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From: Fu Jiantao <fuji246@gmail.com>
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Hi Colin,

   I've got the following comments:

  1."there is little point in providing feedback more often than the codec
can adapt", i think there is no need to send those overlimits data onto the
network if we've got feedback that the network is congested, so i think it
make sense if there is traffic shaper on the sender side, those overlimit
packets can buffered and dropped at the sender side, rather than sending
them to the network which affects other traffics.

  2. How about ack in a given interval, this can be treat as variant of
per-frame ack(several frames per ack). This is another kind which has less
load than per-frame ack.

  3. There can be adaptive feedback which feedback when there is need to do
so, for example, when the given interval is not enough for the adaption(the
queuing delay goes large very fast), the feedback can be accelerated in
this case.

  4. The feedback strategy has some relationship with
sender-based/receiver-based, is this in the scope of this draft?


Thanks

Jeromy


2013/2/26 Colin Perkins <csp@csperkins.org>

> I submitted the following draft that talks about the types of congestion
> control feedback that can be sent using RTCP as is currently specified, and
> their suitability for implementing congestion control for unicast
> multimedia applications. In particular, it talks about the rate of
> congestion feedback that can be supported.
>
> Comments are welcome. I'm also happy to present this in Orlando, if the
> chairs think it useful.
>
> Colin
>
>
>
>
> Begin forwarded message:
> > From: internet-drafts@ietf.org
> > Subject: I-D Action: draft-perkins-rmcat-rtp-cc-feedback-00.txt
> > Date: 18 February 2013 23:28:42 GMT
> > To: i-d-announce@ietf.org
> > Reply-To: internet-drafts@ietf.org
> >
> >
> > A New Internet-Draft is available from the on-line Internet-Drafts
> directories.
> >
> >
> >       Title           : On the Use of RTP Control Protocol (RTCP)
> Feedback for Unicast Multimedia Congestion Control
> >       Author(s)       : Colin Perkins
> >       Filename        : draft-perkins-rmcat-rtp-cc-feedback-00.txt
> >       Pages           : 8
> >       Date            : 2013-02-18
> >
> > Abstract:
> > This memo discusses the types of congestion control feedback that it
> > is possible to send using the RTP Control Protocol (RTCP), and their
> > suitability of use in implementing congestion control for unicast
> > multimedia applications.
> >
> >
> > The IETF datatracker status page for this draft is:
> > https://datatracker.ietf.org/doc/draft-perkins-rmcat-rtp-cc-feedback
> >
> > There's also a htmlized version available at:
> > http://tools.ietf.org/html/draft-perkins-rmcat-rtp-cc-feedback-00
>
>
> --
> Colin Perkins
> http://csperkins.org/
>
>
>
>

--f46d04016975e839cf04d7261766
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<div dir=3D"ltr"><div><div><div><div><div>Hi Colin,<br><br></div>=A0=A0 I&#=
39;ve got the following comments:<br><br>=A0
 1.&quot;there is little point in
   providing feedback more often than the codec can adapt&quot;, i think=20
there is no need to send those overlimits data onto the network if we&#39;v=
e
 got feedback that the network is congested, so i think it make sense if
 there is traffic shaper on the sender side, those overlimit packets can
 buffered and dropped at the sender side, rather than sending them to=20
the network which affects other traffics.<br>
<br></div>=A0 2. How about ack in a given interval, this can be treat as=20
variant of per-frame ack(several frames per ack). This is another kind=20
which has less load than per-frame ack.<br><br></div>=A0 3. There can be=20
adaptive feedback which feedback when there is need to do so, for=20
example, when the given interval is not enough for the adaption(the=20
queuing delay goes large very fast), the feedback can be accelerated in=20
this case.<br>
<br></div>=A0 4. The feedback strategy has some relationship with sender-ba=
sed/receiver-based, is this in the scope of this draft?<br><br><br></div>Th=
anks<br><br>Jeromy</div><div class=3D"gmail_extra"><br><br><div class=3D"gm=
ail_quote">
2013/2/26 Colin Perkins <span dir=3D"ltr">&lt;<a href=3D"mailto:csp@csperki=
ns.org" target=3D"_blank">csp@csperkins.org</a>&gt;</span><br><blockquote c=
lass=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1px #ccc solid;=
padding-left:1ex">
I submitted the following draft that talks about the types of congestion co=
ntrol feedback that can be sent using RTCP as is currently specified, and t=
heir suitability for implementing congestion control for unicast multimedia=
 applications. In particular, it talks about the rate of congestion feedbac=
k that can be supported.<br>

