
From nobody Wed Aug  6 04:31:24 2014
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From: Brett Tate <brett@broadsoft.com>
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Subject: [AVTCORE] status of RFC 3611 errata 3795
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Hi,

Concerning the status of RFC 3611 errata 3795, I recommend that the
proposed change be accepted.

Original Text
-------------
rtcp-xr-attrib = "a=" "rtcp-xr" ":" [xr-format *(SP xr-format)] CRLF

Corrected Text
--------------
rtcp-xr-attrib = "a=" "rtcp-xr" [ ":" xr-format *(SP xr-format)] CRLF

RFC 3611's rtcp-xr-attrib ABNF conflicts with RFC 2327's (and RFC 4566's)
attribute ABNF.  RFC 2327 and RFC 4566 require a byte-string after the
colon.  RFC 3611 requires the colon even if no byte-string after it.  This
is causing interoperability issues.

If you think rfc4566bis should somehow be modified to accommodate RFC
3611, the topic is currently being discussed within mmusic.

http://www.ietf.org/mail-archive/web/mmusic/current/msg13668.html

Thanks,
Brett


From nobody Thu Aug  7 15:50:14 2014
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Subject: [AVTCORE] I-D Action: draft-ietf-avtcore-rtp-topologies-update-03.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 Audio/Video Transport Core Maintenance Working Group of the IETF.

        Title           : RTP Topologies
        Authors         : Magnus Westerlund
                          Stephan Wenger
	Filename        : draft-ietf-avtcore-rtp-topologies-update-03.txt
	Pages           : 44
	Date            : 2014-08-07

Abstract:
   This document discusses point to point and multi-endpoint topologies
   used in Real-time Transport Protocol (RTP)-based environments.  In
   particular, centralized topologies commonly employed in the video
   conferencing industry are mapped to the RTP terminology.

   This document is updated with additional topologies and is intended
   to replace RFC 5117.


The IETF datatracker status page for this draft is:
https://datatracker.ietf.org/doc/draft-ietf-avtcore-rtp-topologies-update/

There's also a htmlized version available at:
http://tools.ietf.org/html/draft-ietf-avtcore-rtp-topologies-update-03

A diff from the previous version is available at:
http://www.ietf.org/rfcdiff?url2=draft-ietf-avtcore-rtp-topologies-update-03


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

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


From nobody Thu Aug  7 16:14:38 2014
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From: Stephan Wenger <stewe@stewe.org>
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Thread-Topic: Topologies-update, reply to Bo's review
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Subject: Re: [AVTCORE] Topologies-update, reply to Bo's review
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Stephan,

The new text in 3.10 confuses me: why can=A1=AFt the subunits be part of =
a common RTP session? You have:

   +---------------------+
   | Endpoint A          |
   | Local Area Network  |
   |      +------------+ |
   |   +->| Audio      |<+-RTP---\
   |   |  +------------+ |        \    +------+
   |   |  +------------+ |         +-->|      |
   |   +->| Video      |<+-RTP-------->|  B   |
   |   |  +------------+ |         +-->|      |
   |   |  +------------+ |        /    +------+
   |   +->| Control    |<+-SIP---/
   |      +------------+ |
   +---------------------+

If B sends reception reports so that the audio part of A can see the =
SSRCs used by the video and Control parts of A, then you have a single =
RTP session.

I agree that a single RTP session isn=A1=AFt required here, but it =
doesn=A1=AFt seem to be prohibited.

Cheers,
Colin




On 8 Aug 2014, at 00:14, Stephan Wenger <stewe@stewe.org> wrote:
> Hi all,
>=20
> You may have noticed that I just posted version -03 of the
> topologies-update draft.  The main changes are alignment with the
> terminology used in avtext=A1=AFs grouping taxonomy draft, and a =
rewrite of
> section 3.10 (split-component terminal).
>=20
> As per guidance received in Toronto, I did include copy-pastes of =
those
> few definitions of the taxonomy draft required herein, so to keep the =
two
> drafts independent.  However, I have also asked the RFC editor to =
replace
> those definitions with a reference to a forthcoming taxonomy-RFC, if =
such
> can be done without undue delay.  See section 2.2.
>=20
> Section 3.10 was close to unparsable--at least very confusing--which
> triggered the rewrite.  I think it=A1=AFs better now, but it certainly =
needs a
> fresh review.
>=20
> Towards the end of this email, find my responses to Bo Burman=A1=AFs =
thorough
> review of version -02, posted to this list on 6/11/14.  I had private
> communication with Bo during the editing process, and he=A1=AFs happy =
with the
> result and found his points addressed.
>=20
> My list of open issues of this draft is now empty.  WGLC would be fine
> with me.  Please review, comment, and get this thing out of the door.
>=20
> Thanks,
> Stephan
>=20
> On 6/11/14, 2:39 AM, "Bo Burman" <bo.burman@ericsson.com> wrote:
>=20
> Hi all,
>=20
> [Bo] I reviewed this as a part of the ongoing update of the taxonomy =
draft
> to
> try to align them, and have the following comments on the
> topologies-update-02:
> * What is meant with "Media Flow" in 3.2.1.3; is it a set of RTP =
Streams?
> In that case, does this set have any specific characteristics?
>=20
> [StW]
>=20
> The comments relates to the description of Fig. 3.  No, the =A9=F8media =
flow=A9=F7
> is really only a single RTP stream (without FEC) to which FEC is added =
by
> the translator.  The relevant paragraph was poorly and overly broadly
> written, and I tightened the language.
>=20
> [Bo]
> * (At least) in 3.2.2 and 3.3.3, the term "flow" is used in "RTP flow" =
or
> "packet flow"; how is that different from taxonomy RTP Stream?
>=20
> [StW]
> Very good catch; the language here (mostly from 5117) was really in =
need
> to be tightened up.  I checked the various occurrences of =A9=F8flow=A9=F7=
, and
> replaced them with =A9=F8RTP stream=A9=F7 where appropriate.  The =
remaining =A9=F8flow=A9=F7s
> are transport flows, which is basically the flow of packets in a =
transport
> association (five tuple).
>=20
> [Bo]
> * The terms "transport flow" or just "flow" are heavily used in 3.4;
> again, as with "Media Flow", is there any specific distinction assumed =
in
> this besides possibly being a set of RTP Streams?
>=20
> [StW]
> Yes, in 3.4, transport flow refers to packets going over a 5 tuple, =
see
> above.  I hope this is now clear from context.
>=20
> [Bo]
> * The term "RTP media stream" is used in many places; is there an
> intended specialization compared to a (generic) RTP Stream?
>=20
> [StW]
> No, RTP media stream and RTP stream are meant to be the same in all =
cases.
> I went through the doc and fixed as appropriate, using =A9=F8RTP =
stream=A9=F7
> (hopefully) consistently.
>=20
> [Bo]
> * What is a "Content source" in 3.6? Is it a Media Source?
>=20
> [StW]
> Tricky.  In 3.6, =A9=F8content source=A9=F7 is used to identify the =
Media sender as
> a whole: uncompressed stream generation (through mixing or =
acquisition),
> encoding, packetizing.   =A9=F8Endpoint=A9=F7 may be more appropriate. =
 However,
> that would imply that a middlebox like a Mixer or MCU are also =
endpoints,
> which is counter-intuitive, at least to me.  I tried to word around.
>=20
> [Bo]
> * What is a "Conference session" in 3.6? Is it a Communication =
Session,
> or some specialization of it?
>=20
> [StW]
> Communication Session seems right.
>=20
> [Bo]
> * Can you explain a bit of what is a "Session participant" in 3.6? =
There
> are discussions on the use of End Point and/or Participant in =
taxonomy,
> so I'm hesitant to suggest any taxonomy term right now since it may
> change.
>=20
> [StW]
> Session participants refers to RTP sending/receiving entities that =
take
> part of the same RTP session.  Insofar, taxonomy-02=A9=F6s End Point =
would be
> appropriate, with the additional remark that these End Points share =
one
> RTP session.
>=20
> The word =A9=F8participant=A9=F7 is used throughout the document in a =
number of
> different contexts, usually but not always with a meaning similar to =
End
> Point in taxonomy.  I have replaced =A9=F8participant=A9=F7 with =A9=F8E=
nd Point=A9=F7 where
> appropriate, and left it (usually with qualifiers) otherwise.
>=20
> [Bo]
> * The stand-alone term "stream" is used in many places throughout the
> document. I think it very reasonable to just assume it is a short for
> "RTP Stream", but you may want to make such comment early on in the
> document. I found no place in the document where such assumption would =
be
> invalid.
>=20
> [StW]
> Fixed by using =A9=F8RTP stream consistently.
>=20
> [Bo]
> * The term "global session" is used in 3.6; I find it a bit hard to
> understand what is meant.
>=20
> [StW]
> I think it=A9=F6s =A9=F8Communication Session=A9=F7 (every Endpoint =
and every middle box
> that are talking to each other).  Fixed.
>=20
> [Bo]
> * Section 3.6.1 could possibly benefit from using the taxonomy term
> "Encoded Stream"
>=20
> [StW]
> Correct.  Clarified.
>=20
> [Bo]
> * Nit: Figure 3 (in 3.2.1.3) and Figure 4 (in 3.2.2) have the same
> caption; is this intended?
>=20
> [StW]
> No.  Fixed.
>=20
> _______________________________________________
> Audio/Video Transport Core Maintenance
> avt@ietf.org
> https://www.ietf.org/mailman/listinfo/avt



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




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From: Stephan Wenger <stewe@stewe.org>
To: Colin Perkins <csp@csperkins.org>
Thread-Topic: [AVTCORE] Topologies-update, reply to Bo's review
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From: Colin Perkins <csp@csperkins.org>
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Subject: Re: [AVTCORE] Topologies-update, reply to Bo's review
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Hi,

They=A1=AFre logically part of the same endpoint, and use a common =
clock, and this makes them semantically coupled anyway, so I think a =
common CNAME and single RTP session makes sense (this is essentially the =
design the old Mbone tools used to lip-sync when audio and video were =
done with separate tools, using a coordination protocol).=20

Colin


On 8 Aug 2014, at 15:30, Stephan Wenger <stewe@stewe.org> wrote:

> Hi Colin,
>=20
> The two RTP subunits (Audio and Video) share the same CNAME, as they =
are
> semantically coupled, and they use distinct transport addresses.  =
These
> are the distinguishing features of this terminal=A1=AFs topology.  In =
that
> case, Audio and Video would at least violate a =A1=B0SHOULD=A1=B1 =
clause of RFC 3550
> (section 6.5.1, first bullet on page 47) if they were in the same RTP
> session, no?
>=20
> A different advise would be to say that if one decomposes an endpoint, =
a
> good choice of the CNAME would be something like
> john.doe.audio@vidconfroom1.example.com and
> john.doe.video@videoconfroom1.example.com, rather than using
> john.doe@videoconfroom1.example.com as CNAME for both Audio and Video.
> So, distinct CNAMEs, but still human-observable semantic coupling =
(even a
> reasonably smart machine could figure it out if configured correctly =
:-)
>=20
> Should I add this as a sensible design alternative?  Or am I in the =
woods?
>=20
> Stephan
>=20
>=20
>=20
>=20
> On 8/8/14, 6:59 AM, "Colin Perkins" <csp@csperkins.org> wrote:
>=20
>> Stephan,
>>=20
>> The new text in 3.10 confuses me: why can=A1=AFt the subunits be part =
of a
>> common RTP session? You have:
>>=20
>>  +---------------------+
>>  | Endpoint A          |
>>  | Local Area Network  |
>>  |      +------------+ |
>>  |   +->| Audio      |<+-RTP---\
>>  |   |  +------------+ |        \    +------+
>>  |   |  +------------+ |         +-->|      |
>>  |   +->| Video      |<+-RTP-------->|  B   |
>>  |   |  +------------+ |         +-->|      |
>>  |   |  +------------+ |        /    +------+
>>  |   +->| Control    |<+-SIP---/
>>  |      +------------+ |
>>  +---------------------+
>>=20
>> If B sends reception reports so that the audio part of A can see the
>> SSRCs used by the video and Control parts of A, then you have a =
single
>> RTP session.
>>=20
>> I agree that a single RTP session isn=A1=AFt required here, but it =
doesn=A1=AFt
>> seem to be prohibited.
>>=20
>> Cheers,
>> Colin
>>=20
>>=20
>>=20
>>=20
>> On 8 Aug 2014, at 00:14, Stephan Wenger <stewe@stewe.org> wrote:
>>> Hi all,
>>>=20
>>> You may have noticed that I just posted version -03 of the
>>> topologies-update draft.  The main changes are alignment with the
>>> terminology used in avtext=A1=AFs grouping taxonomy draft, and a =
rewrite of
>>> section 3.10 (split-component terminal).
>>>=20
>>> As per guidance received in Toronto, I did include copy-pastes of =
those
>>> few definitions of the taxonomy draft required herein, so to keep =
the
>>> two
>>> drafts independent.  However, I have also asked the RFC editor to
>>> replace
>>> those definitions with a reference to a forthcoming taxonomy-RFC, if
>>> such
>>> can be done without undue delay.  See section 2.2.
>>>=20
>>> Section 3.10 was close to unparsable--at least very confusing--which
>>> triggered the rewrite.  I think it=A1=AFs better now, but it =
certainly needs
>>> a
>>> fresh review.
>>>=20
>>> Towards the end of this email, find my responses to Bo Burman=A1=AFs =
thorough
>>> review of version -02, posted to this list on 6/11/14.  I had =
private
>>> communication with Bo during the editing process, and he=A1=AFs =
happy with
>>> the
>>> result and found his points addressed.
>>>=20
>>> My list of open issues of this draft is now empty.  WGLC would be =
fine
>>> with me.  Please review, comment, and get this thing out of the =
door.
>>>=20
>>> Thanks,
>>> Stephan
>>>=20
>>> On 6/11/14, 2:39 AM, "Bo Burman" <bo.burman@ericsson.com> wrote:
>>>=20
>>> Hi all,
>>>=20
>>> [Bo] I reviewed this as a part of the ongoing update of the taxonomy
>>> draft
>>> to
>>> try to align them, and have the following comments on the
>>> topologies-update-02:
>>> * What is meant with "Media Flow" in 3.2.1.3; is it a set of RTP
>>> Streams?
>>> In that case, does this set have any specific characteristics?
>>>=20
>>> [StW]
>>>=20
>>> The comments relates to the description of Fig. 3.  No, the =A9=F8medi=
a flow=A9=F7
>>> is really only a single RTP stream (without FEC) to which FEC is =
added
>>> by
>>> the translator.  The relevant paragraph was poorly and overly =
broadly
>>> written, and I tightened the language.
>>>=20
>>> [Bo]
>>> * (At least) in 3.2.2 and 3.3.3, the term "flow" is used in "RTP =
flow"
>>> or
>>> "packet flow"; how is that different from taxonomy RTP Stream?
>>>=20
>>> [StW]
>>> Very good catch; the language here (mostly from 5117) was really in =
need
>>> to be tightened up.  I checked the various occurrences of =A9=F8flow=A9=
=F7, and
>>> replaced them with =A9=F8RTP stream=A9=F7 where appropriate.  The =
remaining
>>> =A9=F8flow=A9=F7s
>>> are transport flows, which is basically the flow of packets in a
>>> transport
>>> association (five tuple).
>>>=20
>>> [Bo]
>>> * The terms "transport flow" or just "flow" are heavily used in 3.4;
>>> again, as with "Media Flow", is there any specific distinction =
assumed
>>> in
>>> this besides possibly being a set of RTP Streams?
>>>=20
>>> [StW]
>>> Yes, in 3.4, transport flow refers to packets going over a 5 tuple, =
see
>>> above.  I hope this is now clear from context.
>>>=20
>>> [Bo]
>>> * The term "RTP media stream" is used in many places; is there an
>>> intended specialization compared to a (generic) RTP Stream?
>>>=20
>>> [StW]
>>> No, RTP media stream and RTP stream are meant to be the same in all
>>> cases.
>>> I went through the doc and fixed as appropriate, using =A9=F8RTP =
stream=A9=F7
>>> (hopefully) consistently.
>>>=20
>>> [Bo]
>>> * What is a "Content source" in 3.6? Is it a Media Source?
>>>=20
>>> [StW]
>>> Tricky.  In 3.6, =A9=F8content source=A9=F7 is used to identify the =
Media sender
>>> as
>>> a whole: uncompressed stream generation (through mixing or =
acquisition),
>>> encoding, packetizing.   =A9=F8Endpoint=A9=F7 may be more =
appropriate.  However,
>>> that would imply that a middlebox like a Mixer or MCU are also
>>> endpoints,
>>> which is counter-intuitive, at least to me.  I tried to word around.
>>>=20
>>> [Bo]
>>> * What is a "Conference session" in 3.6? Is it a Communication =
Session,
>>> or some specialization of it?
>>>=20
>>> [StW]
>>> Communication Session seems right.
>>>=20
>>> [Bo]
>>> * Can you explain a bit of what is a "Session participant" in 3.6? =
There
>>> are discussions on the use of End Point and/or Participant in =
taxonomy,
>>> so I'm hesitant to suggest any taxonomy term right now since it may
>>> change.
>>>=20
>>> [StW]
>>> Session participants refers to RTP sending/receiving entities that =
take
>>> part of the same RTP session.  Insofar, taxonomy-02=A9=F6s End Point =
would be
>>> appropriate, with the additional remark that these End Points share =
one
>>> RTP session.
>>>=20
>>> The word =A9=F8participant=A9=F7 is used throughout the document in =
a number of
>>> different contexts, usually but not always with a meaning similar to =
End
>>> Point in taxonomy.  I have replaced =A9=F8participant=A9=F7 with =
=A9=F8End Point=A9=F7 where
>>> appropriate, and left it (usually with qualifiers) otherwise.
>>>=20
>>> [Bo]
>>> * The stand-alone term "stream" is used in many places throughout =
the
>>> document. I think it very reasonable to just assume it is a short =
for
>>> "RTP Stream", but you may want to make such comment early on in the
>>> document. I found no place in the document where such assumption =
would
>>> be
>>> invalid.
>>>=20
>>> [StW]
>>> Fixed by using =A9=F8RTP stream consistently.
>>>=20
>>> [Bo]
>>> * The term "global session" is used in 3.6; I find it a bit hard to
>>> understand what is meant.
>>>=20
>>> [StW]
>>> I think it=A9=F6s =A9=F8Communication Session=A9=F7 (every Endpoint =
and every middle
>>> box
>>> that are talking to each other).  Fixed.
>>>=20
>>> [Bo]
>>> * Section 3.6.1 could possibly benefit from using the taxonomy term
>>> "Encoded Stream"
>>>=20
>>> [StW]
>>> Correct.  Clarified.
>>>=20
>>> [Bo]
>>> * Nit: Figure 3 (in 3.2.1.3) and Figure 4 (in 3.2.2) have the same
>>> caption; is this intended?
>>>=20
>>> [StW]
>>> No.  Fixed.
>>>=20
>>> _______________________________________________
>>> Audio/Video Transport Core Maintenance
>>> avt@ietf.org
>>> https://www.ietf.org/mailman/listinfo/avt
>>=20
>>=20
>>=20
>> --=20
>> Colin Perkins
>> http://csperkins.org/
>>=20
>>=20
>>=20
>=20
> _______________________________________________
> Audio/Video Transport Core Maintenance
> avt@ietf.org
> https://www.ietf.org/mailman/listinfo/avt



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




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From: Stephan Wenger <stewe@stewe.org>
To: Stephan Wenger <stewe@stewe.org>, Colin Perkins <csp@csperkins.org>
Thread-Topic: [AVTCORE] Topologies-update, reply to Bo's review
Thread-Index: AQHPspVWg15Q2HN36EeDlcGGcl3C3pvGvOWA//+TPwCAAAG4gA==
Date: Fri, 8 Aug 2014 14:37:06 +0000
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References: <D0095161.459E9%stewe@stewe.org> <3147A00C-8D9D-41FD-8A4B-273972E769B0@csperkins.org> <D00A2960.45AC0%stewe@stewe.org>
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From: Stephan Wenger <stewe@stewe.org>
To: Colin Perkins <csp@csperkins.org>
Thread-Topic: [AVTCORE] Topologies-update, reply to Bo's review
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Subject: Re: [AVTCORE] Topologies-update, reply to Bo's review
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From: Colin Perkins <csp@csperkins.org>
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Subject: Re: [AVTCORE] Topologies-update, reply to Bo's review
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Hi,

On 8 Aug 2014, at 16:20, Stephan Wenger <stewe@stewe.org> wrote:
> On 8/8/14, 7:36 AM, "Colin Perkins" <csp@csperkins.org> wrote:
>> They=A1=AFre logically part of the same endpoint, and use a common =
clock, and
>> this makes them semantically coupled anyway, so I think a common =
CNAME
>> and single RTP session makes sense (this is essentially the design =
the
>> old Mbone tools used to lip-sync when audio and video were done with
>> separate tools, using a coordination protocol).
>=20
> So let=A1=AFs assume a common CNAME, but also separate transport =
addresses per
> subunit.  Both version 02 and 03 of the draft comment on that scenario =
as
> follows:
>=20
> =A1=B0It is not really feasible to decompose one logical end-point =
into two
> different transport nodes in one RTP session.  A third party monitor =
would
> report such an attempt as two entities being two different end-points =
with
> a CNAME collision.  As a result, a fully RTP conformant de-composited
> endpoint is one where the different decomposed parts use separate RTP
> sessions to send and/or receive media streams intended for them."
>=20
> You seem to assert that this is a non-issue.  Why?

I think you=A1=AFre confusing transport-layer flows and RTP sessions, =
and the role of the CNAME. It=A1=AFs a non-issue because the =A1=B0CNAME =
collision=A1=B1 is normal, and orthogonal to the number of RTP sessions =
or transport-layer flows.

Assume I want to send you audio and video flows that can be synchronised =
on playout. I=A1=AFll send two RTP streams, one for the audio and one =
for the video, using a different SSRC for each stream, and mapping their =
media clocks to a common reference clock, which is reported in RTCP SR =
packets. The RTCP SDES packets for each SSRC will use the same CNAME, =
and it=A1=AFs that common CNAME that indicates to the receiver that the =
two flows can be synchronised. The receiver also has its own two SSRC =
values, one for the audio and one for video, that it includes in the =
RTCP RR packets it sends (and also for any audio and video RTP packets =
it sends back).

The RTP streams can be multiplexed onto a single 5-tuple, for example:

 Sender                      Receiver
 +-------------------+       +------------------+   =20
 | IP addr 10.0.0.1  |       | IP addr 10.0.0.2 |
 |                   |       |                  |
 | +-----+SSRC 1111  |       | SSRC 3333+-----+ |
 | |Audio|<-----+    |       |    +---->|Audio| |
 | +-----+      |Port|       |Port|     +-----+ |
 |              +=3D=3D=3D=3D+=3D=3D=3D=3D=3D=3D=3D+=3D=3D=3D=3D+        =
     |
 | +-----+      |5000|       |6000|     +-----+ |
 | |Video|<-----+    |       |    +---->|Video| |
 | +-----+SSRC 2222  |       | SSRC 4444+-----+ |
 |                   |       |                  |
 +-------------------+       +------------------+

There is a single transport-layer flow, from 10.0.0.1:5000 to =
10.0.0.2:6000. RTP packets with two SSRC values are multiplexed on that =
flow from sender to receiver, and RTCP packets with two SSRC values are =
multiplexed on that flow in the reverse direction. In this case, both =
SSRCs at the receiver can see the packets sent by both sender SSRCs, =
since they all arrive on the same port. The RTCP sent by SSRC 3333 will =
contain RTCP RR packets relating both to SSRC 1111 and SSRC 2222, as =
will the RTCP sent by SSRC 4444. This gives a single RTP session, since =
each SSRC can see every other SSRC. The RTCP reports sent by SSRC 1111 =
and SSRC 2222 will both use the same CNAME, to indicate that they can be =
synchronised.

The RTP streams can also be sent using distinct 5-tuples, for example:

 Sender                      Receiver
 +-------------------+       +------------------+   =20
 | IP addr 10.0.0.1  |       | IP addr 10.0.0.2 |
 |                   |       |                  |
 | +-----+ Port 5000 |       |Port 6000 +-----+ |
 | |Audio|-----------+-------+--------->|Audio| |
 | +-----+ SSRC 1111 |       |SSRC 3333 +-----+ |
 |                   |       |             :    |
 | +-----+ Port 5002 |       |Port 6002 +-----+ |
 | |Video|-----------+-------+--------->|Video| |
 | +-----+ SSRC 2222 |       |SSRC 4444 +-----+ |
 |                   |       |                  |
 +-------------------+       +------------------+

This has two transport-layer flows, one from 10.0.0.1:5000 to =
10.0.0.2:6000, and the other from 10.0.0.1:5002 to 10.0.0.2:6002. One of =
those flows carries audio, the other carries video.

If SSRC 3333 only sends RTCP reports relating to SSRC 1111, and SSRC =
4444 only sends RTCP reports relating to SSRC 2222, then this scenario =
comprises two RTP sessions. However, if the receiver is implemented such =
that SSRC 3333 and SSRC 4444 share reception quality information, and =
each of SSRC 3333 and SSRC 4444 send RTCP reports relating to both SSRC =
1111 and SSRC 2222, then there is only a single RTP session, despite the =
use of multiple transport-layer flows, since each SSRC can see every =
other SSRC.=20

No matter whether how the receiver is implemented, RTCP packets sent by =
SSRC 1111 and by SSRC 222 will use the same CNAME, to indicate that they =
can be synchronised, even though they are sent on separate =
transport-layer flows.

We can split things up further, to the distinct 5-tuples come from =
different IP addresses (this is a version of Topo-Split-Terminal):

 Sender                      Receiver
 +-------------------+       +------------------+   =20
 | IP addr 10.0.0.1  |       | IP addr 10.0.0.2 |
 |                   |       |                  |
 | +-----+ Port 5000 |       |Port 6000 +-----+ |
 | |Audio|<----------+-------+--------->|Audio| |
 | +-----+ SSRC 1111 |       |SSRC 3333 +-----+ |
 +---+---------------+       |             :    |
     |clock sync             |             :    |
 +---+---------------+       |             :    |
 | IP addr 10.0.1.1  |       |             :    |
 | +-----+ Port 5002 |       |Port 6004 +-----+ |
 | |Video|<----------+-------+--------->|Video| |
 | +-----+ SSRC 2222 |       |SSRC 4444 +-----+ |
 |                   |       |                  |
 +-------------------+       +------------------+

Again, there are two transport layer flows, and again this can be either =
one or two RTP sessions, depending on whether the two SSRCs in the =
Receiver share information (i.e., if the RTCP packets that SSRC 3333 =
sends include reports relating to both SSRC 1111 and SSRC 2222, and =
similarly for SSRC 4444, then each SSRC can see every other, and there =
is a single RTP session; however, if SSRC 3333 only reports on SSRC =
1111, and SSRC 4444 only reports on SSRC 2222, then there are two RTP =
sessions). Again, irrespective of receiver behaviour, and the number of =
RTP sessions, RTCP packets sent by SSRC 1111 and SSRC 2222 will use the =
same CNAME to indicate that their flows can be synchronised (they need =
to run a clock synchronisation protocol to ensure they have access to =
the common clock).

If you want to decompose an endpoint, sending audio and video from =
separate transport-layer addresses and allowing the receiver to =
synchronise the flows, then the audio and video need to use different =
SSRCs, but the same CNAME, and use some (non-RTP) clock synchronisation =
protocol. Whether those flows form a single RTP session or not depends =
on whether the system is implemented to share a single SSRC space and =
single view of the reception quality, or not. Topo-Split-Terminal can =
therefore be implemented using multiple RTP sessions, or using a single =
RTP session (although neither affects the choice of CNAME).

I would argue that implementing this topology as a single RTP session =
could be beneficial, since the cross reporting provides information =
that=A1=AFs helpful for (potentially automated) debugging of reception =
quality problems. Others would doubtless say the cross-reporting is =
unnecessary overhead, and would prefer to implement this as separate RTP =
sessions. Both are valid views, with different trade-offs.

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




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From: Stephan Wenger <stewe@stewe.org>
To: Colin Perkins <csp@csperkins.org>
Thread-Topic: [AVTCORE] Topologies-update, reply to Bo's review
Thread-Index: AQHPspVWg15Q2HN36EeDlcGGcl3C3pvGvOWA//+TPwCAAHcNAP//lsSAgACXf4D//5EhgA==
Date: Fri, 8 Aug 2014 17:45:38 +0000
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References: <D0095161.459E9%stewe@stewe.org> <3147A00C-8D9D-41FD-8A4B-273972E769B0@csperkins.org> <D00A2960.45AC0%stewe@stewe.org> <25F74D0B-5BB2-480F-B4FA-10D226843952@csperkins.org> <D00A3670.45AF2%stewe@stewe.org> <716C6558-C1B9-499E-A0A2-93F9111802AA@csperkins.org>
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Hi Colin,
I think I start seeing your point.  Let me work on section 3.10 a bit more
(over the weekend most likely).
But please also see inline (towards the top of the thread only).
Thanks,
Stephan
=20

On 8/8/14, 10:22 AM, "Colin Perkins" <csp@csperkins.org> wrote:

>Hi,
>
>On 8 Aug 2014, at 16:20, Stephan Wenger <stewe@stewe.org> wrote:
>> On 8/8/14, 7:36 AM, "Colin Perkins" <csp@csperkins.org> wrote:
>>> They=B9re logically part of the same endpoint, and use a common clock,
>>>and
>>> this makes them semantically coupled anyway, so I think a common CNAME
>>> and single RTP session makes sense (this is essentially the design the
>>> old Mbone tools used to lip-sync when audio and video were done with
>>> separate tools, using a coordination protocol).
>>=20
>> So let=B9s assume a common CNAME, but also separate transport addresses
>>per
>> subunit.  Both version 02 and 03 of the draft comment on that scenario
>>as
>> follows:
>>=20
>> =B3It is not really feasible to decompose one logical end-point into two
>> different transport nodes in one RTP session.

This is the statement you question.  You argue it is possible and still
stay compliant with RTP.

>>A third party monitor would
>> report such an attempt as two entities being two different end-points
>>with
>> a CNAME collision.

Do I understand you correctly that yes, a third party monitor would indeed
report a CNAME collision, but that=B9s a problem of the third party monitor=
.
 Or, better, the third party monitor=B9s lack of awareness of the existence
of decomposed endpoint subunits in the same RTP session, them being a
legal (and, in at least some scenarios, even desirable) design choice.

>>As a result, a fully RTP conformant de-composited
>> endpoint is one where the different decomposed parts use separate RTP
>> sessions to send and/or receive media streams intended for them."

And this is, according to you, an incorrect statement.

