d0@ericsson.com"
      type=3D"cite">Den 2016-07-18 kl. 09:55, skrev Harald Alvestrand:
      <br>
      <blockquote type=3D"cite">David,
        <br>
        <br>
        repeating for transparency what I asked in person:
        <br>
        <br>
        Can you post the links to the measurements that led to the
        conclusion
        <br>
        that non-arrival of DSCP-marked packets is a big enough problem
        that we
        <br>
        should develop specific mechanisms to detect it?
        <br>
        <br>
      </blockquote>
      <br>
      <a class=3D"moz-txt-link-freetext" href=3D"https://irtf.org/anrw/20=
16/anrw16-final17.pdf">https://irtf.org/anrw/2016/anrw16-final17.pdf</a>
      <br>
    </blockquote>
    <br>
    Interesting (especially the drops; remarking to 0 is what I'd
    expect).=C2=A0 10+ % chance of a non-working path definitely requires=
 us
    to anticipate it. <br>
    <br>
    Using ICE makes some sense, but I'm a little concerned at
    multiplying the combinatorial number of connectivity checks yet
    again.<br>
    <br>
    Note this would need to be redone on ICE restart or any connection
    change of course.<br>
    <br>
    Intermediate route changes could also cause it to suddenly fail if
    DSCP is in-use.=C2=A0 Fallback to DSCP 0 (or re-probing) on connectio=
n
    loss (or when doing media retransmits?) may also be needed, though
    restarting ICE should work, if ICE can re-establish quickly.<br>
    <br>
    <pre class=3D"moz-signature" cols=3D"72">--=20
Randell Jesup -- rjesup a t mozilla d o t com
Please please please don't email <a class=3D"moz-txt-link-abbreviated" hr=
ef=3D"mailto:randell-ietf@jesup.org">randell-ietf@jesup.org</a>!  Way too=
 much spam
</pre>
  </body>
</html>

--------------36A82F6212DA3351186BD180--


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Subject: Re: [rtcweb] [tsvwg] Fall-back to DSCP 0 in draft-ietf-tsvwg-rtcweb-qos ?
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See inline.

Den 2016-07-18 kl. 17:24, skrev Randell Jesup:
> On 7/18/2016 4:15 AM, Magnus Westerlund wrote:
>> Den 2016-07-18 kl. 09:55, skrev Harald Alvestrand:
>>> David,
>>>
>>> repeating for transparency what I asked in person:
>>>
>>> Can you post the links to the measurements that led to the conclusion
>>> that non-arrival of DSCP-marked packets is a big enough problem that we
>>> should develop specific mechanisms to detect it?
>>>
>>
>> https://irtf.org/anrw/2016/anrw16-final17.pdf
>
> Interesting (especially the drops; remarking to 0 is what I'd expect).
> 10+ % chance of a non-working path definitely requires us to anticipate it.

Yes, the measured set is small, but the existance proof for black holes 
due to DSCP !=0 is not good.

>
> Using ICE makes some sense, but I'm a little concerned at multiplying
> the combinatorial number of connectivity checks yet again.

This is one of the design questions. For ECN we did the choice to run 
the ECT path verification after we found a working path using the normal 
connectivity checks. The trade-off there was that it was more important 
to find a path with basic connectivity, rather than looking for path 
that supports ECN. For DSCP that choice might be different, but the 
alternative to make the DSCP checks sequential to basic connectivity is 
a clear possibility.


>
> Note this would need to be redone on ICE restart or any connection
> change of course.

Yes
>
> Intermediate route changes could also cause it to suddenly fail if DSCP
> is in-use.  Fallback to DSCP 0 (or re-probing) on connection loss (or
> when doing media retransmits?) may also be needed, though restarting ICE
> should work, if ICE can re-establish quickly.
>

Yes, I think such monitoring is needed.

Cheers

Magnus Westerlund

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From: Harald Alvestrand <harald@alvestrand.no>
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Subject: Re: [rtcweb] [tsvwg] Fall-back to DSCP 0 in draft-ietf-tsvwg-rtcweb-qos ?
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On 07/19/2016 11:10 AM, Magnus Westerlund wrote:
> See inline.
>
> Den 2016-07-18 kl. 17:24, skrev Randell Jesup:
>> On 7/18/2016 4:15 AM, Magnus Westerlund wrote:
>>> Den 2016-07-18 kl. 09:55, skrev Harald Alvestrand:
>>>> David,
>>>>
>>>> repeating for transparency what I asked in person:
>>>>
>>>> Can you post the links to the measurements that led to the conclusion
>>>> that non-arrival of DSCP-marked packets is a big enough problem
>>>> that we
>>>> should develop specific mechanisms to detect it?
>>>>
>>>
>>> https://irtf.org/anrw/2016/anrw16-final17.pdf
>>
>> Interesting (especially the drops; remarking to 0 is what I'd expect).
>> 10+ % chance of a non-working path definitely requires us to
>> anticipate it.
>
> Yes, the measured set is small, but the existance proof for black
> holes due to DSCP !=0 is not good. 

Absolutely.

Note that what has been shown to exist is black holes for TCP with DSCP.
There are arguments that there are TCP-specific design patterns for
cheap firewalls that would cause this, but we have no proof of
non-existence for UDP black holes.

The black hole problem is addressed by the current text in -transport
(section 4.2) as follows:

   The implementation MAY turn off use of DSCP markings if it detects
   symptoms of unexpected behaviour like priority inversion or blocking
   of packets with certain DSCP markings.  The detection of these
   conditions is implementation dependent.

I think it is the last sentence that we're discussing modifications to.
Is that correct?



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Subject: Re: [rtcweb] [tsvwg] Fall-back to DSCP 0 in draft-ietf-tsvwg-rtcweb-qos ?
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> On 20. jul. 2016, at 10.43, Harald Alvestrand <harald@alvestrand.no> =
wrote:
>=20
> On 07/19/2016 11:10 AM, Magnus Westerlund wrote:
>> See inline.
>>=20
>> Den 2016-07-18 kl. 17:24, skrev Randell Jesup:
>>> On 7/18/2016 4:15 AM, Magnus Westerlund wrote:
>>>> Den 2016-07-18 kl. 09:55, skrev Harald Alvestrand:
>>>>> David,
>>>>>=20
>>>>> repeating for transparency what I asked in person:
>>>>>=20
>>>>> Can you post the links to the measurements that led to the =
conclusion
>>>>> that non-arrival of DSCP-marked packets is a big enough problem
>>>>> that we
>>>>> should develop specific mechanisms to detect it?
>>>>>=20
>>>>=20
>>>> https://irtf.org/anrw/2016/anrw16-final17.pdf
>>>=20
>>> Interesting (especially the drops; remarking to 0 is what I'd =
expect).
>>> 10+ % chance of a non-working path definitely requires us to
>>> anticipate it.
>>=20
>> Yes, the measured set is small, but the existance proof for black
>> holes due to DSCP !=3D0 is not good.=20
>=20
> Absolutely.
>=20
> Note that what has been shown to exist is black holes for TCP with =
DSCP.
> There are arguments that there are TCP-specific design patterns for
> cheap firewalls that would cause this, but we have no proof of
> non-existence for UDP black holes.