<br>
Comments are welcome. I&#39;m also happy to present this in Orlando, if the=
 chairs think it useful.<br>
<br>
Colin<br>
<br>
<br>
<br>
<br>
Begin forwarded message:<br>
&gt; From: <a href=3D"mailto:internet-drafts@ietf.org">internet-drafts@ietf=
.org</a><br>
&gt; Subject: I-D Action: draft-perkins-rmcat-rtp-cc-feedback-00.txt<br>
&gt; Date: 18 February 2013 23:28:42 GMT<br>
&gt; To: <a href=3D"mailto:i-d-announce@ietf.org">i-d-announce@ietf.org</a>=
<br>
&gt; Reply-To: <a href=3D"mailto:internet-drafts@ietf.org">internet-drafts@=
ietf.org</a><br>
&gt;<br>
&gt;<br>
&gt; A New Internet-Draft is available from the on-line Internet-Drafts dir=
ectories.<br>
&gt;<br>
&gt;<br>
&gt; =A0 =A0 =A0 Title =A0 =A0 =A0 =A0 =A0 : On the Use of RTP Control Prot=
ocol (RTCP) Feedback for Unicast Multimedia Congestion Control<br>
&gt; =A0 =A0 =A0 Author(s) =A0 =A0 =A0 : Colin Perkins<br>
&gt; =A0 =A0 =A0 Filename =A0 =A0 =A0 =A0: draft-perkins-rmcat-rtp-cc-feedb=
ack-00.txt<br>
&gt; =A0 =A0 =A0 Pages =A0 =A0 =A0 =A0 =A0 : 8<br>
&gt; =A0 =A0 =A0 Date =A0 =A0 =A0 =A0 =A0 =A0: 2013-02-18<br>
&gt;<br>
&gt; Abstract:<br>
&gt; This memo discusses the types of congestion control feedback that it<b=
r>
&gt; is possible to send using the RTP Control Protocol (RTCP), and their<b=
r>
&gt; suitability of use in implementing congestion control for unicast<br>
&gt; multimedia applications.<br>
&gt;<br>
&gt;<br>
&gt; The IETF datatracker status page for this draft is:<br>
&gt; <a href=3D"https://datatracker.ietf.org/doc/draft-perkins-rmcat-rtp-cc=
-feedback" target=3D"_blank">https://datatracker.ietf.org/doc/draft-perkins=
-rmcat-rtp-cc-feedback</a><br>
&gt;<br>
&gt; There&#39;s also a htmlized version available at:<br>
&gt; <a href=3D"http://tools.ietf.org/html/draft-perkins-rmcat-rtp-cc-feedb=
ack-00" target=3D"_blank">http://tools.ietf.org/html/draft-perkins-rmcat-rt=
p-cc-feedback-00</a><br>
<span class=3D"HOEnZb"><font color=3D"#888888"><br>
<br>
--<br>
Colin Perkins<br>
<a href=3D"http://csperkins.org/" target=3D"_blank">http://csperkins.org/</=
a><br>
<br>
<br>
<br>
</font></span></blockquote></div><br></div>

--f46d04016975e839cf04d7261766--

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Subject: Re: [rmcat] requirements (was Re: time to get active)
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Another concern for the requirement

There maybe sometimes UDP is blocked, and ICE may use TCP connection(see
http://tools.ietf.org/html/rfc4571 for how to use rtp/rtcp on TCP), this is
a practical user scenario, is there any consideration for this case? I'm
not sure whether this is in the scope of rmcat.

Thanks!

Jeromy


2013/2/28 ken carlberg <carlberg@g11.org.uk>

> well, perhaps I can get the ball rolling in terms of a headache to chew on
> :-)
>
> The second requirement in draft-jesup-rmcat-reqs-01.txt is...
>
>   2.   The algorithm must be fair to other flows, both realtime flows
>         (such as other instances of itself), and TCP flows, both long-
>         lived and bursts such as the traffic generated by a typical web
>         browsing session.  Note that 'fair' is a rather hard-to-define
>         term.
>
>         A.  The algorithm must not overreact to short-term bursts (such
>             as web-browsing) which can quickly saturate a local-
>             bottleneck router or link, but also clear quickly, and
>             should recover quickly when the burst ends.
>
>         B.  We will need make some evaluation of fairness, but deciding
>             what is "fair" is a tough question and likely to be
>             partially subjective, but we should specify some of the
>             inputs needed in order to select among algorithms and
>             tunings presented as options.
>
> however, the document does not have a definition of the term "fair" or
> "fairness".  And while the document points us to terminology in another
> draft (draft-ietf-rtcweb-overview-06.txt), this other draft doesn't even
> mention "fair" or fairness".  I think one nit is that the RMCAT
> requirements draft needs its own terminology section instead of pointing to
> another draft.
>
> And more importantly, I think we need to bite the bullet and decide what
> is "fair" / "fairness".  It just doesn't seem possible for us to say that
> the algorithms *must* be fair, but then punt in defining what is (or
> perhaps, what is not) fair.  And I'm definitely not meaning to beat up on
> the author, but rather ourselves.
>
> As a starting point, and in the context of real time apps, I don't think
> we need to be fair to a continual series of parallel short transaction
> flows.  Rather, I would prefer to be fair, or focus our attention, to
> longer transaction flows.
>
> thoughts?  Have I missed something?
>
> -ken
>
>
>
> On Feb 27, 2013, at 1:37 PM, "Eggert, Lars" <lars@netapp.com> wrote:
>
> > On Feb 27, 2013, at 19:35, Matt Mathis <mattmathis@google.com> wrote:
> >> Did we perhaps address all of the hard problems in the charter process,
> so
> >> now there is nothing to controversial to chew on?
> >
> > I wish...
> >
> > Lars
>
>