>>=20
>> You seem to assert that this is a non-issue.  Why?
>
>I think you=B9re confusing transport-layer flows and RTP sessions, and the
>role of the CNAME. It=B9s a non-issue because the =B3CNAME collision=B2 is
>normal, and orthogonal to the number of RTP sessions or transport-layer
>flows.
>
>Assume I want to send you audio and video flows that can be synchronised
>on playout. I=B9ll send two RTP streams, one for the audio and one for the
>video, using a different SSRC for each stream, and mapping their media
>clocks to a common reference clock, which is reported in RTCP SR packets.
>The RTCP SDES packets for each SSRC will use the same CNAME, and it=B9s
>that common CNAME that indicates to the receiver that the two flows can
>be synchronised. The receiver also has its own two SSRC values, one for
>the audio and one for video, that it includes in the RTCP RR packets it
>sends (and also for any audio and video RTP packets it sends back).
>
>The RTP streams can be multiplexed onto a single 5-tuple, for example:
>
> Sender                      Receiver
> +-------------------+       +------------------+
> | IP addr 10.0.0.1  |       | IP addr 10.0.0.2 |
> |                   |       |                  |
> | +-----+SSRC 1111  |       | SSRC 3333+-----+ |
> | |Audio|<-----+    |       |    +---->|Audio| |
> | +-----+      |Port|       |Port|     +-----+ |
> |              +=3D=3D=3D=3D+=3D=3D=3D=3D=3D=3D=3D+=3D=3D=3D=3D+         =
    |
> | +-----+      |5000|       |6000|     +-----+ |
> | |Video|<-----+    |       |    +---->|Video| |
> | +-----+SSRC 2222  |       | SSRC 4444+-----+ |
> |                   |       |                  |
> +-------------------+       +------------------+
>
>There is a single transport-layer flow, from 10.0.0.1:5000 to
>10.0.0.2:6000. RTP packets with two SSRC values are multiplexed on that
>flow from sender to receiver, and RTCP packets with two SSRC values are
>multiplexed on that flow in the reverse direction. In this case, both
>SSRCs at the receiver can see the packets sent by both sender SSRCs,
>since they all arrive on the same port. The RTCP sent by SSRC 3333 will
>contain RTCP RR packets relating both to SSRC 1111 and SSRC 2222, as will
>the RTCP sent by SSRC 4444. This gives a single RTP session, since each
>SSRC can see every other SSRC. The RTCP reports sent by SSRC 1111 and
>SSRC 2222 will both use the same CNAME, to indicate that they can be
>synchronised.
>
>The RTP streams can also be sent using distinct 5-tuples, for example:
>
> Sender                      Receiver
> +-------------------+       +------------------+
> | IP addr 10.0.0.1  |       | IP addr 10.0.0.2 |
> |                   |       |                  |
> | +-----+ Port 5000 |       |Port 6000 +-----+ |
> | |Audio|-----------+-------+--------->|Audio| |
> | +-----+ SSRC 1111 |       |SSRC 3333 +-----+ |
> |                   |       |             :    |
> | +-----+ Port 5002 |       |Port 6002 +-----+ |
> | |Video|-----------+-------+--------->|Video| |
> | +-----+ SSRC 2222 |       |SSRC 4444 +-----+ |
> |                   |       |                  |
> +-------------------+       +------------------+
>
>This has two transport-layer flows, one from 10.0.0.1:5000 to
>10.0.0.2:6000, and the other from 10.0.0.1:5002 to 10.0.0.2:6002. One of
>those flows carries audio, the other carries video.
>
>If SSRC 3333 only sends RTCP reports relating to SSRC 1111, and SSRC 4444
>only sends RTCP reports relating to SSRC 2222, then this scenario
>comprises two RTP sessions. However, if the receiver is implemented such
>that SSRC 3333 and SSRC 4444 share reception quality information, and
>each of SSRC 3333 and SSRC 4444 send RTCP reports relating to both SSRC
>1111 and SSRC 2222, then there is only a single RTP session, despite the
>use of multiple transport-layer flows, since each SSRC can see every
>other SSRC.=20
>
>No matter whether how the receiver is implemented, RTCP packets sent by
>SSRC 1111 and by SSRC 222 will use the same CNAME, to indicate that they
>can be synchronised, even though they are sent on separate
>transport-layer flows.
>
>We can split things up further, to the distinct 5-tuples come from
>different IP addresses (this is a version of Topo-Split-Terminal):
>
> Sender                      Receiver
> +-------------------+       +------------------+
> | IP addr 10.0.0.1  |       | IP addr 10.0.0.2 |
> |                   |       |                  |
> | +-----+ Port 5000 |       |Port 6000 +-----+ |
> | |Audio|<----------+-------+--------->|Audio| |
> | +-----+ SSRC 1111 |       |SSRC 3333 +-----+ |
> +---+---------------+       |             :    |
>     |clock sync             |             :    |
> +---+---------------+       |             :    |
> | IP addr 10.0.1.1  |       |             :    |
> | +-----+ Port 5002 |       |Port 6004 +-----+ |
> | |Video|<----------+-------+--------->|Video| |
> | +-----+ SSRC 2222 |       |SSRC 4444 +-----+ |
> |                   |       |                  |
> +-------------------+       +------------------+
>
>Again, there are two transport layer flows, and again this can be either
>one or two RTP sessions, depending on whether the two SSRCs in the
>Receiver share information (i.e., if the RTCP packets that SSRC 3333
>sends include reports relating to both SSRC 1111 and SSRC 2222, and
>similarly for SSRC 4444, then each SSRC can see every other, and there is
>a single RTP session; however, if SSRC 3333 only reports on SSRC 1111,
>and SSRC 4444 only reports on SSRC 2222, then there are two RTP
>sessions). Again, irrespective of receiver behaviour, and the number of
>RTP sessions, RTCP packets sent by SSRC 1111 and SSRC 2222 will use the
>same CNAME to indicate that their flows can be synchronised (they need to
>run a clock synchronisation protocol to ensure they have access to the
>common clock).
>
>If you want to decompose an endpoint, sending audio and video from
>separate transport-layer addresses and allowing the receiver to
>synchronise the flows, then the audio and video need to use different
>SSRCs, but the same CNAME, and use some (non-RTP) clock synchronisation
>protocol. Whether those flows form a single RTP session or not depends on
>whether the system is implemented to share a single SSRC space and single
>view of the reception quality, or not. Topo-Split-Terminal can therefore
>be implemented using multiple RTP sessions, or using a single RTP session
>(although neither affects the choice of CNAME).
>
>I would argue that implementing this topology as a single RTP session
>could be beneficial, since the cross reporting provides information
>that=B9s helpful for (potentially automated) debugging of reception qualit=
y
>problems. Others would doubtless say the cross-reporting is unnecessary
>overhead, and would prefer to implement this as separate RTP sessions.
>Both are valid views, with different trade-offs.
>
>--=20
>Colin Perkins
>http://csperkins.org/
>
>
>


From nobody Sat Aug  9 11:49:21 2014
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Subject: [AVTCORE] Fwd: WGLC: draft-ietf-dart-dscp-rtp-02
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Hi,

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

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

Thanks!

Ben.

Begin forwarded message:

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


From nobody Sat Aug  9 22:57:36 2014
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From: Ross Finlayson <finlayson@live555.com>
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Subject: [AVTCORE] Minor bug in RFC 3711 (SRTP), FYI
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--Apple-Mail=_7BB2D237-67B6-4AE4-ACF0-34F669F18416
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In section 3.4 ("Secure RTCP"), there is a bug in "Figure 2. An example =
of the format of a Secure RTCP packet..."

      0                   1                   2                   3
      0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1
     +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+<+
     |V=3D2|P|    RC   |   PT=3DSR or RR   |             length          =
| |
     +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ |

The boundary between the "PT=3DSR or RR" and the "length" fields is =
wrong: The boundary is shown as being between bits 16 and 17; it should =
be between bits 15 and 16.  I.e., the "PT=3DSR or RR" field should be 8 =
bits long, not 9.

This is just a minor bug, because the equivalent diagram in RFC 3550 =
(the normative reference for RTCP) is correct.  Nonetheless, this bug =
should probably be added to the errata for RFC 3711

Ross Finlayson
Live Networks, Inc.
http://www.live555.com/


--Apple-Mail=_7BB2D237-67B6-4AE4-ACF0-34F669F18416
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<html><head><meta http-equiv=3D"Content-Type" content=3D"text/html =
charset=3Dus-ascii"></head><body style=3D"word-wrap: break-word; =
-webkit-nbsp-mode: space; -webkit-line-break: after-white-space;">In =
section 3.4 ("Secure RTCP"), there is a bug in "Figure 2.&nbsp;An =
example of the format of a Secure RTCP =
packet..."<div><br></div><div><div>&nbsp; &nbsp; &nbsp; 0 &nbsp; &nbsp; =
&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; 1 &nbsp; &nbsp; &nbsp; =
&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; 2 &nbsp; &nbsp; &nbsp; &nbsp; =
&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; 3</div><div>&nbsp; &nbsp; &nbsp; 0 1 =
2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 =
1</div><div>&nbsp; &nbsp; =
&nbsp;+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+&lt=
;+</div><div>&nbsp; &nbsp; &nbsp;|V=3D2|P| &nbsp; &nbsp;RC &nbsp; | =
&nbsp; PT=3DSR or RR &nbsp; | &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; =
length &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;| |</div><div>&nbsp; &nbsp; =
&nbsp;+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ =
|</div></div><div><br></div><div>The boundary between the "PT=3DSR or =
RR" and the "length" fields is wrong: The boundary is shown as being =
between bits 16 and 17; it should be between bits 15 and 16. &nbsp;I.e., =
the "PT=3DSR or RR" field should be 8 bits long, not =
9.</div><div><br></div><div>This is just a minor bug, because the =
equivalent diagram in RFC 3550 (the normative reference for RTCP) is =
correct. &nbsp;Nonetheless, this bug should probably be added to the =
errata for RFC 3711</div><br><div apple-content-edited=3D"true">
<span class=3D"Apple-style-span" style=3D"border-collapse: separate; =
color: rgb(0, 0, 0); font-family: Helvetica; font-style: normal; =
font-variant: normal; font-weight: normal; letter-spacing: normal; =
line-height: normal; orphans: 2; text-align: -webkit-auto; text-indent: =
0px; text-transform: none; white-space: normal; widows: 2; word-spacing: =
0px; -webkit-border-horizontal-spacing: 0px; =
-webkit-border-vertical-spacing: 0px; =
-webkit-text-decorations-in-effect: none; -webkit-text-size-adjust: =
auto; -webkit-text-stroke-width: 0px;  "><span class=3D"Apple-style-span" =
style=3D"border-collapse: separate; color: rgb(0, 0, 0); font-family: =
Helvetica; font-style: normal; font-variant: normal; font-weight: =
normal; letter-spacing: normal; line-height: normal; orphans: 2; =
text-align: -webkit-auto; text-indent: 0px; text-transform: none; =
white-space: normal; widows: 2; word-spacing: 0px; =
-webkit-border-horizontal-spacing: 0px; -webkit-border-vertical-spacing: =
0px; -webkit-text-decorations-in-effect: none; -webkit-text-size-adjust: =
auto; -webkit-text-stroke-width: 0px;  ">Ross Finlayson<br>Live =
Networks, Inc.<br><a =
href=3D"http://www.live555.com/">http://www.live555.com/</a></span></span>=

</div>
<br></body></html>=

--Apple-Mail=_7BB2D237-67B6-4AE4-ACF0-34F669F18416--


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From: Yi Cheng <yi.cheng@ericsson.com>
To: "avt@ietf.org" <avt@ietf.org>
Thread-Topic: Comments on SRTP EKT (cont'd)
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From nobody Mon Aug 11 02:43:51 2014
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From: Colin Perkins <csp@csperkins.org>
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Date: Mon, 11 Aug 2014 10:43:33 +0100
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To: Stephan Wenger <stewe@stewe.org>
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Subject: Re: [AVTCORE] Topologies-update, reply to Bo's review
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Hi Stephan,

[inline]

On 8 Aug 2014, at 18:45, Stephan Wenger <stewe@stewe.org> wrote:
> Hi Colin,
> I think I start seeing your point.  Let me work on section 3.10 a bit =
more
> (over the weekend most likely).
> But please also see inline (towards the top of the thread only).
> Thanks,
> Stephan
>=20
>=20
> On 8/8/14, 10:22 AM, "Colin Perkins" <csp@csperkins.org> wrote:
>=20
>> Hi,
>>=20
>> On 8 Aug 2014, at 16:20, Stephan Wenger <stewe@stewe.org> wrote:
>>> On 8/8/14, 7:36 AM, "Colin Perkins" <csp@csperkins.org> wrote:
>>>> They=B9re logically part of the same endpoint, and use a common =
clock,
>>>> and
>>>> this makes them semantically coupled anyway, so I think a common =
CNAME
>>>> and single RTP session makes sense (this is essentially the design =
the
>>>> old Mbone tools used to lip-sync when audio and video were done =
with
>>>> separate tools, using a coordination protocol).
>>>=20
>>> So let=B9s assume a common CNAME, but also separate transport =
addresses
>>> per subunit.  Both version 02 and 03 of the draft comment on that =
scenario
>>> as follows:
>>>=20
>>> =B3It is not really feasible to decompose one logical end-point into =
two
>>> different transport nodes in one RTP session.
>=20
> This is the statement you question.  You argue it is possible and =
still
> stay compliant with RTP.
>=20
>>> A third party monitor would report such an attempt as two entities =
being two different end-points with a CNAME collision.
>=20
> Do I understand you correctly that yes, a third party monitor would =
indeed
> report a CNAME collision, but that=B9s a problem of the third party =
monitor.

Correct. A =93CNAME collision=94 is not a problem in RTP. Two SSRCs =
sending media with the same CNAME from different source address/port =
pairs is perfectly legal.

> Or, better, the third party monitor=B9s lack of awareness of the =
existence
> of decomposed endpoint subunits in the same RTP session, them being a
> legal (and, in at least some scenarios, even desirable) design choice.

Right. This can also occur with multi-homed endpoints, sending media =
using multiple interfaces.=20

>>> As a result, a fully RTP conformant de-composited
>>> endpoint is one where the different decomposed parts use separate =
RTP
>>> sessions to send and/or receive media streams intended for them."
>=20
> And this is, according to you, an incorrect statement.

No. The RTP streams generated by a decomposed endpoint can from a single =
RTP session, or they can form multiple RTP sessions. Whether they do so =
or not depends on SSRC visibility, and is orthogonal to the choice of =
CNAME (since they=92ll always use a single CNAME) or the number of =
transport layer flows.

Cheers,
Colin


From nobody Wed Aug 13 08:14:16 2014
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Subject: Re: [AVTCORE] [rtcweb] Fwd: WGLC: draft-ietf-dart-dscp-rtp-02
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This is also of some interest to me at least, and perhaps some members
of the AQM working group who are working out weighted fair queuing and
wred like approaches to these needs.

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

Some relevant links:

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

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

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


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

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

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

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

In linux wifi at least:

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

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

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

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

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




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



--=20
Dave T=C3=A4ht

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


From nobody Mon Aug 18 08:17:54 2014
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Subject: Re: [AVTCORE] WGLC: draft-ietf-dart-dscp-rtp-02
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Hi,

[I=92m not on the DART list, so please cc me in replies]

This draft looks good overall, but I did have some comments, primarily =
about the discussion of RTP multiplexing.=20

The four bullet points in Section 2.1 that follow =93The RTCWEB protocol =
suit encompasses a number of forms of multiplexing=94 seem to confuse =
RTP sessions,
RTCP, and transport-layer flows. I suggest they be changed to:

>    The RTP streams that comprise a WebRTC session can be multiplexed=20=

>    together in a number of ways:
>=20
>    1.  Individual source streams are carried in one or more individual
>        RTP streams. These RTP streams can be multiplexed onto a single
>        transport-layer flow or sent as separate transport-layer flows.
>        This memo only considers the case where the RTP streams are to =
be
>        multiplexed onto a single transport-layer flow, forming a =
single
>        RTP session;
>=20
>    2.  The RTP Control Protocol (RTCP) [RFC3550] may be multiplexed =
onto
>        the same transport-layer flow as the RTP stream with which it =
is=20
>        associated, as described in [RFC5761], or it may be sent on a=20=

>        separate transport-layer flow;
>=20
>    3.  The RTP and RTCP traffic can be multiplexed with other =
protocols=20
>        via UDP encapsulation over a common pair of UDP ports as =
described=20
>        in [RFC5764] as updated by
>        [I-D.petithuguenin-avtcore-rfc5764-mux-fixes]; and
>=20
>    4.  The data may be further encapsulated via STUN [RFC5389] and =
TURN
>        [RFC5766] for NAT (Network Address Translator) traversal.

In the following paragraph (penultimate paragraph in Section 2.1), I =
suggest changing =93unidirectional UDP packet flow=94 to =
=93transport-layer flow=94 since it will not be unidirectional (since =
RTCP also flows in the reverse direction), and might not use UDP.=20

In the first paragraph of section 2.2, I suggest changing =93multiplexed =
over RTP sessions=94 to =93multiplexed in a single RTP session=94 and =
=93multiplexing of source streams in RTP sessions=94 to =93multiplexing =
of source streams in a single RTP session=94.=20

In Section 6, the second bullet point refers to =93an RTCP session=94. =
There is no such thing. Rather, within a single RTP session there are =
RTCP packets sent that give information about the RTP streams that are =
being sent, and that report on the reception quality of RTP streams =
being received. Using a single PHB and DSCP for all RTCP packets within =
an RTP session might make sense, but it=92s important to note that one =
role of RTCP is to provide an estimate of the round-trip time seen by =
the media, so the PHB/DSCP will have to be chosen with care to avoid =
biasing that estimate too much.

Editorial: the protocol is called WebRTC, but the working group is =
RTCWEB. Accordingly, I think all instances of =93RTCWEB=94 in this draft =
need to change to =93WebRTC=94.

Cheers,
Colin





On 9 Aug 2014, at 19:48, Ben Campbell <ben@nostrum.com> wrote:
> Hi,
>=20
> The DART working group has started a last call for =
draft-ietf-dart-dscp-rtp-02, ending on August 22. This informational =
draft describes considerations for the use of DiffServ code points in =
situations where one multiplexes multiple RTP packet streams, and =
potentially other protocol streams, that share the same 5-tuple.=20
>=20
> Since this draft is likely of interest to several working groups, I =
would like to solicit additional reviews from participants of TSVWG, =
AVTCORE, MMUSIC, CLUE, RTCWEB, and RMCAT. Please send any feedback =
(including feedback to the effect of "this is ready to progress") to the =
DART working group mailing list.
>=20
> Thanks!
>=20
> Ben.
>=20
> Begin forwarded message:
>=20
>> Resent-From: draft-alias-bounces@tools.ietf.org
>> From: Ben Campbell <ben@nostrum.com>
>> Subject: WGLC: draft-ietf-dart-dscp-rtp-02
>> Date: August 9, 2014 at 12:50:16 PM CDT
>> Resent-To: ben@nostrum.com, david.black@emc.com, =
paulej@packetizer.com
>> To: dart@ietf.org
>> Cc: draft-ietf-dart-dscp-rtp.all@tools.ietf.org, mls.ietf@gmail.com, =
Richard Barnes <rlb@ipv.sx>
>>=20
>> This is a DART Working Group Last Call of =
draft-ietf-dart-dscp-rtp-02. It's available on the data tracker at the =
following URL:
>>=20
>> https://datatracker.ietf.org/doc/draft-ietf-dart-dscp-rtp/
>>=20
>> The WGLC will conclude on 22 August, 2014. Please send your comments =
to the authors and the DART mailing list. If you've reviewed it and =
think it's good to go, please say so.
>>=20
>> Thanks!
>>=20
>> Ben.
>>=20
>=20
> _______________________________________________
> Audio/Video Transport Core Maintenance
> avt@ietf.org
> https://www.ietf.org/mailman/listinfo/avt



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




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From: Ben Campbell <ben@nostrum.com>
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Subject: Re: [AVTCORE] WGLC: draft-ietf-dart-dscp-rtp-02
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Hi,

I concur with all of Colin's comments, save one question below:

On Aug 18, 2014, at 10:17 AM, Colin Perkins <csp@csperkins.org> wrote:

> Hi,
>=20
> [I=92m not on the DART list, so please cc me in replies]
>=20
> This draft looks good overall, but I did have some comments, primarily =
about the discussion of RTP multiplexing.=20
>=20
> The four bullet points in Section 2.1 that follow =93The RTCWEB =
protocol suit encompasses a number of forms of multiplexing=94 seem to =
confuse RTP sessions,
> RTCP, and transport-layer flows. I suggest they be changed to:
>=20
>>   The RTP streams that comprise a WebRTC session can be multiplexed=20=

>>   together in a number of ways:
>>=20
>>   1.  Individual source streams are carried in one or more individual
>>       RTP streams. These RTP streams can be multiplexed onto a single
>>       transport-layer flow or sent as separate transport-layer flows.
>>       This memo only considers the case where the RTP streams are to =
be
>>       multiplexed onto a single transport-layer flow, forming a =
single
>>       RTP session;
>>=20
>>   2.  The RTP Control Protocol (RTCP) [RFC3550] may be multiplexed =
onto
>>       the same transport-layer flow as the RTP stream with which it =
is=20
>>       associated, as described in [RFC5761], or it may be sent on a=20=

>>       separate transport-layer flow;
>>=20
>>   3.  The RTP and RTCP traffic can be multiplexed with other =
protocols=20
>>       via UDP encapsulation over a common pair of UDP ports as =
described=20
>>       in [RFC5764] as updated by
>>       [I-D.petithuguenin-avtcore-rfc5764-mux-fixes]; and

Can we clarify what "other protocols" are expected? Are we just talking =
about WebRTC data-channels over SCTP/DTLS, or other things? If the =
latter, are we talking things we expect to do today, or in immediate =
futures, or extensions that might happen someday?

>>=20
>>   4.  The data may be further encapsulated via STUN [RFC5389] and =
TURN
>>       [RFC5766] for NAT (Network Address Translator) traversal.
>=20
> In the following paragraph (penultimate paragraph in Section 2.1), I =
suggest changing =93unidirectional UDP packet flow=94 to =
=93transport-layer flow=94 since it will not be unidirectional (since =
RTCP also flows in the reverse direction), and might not use UDP.=20
>=20
> In the first paragraph of section 2.2, I suggest changing =93multiplexed=
 over RTP sessions=94 to =93multiplexed in a single RTP session=94 and =
=93multiplexing of source streams in RTP sessions=94 to =93multiplexing =
of source streams in a single RTP session=94.=20
>=20
> In Section 6, the second bullet point refers to =93an RTCP session=94. =
There is no such thing. Rather, within a single RTP session there are =
RTCP packets sent that give information about the RTP streams that are =
being sent, and that report on the reception quality of RTP streams =
being received. Using a single PHB and DSCP for all RTCP packets within =
an RTP session might make sense, but it=92s important to note that one =
role of RTCP is to provide an estimate of the round-trip time seen by =
the media, so the PHB/DSCP will have to be chosen with care to avoid =
biasing that estimate too much.
>=20
> Editorial: the protocol is called WebRTC, but the working group is =
RTCWEB. Accordingly, I think all instances of =93RTCWEB=94 in this draft =
need to change to =93WebRTC=94.
>=20
> Cheers,
> Colin
>=20
>=20
>=20
>=20
>=20
> On 9 Aug 2014, at 19:48, Ben Campbell <ben@nostrum.com> wrote:
>> Hi,
>>=20
>> The DART working group has started a last call for =
draft-ietf-dart-dscp-rtp-02, ending on August 22. This informational =
draft describes considerations for the use of DiffServ code points in =
situations where one multiplexes multiple RTP packet streams, and =
potentially other protocol streams, that share the same 5-tuple.=20
>>=20
>> Since this draft is likely of interest to several working groups, I =
would like to solicit additional reviews from participants of TSVWG, =
AVTCORE, MMUSIC, CLUE, RTCWEB, and RMCAT. Please send any feedback =
(including feedback to the effect of "this is ready to progress") to the =
DART working group mailing list.
>>=20
>> Thanks!
>>=20
>> Ben.
>>=20
>> Begin forwarded message:
>>=20
>>> Resent-From: draft-alias-bounces@tools.ietf.org
>>> From: Ben Campbell <ben@nostrum.com>
>>> Subject: WGLC: draft-ietf-dart-dscp-rtp-02
>>> Date: August 9, 2014 at 12:50:16 PM CDT
>>> Resent-To: ben@nostrum.com, david.black@emc.com, =
paulej@packetizer.com
>>> To: dart@ietf.org
>>> Cc: draft-ietf-dart-dscp-rtp.all@tools.ietf.org, mls.ietf@gmail.com, =
Richard Barnes <rlb@ipv.sx>
>>>=20
>>> This is a DART Working Group Last Call of =
draft-ietf-dart-dscp-rtp-02. It's available on the data tracker at the =
following URL:
>>>=20
>>> https://datatracker.ietf.org/doc/draft-ietf-dart-dscp-rtp/
>>>=20
>>> The WGLC will conclude on 22 August, 2014. Please send your comments =
to the authors and the DART mailing list. If you've reviewed it and =
think it's good to go, please say so.
>>>=20
>>> Thanks!
>>>=20
>>> Ben.
>>>=20
>>=20
>> _______________________________________________
>> Audio/Video Transport Core Maintenance
>> avt@ietf.org
>> https://www.ietf.org/mailman/listinfo/avt
>=20
>=20
>=20
> --=20
> Colin Perkins
> http://csperkins.org/
>=20
>=20
>=20


From nobody Mon Aug 18 23:50:30 2014
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A New Internet-Draft is available from the on-line Internet-Drafts directories.
 This draft is a work item of the Audio/Video Transport Core Maintenance Working Group of the IETF.

        Title           : RTP Topologies
        Authors         : Magnus Westerlund
                          Stephan Wenger
	Filename        : draft-ietf-avtcore-rtp-topologies-update-04.txt
	Pages           : 45
	Date            : 2014-08-18

Abstract:
   This document discusses point to point and multi-endpoint topologies
   used in Real-time Transport Protocol (RTP)-based environments.  In
   particular, centralized topologies commonly employed in the video
   conferencing industry are mapped to the RTP terminology.

   This document is updated with additional topologies and is intended
   to replace RFC 5117.


The IETF datatracker status page for this draft is:
https://datatracker.ietf.org/doc/draft-ietf-avtcore-rtp-topologies-update/

There's also a htmlized version available at:
http://tools.ietf.org/html/draft-ietf-avtcore-rtp-topologies-update-04

A diff from the previous version is available at:
http://www.ietf.org/rfcdiff?url2=draft-ietf-avtcore-rtp-topologies-update-04


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

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


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From: Stephan Wenger <stewe@stewe.org>
To: "avt@ietf.org" <avt@ietf.org>
Thread-Topic: RTP topologies update -04 posted
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--_000_D018427D461C1stewesteweorg_
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Hi,
I just posted version 4 of the RTP topologies update I-D.  The only technic=
al changes are in section 3.10 on the subject of split component terminals,=
 as per discussions on this list over the last two weeks, as well as privat=
e conversations I had with Colin Perkins afterwards.  Thanks again to Colin=
 for his careful review and guidance.
Please review.
Thanks,
Stephan



--_000_D018427D461C1stewesteweorg_
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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; color: rgb(0, 0, 0); font-size: 14px; font-fami=
ly: Calibri, sans-serif;">
<div>Hi,</div>
<div>I just posted version 4 of the RTP topologies update I-D. &nbsp;The on=
ly technical changes are in section 3.10 on the subject of split component =
terminals, as per discussions on this list over the last two weeks, as well=
 as private conversations I had with
 Colin Perkins afterwards. &nbsp;Thanks again to Colin for his careful revi=
ew and guidance.</div>
<div>Please review.</div>
<div>Thanks,</div>
<div>Stephan</div>
<div><br>
</div>
<div><br>
</div>
</body>
</html>

--_000_D018427D461C1stewesteweorg_--


From nobody Fri Aug 22 08:40:57 2014
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From: "Black, David" <david.black@emc.com>
To: Ben Campbell <ben@nostrum.com>, Colin Perkins <csp@csperkins.org>
Date: Thu, 21 Aug 2014 21:06:14 -0400
Thread-Topic: [Dart] [AVTCORE] WGLC: draft-ietf-dart-dscp-rtp-02
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Subject: Re: [AVTCORE] [Dart]  WGLC: draft-ietf-dart-dscp-rtp-02
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Colin - many thanks for the review.  I'll pick up your proposed changes, pe=
rhaps
with minor edits.

Ben,

> >>   3.  The RTP and RTCP traffic can be multiplexed with other protocols
> >>       via UDP encapsulation over a common pair of UDP ports as describ=
ed
> >>       in [RFC5764] as updated by
> >>       [I-D.petithuguenin-avtcore-rfc5764-mux-fixes]; and
>=20
> Can we clarify what "other protocols" are expected? Are we just talking a=
bout
> WebRTC data-channels over SCTP/DTLS, or other things? If the latter, are =
we
> talking things we expect to do today, or in immediate futures, or extensi=
ons
> that might happen someday?

WebRTC will only multiplex its data channels, and this numbered list is abo=
ut
WebRTC, so this text should focus on those data channels - I'll revise the =
draft
accordingly.

Other text elsewhere in the draft is broader (e.g., SCTP/UDP without DTLS, =
which
is of interest, even though WebRTC always uses DTLS).

Thanks,
--David

> -----Original Message-----
> From: Dart [mailto:dart-bounces@ietf.org] On Behalf Of Ben Campbell
> Sent: Monday, August 18, 2014 4:48 PM
> To: Colin Perkins
> Cc: draft-ietf-dart-dscp-rtp.all@tools.ietf.org; dart@ietf.org; avt@ietf.=
org
> WG
> Subject: Re: [Dart] [AVTCORE] WGLC: draft-ietf-dart-dscp-rtp-02
>=20
> Hi,
>=20
> I concur with all of Colin's comments, save one question below:
>=20
> On Aug 18, 2014, at 10:17 AM, Colin Perkins <csp@csperkins.org> wrote:
>=20
> > Hi,
> >
> > [I'm not on the DART list, so please cc me in replies]
> >
> > This draft looks good overall, but I did have some comments, primarily =
about
> the discussion of RTP multiplexing.
> >
> > The four bullet points in Section 2.1 that follow "The RTCWEB protocol =
suit
> encompasses a number of forms of multiplexing" seem to confuse RTP sessio=
ns,
> > RTCP, and transport-layer flows. I suggest they be changed to:
> >=20
> >>   The RTP streams that comprise a WebRTC session can be multiplexed
> >>   together in a number of ways:
> >>
> >>   1.  Individual source streams are carried in one or more individual
> >>       RTP streams. These RTP streams can be multiplexed onto a single
> >>       transport-layer flow or sent as separate transport-layer flows.
> >>       This memo only considers the case where the RTP streams are to b=
e
> >>       multiplexed onto a single transport-layer flow, forming a single
> >>       RTP session;
> >>
> >>   2.  The RTP Control Protocol (RTCP) [RFC3550] may be multiplexed ont=
o
> >>       the same transport-layer flow as the RTP stream with which it is
> >>       associated, as described in [RFC5761], or it may be sent on a
> >>       separate transport-layer flow;
> >>
> >>   3.  The RTP and RTCP traffic can be multiplexed with other protocols
> >>       via UDP encapsulation over a common pair of UDP ports as describ=
ed
> >>       in [RFC5764] as updated by
> >>       [I-D.petithuguenin-avtcore-rfc5764-mux-fixes]; and
>=20
> Can we clarify what "other protocols" are expected? Are we just talking a=
bout
> WebRTC data-channels over SCTP/DTLS, or other things? If the latter, are =
we
> talking things we expect to do today, or in immediate futures, or extensi=
ons
> that might happen someday?
>=20
> >>
> >>   4.  The data may be further encapsulated via STUN [RFC5389] and TURN
> >>       [RFC5766] for NAT (Network Address Translator) traversal.
> >
> > In the following paragraph (penultimate paragraph in Section 2.1), I su=
ggest
> changing "unidirectional UDP packet flow" to "transport-layer flow" since=
 it
> will not be unidirectional (since RTCP also flows in the reverse directio=
n),
> and might not use UDP.
> >
> > In the first paragraph of section 2.2, I suggest changing "multiplexed =
over
> RTP sessions" to "multiplexed in a single RTP session" and "multiplexing =
of
> source streams in RTP sessions" to "multiplexing of source streams in a s=
ingle
> RTP session".
> >
> > In Section 6, the second bullet point refers to "an RTCP session". Ther=
e is
> no such thing. Rather, within a single RTP session there are RTCP packets=
 sent
> that give information about the RTP streams that are being sent, and that
> report on the reception quality of RTP streams being received. Using a si=
ngle
> PHB and DSCP for all RTCP packets within an RTP session might make sense,=
 but
> it's important to note that one role of RTCP is to provide an estimate of=
 the
> round-trip time seen by the media, so the PHB/DSCP will have to be chosen=
 with
> care to avoid biasing that estimate too much.
> >
> > Editorial: the protocol is called WebRTC, but the working group is RTCW=
EB.
> Accordingly, I think all instances of "RTCWEB" in this draft need to chan=
ge to
> "WebRTC".
> >
> > Cheers,
> > Colin
> >
> >
> >
> >
> >
> > On 9 Aug 2014, at 19:48, Ben Campbell <ben@nostrum.com> wrote:
> >> Hi,
> >>
> >> The DART working group has started a last call for draft-ietf-dart-dsc=
p-
> rtp-02, ending on August 22. This informational draft describes considera=
tions
> for the use of DiffServ code points in situations where one multiplexes
> multiple RTP packet streams, and potentially other protocol streams, that
> share the same 5-tuple.
> >>
> >> Since this draft is likely of interest to several working groups, I wo=
uld
> like to solicit additional reviews from participants of TSVWG, AVTCORE,
> MMUSIC, CLUE, RTCWEB, and RMCAT. Please send any feedback (including feed=
back
> to the effect of "this is ready to progress") to the DART working group
> mailing list.
> >>
> >> Thanks!
> >>
> >> Ben.
> >>
> >> Begin forwarded message:
> >>
> >>> Resent-From: draft-alias-bounces@tools.ietf.org
> >>> From: Ben Campbell <ben@nostrum.com>
> >>> Subject: WGLC: draft-ietf-dart-dscp-rtp-02
> >>> Date: August 9, 2014 at 12:50:16 PM CDT
> >>> Resent-To: ben@nostrum.com, david.black@emc.com, paulej@packetizer.co=
m
> >>> To: dart@ietf.org
> >>> Cc: draft-ietf-dart-dscp-rtp.all@tools.ietf.org, mls.ietf@gmail.com,
> Richard Barnes <rlb@ipv.sx>
> >>>
> >>> This is a DART Working Group Last Call of draft-ietf-dart-dscp-rtp-02=
.
> It's available on the data tracker at the following URL:
> >>>
> >>> https://datatracker.ietf.org/doc/draft-ietf-dart-dscp-rtp/
> >>>
> >>> The WGLC will conclude on 22 August, 2014. Please send your comments =
to
> the authors and the DART mailing list. If you've reviewed it and think it=
's
> good to go, please say so.
> >>>
> >>> Thanks!
> >>>
> >>> Ben.
> >>>
> >>
> >> _______________________________________________
> >> Audio/Video Transport Core Maintenance
> >> avt@ietf.org
> >> https://www.ietf.org/mailman/listinfo/avt
> >
> >
> >
> > --
> > Colin Perkins
> > http://csperkins.org/
> >
> >
> >
>=20
> _______________________________________________
> Dart mailing list
> Dart@ietf.org
> https://www.ietf.org/mailman/listinfo/dart


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From: Ben Campbell <ben@nostrum.com>
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Subject: Re: [AVTCORE] WGLC: draft-ietf-dart-dscp-rtp-02 - proposed text for section 2.1 mux items 3 and 4
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I agree except where noted inline:

On Aug 22, 2014, at 3:27 PM, Black, David <david.black@emc.com> wrote:

>=20
> Proposed new text (merging item 4 into item 3, which aligns with the
> structure of RFC 5764 and the rfc5764-mux-fixes draft):
>=20
>   3.  An RTP session could be multiplexed with a single SCTP over DTLS
>       association plus signaling traffic for STUN [RFC5389] and TURN
>       [RFC5766] into a single transport level flow as described in
>       [RFC5764] and [I-D.petithuguenin-avtcore-rfc5764-mux-fixes].
>       STUN and TURN are used for NAT traversal.
>=20

I agree with the direction here--but I'm not sure I would characterize =
STUN and TURN as signaling traffic. And STUN is used for connectivity =
checks, as well as NAT traversal. OTOH, maybe it would be better to just =
for this section to just say that STUN and TURN can be multiplexed in =
the flow without the additional comments about them?