We got that message from Harald at the MAPRG presentation. Agree - we =
should (and are planning to) try to repeat these tests with UDP.=20

Cheers,
Michael


> The black hole problem is addressed by the current text in -transport
> (section 4.2) as follows:
>=20
>   The implementation MAY turn off use of DSCP markings if it detects
>   symptoms of unexpected behaviour like priority inversion or blocking
>   of packets with certain DSCP markings.  The detection of these
>   conditions is implementation dependent.
>=20
> I think it is the last sentence that we're discussing modifications =
to.
> Is that correct?
>=20
>=20
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb


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From: Ted Hardie <ted.ietf@gmail.com>
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RTCWEB has a critical dependency on the output of both the ICE and MMUSIC
drafts.  In the MMUSIC group just now, the ICE chairs asked for review of
ice-bis:
https://datatracker.ietf.org/doc/draft-ietf-ice-rfc5245bis/ and
https://datatracker.ietf.org/doc/draft-ietf-ice-trickle/.

Progress in MMUSIC (and thus in RTCWEB) depends on these.

Please review them and let the chairs of RTCWEB and ICE know you're taking
them on.

Thanks,

Ted

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<div dir=3D"ltr"><div>RTCWEB has a critical dependency on the output of bot=
h the ICE and MMUSIC drafts.=C2=A0 In the MMUSIC group just now, the ICE ch=
airs asked for review of ice-bis:<br><a href=3D"https://datatracker.ietf.or=
g/doc/draft-ietf-ice-rfc5245bis/">https://datatracker.ietf.org/doc/draft-ie=
tf-ice-rfc5245bis/</a> and <a href=3D"https://datatracker.ietf.org/doc/draf=
t-ietf-ice-trickle/">https://datatracker.ietf.org/doc/draft-ietf-ice-trickl=
e/</a>.=C2=A0 <br><br></div><div>Progress in MMUSIC (and thus in RTCWEB) de=
pends on these.=C2=A0 <br><br>Please review them and let the chairs of RTCW=
EB and ICE know you&#39;re taking them on.<br><br></div><div>Thanks,<br><br=
></div><div>Ted<br></div></div>

--001a113cdc9c720bd20538371446--


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I think the draft should point out the privacy issues only happen with =
split tunnel VPNs and many VPNs are not split.





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Subject: Re: [rtcweb] [AVTCORE] I-D Action: draft-ietf-avtcore-rtp-circuit-breakers-17.txt
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AVTCORE WG,
(cc RTCWEB and TSVWG)

The AD plans to approve this at the end of the Week. So if you have any 
issues with the new text, this is the time to raise it.

Cheers

Magnus


Den 2016-07-20 kl. 10:02, skrev 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 Audio/Video Transport Core Maintenance of the IETF.
>
>         Title           : Multimedia Congestion Control: Circuit Breakers for Unicast RTP Sessions
>         Authors         : Colin Perkins
>                           Varun Singh
> 	Filename        : draft-ietf-avtcore-rtp-circuit-breakers-17.txt
> 	Pages           : 25
> 	Date            : 2016-07-20
>
> Abstract:
>    The Real-time Transport Protocol (RTP) is widely used in telephony,
>    video conferencing, and telepresence applications.  Such applications
>    are often run on best-effort UDP/IP networks.  If congestion control
>    is not implemented in these applications, then network congestion can
>    lead to uncontrolled packet loss, and a resulting deterioration of
>    the user's multimedia experience.  The congestion control algorithm
>    acts as a safety measure, stopping RTP flows from using excessive
>    resources, and protecting the network from overload.  At the time of
>    this writing, however, while there are several proprietary solutions,
>    there is no standard algorithm for congestion control of interactive
>    RTP flows.
>
>    This document does not propose a congestion control algorithm.  It
>    instead defines a minimal set of RTP circuit breakers: conditions
>    under which an RTP sender needs to stop transmitting media data, to
>    protect the network from excessive congestion.  It is expected that,
>    in the absence of long-lived excessive congestion, RTP applications
>    running on best-effort IP networks will be able to operate without
>    triggering these circuit breakers.  To avoid triggering the RTP
>    circuit breaker, any standards-track congestion control algorithms
>    defined for RTP will need to operate within the envelope set by these
>    RTP circuit breaker algorithms.
>
>
> The IETF datatracker status page for this draft is:
> https://datatracker.ietf.org/doc/draft-ietf-avtcore-rtp-circuit-breakers/
>
> There's also a htmlized version available at:
> https://tools.ietf.org/html/draft-ietf-avtcore-rtp-circuit-breakers-17
>
> A diff from the previous version is available at:
> https://www.ietf.org/rfcdiff?url2=draft-ietf-avtcore-rtp-circuit-breakers-17
>
>
> Please note that it may take a couple of minutes from the time of submission
> until the htmlized version and diff are available at tools.ietf.org.
>
> Internet-Drafts are also available by anonymous FTP at:
> ftp://ftp.ietf.org/internet-drafts/
>
> _______________________________________________
> Audio/Video Transport Core Maintenance
> avt@ietf.org
> https://www.ietf.org/mailman/listinfo/avt
>


-- 

Magnus Westerlund

----------------------------------------------------------------------
Services, Media and Network features, Ericsson Research EAB/TXM
----------------------------------------------------------------------
Ericsson AB                 | Phone  +46 10 7148287
Färögatan 6                 | Mobile +46 73 0949079
SE-164 80 Stockholm, Sweden | mailto: magnus.westerlund@ericsson.com
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From nobody Tue Jul 26 02:35:41 2016
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From: Christer Holmberg <christer.holmberg@ericsson.com>
To: "rtcweb@ietf.org" <rtcweb@ietf.org>
Thread-Topic: JSEP: Associate answer with offer
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Subject: [rtcweb] JSEP: Associate answer with offer
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Hi,

Assume the following:


1)      Alice calls createOffer(), and sends the generated offer (offer X) =
towards Bob.

2)      Alice calls createOffer() again, and sends the generated offer (off=
er Y) towards Bob.

3)      Alice receives an answer from Bob.

How does Alice know whether the answer is associated with offer X (in which=
 case I assume Alice should discard it) or with offer Y?

Since JSEP allows multiple outstanding offers (per section 3.2), there need=
s to be a mechanism for associating an answer with a specific offer.