--bcaec54b48d083754e04d729f2c2
Content-Type: text/html; charset=ISO-8859-1
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<div dir=3D"ltr"><div><div>Another concern for the requirement<br><br></div=
>There maybe sometimes UDP is blocked, and ICE may use TCP connection(see <=
a href=3D"http://tools.ietf.org/html/rfc4571">http://tools.ietf.org/html/rf=
c4571</a> for how to use rtp/rtcp on TCP), this is a practical user scenari=
o, is there any consideration for this case? I&#39;m not sure whether this =
is in the scope of rmcat.<br>
<br></div><div>Thanks!<br><br></div><div>Jeromy<br></div></div><div class=
=3D"gmail_extra"><br><br><div class=3D"gmail_quote">2013/2/28 ken carlberg =
<span dir=3D"ltr">&lt;<a href=3D"mailto:carlberg@g11.org.uk" target=3D"_bla=
nk">carlberg@g11.org.uk</a>&gt;</span><br>
<blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1p=
x #ccc solid;padding-left:1ex">well, perhaps I can get the ball rolling in =
terms of a headache to chew on :-)<br>
<br>
The second requirement in draft-jesup-rmcat-reqs-01.txt is...<br>
<br>
=A0 2. =A0 The algorithm must be fair to other flows, both realtime flows<b=
r>
=A0 =A0 =A0 =A0 (such as other instances of itself), and TCP flows, both lo=
ng-<br>
=A0 =A0 =A0 =A0 lived and bursts such as the traffic generated by a typical=
 web<br>
=A0 =A0 =A0 =A0 browsing session. =A0Note that &#39;fair&#39; is a rather h=
ard-to-define<br>
=A0 =A0 =A0 =A0 term.<br>
<br>
=A0 =A0 =A0 =A0 A. =A0The algorithm must not overreact to short-term bursts=
 (such<br>
=A0 =A0 =A0 =A0 =A0 =A0 as web-browsing) which can quickly saturate a local=
-<br>
=A0 =A0 =A0 =A0 =A0 =A0 bottleneck router or link, but also clear quickly, =
and<br>
=A0 =A0 =A0 =A0 =A0 =A0 should recover quickly when the burst ends.<br>
<br>
=A0 =A0 =A0 =A0 B. =A0We will need make some evaluation of fairness, but de=
ciding<br>
=A0 =A0 =A0 =A0 =A0 =A0 what is &quot;fair&quot; is a tough question and li=
kely to be<br>
=A0 =A0 =A0 =A0 =A0 =A0 partially subjective, but we should specify some of=
 the<br>
=A0 =A0 =A0 =A0 =A0 =A0 inputs needed in order to select among algorithms a=
nd<br>
=A0 =A0 =A0 =A0 =A0 =A0 tunings presented as options.<br>
<br>
however, the document does not have a definition of the term &quot;fair&quo=
t; or &quot;fairness&quot;. =A0And while the document points us to terminol=
ogy in another draft (draft-ietf-rtcweb-overview-06.txt), this other draft =
doesn&#39;t even mention &quot;fair&quot; or fairness&quot;. =A0I think one=
 nit is that the RMCAT requirements draft needs its own terminology section=
 instead of pointing to another draft.<br>

<br>
And more importantly, I think we need to bite the bullet and decide what is=
 &quot;fair&quot; / &quot;fairness&quot;. =A0It just doesn&#39;t seem possi=
ble for us to say that the algorithms *must* be fair, but then punt in defi=
ning what is (or perhaps, what is not) fair. =A0And I&#39;m definitely not =
meaning to beat up on the author, but rather ourselves.<br>

<br>
As a starting point, and in the context of real time apps, I don&#39;t thin=
k we need to be fair to a continual series of parallel short transaction fl=
ows. =A0Rather, I would prefer to be fair, or focus our attention, to longe=
r transaction flows.<br>