> Thanks,
> --David
>=20
>=20
>> -----Original Message-----
>> From: Black, David
>> Sent: Thursday, August 21, 2014 9:06 PM
>> To: Ben Campbell; Colin Perkins
>> Cc: draft-ietf-dart-dscp-rtp.all@tools.ietf.org; dart@ietf.org; =
avt@ietf.org
>> WG; Black, David
>> Subject: RE: [Dart] [AVTCORE] WGLC: draft-ietf-dart-dscp-rtp-02
>>=20
>> Colin - many thanks for the review.  I'll pick up your proposed =
changes,
>> perhaps
>> with minor edits.
>>=20
>> Ben,
>>=20
>>>>>  3.  The RTP and RTCP traffic can be multiplexed with other =
protocols
>>>>>      via UDP encapsulation over a common pair of UDP ports as =
described
>>>>>      in [RFC5764] as updated by
>>>>>      [I-D.petithuguenin-avtcore-rfc5764-mux-fixes]; and
>>>=20
>>> Can we clarify what "other protocols" are expected? Are we just =
talking
>> about
>>> WebRTC data-channels over SCTP/DTLS, or other things? If the latter, =
are we
>>> talking things we expect to do today, or in immediate futures, or =
extensions
>>> that might happen someday?
>>=20
>> WebRTC will only multiplex its data channels, and this numbered list =
is about
>> WebRTC, so this text should focus on those data channels - I'll =
revise the
>> draft
>> accordingly.
>>=20
>> Other text elsewhere in the draft is broader (e.g., SCTP/UDP without =
DTLS,
>> which
>> is of interest, even though WebRTC always uses DTLS).
>>=20
>> Thanks,
>> --David
>>=20
>>> -----Original Message-----
>>> From: Dart [mailto:dart-bounces@ietf.org] On Behalf Of Ben Campbell
>>> Sent: Monday, August 18, 2014 4:48 PM
>>> To: Colin Perkins
>>> Cc: draft-ietf-dart-dscp-rtp.all@tools.ietf.org; dart@ietf.org; =
avt@ietf.org
>>> WG
>>> Subject: Re: [Dart] [AVTCORE] WGLC: draft-ietf-dart-dscp-rtp-02
>>>=20
>>> Hi,
>>>=20
>>> I concur with all of Colin's comments, save one question below:
>>>=20
>>> On Aug 18, 2014, at 10:17 AM, Colin Perkins <csp@csperkins.org> =
wrote:
>>>=20
>>>> Hi,
>>>>=20
>>>> [I'm not on the DART list, so please cc me in replies]
>>>>=20
>>>> This draft looks good overall, but I did have some comments, =
primarily
>> about
>>> the discussion of RTP multiplexing.
>>>>=20
>>>> The four bullet points in Section 2.1 that follow "The RTCWEB =
protocol
>> suit
>>> encompasses a number of forms of multiplexing" seem to confuse RTP =
sessions,
>>>> RTCP, and transport-layer flows. I suggest they be changed to:
>>>>=20
>>>>>  The RTP streams that comprise a WebRTC session can be multiplexed
>>>>>  together in a number of ways:
>>>>>=20
>>>>>  1.  Individual source streams are carried in one or more =
individual
>>>>>      RTP streams. These RTP streams can be multiplexed onto a =
single
>>>>>      transport-layer flow or sent as separate transport-layer =
flows.
>>>>>      This memo only considers the case where the RTP streams are =
to be
>>>>>      multiplexed onto a single transport-layer flow, forming a =
single
>>>>>      RTP session;
>>>>>=20
>>>>>  2.  The RTP Control Protocol (RTCP) [RFC3550] may be multiplexed =
onto
>>>>>      the same transport-layer flow as the RTP stream with which it =
is
>>>>>      associated, as described in [RFC5761], or it may be sent on a
>>>>>      separate transport-layer flow;
>>>>>=20
>>>>>  3.  The RTP and RTCP traffic can be multiplexed with other =
protocols
>>>>>      via UDP encapsulation over a common pair of UDP ports as =
described
>>>>>      in [RFC5764] as updated by
>>>>>      [I-D.petithuguenin-avtcore-rfc5764-mux-fixes]; and
>>>=20
>>> Can we clarify what "other protocols" are expected? Are we just =
talking
>> about
>>> WebRTC data-channels over SCTP/DTLS, or other things? If the latter, =
are we
>>> talking things we expect to do today, or in immediate futures, or =
extensions
>>> that might happen someday?
>>>=20
>>>>>=20
>>>>>  4.  The data may be further encapsulated via STUN [RFC5389] and =
TURN
>>>>>      [RFC5766] for NAT (Network Address Translator) traversal.
>>>>=20
>>>> In the following paragraph (penultimate paragraph in Section 2.1), =
I
>> suggest
>>> changing "unidirectional UDP packet flow" to "transport-layer flow" =
since it
>>> will not be unidirectional (since RTCP also flows in the reverse =
direction),
>>> and might not use UDP.
>>>>=20
>>>> In the first paragraph of section 2.2, I suggest changing =
"multiplexed
>> over
>>> RTP sessions" to "multiplexed in a single RTP session" and =
"multiplexing of
>>> source streams in RTP sessions" to "multiplexing of source streams =
in a
>> single
>>> RTP session".
>>>>=20
>>>> In Section 6, the second bullet point refers to "an RTCP session". =
There
>> is
>>> no such thing. Rather, within a single RTP session there are RTCP =
packets
>> sent
>>> that give information about the RTP streams that are being sent, and =
that
>>> report on the reception quality of RTP streams being received. Using =
a
>> single
>>> PHB and DSCP for all RTCP packets within an RTP session might make =
sense,
>> but
>>> it's important to note that one role of RTCP is to provide an =
estimate of
>> the
>>> round-trip time seen by the media, so the PHB/DSCP will have to be =
chosen
>> with
>>> care to avoid biasing that estimate too much.
>>>>=20
>>>> Editorial: the protocol is called WebRTC, but the working group is =
RTCWEB.
>>> Accordingly, I think all instances of "RTCWEB" in this draft need to =
change
>> to
>>> "WebRTC".
>>>>=20
>>>> Cheers,
>>>> Colin
>>>>=20
>>>>=20
>>>>=20
>>>>=20
>>>>=20
>>>> On 9 Aug 2014, at 19:48, Ben Campbell <ben@nostrum.com> wrote:
>>>>> Hi,
>>>>>=20
>>>>> The DART working group has started a last call for =
draft-ietf-dart-dscp-
>>> rtp-02, ending on August 22. This informational draft describes
>> considerations
>>> for the use of DiffServ code points in situations where one =
multiplexes
>>> multiple RTP packet streams, and potentially other protocol streams, =
that
>>> share the same 5-tuple.
>>>>>=20
>>>>> Since this draft is likely of interest to several working groups, =
I would
>>> like to solicit additional reviews from participants of TSVWG, =
AVTCORE,
>>> MMUSIC, CLUE, RTCWEB, and RMCAT. Please send any feedback (including
>> feedback
>>> to the effect of "this is ready to progress") to the DART working =
group
>>> mailing list.
>>>>>=20
>>>>> Thanks!
>>>>>=20
>>>>> Ben.
>>>>>=20
>>>>> Begin forwarded message:
>>>>>=20
>>>>>> Resent-From: draft-alias-bounces@tools.ietf.org
>>>>>> From: Ben Campbell <ben@nostrum.com>
>>>>>> Subject: WGLC: draft-ietf-dart-dscp-rtp-02
>>>>>> Date: August 9, 2014 at 12:50:16 PM CDT
>>>>>> Resent-To: ben@nostrum.com, david.black@emc.com, =
paulej@packetizer.com
>>>>>> To: dart@ietf.org
>>>>>> Cc: draft-ietf-dart-dscp-rtp.all@tools.ietf.org, =
mls.ietf@gmail.com,
>>> Richard Barnes <rlb@ipv.sx>
>>>>>>=20
>>>>>> This is a DART Working Group Last Call of =
draft-ietf-dart-dscp-rtp-02.
>>> It's available on the data tracker at the following URL:
>>>>>>=20
>>>>>> https://datatracker.ietf.org/doc/draft-ietf-dart-dscp-rtp/
>>>>>>=20
>>>>>> The WGLC will conclude on 22 August, 2014. Please send your =
comments to
>>> the authors and the DART mailing list. If you've reviewed it and =
think it's
>>> good to go, please say so.
>>>>>>=20
>>>>>> Thanks!
>>>>>>=20
>>>>>> Ben.
>>>>>>=20
>>>>>=20
>>>>> _______________________________________________
>>>>> Audio/Video Transport Core Maintenance
>>>>> avt@ietf.org
>>>>> https://www.ietf.org/mailman/listinfo/avt
>>>>=20
>>>>=20
>>>>=20
>>>> --
>>>> Colin Perkins
>>>> http://csperkins.org/
>>>>=20
>>>>=20
>>>>=20
>>>=20
>>> _______________________________________________
>>> Dart mailing list
>>> Dart@ietf.org
>>> https://www.ietf.org/mailman/listinfo/dart


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From: "Black, David" <david.black@emc.com>
To: Colin Perkins <csp@csperkins.org>, "dart@ietf.org" <dart@ietf.org>
Date: Fri, 22 Aug 2014 16:12:11 -0400
Thread-Topic: Colin Perkins comments -  WGLC: draft-ietf-dart-dscp-rtp-02
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Subject: [AVTCORE] Colin Perkins comments - WGLC: draft-ietf-dart-dscp-rtp-02
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Hi Colin,

I've picked up your changes (including RTCWEB -> WebRTC globally)
in my working draft, with two exceptions:

(1) Items 3 and 4 in the list of what can be multiplexed.

I don't think you suggested any changes to these two items, but
they need significant attention for other reasons:

- Ben's comment that item 3 ought to focus on the WebRTC data channel
- STUN doesn't encapsulate WebRTC traffic (TURN does).

I plan to merge these items 3 and 4 as part of revising them, and
intend to move the topic of TURN encapsulation out to be discussed
elsewhere, as it's independent of multiplexing.  I'll propose
specific text in a separate message.

(2) PHBs and DSCPs for RTCP.

I'm not sure what to say about this, as it looks like I need to set
off a discussion (debate?) between you and my co-author, Paul Jones. =20

Colin (from WGLC comments):

> Rather, within a single RTP session there are RTCP packets sent
> that give information about the RTP streams that are being sent, and that
> report on the reception quality of RTP streams being received. Using a si=
ngle
> PHB and DSCP for all RTCP packets within an RTP session might make sense,=
 but
> it's important to note that one role of RTCP is to provide an estimate of=
 the
> round-trip time seen by the media, so the PHB/DSCP will have to be chosen=
 with
> care to avoid biasing that estimate too much.

Paul (from shortly after the Toronto meeting:

> During the meeting, there was discussion of marking RTCP packets.  Some
> notes I received on this topic suggested that it was proposed that RTCP
> should be marked the same as for RTP.  The argument was that this is used
> for RTT calculations.  If that is what was said, I'd like to state my
> disagreement. :-)
>=20
> The forward and reverse paths are not necessarily the same and there is
> nothing one should assume about the reverse path to provide guidance abou=
t
> the forward path (or vice versa).  As perhaps a gross example, I have the
> ability to download far faster on my home Internet connection than I can
> upload.  Other important traffic characteristics differ in each direction=
.
> =20
> Further, an RTCP packet might provide information related to several diff=
erent
> RTP packets.  I certainly would not want to see one RTCP packet per RTP p=
acket.

I'll watch where that discussion goes before proposing text.

Thanks,
--David


> -----Original Message-----
> From: Colin Perkins [mailto:csp@csperkins.org]
> Sent: Monday, August 18, 2014 11:18 AM
> To: dart@ietf.org
> Cc: avt@ietf.org WG; Ben Campbell; draft-ietf-dart-dscp-rtp.all@tools.iet=
f.org
> Subject: Re: [AVTCORE] WGLC: draft-ietf-dart-dscp-rtp-02
>=20
> Hi,
>=20
> [I'm not on the DART list, so please cc me in replies]
>=20
> This draft looks good overall, but I did have some comments, primarily ab=
out
> the discussion of RTP multiplexing.
>=20
> The four bullet points in Section 2.1 that follow "The RTCWEB protocol su=
it
> encompasses a number of forms of multiplexing" seem to confuse RTP sessio=
ns,
> RTCP, and transport-layer flows. I suggest they be changed to:
>=20
> >    The RTP streams that comprise a WebRTC session can be multiplexed
> >    together in a number of ways:
> >
> >    1.  Individual source streams are carried in one or more individual
> >        RTP streams. These RTP streams can be multiplexed onto a single
> >        transport-layer flow or sent as separate transport-layer flows.
> >        This memo only considers the case where the RTP streams are to b=
e
> >        multiplexed onto a single transport-layer flow, forming a single
> >        RTP session;
> >
> >    2.  The RTP Control Protocol (RTCP) [RFC3550] may be multiplexed ont=
o
> >        the same transport-layer flow as the RTP stream with which it is
> >        associated, as described in [RFC5761], or it may be sent on a
> >        separate transport-layer flow;
> >
> >    3.  The RTP and RTCP traffic can be multiplexed with other protocols
> >        via UDP encapsulation over a common pair of UDP ports as describ=
ed
> >        in [RFC5764] as updated by
> >        [I-D.petithuguenin-avtcore-rfc5764-mux-fixes]; and
> >
> >    4.  The data may be further encapsulated via STUN [RFC5389] and TURN
> >        [RFC5766] for NAT (Network Address Translator) traversal.
>=20
> In the following paragraph (penultimate paragraph in Section 2.1), I sugg=
est
> changing "unidirectional UDP packet flow" to "transport-layer flow" since=
 it
> will not be unidirectional (since RTCP also flows in the reverse directio=
n),
> and might not use UDP.
>=20
> In the first paragraph of section 2.2, I suggest changing "multiplexed ov=
er
> RTP sessions" to "multiplexed in a single RTP session" and "multiplexing =
of
> source streams in RTP sessions" to "multiplexing of source streams in a s=
ingle
> RTP session".
>=20
> In Section 6, the second bullet point refers to "an RTCP session". There =
is no
> such thing. Rather, within a single RTP session there are RTCP packets se=
nt
> that give information about the RTP streams that are being sent, and that
> report on the reception quality of RTP streams being received. Using a si=
ngle
> PHB and DSCP for all RTCP packets within an RTP session might make sense,=
 but
> it's important to note that one role of RTCP is to provide an estimate of=
 the
> round-trip time seen by the media, so the PHB/DSCP will have to be chosen=
 with
> care to avoid biasing that estimate too much.
>=20
> Editorial: the protocol is called WebRTC, but the working group is RTCWEB=
.
> Accordingly, I think all instances of "RTCWEB" in this draft need to chan=
ge to
> "WebRTC".
>=20
> Cheers,
> Colin
>=20
>=20
>=20
>=20
>=20
> On 9 Aug 2014, at 19:48, Ben Campbell <ben@nostrum.com> wrote:
> > Hi,
> >
> > The DART working group has started a last call for draft-ietf-dart-dscp=
-rtp-
> 02, ending on August 22. This informational draft describes consideration=
s for
> the use of DiffServ code points in situations where one multiplexes multi=
ple
> RTP packet streams, and potentially other protocol streams, that share th=
e
> same 5-tuple.
> >
> > Since this draft is likely of interest to several working groups, I wou=
ld
> like to solicit additional reviews from participants of TSVWG, AVTCORE,
> MMUSIC, CLUE, RTCWEB, and RMCAT. Please send any feedback (including feed=
back
> to the effect of "this is ready to progress") to the DART working group
> mailing list.
> >
> > Thanks!
> >
> > Ben.
> >
> > Begin forwarded message:
> >
> >> Resent-From: draft-alias-bounces@tools.ietf.org
> >> From: Ben Campbell <ben@nostrum.com>
> >> Subject: WGLC: draft-ietf-dart-dscp-rtp-02
> >> Date: August 9, 2014 at 12:50:16 PM CDT
> >> Resent-To: ben@nostrum.com, david.black@emc.com, paulej@packetizer.com
> >> To: dart@ietf.org
> >> Cc: draft-ietf-dart-dscp-rtp.all@tools.ietf.org, mls.ietf@gmail.com,
> Richard Barnes <rlb@ipv.sx>
> >>
> >> This is a DART Working Group Last Call of draft-ietf-dart-dscp-rtp-02.=
 It's
> available on the data tracker at the following URL:
> >>
> >> https://datatracker.ietf.org/doc/draft-ietf-dart-dscp-rtp/
> >>
> >> The WGLC will conclude on 22 August, 2014. Please send your comments t=
o the
> authors and the DART mailing list. If you've reviewed it and think it's g=
ood
> to go, please say so.
> >>
> >> Thanks!
> >>
> >> Ben.
> >>
> >
> > _______________________________________________
> > Audio/Video Transport Core Maintenance
> > avt@ietf.org
> > https://www.ietf.org/mailman/listinfo/avt
>=20
>=20
>=20
> --
> Colin Perkins
> http://csperkins.org/
>=20
>=20


From nobody Sat Aug 23 09:09:13 2014
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From: "Black, David" <david.black@emc.com>
To: Ben Campbell <ben@nostrum.com>, Colin Perkins <csp@csperkins.org>
Date: Fri, 22 Aug 2014 16:27:43 -0400
Thread-Topic: WGLC: draft-ietf-dart-dscp-rtp-02 - proposed text for section 2.1 mux items 3 and 4
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Cc: "draft-ietf-dart-dscp-rtp.all@tools.ietf.org" <draft-ietf-dart-dscp-rtp.all@tools.ietf.org>, "Black, David" <david.black@emc.com>, "dart@ietf.org" <dart@ietf.org>, "avt@ietf.org WG" <avt@ietf.org>
Subject: [AVTCORE] WGLC: draft-ietf-dart-dscp-rtp-02 - proposed text for section 2.1 mux items 3 and 4
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Section 2.1 of the draft currently has this text on WebRTC multiplexing:

   3.  An RTP session could be multiplexed with other protocols via UDP
       encapsulation over a common pair of UDP ports as described in
       [RFC5764] as updated by
       [I-D.petithuguenin-avtcore-rfc5764-mux-fixes]; and

   4.  The data may be further encapsulated via STUN [RFC5389] and TURN
       [RFC5766] for NAT (Network Address Translator) traversal.

That text has at least a couple of problems:

	- This list is about WebRTC, so item 3 ought to specifically
		focus on the WebRTC data channel.
	- STUN doesn't encapsulate WebRTC traffic (TURN does).

On the latter, for simplicity, I plan to remove the mention of
encapsulation from here and deal with it elsewhere as part of
discussion of what happens if TURN decides to send all the traffic
over TCP (that'll be in a separate message, stay tuned).

Proposed new text (merging item 4 into item 3, which aligns with the
structure of RFC 5764 and the rfc5764-mux-fixes draft):

   3.  An RTP session could be multiplexed with a single SCTP over DTLS
       association plus signaling traffic for STUN [RFC5389] and TURN
       [RFC5766] into a single transport level flow as described in
       [RFC5764] and [I-D.petithuguenin-avtcore-rfc5764-mux-fixes].
       STUN and TURN are used for NAT traversal.

Thanks,
--David


> -----Original Message-----
> From: Black, David
> Sent: Thursday, August 21, 2014 9:06 PM
> To: Ben Campbell; Colin Perkins
> Cc: draft-ietf-dart-dscp-rtp.all@tools.ietf.org; dart@ietf.org; avt@ietf.=
org
> WG; Black, David
> Subject: RE: [Dart] [AVTCORE] WGLC: draft-ietf-dart-dscp-rtp-02
>=20
> Colin - many thanks for the review.  I'll pick up your proposed changes,
> perhaps
> with minor edits.
>=20
> Ben,
>=20
> > >>   3.  The RTP and RTCP traffic can be multiplexed with other protoco=
ls
> > >>       via UDP encapsulation over a common pair of UDP ports as descr=
ibed
> > >>       in [RFC5764] as updated by
> > >>       [I-D.petithuguenin-avtcore-rfc5764-mux-fixes]; and
> >
> > Can we clarify what "other protocols" are expected? Are we just talking
> about
> > WebRTC data-channels over SCTP/DTLS, or other things? If the latter, ar=
e we
> > talking things we expect to do today, or in immediate futures, or exten=
sions
> > that might happen someday?
>=20
> WebRTC will only multiplex its data channels, and this numbered list is a=
bout
> WebRTC, so this text should focus on those data channels - I'll revise th=
e
> draft
> accordingly.
>=20
> Other text elsewhere in the draft is broader (e.g., SCTP/UDP without DTLS=
,
> which
> is of interest, even though WebRTC always uses DTLS).
>=20
> Thanks,
> --David
>=20
> > -----Original Message-----
> > From: Dart [mailto:dart-bounces@ietf.org] On Behalf Of Ben Campbell
> > Sent: Monday, August 18, 2014 4:48 PM
> > To: Colin Perkins
> > Cc: draft-ietf-dart-dscp-rtp.all@tools.ietf.org; dart@ietf.org; avt@iet=
f.org
> > WG
> > Subject: Re: [Dart] [AVTCORE] WGLC: draft-ietf-dart-dscp-rtp-02
> >
> > Hi,
> >
> > I concur with all of Colin's comments, save one question below:
> >
> > On Aug 18, 2014, at 10:17 AM, Colin Perkins <csp@csperkins.org> wrote:
> >
> > > Hi,
> > >
> > > [I'm not on the DART list, so please cc me in replies]
> > >
> > > This draft looks good overall, but I did have some comments, primaril=
y
> about
> > the discussion of RTP multiplexing.
> > >
> > > The four bullet points in Section 2.1 that follow "The RTCWEB protoco=
l
> suit
> > encompasses a number of forms of multiplexing" seem to confuse RTP sess=
ions,
> > > RTCP, and transport-layer flows. I suggest they be changed to:
> > >
> > >>   The RTP streams that comprise a WebRTC session can be multiplexed
> > >>   together in a number of ways:
> > >>
> > >>   1.  Individual source streams are carried in one or more individua=
l
> > >>       RTP streams. These RTP streams can be multiplexed onto a singl=
e
> > >>       transport-layer flow or sent as separate transport-layer flows=
.
> > >>       This memo only considers the case where the RTP streams are to=
 be
> > >>       multiplexed onto a single transport-layer flow, forming a sing=
le
> > >>       RTP session;
> > >>
> > >>   2.  The RTP Control Protocol (RTCP) [RFC3550] may be multiplexed o=
nto
> > >>       the same transport-layer flow as the RTP stream with which it =
is
> > >>       associated, as described in [RFC5761], or it may be sent on a
> > >>       separate transport-layer flow;
> > >>
> > >>   3.  The RTP and RTCP traffic can be multiplexed with other protoco=
ls
> > >>       via UDP encapsulation over a common pair of UDP ports as descr=
ibed
> > >>       in [RFC5764] as updated by
> > >>       [I-D.petithuguenin-avtcore-rfc5764-mux-fixes]; and
> >
> > Can we clarify what "other protocols" are expected? Are we just talking
> about
> > WebRTC data-channels over SCTP/DTLS, or other things? If the latter, ar=
e we
> > talking things we expect to do today, or in immediate futures, or exten=
sions
> > that might happen someday?
> >
> > >>
> > >>   4.  The data may be further encapsulated via STUN [RFC5389] and TU=
RN
> > >>       [RFC5766] for NAT (Network Address Translator) traversal.
> > >
> > > In the following paragraph (penultimate paragraph in Section 2.1), I
> suggest
> > changing "unidirectional UDP packet flow" to "transport-layer flow" sin=
ce it
> > will not be unidirectional (since RTCP also flows in the reverse direct=
ion),
> > and might not use UDP.
> > >
> > > In the first paragraph of section 2.2, I suggest changing "multiplexe=
d
> over
> > RTP sessions" to "multiplexed in a single RTP session" and "multiplexin=
g of
> > source streams in RTP sessions" to "multiplexing of source streams in a
> single
> > RTP session".
> > >
> > > In Section 6, the second bullet point refers to "an RTCP session". Th=
ere
> is
> > no such thing. Rather, within a single RTP session there are RTCP packe=
ts
> sent
> > that give information about the RTP streams that are being sent, and th=
at
> > report on the reception quality of RTP streams being received. Using a
> single
> > PHB and DSCP for all RTCP packets within an RTP session might make sens=
e,
> but
> > it's important to note that one role of RTCP is to provide an estimate =
of
> the
> > round-trip time seen by the media, so the PHB/DSCP will have to be chos=
en
> with
> > care to avoid biasing that estimate too much.
> > >
> > > Editorial: the protocol is called WebRTC, but the working group is RT=
CWEB.
> > Accordingly, I think all instances of "RTCWEB" in this draft need to ch=
ange
> to
> > "WebRTC".
> > >
> > > Cheers,
> > > Colin
> > >
> > >
> > >
> > >
> > >
> > > On 9 Aug 2014, at 19:48, Ben Campbell <ben@nostrum.com> wrote:
> > >> Hi,
> > >>
> > >> The DART working group has started a last call for draft-ietf-dart-d=
scp-
> > rtp-02, ending on August 22. This informational draft describes
> considerations
> > for the use of DiffServ code points in situations where one multiplexes
> > multiple RTP packet streams, and potentially other protocol streams, th=
at
> > share the same 5-tuple.
> > >>
> > >> Since this draft is likely of interest to several working groups, I =
would
> > like to solicit additional reviews from participants of TSVWG, AVTCORE,
> > MMUSIC, CLUE, RTCWEB, and RMCAT. Please send any feedback (including
> feedback
> > to the effect of "this is ready to progress") to the DART working group
> > mailing list.
> > >>
> > >> Thanks!
> > >>
> > >> Ben.
> > >>
> > >> Begin forwarded message:
> > >>
> > >>> Resent-From: draft-alias-bounces@tools.ietf.org
> > >>> From: Ben Campbell <ben@nostrum.com>
> > >>> Subject: WGLC: draft-ietf-dart-dscp-rtp-02
> > >>> Date: August 9, 2014 at 12:50:16 PM CDT
> > >>> Resent-To: ben@nostrum.com, david.black@emc.com, paulej@packetizer.=
com
> > >>> To: dart@ietf.org
> > >>> Cc: draft-ietf-dart-dscp-rtp.all@tools.ietf.org, mls.ietf@gmail.com=
,
> > Richard Barnes <rlb@ipv.sx>
> > >>>
> > >>> This is a DART Working Group Last Call of draft-ietf-dart-dscp-rtp-=
02.
> > It's available on the data tracker at the following URL:
> > >>>
> > >>> https://datatracker.ietf.org/doc/draft-ietf-dart-dscp-rtp/
> > >>>
> > >>> The WGLC will conclude on 22 August, 2014. Please send your comment=
s to
> > the authors and the DART mailing list. If you've reviewed it and think =
it's
> > good to go, please say so.
> > >>>
> > >>> Thanks!
> > >>>
> > >>> Ben.
> > >>>
> > >>
> > >> _______________________________________________
> > >> Audio/Video Transport Core Maintenance
> > >> avt@ietf.org
> > >> https://www.ietf.org/mailman/listinfo/avt
> > >
> > >
> > >
> > > --
> > > Colin Perkins
> > > http://csperkins.org/
> > >
> > >
> > >
> >
> > _______________________________________________
> > Dart mailing list
> > Dart@ietf.org
> > https://www.ietf.org/mailman/listinfo/dart


From nobody Sat Aug 23 09:09:14 2014
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From: "Black, David" <david.black@emc.com>
To: Ben Campbell <ben@nostrum.com>
Date: Fri, 22 Aug 2014 17:45:20 -0400
Thread-Topic: WGLC: draft-ietf-dart-dscp-rtp-02 - proposed text for section 2.1 mux items 3 and 4
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Cc: "draft-ietf-dart-dscp-rtp.all@tools.ietf.org" <draft-ietf-dart-dscp-rtp.all@tools.ietf.org>, "Black, David" <david.black@emc.com>, "dart@ietf.org" <dart@ietf.org>, Colin Perkins <csp@csperkins.org>, "avt@ietf.org WG" <avt@ietf.org>
Subject: Re: [AVTCORE] WGLC: draft-ietf-dart-dscp-rtp-02 - proposed text for section 2.1 mux items 3 and 4
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Ok, "signaling traffic for STUN [RFC5389] and TURN [RFC5766]"
	--> "STUN [RFC5389] and TURN [RFC5766] traffic"

It's even fewer words ;-) - consider that done.