Regards,

Christer

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<p class=3D"MsoNormal">Hi,<o:p></o:p></p>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
<p class=3D"MsoNormal">Assume the following:<o:p></o:p></p>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
<p class=3D"MsoListParagraph" style=3D"text-indent:-18.0pt;mso-list:l0 leve=
l1 lfo2"><![if !supportLists]><span style=3D"mso-list:Ignore">1)<span style=
=3D"font:7.0pt &quot;Times New Roman&quot;">&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;
</span></span><![endif]>Alice calls createOffer(), and sends the generated =
offer (offer X) towards Bob.<o:p></o:p></p>
<p class=3D"MsoListParagraph" style=3D"text-indent:-18.0pt;mso-list:l0 leve=
l1 lfo2"><![if !supportLists]><span style=3D"mso-list:Ignore">2)<span style=
=3D"font:7.0pt &quot;Times New Roman&quot;">&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;
</span></span><![endif]>Alice calls createOffer() again, and sends the gene=
rated offer (offer Y) towards Bob.<o:p></o:p></p>
<p class=3D"MsoListParagraph" style=3D"text-indent:-18.0pt;mso-list:l0 leve=
l1 lfo2"><![if !supportLists]><span style=3D"mso-list:Ignore">3)<span style=
=3D"font:7.0pt &quot;Times New Roman&quot;">&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;
</span></span><![endif]>Alice receives an answer from Bob.<o:p></o:p></p>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
<p class=3D"MsoNormal">How does Alice know whether the answer is associated=
 with offer X (in which case I assume Alice should discard it) or with offe=
r Y?<o:p></o:p></p>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
<p class=3D"MsoNormal">Since JSEP allows multiple outstanding offers (per s=
ection 3.2), there needs to be a mechanism for associating an answer with a=
 specific offer.<o:p></o:p></p>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
<p class=3D"MsoNormal">Regards,<o:p></o:p></p>
<p class=3D"MsoNormal"><o:p>&nbsp;</o:p></p>
<p class=3D"MsoNormal">Christer<o:p></o:p></p>
</div>
</body>
</html>

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From: Ted Hardie <ted.ietf@gmail.com>
Date: Tue, 26 Jul 2016 11:48:04 -0700
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Many thanks to Bo for the notes from our session.  Please review and
comment on the list, so that we can update them for the mintues,

Thanks,

Ted


Hi!
>
>
>
> Here are my notes from the RTCWEB session July 22.
>
>
>
> Cheers,
>
> Bo
>
------------------------------
RTCWEB Administrivia

Note taker: Bo Burman

Jabber scribe: Olle E. Johansson

The chairs pointed out that the WG progress depends on a few drafts in
other WG (ICE bis
<https://datatracker.ietf.org/doc/draft-ietf-ice-rfc5245bis/>, Trickle ICE
<https://datatracker.ietf.org/doc/draft-ietf-ice-trickle/> and STUN bis
<https://datatracker.ietf.org/doc/draft-ietf-tram-stunbis/>) that urgently
need review and solicited volunteers to review them. Eric Rescorla and
Magnus Westerlund accepted to review.


DSCP Black-holing Issue

David Black (TSVWG co-chair) presented the DSCP black-hole issue with
-rtcweb-transports
<https://datatracker.ietf.org/doc/draft-ietf-rtcweb-transports/> draft that
was recently discussed on the list. This issue needs to be solved and
described, even though both -rtcweb-transports and the referenced
draft-ietf-tsvwg-rtcweb-qos has gone through IESG review. Magnus Westerlund
has suggested a solution to the list, but what should the
-rtcweb-transports draft say about DSCP black-holing and the possibility to
use ICE to avoid it?

The WG discussed this and concluded that the issue should be described by
the -rtcweb-transports draft. Ted Hardie summarized the discussion by
suggesting a text formulation for a resolution that seemed acceptable to
the WG: =E2=80=9CWe will treat DSCP-induced path failure parallel with othe=
r types
of path failures and resolve it by using ICE restart. Note: There is a
problem with multiple DSCP codepoints on a single transport, where one
might be blocked and other might get through. In this case, the ICE probes,
using one DSCP codepoint, may succeed while others fail. This is complex
and should be warned about. A likely viable solution is ICE restart with
DSCP markings turned off, but detection requires watching the
multiple-DCSP-codepoint-using channels for differential failures=E2=80=9D. =
If there
are other proposals for resolution, please contact Harald. Cullen Jennings
asked David if this solution was acceptable, but David wanted to see the
text proposal. The -rtcweb-transports author Harald Alvestrand took on the
action item and will work with Justin Uberti to send a text proposal to the
list.


Draft status update JSEP

A number of tracker issues were discussed and concluded on.

*Codecs in re-offer [PR #269]*

The proposed solution is that codecs in an SDP re-offer can only be a
subset of what was negotiated, not what the remote sent (which will be
strange if the re-offering part is the originally answering one). Paul
Kyzivat (remote) commented that this is an old story in SIP Offer/Answer
and turned out to be a bad answer. The topic is discussed in RFC 6337. In
an SDP re-offer, you should offer everything you are capable of and think
is interesting. A brief discussion concluded that this would cause change
of codec depending on who sends the re-offer, for example if A and B offer
(foo, bar) and (bar, foo), respectively (with order reflecting
preferences). So, the SDP re-offer should, as proposed in the slide,
include a subset of what was negotiated before. Paul was not happy with
this, but did not think it worth pursuing.

*Enforce max-bundle on offer processing [PR #282]*

There was a long discussion on how to enforce max-bundle; the conclusion is
basically to go with the proposal on the slide (only the first m=3D line is
accepted), per design and for symmetry reasons. A clarification to that
rule was found when an offer is received with some m=3D lines bundled and
some not; you accept all m=3D lines that are possible to bundle with the
first m=3D line.

*Rewrite LS handling text to indicate edge cases and that we're living with
them [PR #276 & #263]*

The conclusion was to go with the proposal, even if this is a simple LS
group strategy. Justin Uberti commented (via Jabber) that it is always
possible to let the JavaScript (app) to create new m=3D lines that have a n=
ew
LS group if it wants to.

*addIceCandidate() and ICE restart [#250]*

The conclusion was that we need an unambiguous indication in ICE: =E2=80=9C=
this is
the last candidate=E2=80=9D for a certain ufrag, which should (probably) be
specified in JSEP and not in any ICE WG specification.

*Roll back ICE restart [#250]*

Decision to go with what is described on the slide.

*SDP o=3D line increment [#239]*

While the proposed solution on the slide was accepted, it was noted that
the proposal on the slide goes against what was agreed at last meeting.
Emil Ivov also commented that Trickle ICE for SIP does exactly what is
described on the slide. Jonathan Lennox said that we should make sure this
is compatible with whatever is done in trickle ICE SIP stacks.

*addTrack assignment [#288]*

Peter Thatcher commented that it is necessary to scope this down even more
compared to what is proposed on the slide, to in some cases not allow
re-using m-sections after an ICE restart. There=E2=80=99s still some risk t=
hat
another edge case will turn up. Adam Roach suggested that a better approach
might be to instead detail what =E2=80=9Ccompatible=E2=80=9D means. Justin =
Uberti wondered
how to normatively define that m-sections are =E2=80=9Ccompatible=E2=80=9D,=
 but supports
adding stricter text that allows for better determinism. Peter said that it
is always possible to use addTransceiver to avoid any ambiguity with
addTrack. Ted commented that this is a substantial algorithm change. He
then gave Peter and Justin the task to generate fresh text that captures
Peter=E2=80=99s point for scoping down even more and send it to the mailing=
 list.


draft-ietf-rtcweb-sdp

*Open issue: a=3Drtcp usage*

There are three apparently conflicting statements around use of =E2=80=9Ca=
=3Drtcp=E2=80=9D in
JSEP, BUNDLE-31 and ICE-SIP-SDP-08. This seems more appropriate to discuss
in MMUSIC WG. JSEP should be consistent with whatever solution is specified
by BUNDLE and ICE-SIP-SDP, but should be allowed to make it clear what the
implementation should do. Bug #296 was opened in the JSEP tracker. Ted
suggested to send the entire slide deck to MMUSIC WG and ask for time to
present it during their session on Friday July 22nd.