Thanks,
--David


> -----Original Message-----
> From: Ben Campbell [mailto:ben@nostrum.com]
> Sent: Friday, August 22, 2014 5:24 PM
> To: Black, David
> Cc: Colin Perkins; draft-ietf-dart-dscp-rtp.all@tools.ietf.org; dart@ietf=
.org;
> avt@ietf.org WG
> Subject: Re: WGLC: draft-ietf-dart-dscp-rtp-02 - proposed text for sectio=
n 2.1
> mux items 3 and 4
>=20
> I agree except where noted inline:
>=20
> On Aug 22, 2014, at 3:27 PM, Black, David <david.black@emc.com> wrote:
>=20
> >
> > Proposed new text (merging item 4 into item 3, which aligns with the
> > structure of RFC 5764 and the rfc5764-mux-fixes draft):
> >
> >   3.  An RTP session could be multiplexed with a single SCTP over DTLS
> >       association plus signaling traffic for STUN [RFC5389] and TURN
> >       [RFC5766] into a single transport level flow as described in
> >       [RFC5764] and [I-D.petithuguenin-avtcore-rfc5764-mux-fixes].
> >       STUN and TURN are used for NAT traversal.
> >
>=20
> I agree with the direction here--but I'm not sure I would characterize ST=
UN
> and TURN as signaling traffic. And STUN is used for connectivity checks, =
as
> well as NAT traversal. OTOH, maybe it would be better to just for this se=
ction
> to just say that STUN and TURN can be multiplexed in the flow without the
> additional comments about them?
>=20
>=20
> > Thanks,
> > --David
> >
> >
> >> -----Original Message-----
> >> From: Black, David
> >> Sent: Thursday, August 21, 2014 9:06 PM
> >> To: Ben Campbell; Colin Perkins
> >> Cc: draft-ietf-dart-dscp-rtp.all@tools.ietf.org; dart@ietf.org;
> avt@ietf.org
> >> WG; Black, David
> >> Subject: RE: [Dart] [AVTCORE] WGLC: draft-ietf-dart-dscp-rtp-02
> >>
> >> Colin - many thanks for the review.  I'll pick up your proposed change=
s,
> >> perhaps
> >> with minor edits.
> >>
> >> Ben,
> >>
> >>>>>  3.  The RTP and RTCP traffic can be multiplexed with other protoco=
ls
> >>>>>      via UDP encapsulation over a common pair of UDP ports as descr=
ibed
> >>>>>      in [RFC5764] as updated by
> >>>>>      [I-D.petithuguenin-avtcore-rfc5764-mux-fixes]; and
> >>>
> >>> Can we clarify what "other protocols" are expected? Are we just talki=
ng
> >> about
> >>> WebRTC data-channels over SCTP/DTLS, or other things? If the latter, =
are
> we
> >>> talking things we expect to do today, or in immediate futures, or
> extensions
> >>> that might happen someday?
> >>
> >> WebRTC will only multiplex its data channels, and this numbered list i=
s
> about
> >> WebRTC, so this text should focus on those data channels - I'll revise=
 the
> >> draft
> >> accordingly.
> >>
> >> Other text elsewhere in the draft is broader (e.g., SCTP/UDP without D=
TLS,
> >> which
> >> is of interest, even though WebRTC always uses DTLS).
> >>
> >> Thanks,
> >> --David
> >>
> >>> -----Original Message-----
> >>> From: Dart [mailto:dart-bounces@ietf.org] On Behalf Of Ben Campbell
> >>> Sent: Monday, August 18, 2014 4:48 PM
> >>> To: Colin Perkins
> >>> Cc: draft-ietf-dart-dscp-rtp.all@tools.ietf.org; dart@ietf.org;
> avt@ietf.org
> >>> WG
> >>> Subject: Re: [Dart] [AVTCORE] WGLC: draft-ietf-dart-dscp-rtp-02
> >>>
> >>> Hi,
> >>>
> >>> I concur with all of Colin's comments, save one question below:
> >>>
> >>> On Aug 18, 2014, at 10:17 AM, Colin Perkins <csp@csperkins.org> wrote=
:
> >>>
> >>>> Hi,
> >>>>
> >>>> [I'm not on the DART list, so please cc me in replies]
> >>>>
> >>>> This draft looks good overall, but I did have some comments, primari=
ly
> >> about
> >>> the discussion of RTP multiplexing.
> >>>>
> >>>> The four bullet points in Section 2.1 that follow "The RTCWEB protoc=
ol
> >> suit
> >>> encompasses a number of forms of multiplexing" seem to confuse RTP
> sessions,
> >>>> RTCP, and transport-layer flows. I suggest they be changed to:
> >>>>
> >>>>>  The RTP streams that comprise a WebRTC session can be multiplexed
> >>>>>  together in a number of ways:
> >>>>>
> >>>>>  1.  Individual source streams are carried in one or more individua=
l
> >>>>>      RTP streams. These RTP streams can be multiplexed onto a singl=
e
> >>>>>      transport-layer flow or sent as separate transport-layer flows=
.
> >>>>>      This memo only considers the case where the RTP streams are to=
 be
> >>>>>      multiplexed onto a single transport-layer flow, forming a sing=
le
> >>>>>      RTP session;
> >>>>>
> >>>>>  2.  The RTP Control Protocol (RTCP) [RFC3550] may be multiplexed o=
nto
> >>>>>      the same transport-layer flow as the RTP stream with which it =
is
> >>>>>      associated, as described in [RFC5761], or it may be sent on a
> >>>>>      separate transport-layer flow;
> >>>>>
> >>>>>  3.  The RTP and RTCP traffic can be multiplexed with other protoco=
ls
> >>>>>      via UDP encapsulation over a common pair of UDP ports as descr=
ibed
> >>>>>      in [RFC5764] as updated by
> >>>>>      [I-D.petithuguenin-avtcore-rfc5764-mux-fixes]; and
> >>>
> >>> Can we clarify what "other protocols" are expected? Are we just talki=
ng
> >> about
> >>> WebRTC data-channels over SCTP/DTLS, or other things? If the latter, =
are
> we
> >>> talking things we expect to do today, or in immediate futures, or
> extensions
> >>> that might happen someday?
> >>>
> >>>>>
> >>>>>  4.  The data may be further encapsulated via STUN [RFC5389] and TU=
RN
> >>>>>      [RFC5766] for NAT (Network Address Translator) traversal.
> >>>>
> >>>> In the following paragraph (penultimate paragraph in Section 2.1), I
> >> suggest
> >>> changing "unidirectional UDP packet flow" to "transport-layer flow" s=
ince
> it
> >>> will not be unidirectional (since RTCP also flows in the reverse
> direction),
> >>> and might not use UDP.
> >>>>
> >>>> In the first paragraph of section 2.2, I suggest changing "multiplex=
ed
> >> over
> >>> RTP sessions" to "multiplexed in a single RTP session" and "multiplex=
ing
> of
> >>> source streams in RTP sessions" to "multiplexing of source streams in=
 a
> >> single
> >>> RTP session".
> >>>>
> >>>> In Section 6, the second bullet point refers to "an RTCP session". T=
here
> >> is
> >>> no such thing. Rather, within a single RTP session there are RTCP pac=
kets
> >> sent
> >>> that give information about the RTP streams that are being sent, and =
that
> >>> report on the reception quality of RTP streams being received. Using =
a
> >> single
> >>> PHB and DSCP for all RTCP packets within an RTP session might make se=
nse,
> >> but
> >>> it's important to note that one role of RTCP is to provide an estimat=
e of
> >> the
> >>> round-trip time seen by the media, so the PHB/DSCP will have to be ch=
osen
> >> with
> >>> care to avoid biasing that estimate too much.
> >>>>
> >>>> Editorial: the protocol is called WebRTC, but the working group is
> RTCWEB.
> >>> Accordingly, I think all instances of "RTCWEB" in this draft need to
> change
> >> to
> >>> "WebRTC".
> >>>>
> >>>> Cheers,
> >>>> Colin
> >>>>
> >>>>
> >>>>
> >>>>
> >>>>
> >>>> On 9 Aug 2014, at 19:48, Ben Campbell <ben@nostrum.com> wrote:
> >>>>> Hi,
> >>>>>
> >>>>> The DART working group has started a last call for draft-ietf-dart-=
dscp-
> >>> rtp-02, ending on August 22. This informational draft describes
> >> considerations
> >>> for the use of DiffServ code points in situations where one multiplex=
es
> >>> multiple RTP packet streams, and potentially other protocol streams, =
that
> >>> share the same 5-tuple.
> >>>>>
> >>>>> Since this draft is likely of interest to several working groups, I
> would
> >>> like to solicit additional reviews from participants of TSVWG, AVTCOR=
E,
> >>> MMUSIC, CLUE, RTCWEB, and RMCAT. Please send any feedback (including
> >> feedback
> >>> to the effect of "this is ready to progress") to the DART working gro=
up
> >>> mailing list.
> >>>>>
> >>>>> Thanks!
> >>>>>
> >>>>> Ben.
> >>>>>
> >>>>> Begin forwarded message:
> >>>>>
> >>>>>> Resent-From: draft-alias-bounces@tools.ietf.org
> >>>>>> From: Ben Campbell <ben@nostrum.com>
> >>>>>> Subject: WGLC: draft-ietf-dart-dscp-rtp-02
> >>>>>> Date: August 9, 2014 at 12:50:16 PM CDT
> >>>>>> Resent-To: ben@nostrum.com, david.black@emc.com, paulej@packetizer=
.com
> >>>>>> To: dart@ietf.org
> >>>>>> Cc: draft-ietf-dart-dscp-rtp.all@tools.ietf.org, mls.ietf@gmail.co=
m,
> >>> Richard Barnes <rlb@ipv.sx>
> >>>>>>
> >>>>>> This is a DART Working Group Last Call of draft-ietf-dart-dscp-rtp=
-02.
> >>> It's available on the data tracker at the following URL:
> >>>>>>
> >>>>>> https://datatracker.ietf.org/doc/draft-ietf-dart-dscp-rtp/
> >>>>>>
> >>>>>> The WGLC will conclude on 22 August, 2014. Please send your commen=
ts to
> >>> the authors and the DART mailing list. If you've reviewed it and thin=
k
> it's
> >>> good to go, please say so.
> >>>>>>
> >>>>>> Thanks!
> >>>>>>
> >>>>>> Ben.
> >>>>>>
> >>>>>
> >>>>> _______________________________________________
> >>>>> Audio/Video Transport Core Maintenance
> >>>>> avt@ietf.org
> >>>>> https://www.ietf.org/mailman/listinfo/avt
> >>>>
> >>>>
> >>>>
> >>>> --
> >>>> Colin Perkins
> >>>> http://csperkins.org/
> >>>>
> >>>>
> >>>>
> >>>
> >>> _______________________________________________
> >>> Dart mailing list
> >>> Dart@ietf.org
> >>> https://www.ietf.org/mailman/listinfo/dart


From nobody Mon Aug 25 04:20:04 2014
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Subject: Re: [AVTCORE] Colin Perkins comments - WGLC: draft-ietf-dart-dscp-rtp-02
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Hi,

On 22 Aug 2014, at 21:12, Black, David <david.black@emc.com> wrote:
> Hi Colin,
>=20
> I've picked up your changes (including RTCWEB -> WebRTC globally)
> in my working draft, with two exceptions:
>=20
> (1) Items 3 and 4 in the list of what can be multiplexed.
>=20
> I don't think you suggested any changes to these two items, but
> they need significant attention for other reasons:
>=20
> - Ben's comment that item 3 ought to focus on the WebRTC data channel
> - STUN doesn't encapsulate WebRTC traffic (TURN does).
>=20
> I plan to merge these items 3 and 4 as part of revising them, and
> intend to move the topic of TURN encapsulation out to be discussed
> elsewhere, as it's independent of multiplexing.  I'll propose
> specific text in a separate message.

That=92s fine.

> (2) PHBs and DSCPs for RTCP.
>=20
> I'm not sure what to say about this, as it looks like I need to set
> off a discussion (debate?) between you and my co-author, Paul Jones. =20=

>=20
> Colin (from WGLC comments):
>=20
>> Rather, within a single RTP session there are RTCP packets sent
>> that give information about the RTP streams that are being sent, and =
that
>> report on the reception quality of RTP streams being received. Using =
a single
>> PHB and DSCP for all RTCP packets within an RTP session might make =
sense, but
>> it's important to note that one role of RTCP is to provide an =
estimate of the
>> round-trip time seen by the media, so the PHB/DSCP will have to be =
chosen with
>> care to avoid biasing that estimate too much.
>=20
> Paul (from shortly after the Toronto meeting:
>=20
>> During the meeting, there was discussion of marking RTCP packets.  =
Some
>> notes I received on this topic suggested that it was proposed that =
RTCP
>> should be marked the same as for RTP.  The argument was that this is =
used
>> for RTT calculations.  If that is what was said, I'd like to state my
>> disagreement. :-)
>>=20
>> The forward and reverse paths are not necessarily the same and there =
is
>> nothing one should assume about the reverse path to provide guidance =
about
>> the forward path (or vice versa).  As perhaps a gross example, I have =
the
>> ability to download far faster on my home Internet connection than I =
can
>> upload.  Other important traffic characteristics differ in each =
direction.
>>=20
>> Further, an RTCP packet might provide information related to several =
different
>> RTP packets.  I certainly would not want to see one RTCP packet per =
RTP packet.
>=20
> I'll watch where that discussion goes before proposing text.

One of the uses of RTCP is certainly to estimate the round-trip time, as =
described in Section 6.4.1 (page 41) of RFC 3550. The DSCP marking used =
for RTCP does need to take this use into account. Ideally, this would =
mean RTCP packets were marked the same as the corresponding RTP packets, =
as I suggested in the meeting. However, Paul is correct that this only =
works when all RTP packets have the same marking.

Accordingly, I=92m not sure what the right marking is for RTCP. One =
approach might be to say that an SSRC that sends RTP packets with =
different marking should use marking with the highest(lowest?) =
precedence of those is uses for RTCP, so the RTT estimate calculated =
from RTCP is representative of some RTP packets. I=92m not sure that =
using a single DSCP for all RTCP packets would achieve that.

Colin




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




From nobody Mon Aug 25 08:39:33 2014
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Subject: [AVTCORE] I-D Action: draft-ietf-avtcore-rtp-multi-stream-optimisation-04.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 Audio/Video Transport Core Maintenance Working Group of the IETF.

        Title           : Sending Multiple Media Streams in a Single RTP Session: Grouping RTCP Reception Statistics and Other Feedback
        Authors         : Jonathan Lennox
                          Magnus Westerlund
                          Qin Wu
                          Colin Perkins
	Filename        : draft-ietf-avtcore-rtp-multi-stream-optimisation-04.txt
	Pages           : 17
	Date            : 2014-08-25

Abstract:
   RTP allows multiple media streams to be sent in a single session, but
   requires each Synchronisation Source (SSRC) to send RTCP reception
   quality reports for every other SSRC visible in the session.  This
   causes the number of RTCP reception reports to grow with the number
   of SSRCs, rather than the number of endpoints.  In many cases most of
   these RTCP reception reports are unnecessary, since all SSRCs of an
   endpoint are co-located and see the same reception quality.  This
   memo defines a Reporting Group extension to RTCP to reduce the
   reporting overhead in such scenarios.


The IETF datatracker status page for this draft is:
https://datatracker.ietf.org/doc/draft-ietf-avtcore-rtp-multi-stream-optimisation/

There's also a htmlized version available at:
http://tools.ietf.org/html/draft-ietf-avtcore-rtp-multi-stream-optimisation-04

A diff from the previous version is available at:
http://www.ietf.org/rfcdiff?url2=draft-ietf-avtcore-rtp-multi-stream-optimisation-04


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

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


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This is a minor update, to address Paul Kyzivat=92s SDP directorate =
review comments by clarifying that the =93a=3Drtcp-rgrp:=94 attribute =
takes no value.

Colin



On 25 Aug 2014, at 16:39, Internet-Drafts@ietf.org wrote:
> A New Internet-Draft is available from the on-line Internet-Drafts =
directories.
> This draft is a work item of the Audio/Video Transport Core =
Maintenance Working Group of the IETF.
>=20
>        Title           : Sending Multiple Media Streams in a Single =
RTP Session: Grouping RTCP Reception Statistics and Other Feedback
>        Authors         : Jonathan Lennox
>                          Magnus Westerlund
>                          Qin Wu
>                          Colin Perkins
> 	Filename        : =
draft-ietf-avtcore-rtp-multi-stream-optimisation-04.txt
> 	Pages           : 17
> 	Date            : 2014-08-25
>=20
> Abstract:
>   RTP allows multiple media streams to be sent in a single session, =
but
>   requires each Synchronisation Source (SSRC) to send RTCP reception
>   quality reports for every other SSRC visible in the session.  This
>   causes the number of RTCP reception reports to grow with the number
>   of SSRCs, rather than the number of endpoints.  In many cases most =
of
>   these RTCP reception reports are unnecessary, since all SSRCs of an
>   endpoint are co-located and see the same reception quality.  This
>   memo defines a Reporting Group extension to RTCP to reduce the
>   reporting overhead in such scenarios.
>=20
>=20
> The IETF datatracker status page for this draft is:
> =
https://datatracker.ietf.org/doc/draft-ietf-avtcore-rtp-multi-stream-optim=
isation/
>=20
> There's also a htmlized version available at:
> =
http://tools.ietf.org/html/draft-ietf-avtcore-rtp-multi-stream-optimisatio=
n-04
>=20
> A diff from the previous version is available at:
> =
http://www.ietf.org/rfcdiff?url2=3Ddraft-ietf-avtcore-rtp-multi-stream-opt=
imisation-04
>=20
>=20
> Please note that it may take a couple of minutes from the time of =
submission
> until the htmlized version and diff are available at tools.ietf.org.
>=20
> Internet-Drafts are also available by anonymous FTP at:
> ftp://ftp.ietf.org/internet-drafts/
>=20
> _______________________________________________
> Audio/Video Transport Core Maintenance
> avt@ietf.org
> https://www.ietf.org/mailman/listinfo/avt



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




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Subject: Re: [AVTCORE] [Dart] Colin Perkins comments - WGLC: draft-ietf-dart-dscp-rtp-02
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On Aug 25, 2014, at 6:19 AM, Colin Perkins <csp@csperkins.org> wrote:

>> (2) PHBs and DSCPs for RTCP.
>>=20
>> I'm not sure what to say about this, as it looks like I need to set
>> off a discussion (debate?) between you and my co-author, Paul Jones. =20=

>>=20
>> Colin (from WGLC comments):
>>=20
>>> Rather, within a single RTP session there are RTCP packets sent
>>> that give information about the RTP streams that are being sent, and =
that
>>> report on the reception quality of RTP streams being received. Using =
a single
>>> PHB and DSCP for all RTCP packets within an RTP session might make =
sense, but
>>> it's important to note that one role of RTCP is to provide an =
estimate of the
>>> round-trip time seen by the media, so the PHB/DSCP will have to be =
chosen with
>>> care to avoid biasing that estimate too much.
>>=20
>> Paul (from shortly after the Toronto meeting:
>>=20
>>> During the meeting, there was discussion of marking RTCP packets.  =
Some
>>> notes I received on this topic suggested that it was proposed that =
RTCP
>>> should be marked the same as for RTP.  The argument was that this is =
used
>>> for RTT calculations.  If that is what was said, I'd like to state =
my
>>> disagreement. :-)
>>>=20
>>> The forward and reverse paths are not necessarily the same and there =
is
>>> nothing one should assume about the reverse path to provide guidance =
about
>>> the forward path (or vice versa).  As perhaps a gross example, I =
have the
>>> ability to download far faster on my home Internet connection than I =
can
>>> upload.  Other important traffic characteristics differ in each =
direction.
>>>=20
>>> Further, an RTCP packet might provide information related to several =
different
>>> RTP packets.  I certainly would not want to see one RTCP packet per =
RTP packet.
>>=20
>> I'll watch where that discussion goes before proposing text.
>=20
> One of the uses of RTCP is certainly to estimate the round-trip time, =
as described in Section 6.4.1 (page 41) of RFC 3550. The DSCP marking =
used for RTCP does need to take this use into account. Ideally, this =
would mean RTCP packets were marked the same as the corresponding RTP =
packets, as I suggested in the meeting. However, Paul is correct that =
this only works when all RTP packets have the same marking.
>=20
> Accordingly, I=92m not sure what the right marking is for RTCP. One =
approach might be to say that an SSRC that sends RTP packets with =
different marking should use marking with the highest(lowest?) =
precedence of those is uses for RTCP, so the RTT estimate calculated =
from RTCP is representative of some RTP packets. I=92m not sure that =
using a single DSCP for all RTCP packets would achieve that.

Can this working group reasonably answer this question? If not, we could =
list it as something implementors need to think about, and note that we =
do not have generic guidance.


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From: "Paul E. Jones" <paulej@packetizer.com>
To: "Ben Campbell" <ben@nostrum.com>, "Colin Perkins" <csp@csperkins.org>
Date: Mon, 25 Aug 2014 19:52:08 +0000
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Subject: Re: [AVTCORE] [Dart] Colin Perkins comments - WGLC: draft-ietf-dart-dscp-rtp-02
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Ben, Colin,

On (2) below, it might be a challenge to get agreement quickly, but it=20
might be nice if we could.  As this work relates to RMCAT, determining=20
one-way delay or changes thereto will likely prove important.  In=20
general, I would expect perceived quality to be important and conveying=20
quality metrics and/or providing feedback to the sender so that=20
perceived quality improves (in whatever form) will be important.  I'm=20
suspicious of use of any RTT calculation for any mechanism that might be=
=20
used to improve QoE.

Getting feedback to the sender is important, though.  In an ideal world,=
=20
I would argue that RTCP packets should be marked with whatever DSCP=20
value will deliver RTCP packets in the most expedient way.  Since we=20
don't have an ideal world, I don't know which DSCP value that would be.

There are several options we could consider:
* Mark all RTCP packets as CS0
* Mark RTCP packets using a locally-defined "highest priority" marking
* Mark RTCP packets the same as media flow for the first (active) SDP=20
line (thus, if audio is the first m=3D line in SDP and that flow is=20
active, all RTCP packets will be marked the same as audio)
* Specify a value for all RTCP packets irrespective of any value used in=
=20
RTP flows

I think the right folks on the thread and on the list are present to=20
help make that choice.  We just need to decide if that is something we=20
want to specify.

Paul

------ Original Message ------
From: "Ben Campbell" <ben@nostrum.com>
To: "Colin Perkins" <csp@csperkins.org>
Cc: "Black, David" <david.black@emc.com>; "dart@ietf.org"=20
<dart@ietf.org>; "avt@ietf.org WG" <avt@ietf.org>; "Paul E. Jones=20
(paulej@packetizer.com)" <paulej@packetizer.com>;=20
"draft-ietf-dart-dscp-rtp.all@tools.ietf.org"=20
<draft-ietf-dart-dscp-rtp.all@tools.ietf.org>
Sent: 8/25/2014 2:12:01 PM
Subject: Re: [Dart] Colin Perkins comments - WGLC:=20
draft-ietf-dart-dscp-rtp-02

>
>On Aug 25, 2014, at 6:19 AM, Colin Perkins <csp@csperkins.org> wrote:
>
>>>  (2) PHBs and DSCPs for RTCP.
>>>
>>>  I'm not sure what to say about this, as it looks like I need to set
>>>  off a discussion (debate?) between you and my co-author, Paul Jones.
>>>
>>>  Colin (from WGLC comments):
>>>
>>>>  Rather, within a single RTP session there are RTCP packets sent
>>>>  that give information about the RTP streams that are being sent,=20
>>>>and that
>>>>  report on the reception quality of RTP streams being received.=20
>>>>Using a single
>>>>  PHB and DSCP for all RTCP packets within an RTP session might make=
=20
>>>>sense, but
>>>>  it's important to note that one role of RTCP is to provide an=20
>>>>estimate of the
>>>>  round-trip time seen by the media, so the PHB/DSCP will have to be=
=20
>>>>chosen with
>>>>  care to avoid biasing that estimate too much.
>>>
>>>  Paul (from shortly after the Toronto meeting:
>>>
>>>>  During the meeting, there was discussion of marking RTCP packets.=20
>>>>Some
>>>>  notes I received on this topic suggested that it was proposed that=
=20
>>>>RTCP
>>>>  should be marked the same as for RTP. The argument was that this is=
=20
>>>>used
>>>>  for RTT calculations. If that is what was said, I'd like to state=20
>>>>my
>>>>  disagreement. :-)
>>>>
>>>>  The forward and reverse paths are not necessarily the same and=20
>>>>there is
>>>>  nothing one should assume about the reverse path to provide=20
>>>>guidance about
>>>>  the forward path (or vice versa). As perhaps a gross example, I=20
>>>>have the
>>>>  ability to download far faster on my home Internet connection than=
=20
>>>>I can
>>>>  upload. Other important traffic characteristics differ in each=20
>>>>direction.
>>>>
>>>>  Further, an RTCP packet might provide information related to=20
>>>>several different
>>>>  RTP packets. I certainly would not want to see one RTCP packet per=
=20
>>>>RTP packet.
>>>
>>>  I'll watch where that discussion goes before proposing text.
>>
>>  One of the uses of RTCP is certainly to estimate the round-trip time,=
=20
>>as described in Section 6.4.1 (page 41) of RFC 3550. The DSCP marking=20
>>used for RTCP does need to take this use into account. Ideally, this=20
>>would mean RTCP packets were marked the same as the corresponding RTP=20
>>packets, as I suggested in the meeting. However, Paul is correct that=20
>>this only works when all RTP packets have the same marking.
>>
>>  Accordingly, I=E2=80=99m not sure what the right marking is for RTCP.=
 One=20
>>approach might be to say that an SSRC that sends RTP packets with=20
>>different marking should use marking with the highest(lowest?)=20
>>precedence of those is uses for RTCP, so the RTT estimate calculated=20
>>from RTCP is representative of some RTP packets. I=E2=80=99m not sure =
that=20
>>using a single DSCP for all RTCP packets would achieve that.
>
>Can this working group reasonably answer this question? If not, we=20
>could list it as something implementors need to think about, and note=20
>that we do not have generic guidance.
>


From nobody Mon Aug 25 13:12:23 2014
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From: Ben Campbell <ben@nostrum.com>
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Subject: Re: [AVTCORE] [Dart] Colin Perkins comments - WGLC: draft-ietf-dart-dscp-rtp-02
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Our charter is not to improve QoS or solve congestion issues. It's just =
to give guidance on DSCP choices for multiplexed media. So I think the =
question becomes, has RMCAT made decisions (or looks very likely to make =
decisions) to use RTCP in a way that would be impacted by our DSCP =
recommendations?


On Aug 25, 2014, at 2:52 PM, Paul E. Jones <paulej@packetizer.com> =
wrote:

> Ben, Colin,
>=20
> On (2) below, it might be a challenge to get agreement quickly, but it =
might be nice if we could.  As this work relates to RMCAT, determining =
one-way delay or changes thereto will likely prove important.  In =
general, I would expect perceived quality to be important and conveying =
quality metrics and/or providing feedback to the sender so that =
perceived quality improves (in whatever form) will be important.  I'm =
suspicious of use of any RTT calculation for any mechanism that might be =
used to improve QoE.
>=20

If so, then we should document the potential impact. If _not_, then it =
seems to me the best we could do is mention that RTCP treatments may or =
may not have an impact on RTT estimates, and that the implementer should =
think about it.

> Getting feedback to the sender is important, though.  In an ideal =
world, I would argue that RTCP packets should be marked with whatever =
DSCP value will deliver RTCP packets in the most expedient way.  Since =
we don't have an ideal world, I don't know which DSCP value that would =
be.
>=20

Would the starting positions of "getting feedback is important, even of =
not for RTT estimates" and "we need RTCP for RTT estimates" likely land =
on the same guidance for DSCP values?

The argument to send RTCP packets in the most expedient way sounds =
reasonable. I don't know if we need to recommend a particular DSCP, =
since we already have quite a bit of text on how DSCPs might (or might =
not) map into some predictable PHB treatment.

> There are several options we could consider:
> * Mark all RTCP packets as CS0
> * Mark RTCP packets using a locally-defined "highest priority" marking
> * Mark RTCP packets the same as media flow for the first (active) SDP =
line (thus, if audio is the first m=3D line in SDP and that flow is =
active, all RTCP packets will be marked the same as audio)
> * Specify a value for all RTCP packets irrespective of any value used =
in RTP flows
>=20
> I think the right folks on the thread and on the list are present to =
help make that choice.  We just need to decide if that is something we =
want to specify.
>=20
> Paul
>=20
> ------ Original Message ------
> From: "Ben Campbell" <ben@nostrum.com>
> To: "Colin Perkins" <csp@csperkins.org>
> Cc: "Black, David" <david.black@emc.com>; "dart@ietf.org" =
<dart@ietf.org>; "avt@ietf.org WG" <avt@ietf.org>; "Paul E. Jones =
(paulej@packetizer.com)" <paulej@packetizer.com>; =
"draft-ietf-dart-dscp-rtp.all@tools.ietf.org" =
<draft-ietf-dart-dscp-rtp.all@tools.ietf.org>
> Sent: 8/25/2014 2:12:01 PM
> Subject: Re: [Dart] Colin Perkins comments - WGLC: =
draft-ietf-dart-dscp-rtp-02
>=20
>>=20
>> On Aug 25, 2014, at 6:19 AM, Colin Perkins <csp@csperkins.org> wrote:
>>=20
>>>> (2) PHBs and DSCPs for RTCP.
>>>>=20
>>>> I'm not sure what to say about this, as it looks like I need to set
>>>> off a discussion (debate?) between you and my co-author, Paul =
Jones.
>>>>=20
>>>> Colin (from WGLC comments):
>>>>=20
>>>>> Rather, within a single RTP session there are RTCP packets sent
>>>>> that give information about the RTP streams that are being sent, =
and that
>>>>> report on the reception quality of RTP streams being received. =
Using a single
>>>>> PHB and DSCP for all RTCP packets within an RTP session might make =
sense, but
>>>>> it's important to note that one role of RTCP is to provide an =
estimate of the
>>>>> round-trip time seen by the media, so the PHB/DSCP will have to be =
chosen with
>>>>> care to avoid biasing that estimate too much.
>>>>=20
>>>> Paul (from shortly after the Toronto meeting:
>>>>=20
>>>>> During the meeting, there was discussion of marking RTCP packets. =
Some
>>>>> notes I received on this topic suggested that it was proposed that =
RTCP
>>>>> should be marked the same as for RTP. The argument was that this =
is used
>>>>> for RTT calculations. If that is what was said, I'd like to state =
my
>>>>> disagreement. :-)
>>>>>=20
>>>>> The forward and reverse paths are not necessarily the same and =
there is
>>>>> nothing one should assume about the reverse path to provide =
guidance about
>>>>> the forward path (or vice versa). As perhaps a gross example, I =
have the
>>>>> ability to download far faster on my home Internet connection than =
I can
>>>>> upload. Other important traffic characteristics differ in each =
direction.
>>>>>=20
>>>>> Further, an RTCP packet might provide information related to =
several different
>>>>> RTP packets. I certainly would not want to see one RTCP packet per =
RTP packet.
>>>>=20
>>>> I'll watch where that discussion goes before proposing text.
>>>=20
>>> One of the uses of RTCP is certainly to estimate the round-trip =
time, as described in Section 6.4.1 (page 41) of RFC 3550. The DSCP =
marking used for RTCP does need to take this use into account. Ideally, =
this would mean RTCP packets were marked the same as the corresponding =
RTP packets, as I suggested in the meeting. However, Paul is correct =
that this only works when all RTP packets have the same marking.
>>>=20
>>> Accordingly, I=92m not sure what the right marking is for RTCP. One =
approach might be to say that an SSRC that sends RTP packets with =
different marking should use marking with the highest(lowest?) =
precedence of those is uses for RTCP, so the RTT estimate calculated =
from RTCP is representative of some RTP packets. I=92m not sure that =
using a single DSCP for all RTCP packets would achieve that.
>>=20
>> Can this working group reasonably answer this question? If not, we =
could list it as something implementors need to think about, and note =
that we do not have generic guidance.
>>=20
>=20


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From: "Paul E. Jones" <paulej@packetizer.com>
To: "Ben Campbell" <ben@nostrum.com>
Date: Mon, 25 Aug 2014 21:59:22 +0000
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Subject: Re: [AVTCORE] [Dart] Colin Perkins comments - WGLC: draft-ietf-dart-dscp-rtp-02
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Ben,

>Our charter is not to improve QoS or solve congestion issues. It's just=
=20
>to give guidance on DSCP choices for multiplexed media. So I think the=20
>question becomes, has RMCAT made decisions (or looks very likely to=20
>make decisions) to use RTCP in a way that would be impacted by our DSCP=
=20
>recommendations?

I'm not aware of whether such a decision has been made.


>>  Ben, Colin,
>>
>>  On (2) below, it might be a challenge to get agreement quickly, but=20
>>it might be nice if we could. As this work relates to RMCAT,=20
>>determining one-way delay or changes thereto will likely prove=20
>>important. In general, I would expect perceived quality to be=20
>>important and conveying quality metrics and/or providing feedback to=20
>>the sender so that perceived quality improves (in whatever form) will=20
>>be important. I'm suspicious of use of any RTT calculation for any=20
>>mechanism that might be used to improve QoE.
>>
>
>If so, then we should document the potential impact. If _not_, then it=20
>seems to me the best we could do is mention that RTCP treatments may or=
=20
>may not have an impact on RTT estimates, and that the implementer=20
>should think about it.