*Open issue: a=3Dfingerprint*

BUNDLE-31 was thought to be correct and JSEP should align with that.

*Open issue: a=3Drtcp-mux-only*

The conclusion is to update JSEP to align with BUNDLE-31. JSEP bug #297
opened.

*JSEP SDP Usage vs Generic SDP Usage*

Christer commented that it would be good to note that provisional answer is
used in JSEP, but is not possible in generic (RFC 3264) SDP usage without
specific signaling.

*ICEbis vs RFC 5245 reference*

Cullen suggested that this draft should reference ICE bis, trickle ICE and
STUN bis rather than the predecessors, which was not challenged by the WG.

*Next steps*

This draft is really hard to review and the chairs are really grateful for
Paul=E2=80=99s ongoing review of it, but it would be great if someone else =
was able
to review it too.

Jonathan commented that it would be good if the SDP examples in this draft
could be automatically extracted and used for testing. Cullen explained
that this is the intent. He also clarified that the examples are currently
not machine generated, mainly because none of the current implementations
support all of what is in this draft and it would take much work to support
all of it. If someone is able to provide machine-generated examples that
could replace the hand-generated ones currently in the draft, they would
happily be accepted. Justin Uberti offered via Jabber to write some
JavaScript to machine-generate SDP examples. Tim Panton proposed to have a
look at the possibility having ORTC generate SDP.


Summing Up

Ted stressed once more that review of RTCWEB documents, mainly the JSEP
draft, is really needed if we are going to be able to take all needed
documents to WGLC before IETF 97 in Seoul in November.

Alissa Cooper (ART co-AD) asked if the previously discussed, organized
draft review activities are still planned. The RTCWEB chairs explained that
the situation has proven to be less bad than was originally thought, but
some reviews will likely will needed. We will likely need some review
activity during September, maybe as virtual interim meetings or in some
type of review teams, which will hopefully take us to WGLC for all the
documents before IETF 97 in Seoul.

--001a1147191011b9d505388e5977
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<div dir=3D"ltr"><div><div>Many thanks to Bo for the notes from our session=
.=C2=A0 Please review and comment on the list, so that we can update them f=
or the mintues,<br><br></div>Thanks,<br><br></div>Ted<br><div><div><div><di=
v class=3D"gmail_quote"><br><br>