If accurate RTT measurements could be made, they could be useful.  The=20
problem is that in some environments, we know that RTTs will not be=20
accurate.  Sometimes they are, of course.

We could have some text as you suggest, but I would personally prefer we=
=20
not encourage use of RTT values for any particular purpose. :-)


>>  Getting feedback to the sender is important, though. In an ideal=20
>>world, I would argue that RTCP packets should be marked with whatever=20
>>DSCP value will deliver RTCP packets in the most expedient way. Since=20
>>we don't have an ideal world, I don't know which DSCP value that would=
=20
>>be.
>>
>
>Would the starting positions of "getting feedback is important, even of=
=20
>not for RTT estimates" and "we need RTCP for RTT estimates" likely land=
=20
>on the same guidance for DSCP values?
>
>The argument to send RTCP packets in the most expedient way sounds=20
>reasonable. I don't know if we need to recommend a particular DSCP,=20
>since we already have quite a bit of text on how DSCPs might (or might=20
>not) map into some predictable PHB treatment.

Good question and valid point.  Nowhere in the document do we recommend=20
the use of a particular DSCP value for any particular thing, and we=20
should not recommend a particular value for RTCP in this document.  I'm=20
just not sure what statements should be made.

I suspect we can all agree that RTCP information is important.  It's=20
just the DSCP-related guidance that goes with that that is challenging.

David? :-)

Paul


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From: Colin Perkins <csp@csperkins.org>
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Subject: Re: [AVTCORE] [Dart] Colin Perkins comments - WGLC: draft-ietf-dart-dscp-rtp-02
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On 25 Aug 2014, at 22:59, Paul E. Jones <paulej@packetizer.com> wrote:
> Ben,
>=20
>> Our charter is not to improve QoS or solve congestion issues. It's =
just to give guidance on DSCP choices for multiplexed media. So I think =
the question becomes, has RMCAT made decisions (or looks very likely to =
make decisions) to use RTCP in a way that would be impacted by our DSCP =
recommendations?
>=20
> I=92m not aware of whether such a decision has been made.

I don=92t believe RMCAT has considered this issue.

>>> Ben, Colin,
>>>=20
>>> On (2) below, it might be a challenge to get agreement quickly, but =
it might be nice if we could. As this work relates to RMCAT, determining =
one-way delay or changes thereto will likely prove important. In =
general, I would expect perceived quality to be important and conveying =
quality metrics and/or providing feedback to the sender so that =
perceived quality improves (in whatever form) will be important. I'm =
suspicious of use of any RTT calculation for any mechanism that might be =
used to improve QoE.
>>>=20
>>=20
>> If so, then we should document the potential impact. If _not_, then =
it seems to me the best we could do is mention that RTCP treatments may =
or may not have an impact on RTT estimates, and that the implementer =
should think about it.
>=20
> If accurate RTT measurements could be made, they could be useful.  The =
problem is that in some environments, we know that RTTs will not be =
accurate.  Sometimes they are, of course.
>=20
> We could have some text as you suggest, but I would personally prefer =
we not encourage use of RTT values for any particular purpose. :-)
>=20
>=20
>>> Getting feedback to the sender is important, though. In an ideal =
world, I would argue that RTCP packets should be marked with whatever =
DSCP value will deliver RTCP packets in the most expedient way. Since we =
don't have an ideal world, I don't know which DSCP value that would be.
>>=20
>> Would the starting positions of "getting feedback is important, even =
of not for RTT estimates" and "we need RTCP for RTT estimates" likely =
land on the same guidance for DSCP values?
>>=20
>> The argument to send RTCP packets in the most expedient way sounds =
reasonable. I don't know if we need to recommend a particular DSCP, =
since we already have quite a bit of text on how DSCPs might (or might =
not) map into some predictable PHB treatment.
>=20
> Good question and valid point.  Nowhere in the document do we =
recommend the use of a particular DSCP value for any particular thing, =
and we should not recommend a particular value for RTCP in this =
document.  I'm just not sure what statements should be made.
>=20
> I suspect we can all agree that RTCP information is important.  It's =
just the DSCP-related guidance that goes with that that is challenging.

I don=92t agree that RTCP information should be sent as higher priority =
than the media. Ideally, it should be sent with the same priority as the =
media, so it can be used to sample the RTT. This RTT sample is =
independent of RMCAT. It=92s in base RTP specification, and so is =
something we need to support to the extent possible.=20

Since not all the media sent by a single SSRC has the same marking, my =
suggestion would be that each SSRC mark the RTCP packets it sends with =
one of the same code points as it uses to mark the media. Since RTCP is =
somewhat important, it would make sense for each SSRC to mark the RTCP =
packets it sends using the highest priority code point it uses to mark =
the RTP media packets it sends.

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




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From: "Ali C. Begen (abegen)" <abegen@cisco.com>
To: Colin Perkins <csp@csperkins.org>
Thread-Topic: [AVTCORE] [Dart] Colin Perkins comments - WGLC: draft-ietf-dart-dscp-rtp-02
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Subject: Re: [AVTCORE] [Dart] Colin Perkins comments - WGLC: draft-ietf-dart-dscp-rtp-02
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On Aug 26, 2014, at 1:17 PM, Colin Perkins <csp@csperkins.org> wrote:

>> I suspect we can all agree that RTCP information is important.  It's jus=
t the DSCP-related guidance that goes with that that is challenging.
>=20
> I don=92t agree that RTCP information should be sent as higher priority t=
han the media. Ideally, it should be sent with the same priority as the med=
ia, so it can be used to sample the RTT. This RTT sample is independent of =
RMCAT.=20

Agreed. RTCP should not have any higher priority than RTP media. First, you=
 don't want RTCP traffic to cause damage to RTP traffic. Second, not being =
in the same class, RTCP reported data wont reflect the reality for RTP traf=
fic. If you want protection for RTCP packets, use redundancy or other techn=
iques within the bounds allowed by 3550. =


From nobody Tue Aug 26 07:16:34 2014
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Subject: [AVTCORE] Treatment of RTCP (was Re: [Dart] Colin Perkins comments - WGLC: draft-ietf-dart-dscp-rtp-02)
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On Aug 26, 2014, at 5:17 AM, Colin Perkins <csp@csperkins.org> wrote:

>>>>=20
>>>> Getting feedback to the sender is important, though. In an ideal =
world, I would argue that RTCP packets should be marked with whatever =
DSCP value will deliver RTCP packets in the most expedient way. Since we =
don't have an ideal world, I don't know which DSCP value that would be.
>>>=20
>>> Would the starting positions of "getting feedback is important, even =
of not for RTT estimates" and "we need RTCP for RTT estimates" likely =
land on the same guidance for DSCP values?
>>>=20
>>> The argument to send RTCP packets in the most expedient way sounds =
reasonable. I don't know if we need to recommend a particular DSCP, =
since we already have quite a bit of text on how DSCPs might (or might =
not) map into some predictable PHB treatment.
>>=20
>> Good question and valid point.  Nowhere in the document do we =
recommend the use of a particular DSCP value for any particular thing, =
and we should not recommend a particular value for RTCP in this =
document.  I'm just not sure what statements should be made.
>>=20
>> I suspect we can all agree that RTCP information is important.  It's =
just the DSCP-related guidance that goes with that that is challenging.
>=20
> I don=92t agree that RTCP information should be sent as higher =
priority than the media. Ideally, it should be sent with the same =
priority as the media, so it can be used to sample the RTT. This RTT =
sample is independent of RMCAT. It=92s in base RTP specification, and so =
is something we need to support to the extent possible.=20
>=20
> Since not all the media sent by a single SSRC has the same marking, my =
suggestion would be that each SSRC mark the RTCP packets it sends with =
one of the same code points as it uses to mark the media. Since RTCP is =
somewhat important, it would make sense for each SSRC to mark the RTCP =
packets it sends using the highest priority code point it uses to mark =
the RTP media packets it sends.

That makes sense to me. Paul, and others, do you agree with that last =
paragraph?

/Ben



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From: "Paul E. Jones" <paulej@packetizer.com>
To: "Ben Campbell" <ben@nostrum.com>, "Colin Perkins" <csp@csperkins.org>
Date: Tue, 26 Aug 2014 15:52:57 +0000
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Subject: Re: [AVTCORE] Treatment of RTCP (was Re: [Dart] Colin Perkins comments - WGLC: draft-ietf-dart-dscp-rtp-02)
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  Ben,

>>>>>  Getting feedback to the sender is important, though. In an ideal=20
>>>>>world, I would argue that RTCP packets should be marked with=20
>>>>>whatever DSCP value will deliver RTCP packets in the most expedient=
=20
>>>>>way. Since we don't have an ideal world, I don't know which DSCP=20
>>>>>value that would be.
>>>>
>>>>  Would the starting positions of "getting feedback is important,=20
>>>>even of not for RTT estimates" and "we need RTCP for RTT estimates"=20
>>>>likely land on the same guidance for DSCP values?
>>>>
>>>>  The argument to send RTCP packets in the most expedient way sounds=
=20
>>>>reasonable. I don't know if we need to recommend a particular DSCP,=20
>>>>since we already have quite a bit of text on how DSCPs might (or=20
>>>>might not) map into some predictable PHB treatment.
>>>
>>>  Good question and valid point. Nowhere in the document do we=20
>>>recommend the use of a particular DSCP value for any particular=20
>>>thing, and we should not recommend a particular value for RTCP in=20
>>>this document. I'm just not sure what statements should be made.
>>>
>>>  I suspect we can all agree that RTCP information is important. It's=
=20
>>>just the DSCP-related guidance that goes with that that is=20
>>>challenging.
>>
>>  I don=E2=80=99t agree that RTCP information should be sent as higher=
 priority=20
>>than the media. Ideally, it should be sent with the same priority as=20
>>the media, so it can be used to sample the RTT. This RTT sample is=20
>>independent of RMCAT. It=E2=80=99s in base RTP specification, and so is=
=20
>>something we need to support to the extent possible.
>>
>>  Since not all the media sent by a single SSRC has the same marking,=20
>>my suggestion would be that each SSRC mark the RTCP packets it sends=20
>>with one of the same code points as it uses to mark the media. Since=20
>>RTCP is somewhat important, it would make sense for each SSRC to mark=20
>>the RTCP packets it sends using the highest priority code point it=20
>>uses to mark the RTP media packets it sends.
>
>That makes sense to me. Paul, and others, do you agree with that last=20
>paragraph?

I agree that the value should be one of the values used for media, but=20
the challenge is specifying which DSCP value to use.  If audio is sent=20
using EF, video as AF41, app sharing video as AF31, etc., which of these=
=20
does the endpoint choose to use for RTCP?

Since DSCP values cannot be assumed to any particular PHB behavior in=20
the network, that makes answering that question challenging.

My personal preference in a voice/video conference would be to use the=20
same marking as is used for those flows -- and I would use the same=20
value for both voice and video.  However, if the app sharing video is=20
considered more important by some applications, perhaps that's the value=
=20
that should be used in those applications.

I can accept the suggestion, leaving the specific selection as an=20
implementation matter.  I do wish there was something more concrete so=20
that two devices would always follow the same approach to selecting a=20
value, but I doubt we can get agreement very quickly.

Paul


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Subject: Re: [AVTCORE] Treatment of RTCP (was Re: [Dart] Colin Perkins comments - WGLC: draft-ietf-dart-dscp-rtp-02)
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On 26 Aug 2014, at 16:39, Black, David <david.black@emc.com> wrote:

>>> Since not all the media sent by a single SSRC has the same marking, =
my
>> suggestion would be that each SSRC mark the RTCP packets it sends =
with one of
>> the same code points as it uses to mark the media. Since RTCP is =
somewhat
>> important, it would make sense for each SSRC to mark the RTCP packets =
it sends
>> using the highest priority code point it uses to mark the RTP media =
packets it
>> sends.
>>=20
>> That makes sense to me. Paul, and others, do you agree with that last
>> paragraph?
>=20
> <editor hat off>
>=20
> Sure, when there's a notion of priority or importance.  There won't =
always
> be one (e.g., is CS2 higher priority than AF2x?  That depends ...).  =
OTOH,
> we don't need to say much here, e.g., the class selector codepoints =
(CSx)
> are ordered, and adding a pointer to draft-ietf-tsvwg-rtcweb-qos will =
be
> helpful.
>=20
> </editor hat off>
>=20
> Turning to the other RTCP issue, [F] on multi-stream optimization, I =
wonder
> whether we inadvertently framed that issue backwards.  Given the =
above, a
> single RTCP report on multiple RTP streams that use rather different =
DSCPs
> may not result in representative RTT values for all of the streams, =
because
> the report has to be sent with one DSCP.  If that outcome is a =
problem,
> one should send separate RTCP reports (duh!).=20

Each reporting SSRC sends a separate RTCP report block for each SSRC it =
receives, so this is not a problem. The question is what marking an SSRC =
uses for the RTCP packets it sends, given that it uses several different =
markings for the RTP media packets it sends.=20

> That statement seems valuable to make and can be made with a citation =
of
> RFC 3530, as opposed to the multi-stream optimization draft.
>=20
> Harald and Colin - what do you think?

The discussion of the multi-stream-optimisation draft is a separate =
issue, and is correct as it is.

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




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Subject: Re: [AVTCORE] Treatment of RTCP (was Re: [Dart] Colin Perkins comments - WGLC: draft-ietf-dart-dscp-rtp-02)
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On 26 Aug 2014, at 16:52, Paul E. Jones <paulej@packetizer.com> wrote:
> Ben,
>>>>>> Getting feedback to the sender is important, though. In an ideal =
world, I would argue that RTCP packets should be marked with whatever =
DSCP value will deliver RTCP packets in the most expedient way. Since we =
don't have an ideal world, I don't know which DSCP value that would be.
>>>>>=20
>>>>> Would the starting positions of "getting feedback is important, =
even of not for RTT estimates" and "we need RTCP for RTT estimates" =
likely land on the same guidance for DSCP values?
>>>>>=20
>>>>> The argument to send RTCP packets in the most expedient way sounds =
reasonable. I don't know if we need to recommend a particular DSCP, =
since we already have quite a bit of text on how DSCPs might (or might =
not) map into some predictable PHB treatment.
>>>>=20
>>>> Good question and valid point. Nowhere in the document do we =
recommend the use of a particular DSCP value for any particular thing, =
and we should not recommend a particular value for RTCP in this =
document. I'm just not sure what statements should be made.
>>>>=20
>>>> I suspect we can all agree that RTCP information is important. It's =
just the DSCP-related guidance that goes with that that is challenging.
>>>=20
>>> I don=92t agree that RTCP information should be sent as higher =
priority than the media. Ideally, it should be sent with the same =
priority as the media, so it can be used to sample the RTT. This RTT =
sample is independent of RMCAT. It=92s in base RTP specification, and so =
is something we need to support to the extent possible.
>>>=20
>>> Since not all the media sent by a single SSRC has the same marking, =
my suggestion would be that each SSRC mark the RTCP packets it sends =
with one of the same code points as it uses to mark the media. Since =
RTCP is somewhat important, it would make sense for each SSRC to mark =
the RTCP packets it sends using the highest priority code point it uses =
to mark the RTP media packets it sends.
>>=20
>> That makes sense to me. Paul, and others, do you agree with that last =
paragraph?
>=20
> I agree that the value should be one of the values used for media, but =
the challenge is specifying which DSCP value to use.  If audio is sent =
using EF, video as AF41, app sharing video as AF31, etc., which of these =
does the endpoint choose to use for RTCP?

Such an endpoint will have three SSRC values, and will mark the RTCP =
packets sent by each SSRC in the same way as the media packets sent by =
that SSRC. That is, the RTCP relating to the audio will be marked EF, =
the RTCP relating to the video will be marked AF41, and the RTCP =
relating to the application sharing video will be marked AF31. The RTCP =
packets for each stream are sent separately, so this case seems =
straight-forward.

The more difficult case is when an SSRC is sending video using different =
markings for RTP packets carrying the I- and P-frames. Should that SSRC =
then mark its RTCP packets like the RTP packets carrying I-frames, like =
the RTP packets carrying P-frames, or what?

Colin


=20

> Since DSCP values cannot be assumed to any particular PHB behavior in =
the network, that makes answering that question challenging.
>=20
> My personal preference in a voice/video conference would be to use the =
same marking as is used for those flows -- and I would use the same =
value for both voice and video.  However, if the app sharing video is =
considered more important by some applications, perhaps that's the value =
that should be used in those applications.
>=20
> I can accept the suggestion, leaving the specific selection as an =
implementation matter.  I do wish there was something more concrete so =
that two devices would always follow the same approach to selecting a =
value, but I doubt we can get agreement very quickly.
>=20
> Paul
>=20



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




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From: "Paul E. Jones" <paulej@packetizer.com>
To: "Colin Perkins" <csp@csperkins.org>
Date: Wed, 27 Aug 2014 05:17:38 +0000
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Subject: Re: [AVTCORE] Treatment of RTCP (was Re: [Dart] Colin Perkins comments - WGLC: draft-ietf-dart-dscp-rtp-02)
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Colin,

>On 26 Aug 2014, at 16:52, Paul E. Jones <paulej@packetizer.com> wrote:
>>  Ben,
>>>>>>>  Getting feedback to the sender is important, though. In an ideal=
=20
>>>>>>>world, I would argue that RTCP packets should be marked with=20
>>>>>>>whatever DSCP value will deliver RTCP packets in the most=20
>>>>>>>expedient way. Since we don't have an ideal world, I don't know=20
>>>>>>>which DSCP value that would be.
>>>>>>
>>>>>>  Would the starting positions of "getting feedback is important,=20
>>>>>>even of not for RTT estimates" and "we need RTCP for RTT=20
>>>>>>estimates" likely land on the same guidance for DSCP values?
>>>>>>
>>>>>>  The argument to send RTCP packets in the most expedient way=20
>>>>>>sounds reasonable. I don't know if we need to recommend a=20
>>>>>>particular DSCP, since we already have quite a bit of text on how=20
>>>>>>DSCPs might (or might not) map into some predictable PHB=20
>>>>>>treatment.
>>>>>
>>>>>  Good question and valid point. Nowhere in the document do we=20
>>>>>recommend the use of a particular DSCP value for any particular=20
>>>>>thing, and we should not recommend a particular value for RTCP in=20
>>>>>this document. I'm just not sure what statements should be made.
>>>>>
>>>>>  I suspect we can all agree that RTCP information is important.=20
>>>>>It's just the DSCP-related guidance that goes with that that is=20
>>>>>challenging.
>>>>
>>>>  I don=E2=80=99t agree that RTCP information should be sent as higher=
=20
>>>>priority than the media. Ideally, it should be sent with the same=20
>>>>priority as the media, so it can be used to sample the RTT. This RTT=
=20
>>>>sample is independent of RMCAT. It=E2=80=99s in base RTP specification,=
 and=20
>>>>so is something we need to support to the extent possible.
>>>>
>>>>  Since not all the media sent by a single SSRC has the same marking,=
=20
>>>>my suggestion would be that each SSRC mark the RTCP packets it sends=
=20
>>>>with one of the same code points as it uses to mark the media. Since=
=20
>>>>RTCP is somewhat important, it would make sense for each SSRC to=20
>>>>mark the RTCP packets it sends using the highest priority code point=
=20
>>>>it uses to mark the RTP media packets it sends.
>>>
>>>  That makes sense to me. Paul, and others, do you agree with that=20
>>>last paragraph?
>>
>>  I agree that the value should be one of the values used for media,=20
>>but the challenge is specifying which DSCP value to use. If audio is=20
>>sent using EF, video as AF41, app sharing video as AF31, etc., which=20
>>of these does the endpoint choose to use for RTCP?
>
>Such an endpoint will have three SSRC values, and will mark the RTCP=20
>packets sent by each SSRC in the same way as the media packets sent by=20
>that SSRC. That is, the RTCP relating to the audio will be marked EF,=20
>the RTCP relating to the video will be marked AF41, and the RTCP=20
>relating to the application sharing video will be marked AF31. The RTCP=
=20
>packets for each stream are sent separately, so this case seems=20
>straight-forward.

I'm a little confused here.  These are separate SSRCs, but all part of a=
=20
single RTP session.  So, wouldn't the three SSRCs be reported in=20
different report blocks in the sender report?


>The more difficult case is when an SSRC is sending video using=20
>different markings for RTP packets carrying the I- and P-frames. Should=
=20
>that SSRC then mark its RTCP packets like the RTP packets carrying=20
>I-frames, like the RTP packets carrying P-frames, or what?

For this, I would argue it should be whatever DSCP that has the lower=20
drop precedence (in the case of assured forwarding).  Otherwise, it=20
should be the same as the media, I'd think.

Paul


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From: Colin Perkins <csp@csperkins.org>
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Subject: Re: [AVTCORE] Treatment of RTCP (was Re: [Dart] Colin Perkins comments - WGLC: draft-ietf-dart-dscp-rtp-02)
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On 27 Aug 2014, at 06:17, Paul E. Jones <paulej@packetizer.com> wrote:
>> On 26 Aug 2014, at 16:52, Paul E. Jones <paulej@packetizer.com> =
wrote:
>>>>>>>> Getting feedback to the sender is important, though. In an =
ideal world, I would argue that RTCP packets should be marked with =
whatever DSCP value will deliver RTCP packets in the most expedient way. =
Since we don't have an ideal world, I don't know which DSCP value that =
would be.
>>>>>>>=20
>>>>>>> Would the starting positions of "getting feedback is important, =
even of not for RTT estimates" and "we need RTCP for RTT estimates" =
likely land on the same guidance for DSCP values?
>>>>>>>=20
>>>>>>> The argument to send RTCP packets in the most expedient way =
sounds reasonable. I don't know if we need to recommend a particular =
DSCP, since we already have quite a bit of text on how DSCPs might (or =
might not) map into some predictable PHB treatment.
>>>>>>=20
>>>>>> Good question and valid point. Nowhere in the document do we =
recommend the use of a particular DSCP value for any particular thing, =
and we should not recommend a particular value for RTCP in this =
document. I'm just not sure what statements should be made.
>>>>>>=20
>>>>>> I suspect we can all agree that RTCP information is important. =
It's just the DSCP-related guidance that goes with that that is =
challenging.
>>>>>=20
>>>>> I don=92t agree that RTCP information should be sent as higher =
priority than the media. Ideally, it should be sent with the same =
priority as the media, so it can be used to sample the RTT. This RTT =
sample is independent of RMCAT. It=92s in base RTP specification, and so =
is something we need to support to the extent possible.
>>>>>=20
>>>>> Since not all the media sent by a single SSRC has the same =
marking, my suggestion would be that each SSRC mark the RTCP packets it =
sends with one of the same code points as it uses to mark the media. =
Since RTCP is somewhat important, it would make sense for each SSRC to =
mark the RTCP packets it sends using the highest priority code point it =
uses to mark the RTP media packets it sends.
>>>>=20
>>>> That makes sense to me. Paul, and others, do you agree with that =
last paragraph?
>>>=20
>>> I agree that the value should be one of the values used for media, =
but the challenge is specifying which DSCP value to use. If audio is =
sent using EF, video as AF41, app sharing video as AF31, etc., which of =
these does the endpoint choose to use for RTCP?
>>=20
>> Such an endpoint will have three SSRC values, and will mark the RTCP =
packets sent by each SSRC in the same way as the media packets sent by =
that SSRC. That is, the RTCP relating to the audio will be marked EF, =
the RTCP relating to the video will be marked AF41, and the RTCP =
relating to the application sharing video will be marked AF31. The RTCP =
packets for each stream are sent separately, so this case seems =
straight-forward.
>=20
> I'm a little confused here.  These are separate SSRCs, but all part of =
a single RTP session.  So, wouldn=92t the three SSRCs be reported in =
different report blocks in the sender report?

Each of the SSRCs will report on all the others, unless the =
rtp-multi-stream-optimisation draft is used, but so what?=20

>> The more difficult case is when an SSRC is sending video using =
different markings for RTP packets carrying the I- and P-frames. Should =
that SSRC then mark its RTCP packets like the RTP packets carrying =
I-frames, like the RTP packets carrying P-frames, or what?
>=20
> For this, I would argue it should be whatever DSCP that has the lower =
drop precedence (in the case of assured forwarding).  Otherwise, it =
should be the same as the media, I'd think.

You mean the RTCP for each SSRC should use the DSCP with the lowest drop =
preference of those used for the media sent by that SSRC?

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




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From: "Black, David" <david.black@emc.com>
To: Ben Campbell <ben@nostrum.com>, Colin Perkins <csp@csperkins.org>, "Paul E. Jones" <paulej@packetizer.com>
Date: Tue, 26 Aug 2014 11:39:43 -0400
Thread-Topic: Treatment of RTCP (was Re: [Dart] Colin Perkins comments - WGLC: draft-ietf-dart-dscp-rtp-02)
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Subject: Re: [AVTCORE] Treatment of RTCP (was Re: [Dart] Colin Perkins comments - WGLC: draft-ietf-dart-dscp-rtp-02)
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> > Since not all the media sent by a single SSRC has the same marking, my
> suggestion would be that each SSRC mark the RTCP packets it sends with on=
e of
> the same code points as it uses to mark the media. Since RTCP is somewhat
> important, it would make sense for each SSRC to mark the RTCP packets it =
sends
> using the highest priority code point it uses to mark the RTP media packe=
ts it
> sends.
>=20
> That makes sense to me. Paul, and others, do you agree with that last
> paragraph?

<editor hat off>

Sure, when there's a notion of priority or importance.  There won't always
be one (e.g., is CS2 higher priority than AF2x?  That depends ...).  OTOH,
we don't need to say much here, e.g., the class selector codepoints (CSx)
are ordered, and adding a pointer to draft-ietf-tsvwg-rtcweb-qos will be
helpful.

</editor hat off>

Turning to the other RTCP issue, [F] on multi-stream optimization, I wonder
whether we inadvertently framed that issue backwards.  Given the above, a
single RTCP report on multiple RTP streams that use rather different DSCPs
may not result in representative RTT values for all of the streams, because
the report has to be sent with one DSCP.  If that outcome is a problem,
one should send separate RTCP reports (duh!).=20

That statement seems valuable to make and can be made with a citation of
RFC 3530, as opposed to the multi-stream optimization draft.

Harald and Colin - what do you think?

Thanks,
--David

> -----Original Message-----
> From: Ben Campbell [mailto:ben@nostrum.com]
> Sent: Tuesday, August 26, 2014 10:16 AM
> To: Colin Perkins; Paul E. Jones
> Cc: Black, David; dart@ietf.org; avt@ietf.org WG; draft-ietf-dart-dscp-
> rtp.all@tools.ietf.org
> Subject: Treatment of RTCP (was Re: [Dart] Colin Perkins comments - WGLC:
> draft-ietf-dart-dscp-rtp-02)
>=20
>=20
> On Aug 26, 2014, at 5:17 AM, Colin Perkins <csp@csperkins.org> wrote:
>=20
> >>>>
> >>>> Getting feedback to the sender is important, though. In an ideal wor=
ld, I
> would argue that RTCP packets should be marked with whatever DSCP value w=
ill
> deliver RTCP packets in the most expedient way. Since we don't have an id=
eal
> world, I don't know which DSCP value that would be.
> >>>
> >>> Would the starting positions of "getting feedback is important, even =
of
> not for RTT estimates" and "we need RTCP for RTT estimates" likely land o=
n the
> same guidance for DSCP values?
> >>>
> >>> The argument to send RTCP packets in the most expedient way sounds
> reasonable. I don't know if we need to recommend a particular DSCP, since=
 we
> already have quite a bit of text on how DSCPs might (or might not) map in=
to
> some predictable PHB treatment.
> >>
> >> Good question and valid point.  Nowhere in the document do we recommen=
d the
> use of a particular DSCP value for any particular thing, and we should no=
t
> recommend a particular value for RTCP in this document.  I'm just not sur=
e
> what statements should be made.
> >>
> >> I suspect we can all agree that RTCP information is important.  It's j=
ust
> the DSCP-related guidance that goes with that that is challenging.
> >
> > I don't agree that RTCP information should be sent as higher priority t=
han
> the media. Ideally, it should be sent with the same priority as the media=
, so
> it can be used to sample the RTT. This RTT sample is independent of RMCAT=
.
> It's in base RTP specification, and so is something we need to support to=
 the
> extent possible.
> >
> > Since not all the media sent by a single SSRC has the same marking, my
> suggestion would be that each SSRC mark the RTCP packets it sends with on=
e of
> the same code points as it uses to mark the media. Since RTCP is somewhat
> important, it would make sense for each SSRC to mark the RTCP packets it =
sends
> using the highest priority code point it uses to mark the RTP media packe=
ts it
> sends.
>=20
> That makes sense to me. Paul, and others, do you agree with that last
> paragraph?
>=20
> /Ben
>=20


From nobody Wed Aug 27 08:38:56 2014
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From: "Black, David" <david.black@emc.com>
To: Colin Perkins <csp@csperkins.org>
Date: Tue, 26 Aug 2014 12:27:14 -0400
Thread-Topic: [Dart] draft-ietf-dart-dscp-rtp-02 - Section 5.3 RTCP multi-stream optimization text
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Subject: Re: [AVTCORE] [Dart] draft-ietf-dart-dscp-rtp-02 - Section 5.3 RTCP multi-stream optimization text
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<editor hat OFF>

Colin writes:

   The discussion of the multi-stream-optimisation draft is a separate
   issue, and is correct as it is.

Looking closer, that discussion in the current DART draft currently says:

   RTCP multi-stream reporting optimizations for an RTP session
   [I-D.ietf-avtcore-rtp-multi-stream-optimisation] assume that the RTP
   streams involved experience the same packet loss behavior.  This
   mechanism is highly inappropriate when the RTP streams involved use
   different PHBs, even if those PHBs differ solely in drop precedence.

That's based on this text from the last paragraph of Section 1
(Introduction) of the multi-stream optimization draft:

   This memo defines such an RTCP extension, RTCP Reporting Groups.
   This extension is used to indicate the SSRCs that originate from the
   same endpoint, and therefore have identical reception quality, hence
   allowing the endpoints to suppress unnecessary RTCP reception quality
   reports.

When different DSCPs are used for SSRCs (RTP streams) that originate
from the same source, the statement that they "therefore have identical
reception quality" is clearly incorrect, so it looks like the current
DART draft text is correct, IMHO.

Harald?

</editor hat OFF>

Thanks,
--David

> -----Original Message-----
> From: Colin Perkins [mailto:csp@csperkins.org]
> Sent: Tuesday, August 26, 2014 11:54 AM
> To: Black, David
> Cc: Ben Campbell; Paul E. Jones; dart@ietf.org; avt@ietf.org WG; draft-ie=
tf-
> dart-dscp-rtp.all@tools.ietf.org
> Subject: Re: Treatment of RTCP (was Re: [Dart] Colin Perkins comments - W=
GLC:
> draft-ietf-dart-dscp-rtp-02)
>=20
>=20
> On 26 Aug 2014, at 16:39, Black, David <david.black@emc.com> wrote:
>=20
> >>> Since not all the media sent by a single SSRC has the same marking, m=
y
> >> suggestion would be that each SSRC mark the RTCP packets it sends with=
 one
> of
> >> the same code points as it uses to mark the media. Since RTCP is somew=
hat
> >> important, it would make sense for each SSRC to mark the RTCP packets =
it
> sends
> >> using the highest priority code point it uses to mark the RTP media pa=
ckets
> it
> >> sends.
> >>
> >> That makes sense to me. Paul, and others, do you agree with that last
> >> paragraph?
> >
> > <editor hat off>
> >
> > Sure, when there's a notion of priority or importance.  There won't alw=
ays
> > be one (e.g., is CS2 higher priority than AF2x?  That depends ...).  OT=
OH,
> > we don't need to say much here, e.g., the class selector codepoints (CS=
x)
> > are ordered, and adding a pointer to draft-ietf-tsvwg-rtcweb-qos will b=
e
> > helpful.
> >
> > </editor hat off>
> >
> > Turning to the other RTCP issue, [F] on multi-stream optimization, I wo=
nder
> > whether we inadvertently framed that issue backwards.  Given the above,=
 a
> > single RTCP report on multiple RTP streams that use rather different DS=
CPs
> > may not result in representative RTT values for all of the streams, bec=
ause
> > the report has to be sent with one DSCP.  If that outcome is a problem,
> > one should send separate RTCP reports (duh!).
>=20
> Each reporting SSRC sends a separate RTCP report block for each SSRC it
> receives, so this is not a problem. The question is what marking an SSRC =
uses
> for the RTCP packets it sends, given that it uses several different marki=
ngs
> for the RTP media packets it sends.
>=20
> > That statement seems valuable to make and can be made with a citation o=
f
> > RFC 3530, as opposed to the multi-stream optimization draft.
> >
> > Harald and Colin - what do you think?
>=20
> The discussion of the multi-stream-optimisation draft is a separate issue=
, and
> is correct as it is.
>=20
> --
> Colin Perkins
> http://csperkins.org/
>=20
>=20


From nobody Wed Aug 27 08:38:57 2014
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From: "Black, David" <david.black@emc.com>
To: Colin Perkins <csp@csperkins.org>, "Paul E. Jones" <paulej@packetizer.com>
Date: Tue, 26 Aug 2014 12:38:22 -0400
Thread-Topic: Treatment of RTCP (was Re: [Dart] Colin Perkins comments - WGLC: draft-ietf-dart-dscp-rtp-02)
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Subject: Re: [AVTCORE] Treatment of RTCP (was Re: [Dart] Colin Perkins comments - WGLC: draft-ietf-dart-dscp-rtp-02)
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> The more difficult case is when an SSRC is sending video using different
> markings for RTP packets carrying the I- and P-frames. Should that SSRC t=
hen
> mark its RTCP packets like the RTP packets carrying I-frames, like the RT=
P
> packets carrying P-frames, or what?