<div link=3D"blue" vlink=3D"purple" lang=3D"EN-US">
<div>
<blockquote style=3D"margin:0px 0px 0px 0.8ex;border-left:1px solid rgb(204=
,204,204);padding-left:1ex" class=3D"gmail_quote"><p class=3D"MsoNormal"><s=
pan lang=3D"SV">Hi!</span></p>
<p class=3D"MsoNormal"><span lang=3D"SV">=C2=A0</span></p>
<p class=3D"MsoNormal">Here are my notes from the RTCWEB session July 22.</=
p>
<p class=3D"MsoNormal">=C2=A0</p>
<p class=3D"MsoNormal">Cheers,</p>
<p class=3D"MsoNormal">Bo</p></blockquote>
<div class=3D"MsoNormal" style=3D"text-align:center" align=3D"center">
<hr size=3D"2" width=3D"100%" align=3D"center">
</div>
<h1><span style=3D"color:#365f91" lang=3D"SV">RTCWEB<u></u><u></u></span></=
h1>
<h2><span style=3D"color:#365f91" lang=3D"SV">Administrivia<u></u><u></u></=
span></h2>
<p class=3D"MsoNormal"><span lang=3D"SV">Note taker: Bo Burman<u></u><u></u=
></span></p>
<p class=3D"MsoNormal">Jabber scribe: Olle E. Johansson<u></u><u></u></p>
<p class=3D"MsoNormal">The chairs pointed out that the WG progress depends =
on a few drafts in other WG (<a href=3D"https://datatracker.ietf.org/doc/dr=
aft-ietf-ice-rfc5245bis/" target=3D"_blank">ICE bis</a>,
<a href=3D"https://datatracker.ietf.org/doc/draft-ietf-ice-trickle/" target=
=3D"_blank">Trickle ICE</a> and
<a href=3D"https://datatracker.ietf.org/doc/draft-ietf-tram-stunbis/" targe=
t=3D"_blank">STUN bis</a>) that urgently need review and solicited voluntee=
rs to review them. Eric Rescorla and Magnus Westerlund accepted to review.<=
u></u><u></u></p>
<p class=3D"MsoNormal"><u></u>=C2=A0<u></u></p>
<h2><span style=3D"color:#365f91">DSCP Black-holing Issue<u></u><u></u></sp=
an></h2>
<p class=3D"MsoNormal">David Black (TSVWG co-chair) presented the DSCP blac=
k-hole issue with
<a href=3D"https://datatracker.ietf.org/doc/draft-ietf-rtcweb-transports/" =
target=3D"_blank">-rtcweb-transports</a> draft that was recently discussed =
on the list. This issue needs to be solved and described, even though both =
-rtcweb-transports and the referenced draft-ietf-tsvwg-rtcweb-qos
 has gone through IESG review. Magnus Westerlund has suggested a solution t=
o the list, but what should the -rtcweb-transports draft say about DSCP bla=
ck-holing and the possibility to use ICE to avoid it?<u></u><u></u></p>
<p class=3D"MsoNormal">The WG discussed this and concluded that the issue s=
hould be described by the -rtcweb-transports draft. Ted Hardie summarized t=
he discussion by suggesting a text formulation for a resolution that seemed=
 acceptable to the WG: =E2=80=9CWe will treat
 DSCP-induced path failure parallel with other types of path failures and r=
esolve it by using ICE restart. Note: There is a problem with multiple DSCP=
 codepoints on a single transport, where one might be blocked and other mig=
ht get through. In this case, the
 ICE probes, using one DSCP codepoint, may succeed while others fail. This =
is complex and should be warned about. A likely viable solution is ICE rest=
art with DSCP markings turned off, but detection requires watching the mult=
iple-DCSP-codepoint-using channels
 for differential failures=E2=80=9D. If there are other proposals for resol=
ution, please contact Harald. Cullen Jennings asked David if this solution =
was acceptable, but David wanted to see the text proposal. The -rtcweb-tran=
sports author Harald Alvestrand took on
 the action item and will work with Justin Uberti to send a text proposal t=
o the list.<u></u><u></u></p>
<p class=3D"MsoNormal"><u></u>=C2=A0<u></u></p>
<h2><span style=3D"color:#365f91">Draft status update<u></u><u></u></span><=
/h2>
<h3><span style=3D"color:#243f60">JSEP<u></u><u></u></span></h3>
<p class=3D"MsoNormal">A number of tracker issues were discussed and conclu=
ded on.<u></u><u></u></p>
<p class=3D"MsoNormal"><u>Codecs in re-offer [PR #269]<u></u><u></u></u></p=
>
<p class=3D"MsoNormal">The proposed solution is that codecs in an SDP re-of=
fer can only be a subset of what was negotiated, not what the remote sent (=
which will be strange if the re-offering part is the originally answering o=
ne). Paul Kyzivat (remote) commented
 that this is an old story in SIP Offer/Answer and turned out to be a bad a=
nswer. The topic is discussed in RFC 6337. In an SDP re-offer, you should o=
ffer everything you are capable of and think is interesting. A brief discus=
sion concluded that this would cause
 change of codec depending on who sends the re-offer, for example if A and =
B offer (foo, bar) and (bar, foo), respectively (with order reflecting pref=
erences). So, the SDP re-offer should, as proposed in the slide, include a =
subset of what was negotiated before.
 Paul was not happy with this, but did not think it worth pursuing.<u></u><=
u></u></p>
<p class=3D"MsoNormal"><u>Enforce max-bundle on offer processing [PR #282]<=
u></u><u></u></u></p>
<p class=3D"MsoNormal">There was a long discussion on how to enforce max-bu=
ndle; the conclusion is basically to go with the proposal on the slide (onl=
y the first m=3D line is accepted), per design and for symmetry reasons. A =
clarification to that rule was found
 when an offer is received with some m=3D lines bundled and some not; you a=
ccept all m=3D lines that are possible to bundle with the first m=3D line.<=
u></u><u></u></p>
<p class=3D"MsoNormal"><u>Rewrite LS handling text to indicate edge cases a=
nd that we&#39;re living with them [PR #276 &amp; #263]<u></u><u></u></u></=
p>
<p class=3D"MsoNormal">The conclusion was to go with the proposal, even if =
this is a simple LS group strategy. Justin Uberti commented (via Jabber) th=
at it is always possible to let the JavaScript (app) to create new m=3D lin=
es that have a new LS group if it wants
 to.<u></u><u></u></p>
<p class=3D"MsoNormal"><u>addIceCandidate() and ICE restart [#250]<u></u><u=
></u></u></p>
<p class=3D"MsoNormal">The conclusion was that we need an unambiguous indic=
ation in ICE: =E2=80=9Cthis is the last candidate=E2=80=9D for a certain uf=
rag, which should (probably) be specified in JSEP and not in any ICE WG spe=
cification.<u></u><u></u></p>
<p class=3D"MsoNormal"><u>Roll back ICE restart [#250]<u></u><u></u></u></p=
>
<p class=3D"MsoNormal">Decision to go with what is described on the slide.<=
u></u><u></u></p>
<p class=3D"MsoNormal"><u>SDP o=3D line increment [#239]<u></u><u></u></u><=
/p>
<p class=3D"MsoNormal">While the proposed solution on the slide was accepte=
d, it was noted that the proposal on the slide goes against what was agreed=
 at last meeting. Emil Ivov also commented that Trickle ICE for SIP does ex=
actly what is described on the slide.
 Jonathan Lennox said that we should make sure this is compatible with what=
ever is done in trickle ICE SIP stacks.<u></u><u></u></p>
<p class=3D"MsoNormal"><u>addTrack assignment [#288]<u></u><u></u></u></p>
<p class=3D"MsoNormal">Peter Thatcher commented that it is necessary to sco=
pe this down even more compared to what is proposed on the slide, to in som=
e cases not allow re-using m-sections after an ICE restart. There=E2=80=99s=
 still some risk that another edge case will
 turn up. Adam Roach suggested that a better approach might be to instead d=
etail what =E2=80=9Ccompatible=E2=80=9D means. Justin Uberti wondered how t=
o normatively define that m-sections are =E2=80=9Ccompatible=E2=80=9D, but =
supports adding stricter text that allows for better determinism.
 Peter said that it is always possible to use addTransceiver to avoid any a=
mbiguity with addTrack. Ted commented that this is a substantial algorithm =
change. He then gave Peter and Justin the task to generate fresh text that =
captures Peter=E2=80=99s point for scoping
 down even more and send it to the mailing list.<u></u><u></u></p>
<p class=3D"MsoNormal"><u></u>=C2=A0<u></u></p>
<h3><span style=3D"color:#243f60">draft-ietf-rtcweb-sdp<u></u><u></u></span=
></h3>
<p class=3D"MsoNormal"><u>Open issue: a=3Drtcp usage<u></u><u></u></u></p>
<p class=3D"MsoNormal">There are three apparently conflicting statements ar=
ound use of =E2=80=9Ca=3Drtcp=E2=80=9D in JSEP, BUNDLE-31 and ICE-SIP-SDP-0=
8. This seems more appropriate to discuss in MMUSIC WG. JSEP should be cons=
istent with whatever solution is specified by BUNDLE
 and ICE-SIP-SDP, but should be allowed to make it clear what the implement=
ation should do. Bug #296 was opened in the JSEP tracker. Ted suggested to =
send the entire slide deck to MMUSIC WG and ask for time to present it duri=
ng their session on Friday July
 22nd.<u></u><u></u></p>
<p class=3D"MsoNormal"><u>Open issue: a=3Dfingerprint<u></u><u></u></u></p>
<p class=3D"MsoNormal">BUNDLE-31 was thought to be correct and JSEP should =
align with that.<u></u><u></u></p>
<p class=3D"MsoNormal"><u>Open issue: a=3Drtcp-mux-only<u></u><u></u></u></=
p>
<p class=3D"MsoNormal">The conclusion is to update JSEP to align with BUNDL=
E-31. JSEP bug #297 opened.<u></u><u></u></p>
<p class=3D"MsoNormal"><u>JSEP SDP Usage vs Generic SDP Usage<u></u><u></u>=
</u></p>
<p class=3D"MsoNormal">Christer commented that it would be good to note tha=
t provisional answer is used in JSEP, but is not possible in generic (RFC 3=
264) SDP usage without specific signaling.<u></u><u></u></p>
<p class=3D"MsoNormal"><u>ICEbis vs RFC 5245 reference<u></u><u></u></u></p=
>
<p class=3D"MsoNormal">Cullen suggested that this draft should reference IC=
E bis, trickle ICE and STUN bis rather than the predecessors, which was not=
 challenged by the WG.<u></u><u></u></p>
<p class=3D"MsoNormal"><u>Next steps<u></u><u></u></u></p>
<p class=3D"MsoNormal">This draft is really hard to review and the chairs a=
re really grateful for Paul=E2=80=99s ongoing review of it, but it would be=
 great if someone else was able to review it too.<u></u><u></u></p>
<p class=3D"MsoNormal">Jonathan commented that it would be good if the SDP =
examples in this draft could be automatically extracted and used for testin=
g. Cullen explained that this is the intent. He also clarified that the exa=
mples are currently not machine generated,
 mainly because none of the current implementations support all of what is =
in this draft and it would take much work to support all of it. If someone =
is able to provide machine-generated examples that could replace the hand-g=
enerated ones currently in the draft,
 they would happily be accepted. Justin Uberti offered via Jabber to write =
some JavaScript to machine-generate SDP examples. Tim Panton proposed to ha=
ve a look at the possibility having ORTC generate SDP.<u></u><u></u></p>
<p class=3D"MsoNormal"><u></u>=C2=A0<u></u></p>
<h2><span style=3D"color:#365f91">Summing Up<u></u><u></u></span></h2>
<p class=3D"MsoNormal">Ted stressed once more that review of RTCWEB documen=
ts, mainly the JSEP draft, is really needed if we are going to be able to t=
ake all needed documents to WGLC before IETF 97 in Seoul in November.<u></u=
><u></u></p>
<p class=3D"MsoNormal">Alissa Cooper (ART co-AD) asked if the previously di=
scussed, organized draft review activities are still planned. The RTCWEB ch=
airs explained that the situation has proven to be less bad than was origin=
ally thought, but some reviews will
 likely will needed. We will likely need some review activity during Septem=
ber, maybe as virtual interim meetings or in some type of review teams, whi=
ch will hopefully take us to WGLC for all the documents before IETF 97 in S=
eoul.<u></u><u></u></p>
<p class=3D"MsoNormal"><u></u>=C2=A0<u></u></p>
</div>
</div>