Answering a question w/a question :-), how are those reports likely to be u=
sed?

For example, if the primary use of these reports is to adjust a variable ra=
te
codec's sending rate, the P-frame info is probably more useful as indicativ=
e
of what's happening to the traffic that the network drops first when the go=
ing
gets rough (or whose delivery w/o loss indicates that a sending rate increa=
se
may be reasonable), which suggests P-frame-like RTCP report marking.

Thanks,
--David


> -----Original Message-----
> From: Colin Perkins [mailto:csp@csperkins.org]
> Sent: Tuesday, August 26, 2014 12:19 PM
> To: Paul E. Jones
> Cc: Ben Campbell; Black, David; dart@ietf.org; avt@ietf.org WG; draft-iet=
f-
> dart-dscp-rtp.all@tools.ietf.org
> Subject: Re: Treatment of RTCP (was Re: [Dart] Colin Perkins comments - W=
GLC:
> draft-ietf-dart-dscp-rtp-02)
>=20
> On 26 Aug 2014, at 16:52, Paul E. Jones <paulej@packetizer.com> wrote:
> > Ben,
> >>>>>> Getting feedback to the sender is important, though. In an ideal w=
orld,
> I would argue that RTCP packets should be marked with whatever DSCP value=
 will
> deliver RTCP packets in the most expedient way. Since we don't have an id=
eal
> world, I don't know which DSCP value that would be.
> >>>>>
> >>>>> Would the starting positions of "getting feedback is important, eve=
n of
> not for RTT estimates" and "we need RTCP for RTT estimates" likely land o=
n the
> same guidance for DSCP values?
> >>>>>
> >>>>> The argument to send RTCP packets in the most expedient way sounds
> reasonable. I don't know if we need to recommend a particular DSCP, since=
 we
> already have quite a bit of text on how DSCPs might (or might not) map in=
to
> some predictable PHB treatment.
> >>>>
> >>>> Good question and valid point. Nowhere in the document do we recomme=
nd
> the use of a particular DSCP value for any particular thing, and we shoul=
d not
> recommend a particular value for RTCP in this document. I'm just not sure=
 what
> statements should be made.
> >>>>
> >>>> I suspect we can all agree that RTCP information is important. It's =
just
> the DSCP-related guidance that goes with that that is challenging.
> >>>
> >>> I don't agree that RTCP information should be sent as higher priority=
 than
> the media. Ideally, it should be sent with the same priority as the media=
, so
> it can be used to sample the RTT. This RTT sample is independent of RMCAT=
.
> It's in base RTP specification, and so is something we need to support to=
 the
> extent possible.
> >>>
> >>> Since not all the media sent by a single SSRC has the same marking, m=
y
> suggestion would be that each SSRC mark the RTCP packets it sends with on=
e of
> the same code points as it uses to mark the media. Since RTCP is somewhat
> important, it would make sense for each SSRC to mark the RTCP packets it =
sends
> using the highest priority code point it uses to mark the RTP media packe=
ts it
> sends.
> >>
> >> That makes sense to me. Paul, and others, do you agree with that last
> paragraph?
> >
> > I agree that the value should be one of the values used for media, but =
the
> challenge is specifying which DSCP value to use.  If audio is sent using =
EF,
> video as AF41, app sharing video as AF31, etc., which of these does the
> endpoint choose to use for RTCP?
>=20
> Such an endpoint will have three SSRC values, and will mark the RTCP pack=
ets
> sent by each SSRC in the same way as the media packets sent by that SSRC.=
 That
> is, the RTCP relating to the audio will be marked EF, the RTCP relating t=
o the
> video will be marked AF41, and the RTCP relating to the application shari=
ng
> video will be marked AF31. The RTCP packets for each stream are sent
> separately, so this case seems straight-forward.
>=20
> The more difficult case is when an SSRC is sending video using different
> markings for RTP packets carrying the I- and P-frames. Should that SSRC t=
hen
> mark its RTCP packets like the RTP packets carrying I-frames, like the RT=
P
> packets carrying P-frames, or what?
>=20
> Colin
>=20
>=20
>=20
>=20
> > Since DSCP values cannot be assumed to any particular PHB behavior in t=
he
> network, that makes answering that question challenging.
> >
> > My personal preference in a voice/video conference would be to use the =
same
> marking as is used for those flows -- and I would use the same value for =
both
> voice and video.  However, if the app sharing video is considered more
> important by some applications, perhaps that's the value that should be u=
sed
> in those applications.
> >
> > I can accept the suggestion, leaving the specific selection as an
> implementation matter.  I do wish there was something more concrete so th=
at
> two devices would always follow the same approach to selecting a value, b=
ut I
> doubt we can get agreement very quickly.
> >
> > Paul
> >
>=20
>=20
>=20
> --
> Colin Perkins
> http://csperkins.org/
>=20
>=20


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Subject: Re: [AVTCORE] Treatment of RTCP (was Re: [Dart] Colin Perkins comments - WGLC: draft-ietf-dart-dscp-rtp-02)
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On 26 Aug 2014, at 17:38, Black, David <david.black@emc.com> wrote:
>> The more difficult case is when an SSRC is sending video using =
different
>> markings for RTP packets carrying the I- and P-frames. Should that =
SSRC then
>> mark its RTCP packets like the RTP packets carrying I-frames, like =
the RTP
>> packets carrying P-frames, or what?
>=20
> Answering a question w/a question :-), how are those reports likely to =
be used?
>=20
> For example, if the primary use of these reports is to adjust a =
variable rate
> codec's sending rate, the P-frame info is probably more useful as =
indicative
> of what's happening to the traffic that the network drops first when =
the going
> gets rough (or whose delivery w/o loss indicates that a sending rate =
increase
> may be reasonable), which suggests P-frame-like RTCP report marking.

I doubt the RTT estimate derived from RTCP is used for congestion =
control, since it=92s too infrequent to get insight into the dynamics. =
It=92s for user experience reporting, maybe rough clustering of =
receivers, that sort of thing.=20

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




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From: Colin Perkins <csp@csperkins.org>
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Subject: Re: [AVTCORE] [Dart] Treatment of RTCP (was Re: Colin Perkins comments - WGLC: draft-ietf-dart-dscp-rtp-02)
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On 27 Aug 2014, at 19:39, Michael Welzl <michawe@ifi.uio.no> wrote:
> On 27. aug. 2014, at 18:00, Colin Perkins <csp@csperkins.org> wrote:
>> On 26 Aug 2014, at 17:38, Black, David <david.black@emc.com> wrote:
>>>> The more difficult case is when an SSRC is sending video using =
different
>>>> markings for RTP packets carrying the I- and P-frames. Should that =
SSRC then
>>>> mark its RTCP packets like the RTP packets carrying I-frames, like =
the RTP
>>>> packets carrying P-frames, or what?
>>>=20
>>> Answering a question w/a question :-), how are those reports likely =
to be used?
>>>=20
>>> For example, if the primary use of these reports is to adjust a =
variable rate
>>> codec's sending rate, the P-frame info is probably more useful as =
indicative
>>> of what's happening to the traffic that the network drops first when =
the going
>>> gets rough (or whose delivery w/o loss indicates that a sending rate =
increase
>>> may be reasonable), which suggests P-frame-like RTCP report marking.
>>=20
>> I doubt the RTT estimate derived from RTCP is used for congestion =
control, since it=92s too infrequent to get insight into the dynamics. =
It=92s for user experience reporting, maybe rough clustering of =
receivers, that sort of thing.=20
>=20
> I'd agree if I didn't have the impression, in RMCAT, that nothing but =
RTP / RTCP is allowed?!  So can we send extra packets to probe for the =
RTT?

I=92d expect RMCAT will probe the one-way delay variation by measuring =
the arrival times of RTP packets, and report that in RTCP once per video =
frame (or so). That would give a much more accurate measurement than a =
single-packet sample of the RTT derived from RTCP, and would still allow =
per-frame adaptation.

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




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Subject: Re: [AVTCORE] [Dart] Treatment of RTCP (was Re: Colin Perkins comments - WGLC: draft-ietf-dart-dscp-rtp-02)
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On 27 Aug 2014, at 19:47, Michael Welzl <michawe@ifi.uio.no> wrote:
> On 27. aug. 2014, at 20:42, Colin Perkins <csp@csperkins.org> wrote:
>> On 27 Aug 2014, at 19:39, Michael Welzl <michawe@ifi.uio.no> wrote:
>>> On 27. aug. 2014, at 18:00, Colin Perkins <csp@csperkins.org> wrote:
>>>> On 26 Aug 2014, at 17:38, Black, David <david.black@emc.com> wrote:
>>>>>> The more difficult case is when an SSRC is sending video using =
different
>>>>>> markings for RTP packets carrying the I- and P-frames. Should =
that SSRC then
>>>>>> mark its RTCP packets like the RTP packets carrying I-frames, =
like the RTP
>>>>>> packets carrying P-frames, or what?
>>>>>=20
>>>>> Answering a question w/a question :-), how are those reports =
likely to be used?
>>>>>=20
>>>>> For example, if the primary use of these reports is to adjust a =
variable rate
>>>>> codec's sending rate, the P-frame info is probably more useful as =
indicative
>>>>> of what's happening to the traffic that the network drops first =
when the going
>>>>> gets rough (or whose delivery w/o loss indicates that a sending =
rate increase
>>>>> may be reasonable), which suggests P-frame-like RTCP report =
marking.
>>>>=20
>>>> I doubt the RTT estimate derived from RTCP is used for congestion =
control, since it=92s too infrequent to get insight into the dynamics. =
It=92s for user experience reporting, maybe rough clustering of =
receivers, that sort of thing.=20
>>>=20
>>> I'd agree if I didn't have the impression, in RMCAT, that nothing =
but RTP / RTCP is allowed?!  So can we send extra packets to probe for =
the RTT?
>>=20
>> I=92d expect RMCAT will probe the one-way delay variation by =
measuring the arrival times of RTP packets, and report that in RTCP once =
per video frame (or so). That would give a much more accurate =
measurement than a single-packet sample of the RTT derived from RTCP, =
and would still allow per-frame adaptation.
>=20
> I agree that this is a useful signal to use - but I think you still =
need a rough idea of the RTT to be able to interoperate with TCP, else =
you may cause TCP to go into RTOs and be ignorant of that. But this =
could perhaps be a more coarse signal - maybe that's okay. I think this =
is reasonable.

You have the RTT estimate, irrespective of the other information you =
include along with the regular RTCP packets, although it=92s relatively =
infrequent and coarse-grained.

> I wonder if your RTCP draft in RMCAT should clarify this?  (sorry if =
it does already, didn=92t check)

It probably should talk about this, I agree.


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




--Apple-Mail=_C67B8F62-80F6-47E2-8469-2AFB250D77B7
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<html><head><meta http-equiv=3D"Content-Type" content=3D"text/html =
charset=3Dwindows-1252"></head><body style=3D"word-wrap: break-word; =
-webkit-nbsp-mode: space; -webkit-line-break: after-white-space;">On 27 =
Aug 2014, at 19:47, Michael Welzl &lt;<a =
href=3D"mailto:michawe@ifi.uio.no">michawe@ifi.uio.no</a>&gt; =
wrote:<br><div><blockquote type=3D"cite"><div style=3D"word-wrap: =
break-word; -webkit-nbsp-mode: space; -webkit-line-break: =
after-white-space;"><div><div>On 27. aug. 2014, at 20:42, Colin Perkins =
&lt;<a href=3D"mailto:csp@csperkins.org">csp@csperkins.org</a>&gt; =
wrote:</div><blockquote type=3D"cite"><div style=3D"font-size: 12px; =
font-style: normal; font-variant: normal; font-weight: normal; =
letter-spacing: normal; line-height: normal; orphans: auto; text-align: =
start; text-indent: 0px; text-transform: none; white-space: normal; =
widows: auto; word-spacing: 0px; -webkit-text-stroke-width: 0px;">On 27 =
Aug 2014, at 19:39, Michael Welzl &lt;<a =
href=3D"mailto:michawe@ifi.uio.no">michawe@ifi.uio.no</a>&gt; =
wrote:<br><blockquote type=3D"cite">On 27. aug. 2014, at 18:00, Colin =
Perkins &lt;<a href=3D"mailto:csp@csperkins.org">csp@csperkins.org</a>&gt;=
 wrote:<br><blockquote type=3D"cite">On 26 Aug 2014, at 17:38, Black, =
David &lt;<a =
href=3D"mailto:david.black@emc.com">david.black@emc.com</a>&gt; =
wrote:<br><blockquote type=3D"cite"><blockquote type=3D"cite">The more =
difficult case is when an SSRC is sending video using =
different<br>markings for RTP packets carrying the I- and P-frames. =
Should that SSRC then<br>mark its RTCP packets like the RTP packets =
carrying I-frames, like the RTP<br>packets carrying P-frames, or =
what?<br></blockquote><br>Answering a question w/a question :-), how are =
those reports likely to be used?<br><br>For example, if the primary use =
of these reports is to adjust a variable rate<br>codec's sending rate, =
the P-frame info is probably more useful as indicative<br>of what's =
happening to the traffic that the network drops first when the =
going<br>gets rough (or whose delivery w/o loss indicates that a sending =
rate increase<br>may be reasonable), which suggests P-frame-like RTCP =
report marking.<br></blockquote><br>I doubt the RTT estimate derived =
from RTCP is used for congestion control, since it=92s too infrequent to =
get insight into the dynamics. It=92s for user experience reporting, =
maybe rough clustering of receivers, that sort of thing.<span =
class=3D"Apple-converted-space">&nbsp;</span><br></blockquote><br>I'd =
agree if I didn't have the impression, in RMCAT, that nothing but RTP / =
RTCP is allowed?! &nbsp;So can we send extra packets to probe for the =
RTT?<br></blockquote><br>I=92d expect RMCAT will probe the one-way delay =
variation by measuring the arrival times of RTP packets, and report that =
in RTCP once per video frame (or so). That would give a much more =
accurate measurement than a single-packet sample of the RTT derived from =
RTCP, and would still allow per-frame =
adaptation.<br></div></blockquote><div><br></div>I agree that this is a =
useful signal to use - but I think you still need a rough idea of the =
RTT to be able to interoperate with TCP, else you may cause TCP to go =
into RTOs and be ignorant of that. But this could perhaps be a more =
coarse signal - maybe that's okay. I think this is =
reasonable.</div></div></blockquote><div><br></div><div>You have the RTT =
estimate, irrespective of the other information you include along with =
the regular RTCP packets, although it=92s relatively infrequent and =
coarse-grained.</div><br><blockquote type=3D"cite"><div =
style=3D"word-wrap: break-word; -webkit-nbsp-mode: space; =
-webkit-line-break: after-white-space;"><div>I wonder if your RTCP draft =
in RMCAT should clarify this? &nbsp;(sorry if it does already, didn=92t =
check)</div></div></blockquote><br></div><div>It probably should talk =
about this, I agree.</div><div apple-content-edited=3D"true"><span =
class=3D"Apple-style-span" style=3D"border-collapse: separate; =
border-spacing: 0px;"><div><br class=3D"Apple-interchange-newline"><br =
class=3D"khtml-block-placeholder"></div><div>--&nbsp;</div><div></div><div=
>Colin Perkins</div><div><a =
href=3D"http://csperkins.org/">http://csperkins.org/</a></div><div><br></d=
iv></span><br class=3D"Apple-interchange-newline">

</div>
<br></body></html>=

--Apple-Mail=_C67B8F62-80F6-47E2-8469-2AFB250D77B7--


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Subject: Re: [AVTCORE] [Dart] Treatment of RTCP (was Re: Colin Perkins comments - WGLC: draft-ietf-dart-dscp-rtp-02)
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Hi,

Where are we with this discussion? Are there conclusions buried in there =
somewhere?

Thanks!

Ben.

On Aug 27, 2014, at 1:50 PM, Colin Perkins <csp@csperkins.org> wrote:

> On 27 Aug 2014, at 19:47, Michael Welzl <michawe@ifi.uio.no> wrote:
>> On 27. aug. 2014, at 20:42, Colin Perkins <csp@csperkins.org> wrote:
>>> On 27 Aug 2014, at 19:39, Michael Welzl <michawe@ifi.uio.no> wrote:
>>>> On 27. aug. 2014, at 18:00, Colin Perkins <csp@csperkins.org> =
wrote:
>>>>> On 26 Aug 2014, at 17:38, Black, David <david.black@emc.com> =
wrote:
>>>>>>> The more difficult case is when an SSRC is sending video using =
different
>>>>>>> markings for RTP packets carrying the I- and P-frames. Should =
that SSRC then
>>>>>>> mark its RTCP packets like the RTP packets carrying I-frames, =
like the RTP
>>>>>>> packets carrying P-frames, or what?
>>>>>>=20
>>>>>> Answering a question w/a question :-), how are those reports =
likely to be used?
>>>>>>=20
>>>>>> For example, if the primary use of these reports is to adjust a =
variable rate
>>>>>> codec's sending rate, the P-frame info is probably more useful as =
indicative
>>>>>> of what's happening to the traffic that the network drops first =
when the going
>>>>>> gets rough (or whose delivery w/o loss indicates that a sending =
rate increase
>>>>>> may be reasonable), which suggests P-frame-like RTCP report =
marking.
>>>>>=20
>>>>> I doubt the RTT estimate derived from RTCP is used for congestion =
control, since it=92s too infrequent to get insight into the dynamics. =
It=92s for user experience reporting, maybe rough clustering of =
receivers, that sort of thing.=20
>>>>=20
>>>> I'd agree if I didn't have the impression, in RMCAT, that nothing =
but RTP / RTCP is allowed?!  So can we send extra packets to probe for =
the RTT?
>>>=20
>>> I=92d expect RMCAT will probe the one-way delay variation by =
measuring the arrival times of RTP packets, and report that in RTCP once =
per video frame (or so). That would give a much more accurate =
measurement than a single-packet sample of the RTT derived from RTCP, =
and would still allow per-frame adaptation.
>>=20
>> I agree that this is a useful signal to use - but I think you still =
need a rough idea of the RTT to be able to interoperate with TCP, else =
you may cause TCP to go into RTOs and be ignorant of that. But this =
could perhaps be a more coarse signal - maybe that's okay. I think this =
is reasonable.
>=20
> You have the RTT estimate, irrespective of the other information you =
include along with the regular RTCP packets, although it=92s relatively =
infrequent and coarse-grained.
>=20
>> I wonder if your RTCP draft in RMCAT should clarify this?  (sorry if =
it does already, didn=92t check)
>=20
> It probably should talk about this, I agree.
>=20
>=20
> --=20
> Colin Perkins
> http://csperkins.org/
>=20
>=20
>=20
> _______________________________________________
> Dart mailing list
> Dart@ietf.org
> https://www.ietf.org/mailman/listinfo/dart


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References: <embac59e09-6dad-42df-94b2-7daa46d31d5d@sydney> <704DAEE2-C26F-48C8-8C75-548FE115B91F@csperkins.org> <8D3D17ACE214DC429325B2B98F3AE712077BB42E1F@MX15A.corp.emc.com> <22E25F9C-E9B1-4C3C-989E-570BAAF58018@csperkins.org> <A4501649-B293-4B6F-A4EB-A08B30EF922C@ifi.uio.no> <1B4C6B70-B3EB-45C0-892D-328954E3AD4E@csperkins.org> <B43C20C1-C34E-4441-80EB-24C89DCE430D@ifi.uio.no> <0F1C9E57-EDA5-4A44-9F8A-87A76C159AED@csperkins.org> <DB56BB81-4CD0-4A2A-A664-67103CED72BC@nostrum.com> <ABC20E38-5908-400D-BAF2-FEA46931DC76@ifi.uio.no>
To: Michael Welzl <michawe@ifi.uio.no>, Colin Perkins <csp@csperkins.org>, "Paul E. Jones" <paulej@packetizer.com>, "Black, David" <david.black@emc.com>
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Subject: Re: [AVTCORE] [Dart] Treatment of RTCP (was Re: Colin Perkins comments - WGLC: draft-ietf-dart-dscp-rtp-02)
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On Aug 27, 2014, at 3:45 PM, Michael Welzl <michawe@ifi.uio.no> wrote:

> I don't know the beginning of this but I can help go up the stack:
>=20
> for the short dialogue between Colin and me, the conclusion is: Colin =
is right. RTCP probably won't be used by congestion control to estimate =
the RTT.
>=20

Thanks, Michael, I think that fits with some earlier conversations as =
well.

Colin and Paul (and David):

Are we converging on what guidance to put into the DART draft?

Thanks!

Ben.



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From: "Paul E. Jones" <paulej@packetizer.com>
To: "Colin Perkins" <csp@csperkins.org>
Date: Wed, 27 Aug 2014 21:41:53 +0000
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Subject: Re: [AVTCORE] Treatment of RTCP (was Re: [Dart] Colin Perkins comments - WGLC: draft-ietf-dart-dscp-rtp-02)
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Colin

>>>>  I agree that the value should be one of the values used for media,=
=20
>>>>but the challenge is specifying which DSCP value to use. If audio is=
=20
>>>>sent using EF, video as AF41, app sharing video as AF31, etc., which=
=20
>>>>of these does the endpoint choose to use for RTCP?
>>>
>>>  Such an endpoint will have three SSRC values, and will mark the RTCP=
=20
>>>packets sent by each SSRC in the same way as the media packets sent=20
>>>by that SSRC. That is, the RTCP relating to the audio will be marked=20
>>>EF, the RTCP relating to the video will be marked AF41, and the RTCP=20
>>>relating to the application sharing video will be marked AF31. The=20
>>>RTCP packets for each stream are sent separately, so this case seems=20
>>>straight-forward.
>>
>>  I'm a little confused here. These are separate SSRCs, but all part of=
=20
>>a single RTP session. So, wouldn=E2=80=99t the three SSRCs be reported=
 in=20
>>different report blocks in the sender report?
>
>Each of the SSRCs will report on all the others, unless the=20
>rtp-multi-stream-optimisation draft is used, but so what?

Oh, I think I understand what you are saying.  You're saying that each=20
media flow (SSRC) would be responsible for sending its own RTCP packets=20
and those packets would be marked in the same way as that media flow. =20
In that case, I agree with you.

That said, would we really require the sending SSRC to include any data=20
for the other SSRCs if they're each sending RTCP packets separately? =20
That seems redundant.  Does RFC 3550 require that?  (It's been a=20
while...)

>>>  The more difficult case is when an SSRC is sending video using=20
>>>different markings for RTP packets carrying the I- and P-frames.=20
>>>Should that SSRC then mark its RTCP packets like the RTP packets=20
>>>carrying I-frames, like the RTP packets carrying P-frames, or what?
>>
>>  For this, I would argue it should be whatever DSCP that has the lower=
=20
>>drop precedence (in the case of assured forwarding). Otherwise, it=20
>>should be the same as the media, I'd think.
>
>You mean the RTCP for each SSRC should use the DSCP with the lowest=20
>drop preference of those used for the media sent by that SSRC?

That's what I'm suggesting. So, a video flow using AF4x would use AF41=20
for RTCP.  I suggest that, since I assume we don't want to lose feedback=
=20
information if we can avoid it.

Paul


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From: "Paul E. Jones" <paulej@packetizer.com>
To: "Ben Campbell" <ben@nostrum.com>, "Michael Welzl" <michawe@ifi.uio.no>, "Colin Perkins" <csp@csperkins.org>, "Black, David" <david.black@emc.com>
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Subject: Re: [AVTCORE] [Dart] Treatment of RTCP (was Re: Colin Perkins comments - WGLC: draft-ietf-dart-dscp-rtp-02)
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RTCP might not be used for RTT calculations (it might; I just doubt it's=
=20
terribly useful),  but RTCP might be used to convey QoE measurements or=20
one-way delay information useful for RMCAT.

I think there is agreement that the DSCP value applied to the RTCP=20
packet for a given SSRC sender should align with the DSCP value used for=
=20
the corresponding RTP packets of that sender. In the case where an AFxx=20
class is used, I'm not sure there is agreement (though no disagreement).=
=20
  I stated my view that we should use the lowest drop precedence in that=
=20
case, but others should weigh in.

Paul

------ Original Message ------
From: "Ben Campbell" <ben@nostrum.com>
To: "Michael Welzl" <michawe@ifi.uio.no>; "Colin Perkins"=20
<csp@csperkins.org>; "Paul E. Jones" <paulej@packetizer.com>; "Black,=20
David" <david.black@emc.com>
Cc: "draft-ietf-dart-dscp-rtp.all@tools.ietf.org"=20
<draft-ietf-dart-dscp-rtp.all@tools.ietf.org>; "dart@ietf.org"=20
<dart@ietf.org>; "avt@ietf.org WG" <avt@ietf.org>
Sent: 8/27/2014 4:48:37 PM
Subject: Re: [Dart] [AVTCORE] Treatment of RTCP (was Re: Colin Perkins=20
comments - WGLC: draft-ietf-dart-dscp-rtp-02)

>On Aug 27, 2014, at 3:45 PM, Michael Welzl <michawe@ifi.uio.no> wrote:
>
>>  I don't know the beginning of this but I can help go up the stack:
>>
>>  for the short dialogue between Colin and me, the conclusion is: Colin=
=20
>>is right. RTCP probably won't be used by congestion control to=20
>>estimate the RTT.
>>
>
>Thanks, Michael, I think that fits with some earlier conversations as=20
>well.
>
>Colin and Paul (and David):
>
>Are we converging on what guidance to put into the DART draft?
>
>Thanks!
>
>Ben.
>
>
>_______________________________________________
>Dart mailing list
>Dart@ietf.org
>https://www.ietf.org/mailman/listinfo/dart


From nobody Wed Aug 27 18:59:15 2014
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From: "Black, David" <david.black@emc.com>
To: "Paul E. Jones" <paulej@packetizer.com>, Ben Campbell <ben@nostrum.com>, Michael Welzl <michawe@ifi.uio.no>, Colin Perkins <csp@csperkins.org>
Date: Wed, 27 Aug 2014 21:58:34 -0400
Thread-Topic: Treatment of RTCP - proposed text
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Subject: [AVTCORE] Treatment of RTCP - proposed text
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From: Ben Campbell <ben@nostrum.com>
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To: "Black, David" <david.black@emc.com>
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Cc: Michael Welzl <michawe@ifi.uio.no>, "draft-ietf-dart-dscp-rtp.all@tools.ietf.org" <draft-ietf-dart-dscp-rtp.all@tools.ietf.org>, "dart@ietf.org" <dart@ietf.org>, "avt@ietf.org WG" <avt@ietf.org>, Colin Perkins <csp@csperkins.org>
Subject: Re: [AVTCORE] Treatment of RTCP - proposed text
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I do not object to the text per se.=20

But I do have a bit of a nagging concern that we may be offering too =
strong of guidance, given the thought process so far.  Do we believe we =
understand this well enough to say the sender should to this or that, as =
opposed to say that these are things the implementer should think about, =
and that these are probably reasonable choices?=20

Is this an area where we need more deployment experience to offer real =
guidance? (I think it's okay for the doc to say that, if needed.)

But in any case, I will defer to Colin, Paul, and David.

On Aug 27, 2014, at 8:58 PM, Black, David <david.black@emc.com> wrote:

> Trying to propose some text here:
>=20
> OLD
>   o  Should use a single DSCP for an RTCP session, primarily to avoid
>      RTCP reordering (and because there is no compelling reason for =
use
>      of different drop precedences).  One of the PHBs and associated
>      DSCP used for the associated RTP traffic would be an appropriate
>      choice.  [Editor's note: This bullet is an open technical issue.]
> NEW
>   o  Should use the same DSCP for RTCP reports as used for the RTP =
stream
>      that is being reported on when that DSCP is known by the RTCP =
sender.
>      When an RTP stream uses multiple DSCPs that differ only in drop
>      precedence, RTCP reports on that RTP stream should use the DSCP =
with
>      the least likelihood of drop to favor delivery of the RTCP =
reports.
>      When a single RTCP message reports on multiple RTP streams that
>      are sent with different DSCPs, the RTCP sender should choose one
>      of those DSCPs.  When the RTCP sender does not know what
> 	DSCP or DSCPs were used to send an RTP stream, it should choose
> 	a DSCP that it would use to send a similar RTP stream.
>=20
> The latter sentence is courtesy of the fact that there will be cases
> where the RTCP sender will not know what DSCP or DSCPs were used to =
send
> the RTP stream, due to en-route remarking.
>=20
> If that text looks close, I'll edit further and try to get the -05 =
submitted
> by the end of this week.
>=20
> Thanks,
> --David
>=20
>> -----Original Message-----
>> From: Paul E. Jones [mailto:paulej@packetizer.com]
>> Sent: Wednesday, August 27, 2014 6:53 PM
>> To: Ben Campbell; Michael Welzl; Colin Perkins; Black, David
>> Cc: draft-ietf-dart-dscp-rtp.all@tools.ietf.org; dart@ietf.org; =
avt@ietf.org
>> WG
>> Subject: Re: [Dart] [AVTCORE] Treatment of RTCP (was Re: Colin =
Perkins
>> comments - WGLC: draft-ietf-dart-dscp-rtp-02)
>>=20
>> RTCP might not be used for RTT calculations (it might; I just doubt =
it's
>> terribly useful),  but RTCP might be used to convey QoE measurements =
or
>> one-way delay information useful for RMCAT.
>>=20
>> I think there is agreement that the DSCP value applied to the RTCP
>> packet for a given SSRC sender should align with the DSCP value used =
for
>> the corresponding RTP packets of that sender. In the case where an =
AFxx
>> class is used, I'm not sure there is agreement (though no =
disagreement).
>>  I stated my view that we should use the lowest drop precedence in =
that
>> case, but others should weigh in.
>>=20
>> Paul
>>=20
>> ------ Original Message ------
>> From: "Ben Campbell" <ben@nostrum.com>
>> To: "Michael Welzl" <michawe@ifi.uio.no>; "Colin Perkins"
>> <csp@csperkins.org>; "Paul E. Jones" <paulej@packetizer.com>; "Black,
>> David" <david.black@emc.com>
>> Cc: "draft-ietf-dart-dscp-rtp.all@tools.ietf.org"
>> <draft-ietf-dart-dscp-rtp.all@tools.ietf.org>; "dart@ietf.org"
>> <dart@ietf.org>; "avt@ietf.org WG" <avt@ietf.org>
>> Sent: 8/27/2014 4:48:37 PM
>> Subject: Re: [Dart] [AVTCORE] Treatment of RTCP (was Re: Colin =
Perkins
>> comments - WGLC: draft-ietf-dart-dscp-rtp-02)
>>=20
>>> On Aug 27, 2014, at 3:45 PM, Michael Welzl <michawe@ifi.uio.no> =
wrote:
>>>=20
>>>> I don't know the beginning of this but I can help go up the stack:
>>>>=20
>>>> for the short dialogue between Colin and me, the conclusion is: =
Colin
>>>> is right. RTCP probably won't be used by congestion control to
>>>> estimate the RTT.
>>>>=20
>>>=20
>>> Thanks, Michael, I think that fits with some earlier conversations =
as
>>> well.
>>>=20
>>> Colin and Paul (and David):
>>>=20
>>> Are we converging on what guidance to put into the DART draft?
>>>=20
>>> Thanks!
>>>=20
>>> Ben.
>>>=20
>>>=20
>>> _______________________________________________
>>> Dart mailing list
>>> Dart@ietf.org
>>> https://www.ietf.org/mailman/listinfo/dart
>=20


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From: "Paul E. Jones" <paulej@packetizer.com>
To: "Black, David" <david.black@emc.com>, "Ben Campbell" <ben@nostrum.com>, "Michael Welzl" <michawe@ifi.uio.no>, "Colin Perkins" <csp@csperkins.org>
Date: Thu, 28 Aug 2014 03:13:53 +0000
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Subject: Re: [AVTCORE] Treatment of RTCP - proposed text
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David,

Should it be the DSCP value of the SSRC being reported on or the DSCP of=
=20
the SSRC sender?  My thought that was if Alice and Bob are sending flows=
=20
to each other and Bob is sending audio using EF, then when Bob sends out=
=20
a Sender Report, the packet would be marked EF.  The way this is worded,=
=20
if Alice is sending packets using CS0, then Bob would mark his SR's=20
using CS0.