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

--001a1147191011b9d505388e5977--


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From: Andy Hutton <andyhutton.ietf@gmail.com>
Date: Wed, 27 Jul 2016 15:22:05 +0100
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Subject: [rtcweb] JSEP Direction in offers.
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https://tools.ietf.org/html/draft-ietf-rtcweb-jsep-15#section-5.2.4 states:

"[RFC3264] direction attributes (defined in Section 6.1) in offers
are chosen according to the states of the RtpSender and RtpReceiver of
a given RtpTransceiver"

I spent some time thinking about this and trying to work out whether it
worked for re-offers and my conclusion is that the above statement is not
strictly correct but hopefully this is not a  big problem. It all depends
on what the above statement means by "the states of the
RtpSender/Receiver". My understanding is that to change the direction an
application calls setDirection() on the transceiver before createOffer and
this would cause the direction to be set accordingly in the offer but
actually the RTCRtpTransceiverDirection does not change until setLocal or
setRemote is called.

Would it make sense to change the statement in 5.2.4 to:

"[RFC3264] direction attributes (defined in Section 6.1) in offers
are chosen according to the currently preferred direction requested using
setDirection() or if no change is requested the current direction of the
RtpTransceiver"

Regards
Andy

--001a1145a578ea4bee05389ebdb4
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<div dir=3D"ltr"><br><a href=3D"https://tools.ietf.org/html/draft-ietf-rtcw=
eb-jsep-15#section-5.2.4">https://tools.ietf.org/html/draft-ietf-rtcweb-jse=
p-15#section-5.2.4</a> states:<br><br>&quot;[RFC3264] direction attributes =
(defined in Section 6.1) in offers are=C2=A0chosen according to the states =
of the RtpSender and RtpReceiver of a=C2=A0given RtpTransceiver&quot;<br><d=
iv><br></div><div>I spent some time thinking about this and trying to work =
out whether it worked for re-offers and my conclusion is that the above sta=
tement is not strictly correct but hopefully this is not a =C2=A0big proble=
m. It all depends on what the above statement means by &quot;the states of =
the RtpSender/Receiver&quot;. My understanding is that to change the direct=
ion an application calls setDirection() on the transceiver before createOff=
er and this would cause the direction to be set accordingly in the offer bu=
t actually the RTCRtpTransceiverDirection does not change until setLocal or=
 setRemote is called. =C2=A0</div><div><br></div><div>Would it make sense t=
o change the statement in 5.2.4 to:</div><div><br></div><div>&quot;[RFC3264=
] direction attributes (defined in Section 6.1) in offers are=C2=A0chosen a=
ccording to the currently preferred direction requested using setDirection(=
) or if no change is requested the current direction of the RtpTransceiver&=
quot;<br></div><div><br></div><div>Regards</div><div>Andy</div></div>

--001a1145a578ea4bee05389ebdb4--


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Subject: [rtcweb] Codecs in re-offer [PR #269]
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The issue of whether a re-offer in WebRTC must be limited to the set of
codec's within the previously negotiated envelope was discussed during
IETF96 but I don't think a conclusion was reached.

Justin mentioned that this had been discussed previously so sorry for
bringing it up again I cannot remember all the old arguments and a quick
search did not find that discussion.

Personally I don't get the argument for making this a restriction in WebRTC
even if there are potential workaround solutions like creating a new PC. As
was mentioned in the meeting there are ways around the toggling of codec's
depending on who creates the next offer without imposing a restriction
which is likely to be a cause of interoperability issues.

At a minimum I would suggest that WebRTC implementations must be able to
receive and process and offer/answer which is not within the previously
negotiated envelop and that should be stated something in JSEP.  However I
think it would also be preferable if the re-offer could contain the full
set of capabilities.

Regards
Andy

--001a1145a57834ce5b05389ef3b7
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<div dir=3D"ltr"><br><div>The issue of whether a re-offer in WebRTC must be=
 limited to the set of codec&#39;s within the previously negotiated envelop=
e was discussed during IETF96 but I don&#39;t think a conclusion was reache=
d.</div><div><br></div><div>Justin mentioned that this had been discussed p=
reviously so sorry for bringing it up again I cannot remember all the old a=
rguments and a quick search did not find that discussion.</div><div><br></d=
iv><div>Personally I don&#39;t get the argument for making this a restricti=
on in WebRTC even if there are potential workaround solutions like creating=
 a new PC. As was mentioned in the meeting there are ways around the toggli=
ng of codec&#39;s depending on who creates the next offer without imposing =
a restriction which is likely to be a cause of interoperability issues.=C2=
=A0</div><div><br></div><div>At a minimum I would suggest that WebRTC imple=
mentations must be able to receive and process and offer/answer which is no=
t within the previously negotiated envelop and that should be stated someth=
ing in JSEP.=C2=A0 However I think it would also be preferable if the re-of=
fer could contain the full set of capabilities.</div><div><br></div><div>Re=
gards</div><div>Andy</div></div>

--001a1145a57834ce5b05389ef3b7--


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Subject: Re: [rtcweb] JSEP Direction in offers.
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This text needs to be revised to reflect the operation of the setDirection A=
PI. In particular, neither RtpReceiver nor RtpSender have a state attribute a=
s referred to in this text.

See the slides from the last WebRTC WG virtual interim (starting at slide 19=
):
https://www.w3.org/2011/04/webrtc/wiki/images/1/12/WebRTCWG-2016-06-28.pdf

> On Jul 27, 2016, at 7:22 AM, Andy Hutton <andyhutton.ietf@gmail.com> wrote=
:
>=20
>=20
> https://tools.ietf.org/html/draft-ietf-rtcweb-jsep-15#section-5.2.4 states=
:
>=20
> "[RFC3264] direction attributes (defined in Section 6.1) in offers are cho=
sen according to the states of the RtpSender and RtpReceiver of a given RtpT=
ransceiver"
>=20
> I spent some time thinking about this and trying to work out whether it wo=
rked for re-offers and my conclusion is that the above statement is not stri=
ctly correct but hopefully this is not a  big problem. It all depends on wha=
t the above statement means by "the states of the RtpSender/Receiver". My un=
derstanding is that to change the direction an application calls setDirectio=
n() on the transceiver before createOffer and this would cause the direction=
 to be set accordingly in the offer but actually the RTCRtpTransceiverDirect=
ion does not change until setLocal or setRemote is called. =20
>=20
> Would it make sense to change the statement in 5.2.4 to:
>=20
> "[RFC3264] direction attributes (defined in Section 6.1) in offers are cho=
sen according to the currently preferred direction requested using setDirect=
ion() or if no change is requested the current direction of the RtpTransceiv=
er"
>=20
> Regards
> Andy
> _______________________________________________
> 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=3D=
utf-8"></head><body dir=3D"auto"><div></div><div>This text needs to be revis=
ed to reflect the operation of the setDirection API. In particular, neither R=
tpReceiver nor RtpSender have a state attribute as referred to in this text.=
</div><div><br></div><div>See the slides from the last WebRTC WG virtual int=
erim (starting at slide 19):</div><div><a href=3D"https://www.w3.org/2011/04=
/webrtc/wiki/images/1/12/WebRTCWG-2016-06-28.pdf">https://www.w3.org/2011/04=
/webrtc/wiki/images/1/12/WebRTCWG-2016-06-28.pdf</a></div><div><br>On Jul 27=
, 2016, at 7:22 AM, Andy Hutton &lt;<a href=3D"mailto:andyhutton.ietf@gmail.=
com">andyhutton.ietf@gmail.com</a>&gt; wrote:<br><br></div><blockquote type=3D=
"cite"><div><div dir=3D"ltr"><br><a href=3D"https://tools.ietf.org/html/draf=
t-ietf-rtcweb-jsep-15#section-5.2.4">https://tools.ietf.org/html/draft-ietf-=
rtcweb-jsep-15#section-5.2.4</a> states:<br><br>"[RFC3264] direction attribu=
tes (defined in Section 6.1) in offers are&nbsp;chosen according to the stat=
es of the RtpSender and RtpReceiver of a&nbsp;given RtpTransceiver"<br><div>=
<br></div><div>I spent some time thinking about this and trying to work out w=
hether it worked for re-offers and my conclusion is that the above statement=
 is not strictly correct but hopefully this is not a &nbsp;big problem. It a=
ll depends on what the above statement means by "the states of the RtpSender=
/Receiver". My understanding is that to change the direction an application c=
alls setDirection() on the transceiver before createOffer and this would cau=
se the direction to be set accordingly in the offer but actually the RTCRtpT=
ransceiverDirection does not change until setLocal or setRemote is called. &=
nbsp;</div><div><br></div><div>Would it make sense to change the statement i=
n 5.2.4 to:</div><div><br></div><div>"[RFC3264] direction attributes (define=
d in Section 6.1) in offers are&nbsp;chosen according to the currently prefe=
rred direction requested using setDirection() or if no change is requested t=
he current direction of the RtpTransceiver"<br></div><div><br></div><div>Reg=
ards</div><div>Andy</div></div>
</div></blockquote><blockquote type=3D"cite"><div><span>____________________=
___________________________</span><br><span>rtcweb mailing list</span><br><s=
pan><a href=3D"mailto:rtcweb@ietf.org">rtcweb@ietf.org</a></span><br><span><=
a href=3D"https://www.ietf.org/mailman/listinfo/rtcweb">https://www.ietf.org=
/mailman/listinfo/rtcweb</a></span><br></div></blockquote></body></html>=

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From: Roman Shpount <roman@telurix.com>
Date: Wed, 27 Jul 2016 12:00:29 -0400
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Subject: Re: [rtcweb] Codecs in re-offer [PR #269]
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I agree that re-offer should contain the full set of codecs or this will
cause interesting problems with 3pcc or SIP interop. The order of codecs
should ideally be the same as currently negotiated to avoid codec flips.
Essentially, what this means all supported codecs not currently negotiated
should be added to the end of the offer after the currently negotiated ones.

Regards,

_____________
Roman Shpount

On Wed, Jul 27, 2016 at 10:36 AM, Andy Hutton <andyhutton.ietf@gmail.com>
wrote:

>
> The issue of whether a re-offer in WebRTC must be limited to the set of
> codec's within the previously negotiated envelope was discussed during
> IETF96 but I don't think a conclusion was reached.
>
> Justin mentioned that this had been discussed previously so sorry for
> bringing it up again I cannot remember all the old arguments and a quick
> search did not find that discussion.
>
> Personally I don't get the argument for making this a restriction in
> WebRTC even if there are potential workaround solutions like creating a new
> PC. As was mentioned in the meeting there are ways around the toggling of
> codec's depending on who creates the next offer without imposing a
> restriction which is likely to be a cause of interoperability issues.
>
> At a minimum I would suggest that WebRTC implementations must be able to
> receive and process and offer/answer which is not within the previously
> negotiated envelop and that should be stated something in JSEP.  However I
> think it would also be preferable if the re-offer could contain the full
> set of capabilities.
>
> Regards
> Andy
>
> _______________________________________________
> rtcweb mailing list
> rtcweb@ietf.org
> https://www.ietf.org/mailman/listinfo/rtcweb
>
>

--001a113ec7ecd1c6120538a01dc1
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<div dir=3D"ltr">I agree that re-offer should contain the full set of codec=
s or this will cause interesting problems with 3pcc or SIP interop. The ord=
er of codecs should ideally be the same as currently negotiated to avoid co=
dec flips. Essentially, what this means all supported codecs not currently =
negotiated should be added to the end of the offer after the currently nego=
tiated ones.<div><br></div><div>Regards,</div></div><div class=3D"gmail_ext=
ra"><br clear=3D"all"><div><div class=3D"gmail_signature" data-smartmail=3D=
"gmail_signature">_____________<br>Roman Shpount</div></div>
<br><div class=3D"gmail_quote">On Wed, Jul 27, 2016 at 10:36 AM, Andy Hutto=
n <span dir=3D"ltr">&lt;<a href=3D"mailto:andyhutton.ietf@gmail.com" target=
=3D"_blank">andyhutton.ietf@gmail.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 dir=3D"ltr"><br><div>The issue of whether a re-offe=
r in WebRTC must be limited to the set of codec&#39;s within the previously=
 negotiated envelope was discussed during IETF96 but I don&#39;t think a co=
nclusion was reached.</div><div><br></div><div>Justin mentioned that this h=
ad been discussed previously so sorry for bringing it up again I cannot rem=
ember all the old arguments and a quick search did not find that discussion=
.</div><div><br></div><div>Personally I don&#39;t get the argument for maki=
ng this a restriction in WebRTC even if there are potential workaround solu=
tions like creating a new PC. As was mentioned in the meeting there are way=
s around the toggling of codec&#39;s depending on who creates the next offe=
r without imposing a restriction which is likely to be a cause of interoper=
ability issues.=C2=A0</div><div><br></div><div>At a minimum I would suggest=
 that WebRTC implementations must be able to receive and process and offer/=
answer which is not within the previously negotiated envelop and that shoul=
d be stated something in JSEP.=C2=A0 However I think it would also be prefe=
rable if the re-offer could contain the full set of capabilities.</div><div=
><br></div><div>Regards</div><span class=3D"HOEnZb"><font color=3D"#888888"=
><div>Andy</div></font></span></div>
<br>_______________________________________________<br>
rtcweb mailing list<br>
<a href=3D"mailto:rtcweb@ietf.org">rtcweb@ietf.org</a><br>
<a href=3D"https://www.ietf.org/mailman/listinfo/rtcweb" rel=3D"noreferrer"=
 target=3D"_blank">https://www.ietf.org/mailman/listinfo/rtcweb</a><br>
<br></blockquote></div><br></div>