I'm not sure how to handle the RR's, though, particularly of a=20
receive-only device.  We could say it would be the same as the flow=20
being reported on, but there might be 31 such flows being reported on. =20
Perhaps if there is agreement that the RTCP packets will be marked in=20
the same way as the sender would send its media, then we say the reports=
=20
should be marked in the same was as it would if it did send media.

So, do we want RTCP packets to be marked the same as the media flow=20
transmitted or the same as the received media flow?

Paul

------ Original Message ------
From: "Black, David" <david.black@emc.com>
To: "Paul E. Jones" <paulej@packetizer.com>; "Ben Campbell"=20
<ben@nostrum.com>; "Michael Welzl" <michawe@ifi.uio.no>; "Colin Perkins"=
=20
<csp@csperkins.org>
Cc: "draft-ietf-dart-dscp-rtp.all@tools.ietf.org"=20
<draft-ietf-dart-dscp-rtp.all@tools.ietf.org>; "dart@ietf.org"=20
<dart@ietf.org>; "avt@ietf.org WG" <avt@ietf.org>; "Black, David"=20
<david.black@emc.com>
Sent: 8/27/2014 9:58:34 PM
Subject: Treatment of RTCP - proposed text

>Trying to propose some text here:
>
>OLD
>    o Should use a single DSCP for an RTCP session, primarily to avoid
>       RTCP reordering (and because there is no compelling reason for=20
>use
>       of different drop precedences). One of the PHBs and associated
>       DSCP used for the associated RTP traffic would be an appropriate
>       choice. [Editor's note: This bullet is an open technical issue.]
>NEW
>    o Should use the same DSCP for RTCP reports as used for the RTP=20
>stream
>       that is being reported on when that DSCP is known by the RTCP=20
>sender.
>       When an RTP stream uses multiple DSCPs that differ only in drop
>       precedence, RTCP reports on that RTP stream should use the DSCP=20
>with
>       the least likelihood of drop to favor delivery of the RTCP=20
>reports.
>       When a single RTCP message reports on multiple RTP streams that
>       are sent with different DSCPs, the RTCP sender should choose one
>       of those DSCPs. When the RTCP sender does not know what
>  DSCP or DSCPs were used to send an RTP stream, it should choose
>  a DSCP that it would use to send a similar RTP stream.
>
>The latter sentence is courtesy of the fact that there will be cases
>where the RTCP sender will not know what DSCP or DSCPs were used to=20
>send
>the RTP stream, due to en-route remarking.
>
>If that text looks close, I'll edit further and try to get the -05=20
>submitted
>by the end of this week.
>
>Thanks,
>--David
>
>>  -----Original Message-----
>>  From: Paul E. Jones [mailto:paulej@packetizer.com]
>>  Sent: Wednesday, August 27, 2014 6:53 PM
>>  To: Ben Campbell; Michael Welzl; Colin Perkins; Black, David
>>  Cc: draft-ietf-dart-dscp-rtp.all@tools.ietf.org; dart@ietf.org;=20
>>avt@ietf.org
>>  WG
>>  Subject: Re: [Dart] [AVTCORE] Treatment of RTCP (was Re: Colin=20
>>Perkins
>>  comments - WGLC: draft-ietf-dart-dscp-rtp-02)
>>
>>  RTCP might not be used for RTT calculations (it might; I just doubt=20
>>it's
>>  terribly useful), but RTCP might be used to convey QoE measurements=20
>>or
>>  one-way delay information useful for RMCAT.
>>
>>  I think there is agreement that the DSCP value applied to the RTCP
>>  packet for a given SSRC sender should align with the DSCP value used=
=20
>>for
>>  the corresponding RTP packets of that sender. In the case where an=20
>>AFxx
>>  class is used, I'm not sure there is agreement (though no=20
>>disagreement).
>>    I stated my view that we should use the lowest drop precedence in=20
>>that
>>  case, but others should weigh in.
>>
>>  Paul
>>
>>  ------ Original Message ------
>>  From: "Ben Campbell" <ben@nostrum.com>
>>  To: "Michael Welzl" <michawe@ifi.uio.no>; "Colin Perkins"
>>  <csp@csperkins.org>; "Paul E. Jones" <paulej@packetizer.com>; "Black,
>>  David" <david.black@emc.com>
>>  Cc: "draft-ietf-dart-dscp-rtp.all@tools.ietf.org"
>>  <draft-ietf-dart-dscp-rtp.all@tools.ietf.org>; "dart@ietf.org"
>>  <dart@ietf.org>; "avt@ietf.org WG" <avt@ietf.org>
>>  Sent: 8/27/2014 4:48:37 PM
>>  Subject: Re: [Dart] [AVTCORE] Treatment of RTCP (was Re: Colin=20
>>Perkins
>>  comments - WGLC: draft-ietf-dart-dscp-rtp-02)
>>
>>  >On Aug 27, 2014, at 3:45 PM, Michael Welzl <michawe@ifi.uio.no>=20
>>wrote:
>>  >
>>  >> I don't know the beginning of this but I can help go up the stack:
>>  >>
>>  >> for the short dialogue between Colin and me, the conclusion is:=20
>>Colin
>>  >>is right. RTCP probably won't be used by congestion control to
>>  >>estimate the RTT.
>>  >>
>>  >
>>  >Thanks, Michael, I think that fits with some earlier conversations=20
>>as
>>  >well.
>>  >
>>  >Colin and Paul (and David):
>>  >
>>  >Are we converging on what guidance to put into the DART draft?
>>  >
>>  >Thanks!
>>  >
>>  >Ben.
>>  >
>>  >
>>  >_______________________________________________
>>  >Dart mailing list
>>  >Dart@ietf.org
>>  >https://www.ietf.org/mailman/listinfo/dart
>


From nobody Wed Aug 27 20:20:54 2014
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From: "Paul E. Jones" <paulej@packetizer.com>
To: "Ben Campbell" <ben@nostrum.com>, "Black, David" <david.black@emc.com>
Date: Thu, 28 Aug 2014 03:21:13 +0000
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Cc: "draft-ietf-dart-dscp-rtp.all@tools.ietf.org" <draft-ietf-dart-dscp-rtp.all@tools.ietf.org>, "dart@ietf.org" <dart@ietf.org>, Colin Perkins <csp@csperkins.org>, Michael Welzl <michawe@ifi.uio.no>, "avt@ietf.org WG" <avt@ietf.org>
Subject: Re: [AVTCORE] Treatment of RTCP - proposed text
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Ben,

I think the answer is really whether we can quickly reach consensus.  I=20
have an idea how I'd implement it if I didn't have guidance otherwise,=20
which I explained in my last email.

I think this can be resolved pretty quickly.  If not, we can certainly=20
leave it open.

Paul

------ Original Message ------
From: "Ben Campbell" <ben@nostrum.com>
To: "Black, David" <david.black@emc.com>
Cc: "Paul E. Jones" <paulej@packetizer.com>; "Michael Welzl"=20
<michawe@ifi.uio.no>; "Colin Perkins" <csp@csperkins.org>;=20
"draft-ietf-dart-dscp-rtp.all@tools.ietf.org"=20
<draft-ietf-dart-dscp-rtp.all@tools.ietf.org>; "dart@ietf.org"=20
<dart@ietf.org>; "avt@ietf.org WG" <avt@ietf.org>
Sent: 8/27/2014 10:28:07 PM
Subject: Re: Treatment of RTCP - proposed text

>I do not object to the text per se.
>
>But I do have a bit of a nagging concern that we may be offering too=20
>strong of guidance, given the thought process so far. Do we believe we=20
>understand this well enough to say the sender should to this or that,=20
>as opposed to say that these are things the implementer should think=20
>about, and that these are probably reasonable choices?
>
>Is this an area where we need more deployment experience to offer real=20
>guidance? (I think it's okay for the doc to say that, if needed.)
>
>But in any case, I will defer to Colin, Paul, and David.
>
>On Aug 27, 2014, at 8:58 PM, Black, David <david.black@emc.com> wrote:
>
>>  Trying to propose some text here:
>>
>>  OLD
>>    o Should use a single DSCP for an RTCP session, primarily to avoid
>>       RTCP reordering (and because there is no compelling reason for=20
>>use
>>       of different drop precedences). One of the PHBs and associated
>>       DSCP used for the associated RTP traffic would be an appropriate
>>       choice. [Editor's note: This bullet is an open technical issue.]
>>  NEW
>>    o Should use the same DSCP for RTCP reports as used for the RTP=20
>>stream
>>       that is being reported on when that DSCP is known by the RTCP=20
>>sender.
>>       When an RTP stream uses multiple DSCPs that differ only in drop
>>       precedence, RTCP reports on that RTP stream should use the DSCP=
=20
>>with
>>       the least likelihood of drop to favor delivery of the RTCP=20
>>reports.
>>       When a single RTCP message reports on multiple RTP streams that
>>       are sent with different DSCPs, the RTCP sender should choose one
>>       of those DSCPs. When the RTCP sender does not know what
>>   DSCP or DSCPs were used to send an RTP stream, it should choose
>>   a DSCP that it would use to send a similar RTP stream.
>>
>>  The latter sentence is courtesy of the fact that there will be cases
>>  where the RTCP sender will not know what DSCP or DSCPs were used to=20
>>send
>>  the RTP stream, due to en-route remarking.
>>
>>  If that text looks close, I'll edit further and try to get the -05=20
>>submitted
>>  by the end of this week.
>>
>>  Thanks,
>>  --David
>>
>>>  -----Original Message-----
>>>  From: Paul E. Jones [mailto:paulej@packetizer.com]
>>>  Sent: Wednesday, August 27, 2014 6:53 PM
>>>  To: Ben Campbell; Michael Welzl; Colin Perkins; Black, David
>>>  Cc: draft-ietf-dart-dscp-rtp.all@tools.ietf.org; dart@ietf.org;=20
>>>avt@ietf.org
>>>  WG
>>>  Subject: Re: [Dart] [AVTCORE] Treatment of RTCP (was Re: Colin=20
>>>Perkins
>>>  comments - WGLC: draft-ietf-dart-dscp-rtp-02)
>>>
>>>  RTCP might not be used for RTT calculations (it might; I just doubt=
=20
>>>it's
>>>  terribly useful), but RTCP might be used to convey QoE measurements=
=20
>>>or
>>>  one-way delay information useful for RMCAT.
>>>
>>>  I think there is agreement that the DSCP value applied to the RTCP
>>>  packet for a given SSRC sender should align with the DSCP value used=
=20
>>>for
>>>  the corresponding RTP packets of that sender. In the case where an=20
>>>AFxx
>>>  class is used, I'm not sure there is agreement (though no=20
>>>disagreement).
>>>   I stated my view that we should use the lowest drop precedence in=20
>>>that
>>>  case, but others should weigh in.
>>>
>>>  Paul
>>>
>>>  ------ Original Message ------
>>>  From: "Ben Campbell" <ben@nostrum.com>
>>>  To: "Michael Welzl" <michawe@ifi.uio.no>; "Colin Perkins"
>>>  <csp@csperkins.org>; "Paul E. Jones" <paulej@packetizer.com>;=20
>>>"Black,
>>>  David" <david.black@emc.com>
>>>  Cc: "draft-ietf-dart-dscp-rtp.all@tools.ietf.org"
>>>  <draft-ietf-dart-dscp-rtp.all@tools.ietf.org>; "dart@ietf.org"
>>>  <dart@ietf.org>; "avt@ietf.org WG" <avt@ietf.org>
>>>  Sent: 8/27/2014 4:48:37 PM
>>>  Subject: Re: [Dart] [AVTCORE] Treatment of RTCP (was Re: Colin=20
>>>Perkins
>>>  comments - WGLC: draft-ietf-dart-dscp-rtp-02)
>>>
>>>>  On Aug 27, 2014, at 3:45 PM, Michael Welzl <michawe@ifi.uio.no>=20
>>>>wrote:
>>>>
>>>>>  I don't know the beginning of this but I can help go up the stack:
>>>>>
>>>>>  for the short dialogue between Colin and me, the conclusion is:=20
>>>>>Colin
>>>>>  is right. RTCP probably won't be used by congestion control to
>>>>>  estimate the RTT.
>>>>>
>>>>
>>>>  Thanks, Michael, I think that fits with some earlier conversations=
=20
>>>>as
>>>>  well.
>>>>
>>>>  Colin and Paul (and David):
>>>>
>>>>  Are we converging on what guidance to put into the DART draft?
>>>>
>>>>  Thanks!
>>>>
>>>>  Ben.
>>>>
>>>>
>>>>  _______________________________________________
>>>>  Dart mailing list
>>>>  Dart@ietf.org
>>>>  https://www.ietf.org/mailman/listinfo/dart
>>
>


From nobody Wed Aug 27 21:48:16 2014
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From: "Black, David" <david.black@emc.com>
To: "Paul E. Jones" <paulej@packetizer.com>, Ben Campbell <ben@nostrum.com>, Michael Welzl <michawe@ifi.uio.no>, Colin Perkins <csp@csperkins.org>
Date: Thu, 28 Aug 2014 00:47:37 -0400
Thread-Topic: Treatment of RTCP - proposed text
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Cc: "draft-ietf-dart-dscp-rtp.all@tools.ietf.org" <draft-ietf-dart-dscp-rtp.all@tools.ietf.org>, "Black, David" <david.black@emc.com>, "dart@ietf.org" <dart@ietf.org>, "avt@ietf.org WG" <avt@ietf.org>
Subject: Re: [AVTCORE] Treatment of RTCP - proposed text
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Subject: Re: [AVTCORE] Treatment of RTCP - proposed text
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Paul, David,

On 28 Aug 2014, at 04:13, Paul E. Jones <paulej@packetizer.com> wrote:
> Should it be the DSCP value of the SSRC being reported on or the DSCP =
of the SSRC sender?  My thought that was if Alice and Bob are sending =
flows to each other and Bob is sending audio using EF, then when Bob =
sends out a Sender Report, the packet would be marked EF. =20

Agree. An SSRC should use the same marking it would use to send packets, =
and not the marking of the received packets.

> The way this is worded, if Alice is sending packets using CS0, then =
Bob would mark his SR's using CS0.
>=20
> I'm not sure how to handle the RR's, though, particularly of a =
receive-only device.  We could say it would be the same as the flow =
being reported on, but there might be 31 such flows being reported on.  =
Perhaps if there is agreement that the RTCP packets will be marked in =
the same way as the sender would send its media, then we say the reports =
should be marked in the same was as it would if it did send media.

A receive-only device should mark in the way it would if it sent media, =
I agree.

> So, do we want RTCP packets to be marked the same as the media flow =
transmitted or the same as the received media flow?

Transmitted, I think.

Colin



> Paul
>=20
> ------ Original Message ------
> From: "Black, David" <david.black@emc.com>
> To: "Paul E. Jones" <paulej@packetizer.com>; "Ben Campbell" =
<ben@nostrum.com>; "Michael Welzl" <michawe@ifi.uio.no>; "Colin Perkins" =
<csp@csperkins.org>
> Cc: "draft-ietf-dart-dscp-rtp.all@tools.ietf.org" =
<draft-ietf-dart-dscp-rtp.all@tools.ietf.org>; "dart@ietf.org" =
<dart@ietf.org>; "avt@ietf.org WG" <avt@ietf.org>; "Black, David" =
<david.black@emc.com>
> Sent: 8/27/2014 9:58:34 PM
> Subject: Treatment of RTCP - proposed text
>=20
>> Trying to propose some text here:
>>=20
>> OLD
>>   o Should use a single DSCP for an RTCP session, primarily to avoid
>>      RTCP reordering (and because there is no compelling reason for =
use
>>      of different drop precedences). One of the PHBs and associated
>>      DSCP used for the associated RTP traffic would be an appropriate
>>      choice. [Editor's note: This bullet is an open technical issue.]
>> NEW
>>   o Should use the same DSCP for RTCP reports as used for the RTP =
stream
>>      that is being reported on when that DSCP is known by the RTCP =
sender.
>>      When an RTP stream uses multiple DSCPs that differ only in drop
>>      precedence, RTCP reports on that RTP stream should use the DSCP =
with
>>      the least likelihood of drop to favor delivery of the RTCP =
reports.
>>      When a single RTCP message reports on multiple RTP streams that
>>      are sent with different DSCPs, the RTCP sender should choose one
>>      of those DSCPs. When the RTCP sender does not know what
>> DSCP or DSCPs were used to send an RTP stream, it should choose
>> a DSCP that it would use to send a similar RTP stream.
>>=20
>> The latter sentence is courtesy of the fact that there will be cases
>> where the RTCP sender will not know what DSCP or DSCPs were used to =
send
>> the RTP stream, due to en-route remarking.
>>=20
>> If that text looks close, I'll edit further and try to get the -05 =
submitted
>> by the end of this week.
>>=20
>> Thanks,
>> --David
>>=20
>>> -----Original Message-----
>>> From: Paul E. Jones [mailto:paulej@packetizer.com]
>>> Sent: Wednesday, August 27, 2014 6:53 PM
>>> To: Ben Campbell; Michael Welzl; Colin Perkins; Black, David
>>> Cc: draft-ietf-dart-dscp-rtp.all@tools.ietf.org; dart@ietf.org; =
avt@ietf.org
>>> WG
>>> Subject: Re: [Dart] [AVTCORE] Treatment of RTCP (was Re: Colin =
Perkins
>>> comments - WGLC: draft-ietf-dart-dscp-rtp-02)
>>>=20
>>> RTCP might not be used for RTT calculations (it might; I just doubt =
it's
>>> terribly useful), but RTCP might be used to convey QoE measurements =
or
>>> one-way delay information useful for RMCAT.
>>>=20
>>> I think there is agreement that the DSCP value applied to the RTCP
>>> packet for a given SSRC sender should align with the DSCP value used =
for
>>> the corresponding RTP packets of that sender. In the case where an =
AFxx
>>> class is used, I'm not sure there is agreement (though no =
disagreement).
>>>   I stated my view that we should use the lowest drop precedence in =
that
>>> case, but others should weigh in.
>>>=20
>>> Paul
>>>=20
>>> ------ Original Message ------
>>> From: "Ben Campbell" <ben@nostrum.com>
>>> To: "Michael Welzl" <michawe@ifi.uio.no>; "Colin Perkins"
>>> <csp@csperkins.org>; "Paul E. Jones" <paulej@packetizer.com>; =
"Black,
>>> David" <david.black@emc.com>
>>> Cc: "draft-ietf-dart-dscp-rtp.all@tools.ietf.org"
>>> <draft-ietf-dart-dscp-rtp.all@tools.ietf.org>; "dart@ietf.org"
>>> <dart@ietf.org>; "avt@ietf.org WG" <avt@ietf.org>
>>> Sent: 8/27/2014 4:48:37 PM
>>> Subject: Re: [Dart] [AVTCORE] Treatment of RTCP (was Re: Colin =
Perkins
>>> comments - WGLC: draft-ietf-dart-dscp-rtp-02)
>>>=20
>>> >On Aug 27, 2014, at 3:45 PM, Michael Welzl <michawe@ifi.uio.no> =
wrote:
>>> >
>>> >> I don't know the beginning of this but I can help go up the =
stack:
>>> >>
>>> >> for the short dialogue between Colin and me, the conclusion is: =
Colin
>>> >>is right. RTCP probably won't be used by congestion control to
>>> >>estimate the RTT.
>>> >>
>>> >
>>> >Thanks, Michael, I think that fits with some earlier conversations =
as
>>> >well.
>>> >
>>> >Colin and Paul (and David):
>>> >
>>> >Are we converging on what guidance to put into the DART draft?
>>> >
>>> >Thanks!
>>> >
>>> >Ben.
>>> >
>>> >
>>> >_______________________________________________
>>> >Dart mailing list
>>> >Dart@ietf.org
>>> >https://www.ietf.org/mailman/listinfo/dart
>>=20
>=20



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




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Subject: Re: [AVTCORE] [Dart] draft-ietf-dart-dscp-rtp-02 - Section 5.3 RTCP multi-stream optimization text
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David,

On 26 Aug 2014, at 17:27, Black, David <david.black@emc.com> wrote:
> <editor hat OFF>
>=20
> Colin writes:
>=20
>   The discussion of the multi-stream-optimisation draft is a separate
>   issue, and is correct as it is.
>=20
> Looking closer, that discussion in the current DART draft currently =
says:
>=20
>   RTCP multi-stream reporting optimizations for an RTP session
>   [I-D.ietf-avtcore-rtp-multi-stream-optimisation] assume that the RTP
>   streams involved experience the same packet loss behavior.  This
>   mechanism is highly inappropriate when the RTP streams involved use
>   different PHBs, even if those PHBs differ solely in drop precedence.

A quick follow-up to ensure this is clear: the multi-stream-optimisation =
cannot be used when reporting on received RTP streams where the =
different streams have different markings, since those streams are =
treated differently in the network.=20

The marking I use for the RTP streams I send (and hence the marking I =
use for my RTCP packets) doesn=92t influence whether I can use the =
multi-stream-optimisation, however.

Colin




> That's based on this text from the last paragraph of Section 1
> (Introduction) of the multi-stream optimization draft:
>=20
>   This memo defines such an RTCP extension, RTCP Reporting Groups.
>   This extension is used to indicate the SSRCs that originate from the
>   same endpoint, and therefore have identical reception quality, hence
>   allowing the endpoints to suppress unnecessary RTCP reception =
quality
>   reports.
>=20
> When different DSCPs are used for SSRCs (RTP streams) that originate
> from the same source, the statement that they "therefore have =
identical
> reception quality" is clearly incorrect, so it looks like the current
> DART draft text is correct, IMHO.
>=20
> Harald?
>=20
> </editor hat OFF>
>=20
> Thanks,
> --David
>=20
>> -----Original Message-----
>> From: Colin Perkins [mailto:csp@csperkins.org]
>> Sent: Tuesday, August 26, 2014 11:54 AM
>> To: Black, David
>> Cc: Ben Campbell; Paul E. Jones; dart@ietf.org; avt@ietf.org WG; =
draft-ietf-
>> dart-dscp-rtp.all@tools.ietf.org
>> Subject: Re: Treatment of RTCP (was Re: [Dart] Colin Perkins comments =
- WGLC:
>> draft-ietf-dart-dscp-rtp-02)
>>=20
>>=20
>> On 26 Aug 2014, at 16:39, Black, David <david.black@emc.com> wrote:
>>=20
>>>>> Since not all the media sent by a single SSRC has the same =
marking, my
>>>> suggestion would be that each SSRC mark the RTCP packets it sends =
with one
>> of
>>>> the same code points as it uses to mark the media. Since RTCP is =
somewhat
>>>> important, it would make sense for each SSRC to mark the RTCP =
packets it
>> sends
>>>> using the highest priority code point it uses to mark the RTP media =
packets
>> it
>>>> sends.
>>>>=20
>>>> That makes sense to me. Paul, and others, do you agree with that =
last
>>>> paragraph?
>>>=20
>>> <editor hat off>
>>>=20
>>> Sure, when there's a notion of priority or importance.  There won't =
always
>>> be one (e.g., is CS2 higher priority than AF2x?  That depends ...).  =
OTOH,
>>> we don't need to say much here, e.g., the class selector codepoints =
(CSx)
>>> are ordered, and adding a pointer to draft-ietf-tsvwg-rtcweb-qos =
will be
>>> helpful.
>>>=20
>>> </editor hat off>
>>>=20
>>> Turning to the other RTCP issue, [F] on multi-stream optimization, I =
wonder
>>> whether we inadvertently framed that issue backwards.  Given the =
above, a
>>> single RTCP report on multiple RTP streams that use rather different =
DSCPs
>>> may not result in representative RTT values for all of the streams, =
because
>>> the report has to be sent with one DSCP.  If that outcome is a =
problem,
>>> one should send separate RTCP reports (duh!).
>>=20
>> Each reporting SSRC sends a separate RTCP report block for each SSRC =
it
>> receives, so this is not a problem. The question is what marking an =
SSRC uses
>> for the RTCP packets it sends, given that it uses several different =
markings
>> for the RTP media packets it sends.
>>=20
>>> That statement seems valuable to make and can be made with a =
citation of
>>> RFC 3530, as opposed to the multi-stream optimization draft.
>>>=20
>>> Harald and Colin - what do you think?
>>=20
>> The discussion of the multi-stream-optimisation draft is a separate =
issue, and
>> is correct as it is.
>>=20
>> --
>> Colin Perkins
>> http://csperkins.org/
>>=20
>>=20
>=20



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




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Subject: Re: [AVTCORE] Treatment of RTCP (was Re: [Dart] Colin Perkins comments - WGLC: draft-ietf-dart-dscp-rtp-02)
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Paul,

On 27 Aug 2014, at 22:41, Paul E. Jones <paulej@packetizer.com> wrote:
>>>>> I agree that the value should be one of the values used for media, =
but the challenge is specifying which DSCP value to use. If audio is =
sent using EF, video as AF41, app sharing video as AF31, etc., which of =
these does the endpoint choose to use for RTCP?
>>>>=20
>>>> Such an endpoint will have three SSRC values, and will mark the =
RTCP packets sent by each SSRC in the same way as the media packets sent =
by that SSRC. That is, the RTCP relating to the audio will be marked EF, =
the RTCP relating to the video will be marked AF41, and the RTCP =
relating to the application sharing video will be marked AF31. The RTCP =
packets for each stream are sent separately, so this case seems =
straight-forward.
>>>=20
>>> I'm a little confused here. These are separate SSRCs, but all part =
of a single RTP session. So, wouldn=92t the three SSRCs be reported in =
different report blocks in the sender report?
>>=20
>> Each of the SSRCs will report on all the others, unless the =
rtp-multi-stream-optimisation draft is used, but so what?
>=20
> Oh, I think I understand what you are saying.  You're saying that each =
media flow (SSRC) would be responsible for sending its own RTCP packets =
and those packets would be marked in the same way as that media flow.  =
In that case, I agree with you.

Yes.

> That said, would we really require the sending SSRC to include any =
data for the other SSRCs if they're each sending RTCP packets =
separately?  That seems redundant.  Does RFC 3550 require that?  (It=92s =
been a while...)

RFC 3550 does require that, and it is redundant (except that it =
indicates that the SSRCs are co-located); =
draft-ietf-avtcore-rtp-multi-stream-optimisation-04 offers an extension =
to avoid it in some cases.

>>>> The more difficult case is when an SSRC is sending video using =
different markings for RTP packets carrying the I- and P-frames. Should =
that SSRC then mark its RTCP packets like the RTP packets carrying =
I-frames, like the RTP packets carrying P-frames, or what?
>>>=20
>>> For this, I would argue it should be whatever DSCP that has the =
lower drop precedence (in the case of assured forwarding). Otherwise, it =
should be the same as the media, I'd think.
>>=20
>> You mean the RTCP for each SSRC should use the DSCP with the lowest =
drop preference of those used for the media sent by that SSRC?
>=20
> That's what I'm suggesting. So, a video flow using AF4x would use AF41 =
for RTCP.  I suggest that, since I assume we don't want to lose feedback =
information if we can avoid it.

Makes sense.

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




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Subject: Re: [AVTCORE] [Dart] Treatment of RTCP (was Re: Colin Perkins comments - WGLC: draft-ietf-dart-dscp-rtp-02)
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On 27. aug. 2014, at 18:00, Colin Perkins <csp@csperkins.org> wrote:

> On 26 Aug 2014, at 17:38, Black, David <david.black@emc.com> wrote:
>>> The more difficult case is when an SSRC is sending video using =
different
>>> markings for RTP packets carrying the I- and P-frames. Should that =
SSRC then
>>> mark its RTCP packets like the RTP packets carrying I-frames, like =
the RTP
>>> packets carrying P-frames, or what?
>>=20
>> Answering a question w/a question :-), how are those reports likely =
to be used?
>>=20
>> For example, if the primary use of these reports is to adjust a =
variable rate
>> codec's sending rate, the P-frame info is probably more useful as =
indicative
>> of what's happening to the traffic that the network drops first when =
the going
>> gets rough (or whose delivery w/o loss indicates that a sending rate =
increase
>> may be reasonable), which suggests P-frame-like RTCP report marking.
>=20
> I doubt the RTT estimate derived from RTCP is used for congestion =
control, since it=92s too infrequent to get insight into the dynamics. =
It=92s for user experience reporting, maybe rough clustering of =
receivers, that sort of thing.=20

I'd agree if I didn't have the impression, in RMCAT, that nothing but =
RTP / RTCP is allowed?!  So can we send extra packets to probe for the =
RTT?

Cheers,
Michael


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Subject: Re: [AVTCORE] [Dart] Treatment of RTCP (was Re: Colin Perkins comments - WGLC: draft-ietf-dart-dscp-rtp-02)
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On 27. aug. 2014, at 20:42, Colin Perkins <csp@csperkins.org> wrote:

> On 27 Aug 2014, at 19:39, Michael Welzl <michawe@ifi.uio.no> wrote:
>> On 27. aug. 2014, at 18:00, Colin Perkins <csp@csperkins.org> wrote:
>>> On 26 Aug 2014, at 17:38, Black, David <david.black@emc.com> wrote:
>>>>> The more difficult case is when an SSRC is sending video using =
different
>>>>> markings for RTP packets carrying the I- and P-frames. Should that =
SSRC then
>>>>> mark its RTCP packets like the RTP packets carrying I-frames, like =
the RTP
>>>>> packets carrying P-frames, or what?
>>>>=20
>>>> Answering a question w/a question :-), how are those reports likely =
to be used?
>>>>=20
>>>> For example, if the primary use of these reports is to adjust a =
variable rate
>>>> codec's sending rate, the P-frame info is probably more useful as =
indicative
>>>> of what's happening to the traffic that the network drops first =
when the going
>>>> gets rough (or whose delivery w/o loss indicates that a sending =
rate increase
>>>> may be reasonable), which suggests P-frame-like RTCP report =
marking.
>>>=20
>>> I doubt the RTT estimate derived from RTCP is used for congestion =
control, since it=92s too infrequent to get insight into the dynamics. =
It=92s for user experience reporting, maybe rough clustering of =
receivers, that sort of thing.=20
>>=20
>> I'd agree if I didn't have the impression, in RMCAT, that nothing but =
RTP / RTCP is allowed?!  So can we send extra packets to probe for the =
RTT?
>=20
> I=92d expect RMCAT will probe the one-way delay variation by measuring =
the arrival times of RTP packets, and report that in RTCP once per video =
frame (or so). That would give a much more accurate measurement than a =
single-packet sample of the RTT derived from RTCP, and would still allow =
per-frame adaptation.

I agree that this is a useful signal to use - but I think you still need =
a rough idea of the RTT to be able to interoperate with TCP, else you =
may cause TCP to go into RTOs and be ignorant of that. But this could =
perhaps be a more coarse signal - maybe that's okay. I think this is =
reasonable.