--001a113ec7ecd1c6120538a01dc1--


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Subject: Re: [rtcweb] Codecs in re-offer [PR #269]
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I found the original discussion which mainly occurred in GitHub at
https://github.com/rtcweb-wg/jsep/issues/266 but I could not find any
conclusion to this on the mailing list.

IMHO we have got this wrong and need to think again the default case should
be that an offer contains the full set of capabilities not what was
previously negotiated. In the GitHUB #266 discussion a use case is
mentioned in which the application (SFU) wants to initiate a new
offer/answer because something changed such as a new party joining a
conference.  In this scenario it may be that the applications wants the
endpoint and not the SFU to generate the offer because it wants the full
set of capabilities to be re-negotiated, this is common in existing SIP
scenarios.

The default behavior in JSEP should be for the offer to contain the full
set of capabilities of the endpoint.

Regards
Andy




On Wed, Jul 27, 2016 at 5:00 PM, Roman Shpount <roman@telurix.com> wrote:
>
> I agree that re-offer should contain the full set of codecs or this will
cause interesting problems with 3pcc or SIP interop. The order of codecs
should ideally be the same as currently negotiated to avoid codec flips.
Essentially, what this means all supported codecs not currently negotiated
should be added to the end of the offer after the currently negotiated ones.
>
> Regards,
>
> _____________
> Roman Shpount
>
> On Wed, Jul 27, 2016 at 10:36 AM, Andy Hutton <andyhutton.ietf@gmail.com>
wrote:
>>
>>
>> The issue of whether a re-offer in WebRTC must be limited to the set of
codec's within the previously negotiated envelope was discussed during
IETF96 but I don't think a conclusion was reached.
>>
>> Justin mentioned that this had been discussed previously so sorry for
bringing it up again I cannot remember all the old arguments and a quick
search did not find that discussion.
>>
>> Personally I don't get the argument for making this a restriction in
WebRTC even if there are potential workaround solutions like creating a new
PC. As was mentioned in the meeting there are ways around the toggling of
codec's depending on who creates the next offer without imposing a
restriction which is likely to be a cause of interoperability issues.
>>
>> At a minimum I would suggest that WebRTC implementations must be able to
receive and process and offer/answer which is not within the previously
negotiated envelop and that should be stated something in JSEP.  However I
think it would also be preferable if the re-offer could contain the full
set of capabilities.
>>
>> Regards
>> Andy
>>
>> _______________________________________________
>> rtcweb mailing list
>> rtcweb@ietf.org
>> https://www.ietf.org/mailman/listinfo/rtcweb
>>
>

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<div dir=3D"ltr"><div>I found the original discussion which mainly occurred=
 in GitHub at=C2=A0<a href=3D"https://github.com/rtcweb-wg/jsep/issues/266"=
>https://github.com/rtcweb-wg/jsep/issues/266</a> but I could not find any =
conclusion to this on the mailing list.</div><div><br></div><div>IMHO we ha=
ve got this wrong and need to think again the default case should be that a=
n offer contains the full set of capabilities not what was previously negot=
iated. In the GitHUB #266 discussion a use case is mentioned in which the a=
pplication (SFU) wants to initiate a new offer/answer because something cha=
nged such as a new party joining a conference.=C2=A0 In this scenario it ma=
y be that the applications wants the endpoint and not the SFU to generate t=
he offer because it wants the full set of capabilities to be re-negotiated,=
 this is common in existing SIP scenarios.</div><div><br></div><div>The def=
ault behavior in JSEP should be for the offer to contain the full set of ca=
pabilities of the endpoint.</div><div><br></div><div>Regards</div><div>Andy=
</div><div><br></div><div><br></div><br><br>On Wed, Jul 27, 2016 at 5:00 PM=
, Roman Shpount &lt;<a href=3D"mailto:roman@telurix.com">roman@telurix.com<=
/a>&gt; wrote:<br>&gt;<br>&gt; I agree that re-offer should contain the ful=
l set of codecs or this will cause interesting problems with 3pcc or SIP in=
terop. The order of codecs should ideally be the same as currently negotiat=
ed to avoid codec flips. Essentially, what this means all supported codecs =
not currently negotiated should be added to the end of the offer after the =
currently negotiated ones.<br>&gt;<br>&gt; Regards,<br>&gt;<br>&gt; _______=
______<br>&gt; Roman Shpount<br>&gt;<br>&gt; On Wed, Jul 27, 2016 at 10:36 =
AM, Andy Hutton &lt;<a href=3D"mailto:andyhutton.ietf@gmail.com">andyhutton=
.ietf@gmail.com</a>&gt; wrote:<br>&gt;&gt;<br>&gt;&gt;<br>&gt;&gt; The issu=
e of whether a re-offer in WebRTC must be limited to the set of codec&#39;s=
 within the previously negotiated envelope was discussed during IETF96 but =
I don&#39;t think a conclusion was reached.<br>&gt;&gt;<br>&gt;&gt; Justin =
mentioned that this had been discussed previously so sorry for bringing it =
up again I cannot remember all the old arguments and a quick search did not=
 find that discussion.<br>&gt;&gt;<br>&gt;&gt; Personally I don&#39;t get t=
he argument for making this a restriction in WebRTC even if there are poten=
tial workaround solutions like creating a new PC. As was mentioned in the m=
eeting there are ways around the toggling of codec&#39;s depending on who c=
reates the next offer without imposing a restriction which is likely to be =
a cause of interoperability issues. <br>&gt;&gt;<br>&gt;&gt; At a minimum I=
 would suggest that WebRTC implementations must be able to receive and proc=
ess and offer/answer which is not within the previously negotiated envelop =
and that should be stated something in JSEP.=C2=A0 However I think it would=
 also be preferable if the re-offer could contain the full set of capabilit=
ies.<br>&gt;&gt;<br>&gt;&gt; Regards<br>&gt;&gt; Andy<br>&gt;&gt;<br>&gt;&g=
t; _______________________________________________<br>&gt;&gt; rtcweb maili=
ng list<br>&gt;&gt; <a href=3D"mailto:rtcweb@ietf.org">rtcweb@ietf.org</a><=
br>&gt;&gt; <a href=3D"https://www.ietf.org/mailman/listinfo/rtcweb">https:=
//www.ietf.org/mailman/listinfo/rtcweb</a><br>&gt;&gt;<br>&gt;<br></div>

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