I wonder if your RTCP draft in RMCAT should clarify this?  (sorry if it =
does already, didn't check)

Cheers,
Michael


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	charset=windows-1252

<html><head><meta http-equiv=3D"Content-Type" content=3D"text/html =
charset=3Dwindows-1252"></head><body style=3D"word-wrap: break-word; =
-webkit-nbsp-mode: space; -webkit-line-break: =
after-white-space;"><br><div><div>On 27. aug. 2014, at 20:42, Colin =
Perkins &lt;<a href=3D"mailto:csp@csperkins.org">csp@csperkins.org</a>&gt;=
 wrote:</div><br class=3D"Apple-interchange-newline"><blockquote =
type=3D"cite"><div style=3D"font-size: 12px; font-style: normal; =
font-variant: normal; font-weight: normal; letter-spacing: normal; =
line-height: normal; orphans: auto; text-align: start; text-indent: 0px; =
text-transform: none; white-space: normal; widows: auto; word-spacing: =
0px; -webkit-text-stroke-width: 0px;">On 27 Aug 2014, at 19:39, Michael =
Welzl &lt;<a href=3D"mailto:michawe@ifi.uio.no">michawe@ifi.uio.no</a>&gt;=
 wrote:<br><blockquote type=3D"cite">On 27. aug. 2014, at 18:00, Colin =
Perkins &lt;<a href=3D"mailto:csp@csperkins.org">csp@csperkins.org</a>&gt;=
 wrote:<br><blockquote type=3D"cite">On 26 Aug 2014, at 17:38, Black, =
David &lt;<a =
href=3D"mailto:david.black@emc.com">david.black@emc.com</a>&gt; =
wrote:<br><blockquote type=3D"cite"><blockquote type=3D"cite">The more =
difficult case is when an SSRC is sending video using =
different<br>markings for RTP packets carrying the I- and P-frames. =
Should that SSRC then<br>mark its RTCP packets like the RTP packets =
carrying I-frames, like the RTP<br>packets carrying P-frames, or =
what?<br></blockquote><br>Answering a question w/a question :-), how are =
those reports likely to be used?<br><br>For example, if the primary use =
of these reports is to adjust a variable rate<br>codec's sending rate, =
the P-frame info is probably more useful as indicative<br>of what's =
happening to the traffic that the network drops first when the =
going<br>gets rough (or whose delivery w/o loss indicates that a sending =
rate increase<br>may be reasonable), which suggests P-frame-like RTCP =
report marking.<br></blockquote><br>I doubt the RTT estimate derived =
from RTCP is used for congestion control, since it=92s too infrequent to =
get insight into the dynamics. It=92s for user experience reporting, =
maybe rough clustering of receivers, that sort of thing.<span =
class=3D"Apple-converted-space">&nbsp;</span><br></blockquote><br>I'd =
agree if I didn't have the impression, in RMCAT, that nothing but RTP / =
RTCP is allowed?! &nbsp;So can we send extra packets to probe for the =
RTT?<br></blockquote><br>I=92d expect RMCAT will probe the one-way delay =
variation by measuring the arrival times of RTP packets, and report that =
in RTCP once per video frame (or so). That would give a much more =
accurate measurement than a single-packet sample of the RTT derived from =
RTCP, and would still allow per-frame =
adaptation.<br></div></blockquote><div><br></div>I agree that this is a =
useful signal to use - but I think you still need a rough idea of the =
RTT to be able to interoperate with TCP, else you may cause TCP to go =
into RTOs and be ignorant of that. But this could perhaps be a more =
coarse signal - maybe that's okay. I think this is =
reasonable.</div><div><br></div><div>I wonder if your RTCP draft in =
RMCAT should clarify this? &nbsp;(sorry if it does already, didn't =
check)</div><div><br></div><div>Cheers,</div><div>Michael</div><div><br></=
div></body></html>=

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Subject: Re: [AVTCORE] [Dart] Treatment of RTCP (was Re: Colin Perkins comments - WGLC: draft-ietf-dart-dscp-rtp-02)
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I don't know the beginning of this but I can help go up the stack:

for the short dialogue between Colin and me, the conclusion is: Colin is =
right. RTCP probably won't be used by congestion control to estimate the =
RTT.

Cheers,
Michael


On 27. aug. 2014, at 21:46, Ben Campbell <ben@nostrum.com> wrote:

> Hi,
>=20
> Where are we with this discussion? Are there conclusions buried in =
there somewhere?
>=20
> Thanks!
>=20
> Ben.
>=20
> On Aug 27, 2014, at 1:50 PM, Colin Perkins <csp@csperkins.org> wrote:
>=20
>> On 27 Aug 2014, at 19:47, Michael Welzl <michawe@ifi.uio.no> wrote:
>>> On 27. aug. 2014, at 20:42, Colin Perkins <csp@csperkins.org> wrote:
>>>> On 27 Aug 2014, at 19:39, Michael Welzl <michawe@ifi.uio.no> wrote:
>>>>> On 27. aug. 2014, at 18:00, Colin Perkins <csp@csperkins.org> =
wrote:
>>>>>> On 26 Aug 2014, at 17:38, Black, David <david.black@emc.com> =
wrote:
>>>>>>>> The more difficult case is when an SSRC is sending video using =
different
>>>>>>>> markings for RTP packets carrying the I- and P-frames. Should =
that SSRC then
>>>>>>>> mark its RTCP packets like the RTP packets carrying I-frames, =
like the RTP
>>>>>>>> packets carrying P-frames, or what?
>>>>>>>=20
>>>>>>> Answering a question w/a question :-), how are those reports =
likely to be used?
>>>>>>>=20
>>>>>>> For example, if the primary use of these reports is to adjust a =
variable rate
>>>>>>> codec's sending rate, the P-frame info is probably more useful =
as indicative
>>>>>>> of what's happening to the traffic that the network drops first =
when the going
>>>>>>> gets rough (or whose delivery w/o loss indicates that a sending =
rate increase
>>>>>>> may be reasonable), which suggests P-frame-like RTCP report =
marking.
>>>>>>=20
>>>>>> I doubt the RTT estimate derived from RTCP is used for congestion =
control, since it=92s too infrequent to get insight into the dynamics. =
It=92s for user experience reporting, maybe rough clustering of =
receivers, that sort of thing.=20
>>>>>=20
>>>>> I'd agree if I didn't have the impression, in RMCAT, that nothing =
but RTP / RTCP is allowed?!  So can we send extra packets to probe for =
the RTT?
>>>>=20
>>>> I=92d expect RMCAT will probe the one-way delay variation by =
measuring the arrival times of RTP packets, and report that in RTCP once =
per video frame (or so). That would give a much more accurate =
measurement than a single-packet sample of the RTT derived from RTCP, =
and would still allow per-frame adaptation.
>>>=20
>>> I agree that this is a useful signal to use - but I think you still =
need a rough idea of the RTT to be able to interoperate with TCP, else =
you may cause TCP to go into RTOs and be ignorant of that. But this =
could perhaps be a more coarse signal - maybe that's okay. I think this =
is reasonable.
>>=20
>> You have the RTT estimate, irrespective of the other information you =
include along with the regular RTCP packets, although it=92s relatively =
infrequent and coarse-grained.
>>=20
>>> I wonder if your RTCP draft in RMCAT should clarify this?  (sorry if =
it does already, didn=92t check)
>>=20
>> It probably should talk about this, I agree.
>>=20
>>=20
>> --=20
>> Colin Perkins
>> http://csperkins.org/
>>=20
>>=20
>>=20
>> _______________________________________________
>> Dart mailing list
>> Dart@ietf.org
>> https://www.ietf.org/mailman/listinfo/dart
>=20


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Date: Thu, 28 Aug 2014 18:31:01 +0100
To: Michael Welzl <michawe@ifi.uio.no>
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Subject: Re: [AVTCORE] [Dart] Treatment of RTCP (was Re: Colin Perkins comments - WGLC: draft-ietf-dart-dscp-rtp-02)
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> On 27 Aug 2014, at 19:39, Michael Welzl <michawe@ifi.uio.no> wrote:
>=20
>=20
>> On 27. aug. 2014, at 18:00, Colin Perkins <csp@csperkins.org> wrote:
>>=20
>> On 26 Aug 2014, at 17:38, Black, David <david.black@emc.com> wrote:
>>>> The more difficult case is when an SSRC is sending video using differen=
t
>>>> markings for RTP packets carrying the I- and P-frames. Should that SSRC=
 then
>>>> mark its RTCP packets like the RTP packets carrying I-frames, like the R=
TP
>>>> packets carrying P-frames, or what?
>>>=20
>>> Answering a question w/a question :-), how are those reports likely to b=
e used?
>>>=20
>>> For example, if the primary use of these reports is to adjust a variable=
 rate
>>> codec's sending rate, the P-frame info is probably more useful as indica=
tive
>>> of what's happening to the traffic that the network drops first when the=
 going
>>> gets rough (or whose delivery w/o loss indicates that a sending rate inc=
rease
>>> may be reasonable), which suggests P-frame-like RTCP report marking.
>>=20
>> I doubt the RTT estimate derived from RTCP is used for congestion control=
, since it=E2=80=99s too infrequent to get insight into the dynamics. It=E2=80=
=99s for user experience reporting, maybe rough clustering of receivers, tha=
t sort of thing.
>=20
> I'd agree if I didn't have the impression, in RMCAT, that nothing but RTP /=
 RTCP is allowed?!  So can we send extra packets to probe for the RTT?

Yes, as I replied a couple of days ago.=20

Colin=


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Subject: Re: [AVTCORE] [Dart] Treatment of RTCP (was Re: Colin Perkins comments - WGLC: draft-ietf-dart-dscp-rtp-02)
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On 28 Aug 2014, at 18:31, Colin Perkins <csp@csperkins.org> wrote:
>> On 27 Aug 2014, at 19:39, Michael Welzl <michawe@ifi.uio.no> wrote:
>>=20
>>> On 27. aug. 2014, at 18:00, Colin Perkins <csp@csperkins.org> wrote:
>>>=20
>>> On 26 Aug 2014, at 17:38, Black, David <david.black@emc.com> wrote:
>>>>> The more difficult case is when an SSRC is sending video using =
different
>>>>> markings for RTP packets carrying the I- and P-frames. Should that =
SSRC then
>>>>> mark its RTCP packets like the RTP packets carrying I-frames, like =
the RTP
>>>>> packets carrying P-frames, or what?
>>>>=20
>>>> Answering a question w/a question :-), how are those reports likely =
to be used?
>>>>=20
>>>> For example, if the primary use of these reports is to adjust a =
variable rate
>>>> codec's sending rate, the P-frame info is probably more useful as =
indicative
>>>> of what's happening to the traffic that the network drops first =
when the going
>>>> gets rough (or whose delivery w/o loss indicates that a sending =
rate increase
>>>> may be reasonable), which suggests P-frame-like RTCP report =
marking.
>>>=20
>>> I doubt the RTT estimate derived from RTCP is used for congestion =
control, since it=92s too infrequent to get insight into the dynamics. =
It=92s for user experience reporting, maybe rough clustering of =
receivers, that sort of thing.
>>=20
>> I'd agree if I didn't have the impression, in RMCAT, that nothing but =
RTP / RTCP is allowed?!  So can we send extra packets to probe for the =
RTT?
>=20
> Yes, as I replied a couple of days ago.=20

Wait, sorry, that was the email I replied to a couple of days ago =
reappearing=85 RTT probing, I guess :-)

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




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Subject: Re: [AVTCORE] [Dart] Treatment of RTCP (was Re: Colin Perkins comments - WGLC: draft-ietf-dart-dscp-rtp-02)
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On 28/08/2014 2:02 PM, Colin Perkins wrote:
...

>
>>> I'd agree if I didn't have the impression, in RMCAT, that nothing but RTP / RTCP is allowed?!  So can we send extra packets to probe for the RTT?
>>
>> Yes, as I replied a couple of days ago.
>
> Wait, sorry, that was the email I replied to a couple of days ago reappearing… RTT probing, I guess :-)
>

Sorry, the new messages were released from non-member intercept a bit 
late. I can add Michael to the "always accept" list if he is likely to 
send a few more messages.

Tom Taylor


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To: "dart@ietf.org" <dart@ietf.org>, "avt@ietf.org WG (avt@ietf.org)" <avt@ietf.org>
Date: Thu, 28 Aug 2014 15:45:10 -0400
Thread-Topic: DART dscp-rtp draft: Proposed RTCP text
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Subject: [AVTCORE] DART dscp-rtp draft: Proposed RTCP text
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Here's the proposed new section 5.4, with the RTCP multi-stream reporting
optimizations text moved into that new section from its current location
at the end of 5.3, plus the rewritten Section 6 guideline bullet:

5.4.  DiffServ and RTCP

   The RTP Control Protocol (RTCP) [RFC3550] is used with RTP to monitor
   quality of service and convey information about RTP session
   participants.  A sender of RTCP packets that also sends RTP packets
   (i.e., originates an RTP stream) should use the same DSCP marking for
   both types of packets.  If an RTCP sender doesn't send any RTP
   packets, it should mark its RTCP packets with the DSCP that it would
   use if it did send RTP packets.  If the RTCP sender uses or would use
   multiple DSCPs that differ only in drop precedence for RTP, then it
   should use the DSCP with the least likelihood of drop for RTCP to
   increase the likelihood of RTCP packet delivery.

   This guidance may result in different DSCP markings for RTP streams
   and RTCP receiver reports about those RTP streams.  The resulting
   variation in network QoS treatment by traffic direction could result
   in unrepresentative round trip time (RTT) estimates that don't
   correspond to consistent network QoS treatment in both directions.
   RTCP receiver reports may be relatively infrequent (e.g., may be sent
   only once per video frame rendered) and hence the resulting RTT
   estimates are of limited utility for congestion control (although
   they have other important uses, see [RFC3550]).  For this reason, it
   is not important that RTCP receiver reports receive the same network
   QoS treatment as the RTP stream or streams being reported on.

   RTCP multi-stream reporting optimizations for an RTP session
   [I-D.ietf-avtcore-rtp-multi-stream-optimisation] assume that the RTP
   streams involved experience the same network behavior, including
   packet loss.  This mechanism is highly inappropriate when the RTP
   streams involved use different DSCPs, even if the corresponding PHBs
   differ solely in drop precedence.

6.  Guidelines

[... snip ...]

   o  Should use a single DSCP for RTCP packets, which should be a DSCP
      used for RTP packets that are or would be sent by the same sender,
      see Section 5.4.

Thanks,
--David
----------------------------------------------------
David L. Black, Distinguished Engineer
EMC Corporation, 176 South St., Hopkinton, MA=A0 01748
+1 (508) 293-7953=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0 FAX: +1 (508) 293-778=
6
david.black@emc.com=A0=A0=A0=A0=A0=A0=A0 Mobile: +1 (978) 394-7754
----------------------------------------------------



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From: "Black, David" <david.black@emc.com>
To: "dart@ietf.org" <dart@ietf.org>, "avt@ietf.org WG (avt@ietf.org)" <avt@ietf.org>
Date: Thu, 28 Aug 2014 16:36:25 -0400
Thread-Topic: DART dscp-rtp draft: Proposed RTCP text
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Subject: Re: [AVTCORE] DART dscp-rtp draft: Proposed RTCP text
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Minor edit to proposed text below:

OLD
    If an RTCP sender doesn't send any RTP
    packets, it should mark its RTCP packets with the DSCP that it would
    use if it did send RTP packets.
NEW
   If an RTCP sender doesn't send any RTP
   packets, it should mark its RTCP packets with the DSCP that it would
   use if it did send RTP packets with media similar to the RTP traffic
   that it receives.

Thanks,
--David


> -----Original Message-----
> From: Black, David
> Sent: Thursday, August 28, 2014 3:45 PM
> To: dart@ietf.org; avt@ietf.org WG (avt@ietf.org)
> Cc: Black, David; Colin Perkins (csp@csperkins.org); ben@nostrum.com
> Subject: DART dscp-rtp draft: Proposed RTCP text
>=20
> Here's the proposed new section 5.4, with the RTCP multi-stream reporting
> optimizations text moved into that new section from its current location
> at the end of 5.3, plus the rewritten Section 6 guideline bullet:
>=20
> 5.4.  DiffServ and RTCP
>=20
>    The RTP Control Protocol (RTCP) [RFC3550] is used with RTP to monitor
>    quality of service and convey information about RTP session
>    participants.  A sender of RTCP packets that also sends RTP packets
>    (i.e., originates an RTP stream) should use the same DSCP marking for
>    both types of packets.  If an RTCP sender doesn't send any RTP
>    packets, it should mark its RTCP packets with the DSCP that it would
>    use if it did send RTP packets.  If the RTCP sender uses or would use
>    multiple DSCPs that differ only in drop precedence for RTP, then it
>    should use the DSCP with the least likelihood of drop for RTCP to
>    increase the likelihood of RTCP packet delivery.
>=20
>    This guidance may result in different DSCP markings for RTP streams
>    and RTCP receiver reports about those RTP streams.  The resulting
>    variation in network QoS treatment by traffic direction could result
>    in unrepresentative round trip time (RTT) estimates that don't
>    correspond to consistent network QoS treatment in both directions.
>    RTCP receiver reports may be relatively infrequent (e.g., may be sent
>    only once per video frame rendered) and hence the resulting RTT
>    estimates are of limited utility for congestion control (although
>    they have other important uses, see [RFC3550]).  For this reason, it
>    is not important that RTCP receiver reports receive the same network
>    QoS treatment as the RTP stream or streams being reported on.
>=20
>    RTCP multi-stream reporting optimizations for an RTP session
>    [I-D.ietf-avtcore-rtp-multi-stream-optimisation] assume that the RTP
>    streams involved experience the same network behavior, including
>    packet loss.  This mechanism is highly inappropriate when the RTP
>    streams involved use different DSCPs, even if the corresponding PHBs
>    differ solely in drop precedence.
>=20
> 6.  Guidelines
>=20
> [... snip ...]
>=20
>    o  Should use a single DSCP for RTCP packets, which should be a DSCP
>       used for RTP packets that are or would be sent by the same sender,
>       see Section 5.4.
>=20
> Thanks,
> --David
> ----------------------------------------------------
> David L. Black, Distinguished Engineer
> EMC Corporation, 176 South St., Hopkinton, MA=A0 01748
> +1 (508) 293-7953=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0=A0 FAX: +1 (508) 293-7=
786
> david.black@emc.com=A0=A0=A0=A0=A0=A0=A0 Mobile: +1 (978) 394-7754
> ----------------------------------------------------
>=20


From nobody Thu Aug 28 13:45:04 2014
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From: Ben Campbell <ben@nostrum.com>
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To: "Black, David" <david.black@emc.com>
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Subject: Re: [AVTCORE] DART dscp-rtp draft: Proposed RTCP text
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That works for me, thanks!

On Aug 28, 2014, at 3:36 PM, Black, David <david.black@emc.com> wrote:

> Minor edit to proposed text below:
> 
> OLD
>    If an RTCP sender doesn't send any RTP
>    packets, it should mark its RTCP packets with the DSCP that it would
>    use if it did send RTP packets.
> NEW
>   If an RTCP sender doesn't send any RTP
>   packets, it should mark its RTCP packets with the DSCP that it would
>   use if it did send RTP packets with media similar to the RTP traffic
>   that it receives.
> 
> Thanks,
> --David
> 
> 
>> -----Original Message-----
>> From: Black, David
>> Sent: Thursday, August 28, 2014 3:45 PM
>> To: dart@ietf.org; avt@ietf.org WG (avt@ietf.org)
>> Cc: Black, David; Colin Perkins (csp@csperkins.org); ben@nostrum.com
>> Subject: DART dscp-rtp draft: Proposed RTCP text
>> 
>> Here's the proposed new section 5.4, with the RTCP multi-stream reporting
>> optimizations text moved into that new section from its current location
>> at the end of 5.3, plus the rewritten Section 6 guideline bullet:
>> 
>> 5.4.  DiffServ and RTCP
>> 
>>   The RTP Control Protocol (RTCP) [RFC3550] is used with RTP to monitor
>>   quality of service and convey information about RTP session
>>   participants.  A sender of RTCP packets that also sends RTP packets
>>   (i.e., originates an RTP stream) should use the same DSCP marking for
>>   both types of packets.  If an RTCP sender doesn't send any RTP
>>   packets, it should mark its RTCP packets with the DSCP that it would
>>   use if it did send RTP packets.  If the RTCP sender uses or would use
>>   multiple DSCPs that differ only in drop precedence for RTP, then it
>>   should use the DSCP with the least likelihood of drop for RTCP to
>>   increase the likelihood of RTCP packet delivery.
>> 
>>   This guidance may result in different DSCP markings for RTP streams
>>   and RTCP receiver reports about those RTP streams.  The resulting
>>   variation in network QoS treatment by traffic direction could result
>>   in unrepresentative round trip time (RTT) estimates that don't
>>   correspond to consistent network QoS treatment in both directions.
>>   RTCP receiver reports may be relatively infrequent (e.g., may be sent
>>   only once per video frame rendered) and hence the resulting RTT
>>   estimates are of limited utility for congestion control (although
>>   they have other important uses, see [RFC3550]).  For this reason, it
>>   is not important that RTCP receiver reports receive the same network
>>   QoS treatment as the RTP stream or streams being reported on.
>> 
>>   RTCP multi-stream reporting optimizations for an RTP session
>>   [I-D.ietf-avtcore-rtp-multi-stream-optimisation] assume that the RTP
>>   streams involved experience the same network behavior, including
>>   packet loss.  This mechanism is highly inappropriate when the RTP
>>   streams involved use different DSCPs, even if the corresponding PHBs
>>   differ solely in drop precedence.
>> 
>> 6.  Guidelines
>> 
>> [... snip ...]
>> 
>>   o  Should use a single DSCP for RTCP packets, which should be a DSCP
>>      used for RTP packets that are or would be sent by the same sender,
>>      see Section 5.4.
>> 
>> Thanks,
>> --David
>> ----------------------------------------------------
>> David L. Black, Distinguished Engineer
>> EMC Corporation, 176 South St., Hopkinton, MA  01748
>> +1 (508) 293-7953             FAX: +1 (508) 293-7786
>> david.black@emc.com        Mobile: +1 (978) 394-7754
>> ----------------------------------------------------
>> 
> 


From nobody Thu Aug 28 13:59:37 2014
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Subject: Re: [AVTCORE] DART dscp-rtp draft: Proposed RTCP text
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On 28 Aug 2014, at 20:45, Black, David <david.black@emc.com> wrote:
> Here's the proposed new section 5.4, with the RTCP multi-stream =
reporting
> optimizations text moved into that new section from its current =
location
> at the end of 5.3, plus the rewritten Section 6 guideline bullet:
>=20
> 5.4.  DiffServ and RTCP
>=20
>   The RTP Control Protocol (RTCP) [RFC3550] is used with RTP to =
monitor
>   quality of service and convey information about RTP session
>   participants.  A sender of RTCP packets that also sends RTP packets
>   (i.e., originates an RTP stream) should use the same DSCP marking =
for
>   both types of packets.  If an RTCP sender doesn't send any RTP
>   packets, it should mark its RTCP packets with the DSCP that it would
>   use if it did send RTP packets.  If the RTCP sender uses or would =
use
>   multiple DSCPs that differ only in drop precedence for RTP, then it
>   should use the DSCP with the least likelihood of drop for RTCP to
>   increase the likelihood of RTCP packet delivery.

This looks okay (the correction in your next message is good also).=20

You might also add something along the lines of =93If the SDP bundle =
extension [cite] is used to negotiate sending multiple types of media in =
a single RTP session, then the receivers will send separate RTCP reports =
for each type of media, using a separate SSRC for each media type; those =
RTCP reports need to be marked with the DSCP corresponding to the type =
of media handled by the reporting SSRC.=94=20

>   This guidance may result in different DSCP markings for RTP streams
>   and RTCP receiver reports about those RTP streams.  The resulting
>   variation in network QoS treatment by traffic direction could result
>   in unrepresentative round trip time (RTT) estimates that don't
>   correspond to consistent network QoS treatment in both directions.

I don=92t think this is a problem. The media doesn=92t get consistent =
network QoS treatment in both directions, and the RTCP is trying to =
sample the RTT seen by the media, so this is fine.

>   RTCP receiver reports may be relatively infrequent (e.g., may be =
sent
>   only once per video frame rendered) and hence the resulting RTT
>   estimates are of limited utility for congestion control (although
>   they have other important uses, see [RFC3550]).  For this reason, it
>   is not important that RTCP receiver reports receive the same network
>   QoS treatment as the RTP stream or streams being reported on.
>=20
>   RTCP multi-stream reporting optimizations for an RTP session
>   [I-D.ietf-avtcore-rtp-multi-stream-optimisation] assume that the RTP
>   streams involved experience the same network behavior, including
>   packet loss.  This mechanism is highly inappropriate when the RTP
>   streams involved use different DSCPs, even if the corresponding PHBs
>   differ solely in drop precedence.

This could maybe say =93received RTP streams=94, since it=92s okay to =
use the rtp-multi-stream-optimisation if you send with different DSCPs =
(the optimisation is optimising how you send reports about what you =
receive).

> 6.  Guidelines
>=20
> [... snip ...]
>=20
>   o  Should use a single DSCP for RTCP packets, which should be a DSCP
>      used for RTP packets that are or would be sent by the same =
sender,
>      see Section 5.4.

Perhaps =93Each SSRC should use a single DSCP for RTCP =85 used by RTP =
packets that are or would be sent by that SSRC=85=94? Just =93sender" =
could be ambiguous, since an endpoint could be a sender for several =
streams, each of which has a separate SSRC and could use a different =
DSCP.

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




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Subject: [AVTCORE] Last Call: <draft-ietf-avtcore-aria-srtp-06.txt> (The ARIA Algorithm and Its Use with the Secure Real-time Transport Protocol(SRTP)) to Proposed Standard
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The IESG has received a request from the Audio/Video Transport Core
Maintenance WG (avtcore) to consider the following document:
- 'The ARIA Algorithm and Its Use with the Secure Real-time Transport
   Protocol(SRTP)'
  <draft-ietf-avtcore-aria-srtp-06.txt> as Proposed Standard

The IESG plans to make a decision in the next few weeks, and solicits
final comments on this action. Please send substantive comments to the
ietf@ietf.org mailing lists by 2014-09-11. 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 defines the use of the ARIA block cipher algorithm
   within the Secure Real-time Transport Protocol (SRTP) for providing
   confidentiality for the Real-time Transport Protocol (RTP) traffic
   and for the control traffic for RTP, the RTP Control Protocol (RTCP).
   It details three modes of operation (CTR, CCM, GCM) and a SRTP Key
   Derivation Function for ARIA.




The file can be obtained via
http://datatracker.ietf.org/doc/draft-ietf-avtcore-aria-srtp/

IESG discussion can be tracked via
http://datatracker.ietf.org/doc/draft-ietf-avtcore-aria-srtp/ballot/


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



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Subject: [AVTCORE] Last Call: <draft-ietf-avtcore-srtp-aes-gcm-14.txt> (AES-GCM and AES-CCM Authenticated Encryption in Secure RTP (SRTP)) to Proposed Standard
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The IESG has received a request from the Audio/Video Transport Core
Maintenance WG (avtcore) to consider the following document:
- 'AES-GCM and AES-CCM Authenticated Encryption in Secure RTP (SRTP)'
  <draft-ietf-avtcore-srtp-aes-gcm-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 2014-09-11. 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 defines how AES-GCM and AES-CCM Authenticated
   Encryption with Associated Data algorithms can be used to provide
   confidentiality and data authentication in the SRTP protocol.





The file can be obtained via
http://datatracker.ietf.org/doc/draft-ietf-avtcore-srtp-aes-gcm/

IESG discussion can be tracked via
http://datatracker.ietf.org/doc/draft-ietf-avtcore-srtp-aes-gcm/ballot/


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



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From: "Black, David" <david.black@emc.com>
To: Colin Perkins <csp@csperkins.org>
Date: Fri, 29 Aug 2014 11:17:19 -0400
Thread-Topic: [Dart] DART dscp-rtp draft: Proposed RTCP text
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Subject: Re: [AVTCORE] [Dart] DART dscp-rtp draft: Proposed RTCP text
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Colin,

Many thanks, once again!

I'm about to submit a -05 version of the DART draft that reflects
this feedback, and also says that we're using SSRC as both the
identifier of an RTP stream and to designate the sender of that
RTP stream  (the new terminology draft uses Media Packetizer
as the term for the sender of an RTP stream).

Thanks,
--David


> -----Original Message-----
> From: Dart [mailto:dart-bounces@ietf.org] On Behalf Of Colin Perkins
> Sent: Thursday, August 28, 2014 4:59 PM
> To: Black, David
> Cc: ben@nostrum.com; dart@ietf.org; avt@ietf.org WG (avt@ietf.org)
> Subject: Re: [Dart] DART dscp-rtp draft: Proposed RTCP text
>=20
> On 28 Aug 2014, at 20:45, Black, David <david.black@emc.com> wrote:
> > Here's the proposed new section 5.4, with the RTCP multi-stream reporti=
ng
> > optimizations text moved into that new section from its current locatio=
n
> > at the end of 5.3, plus the rewritten Section 6 guideline bullet:
> >
> > 5.4.  DiffServ and RTCP
> >
> >   The RTP Control Protocol (RTCP) [RFC3550] is used with RTP to monitor
> >   quality of service and convey information about RTP session
> >   participants.  A sender of RTCP packets that also sends RTP packets
> >   (i.e., originates an RTP stream) should use the same DSCP marking for
> >   both types of packets.  If an RTCP sender doesn't send any RTP
> >   packets, it should mark its RTCP packets with the DSCP that it would
> >   use if it did send RTP packets.  If the RTCP sender uses or would use
> >   multiple DSCPs that differ only in drop precedence for RTP, then it
> >   should use the DSCP with the least likelihood of drop for RTCP to
> >   increase the likelihood of RTCP packet delivery.
>=20
> This looks okay (the correction in your next message is good also).
>=20
> You might also add something along the lines of "If the SDP bundle extens=
ion
> [cite] is used to negotiate sending multiple types of media in a single R=
TP
> session, then the receivers will send separate RTCP reports for each type=
 of
> media, using a separate SSRC for each media type; those RTCP reports need=
 to
> be marked with the DSCP corresponding to the type of media handled by the
> reporting SSRC."
>=20
> >   This guidance may result in different DSCP markings for RTP streams
> >   and RTCP receiver reports about those RTP streams.  The resulting
> >   variation in network QoS treatment by traffic direction could result
> >   in unrepresentative round trip time (RTT) estimates that don't
> >   correspond to consistent network QoS treatment in both directions.
>=20
> I don't think this is a problem. The media doesn't get consistent network=
 QoS
> treatment in both directions, and the RTCP is trying to sample the RTT se=
en by
> the media, so this is fine.
>=20
> >   RTCP receiver reports may be relatively infrequent (e.g., may be sent
> >   only once per video frame rendered) and hence the resulting RTT
> >   estimates are of limited utility for congestion control (although
> >   they have other important uses, see [RFC3550]).  For this reason, it
> >   is not important that RTCP receiver reports receive the same network
> >   QoS treatment as the RTP stream or streams being reported on.
> >
> >   RTCP multi-stream reporting optimizations for an RTP session
> >   [I-D.ietf-avtcore-rtp-multi-stream-optimisation] assume that the RTP
> >   streams involved experience the same network behavior, including
> >   packet loss.  This mechanism is highly inappropriate when the RTP
> >   streams involved use different DSCPs, even if the corresponding PHBs
> >   differ solely in drop precedence.
>=20
> This could maybe say "received RTP streams", since it's okay to use the r=
tp-
> multi-stream-optimisation if you send with different DSCPs (the optimisat=
ion
> is optimising how you send reports about what you receive).
>=20
> > 6.  Guidelines
> >
> > [... snip ...]
> >
> >   o  Should use a single DSCP for RTCP packets, which should be a DSCP
> >      used for RTP packets that are or would be sent by the same sender,
> >      see Section 5.4.
>=20
> Perhaps "Each SSRC should use a single DSCP for RTCP ... used by RTP pack=
ets
> that are or would be sent by that SSRC..."? Just "sender" could be ambigu=
ous,
> since an endpoint could be a sender for several streams, each of which ha=
s a
> separate SSRC and could use a different DSCP.
>=20
> --
> Colin Perkins
> http://csperkins.org/
>=20
>=20
>=20
> _______________________________________________
> Dart mailing list
> Dart@ietf.org
> https://www.ietf.org/mailman/listinfo/dart