<br>
thoughts? =A0Have I missed something?<br>
<br>
-ken<br>
<br>
<br>
<br>
On Feb 27, 2013, at 1:37 PM, &quot;Eggert, Lars&quot; &lt;<a href=3D"mailto=
:lars@netapp.com">lars@netapp.com</a>&gt; wrote:<br>
<br>
&gt; On Feb 27, 2013, at 19:35, Matt Mathis &lt;<a href=3D"mailto:mattmathi=
s@google.com">mattmathis@google.com</a>&gt; wrote:<br>
&gt;&gt; Did we perhaps address all of the hard problems in the charter pro=
cess, so<br>
&gt;&gt; now there is nothing to controversial to chew on?<br>
&gt;<br>
&gt; I wish...<br>
&gt;<br>
&gt; Lars<br>
<br>
</blockquote></div><br></div>

--bcaec54b48d083754e04d729f2c2--

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From: Zaheduzzaman Sarker <zaheduzzaman.sarker@ericsson.com>
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Subject: [rmcat] Initial comments on draft-jesup-rmcat-reqs-01
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Hi,

Thanks to the author for updating the doc. This version has clarified 
couple of issues but still I think there is lot left to do for the whole 
group. My thoughts/comments below-

General :

# I think  it will be very good if the document includes some 
explanation text on why one particular requirement is listed. Those who 
are involved in the discussion from the very beginning perhaps 
understands the reasons but still I think it will be nice thing to add 
(to reduce latecomer effect :-)).

# Now, we will have to decide on the "fairness" (everybody's favorite, 
right?). Yes, it is difficult and yes, we can spend huge amount of time 
discussing on it. However, IMHO the discussion should short and we 
should just pick one that is available or define something ( for 
example, something that Harald had mentioned in one of IAB CC workshop 
paper).

# This draft should somewhere mentioned the circuit breaker draft. I 
would even say that it should be a requirement on the RMCAT Congestion 
Control (CC) algorithm that it works "within the envelope" of circuit 
breaker.

# The applications that will use RMCAT solution for congestion control 
may need to have a way to configure certain behavior related to the 
service it provides. For example, the application may like to set the 
MIN and MAX bitrate rate, certain behavior if the MIN bitrate is not 
sustainable even if after adapting rate etc. How would we like to 
realize this? One way is to , set the MAX and MIN rate as configuration 
to the adaptation algorithm so that if it suggest , for example , 
bitrate to the application then it never goes above MAX and below MIN. 
Another way is to not set any limit to the CC algorithm and let it send 
feedback to the application on what will be the sustainable bitrate at a 
certain time and the application decides what to do with that feedback. 
The first one is perhaps the most commonly implemented approach while 
the second options gives lot of power to the application. Depending what 
we think is the right way to realize this, puts some requirements on the 
RMCAT CC algorithm. Requirement 1 touches upon it but it also created 
confusion in my mind and I think we can we a better job there.

Requirement Specific:

   * Req 4: it says "Extra information could be added to the packets to 
provide more detailed information on actual send times (as opposed to 
sampling times), but should not be required." I think it can say 
something like this "Extra information could be added to the packets to 
provide more detailed information to improve the performance of the 
algorithm but the algorithm should justify the additional information 
added". Then an algorithm can use, for example, actual send times 
instead of sampling times or something else to improve its performance 
or use something else.
   * Req 7: this is an important requirements. Very ofter I find papers 
where they kind of forget about the congestion in feedback channel.
   * Req 8: This requirement focuses on long-lived TCP flows but I think 
it should focus on flows with known congestion control algorithm. in the 
fast phase, RMCAT will publish (hopefully) more than one experimental CC 
algorithms and I think it is important to look how one of the candidate 
is behaving when competing with other candidate.  But if we want to 
focus on the behavior against competing TCP flows then the requirement 
should say "It should attempt to avoid bandwidth 'collapse' when facing 
a saturating TCP flow or flows but it should not starve TCP flow (should 
not grab more than necessary)" or something along this theme. And I 
would also like to get rid of the prefix terms "short-lived" and 
"Long-lived" TCP flow as I think from a congestion controller point of 
view it is hardly possible to distinguish between them unless we have 
some explicit information from the shared bottleneck.

So far, this is what i think.

BR

-- 

Zahed

==================================================
ANM ZAHEDUZZAMAN SARKER


Ericsson AB
Multimedia Technologies
Laboratoriegränd 11
97128 Luleå, Sweden
Phone +46 10 717 37 43
Fax +46 920 996 21
SMS/MMS +46 76 115 37 43
zaheduzzaman.sarker@ericsson.com
www.ericsson.com

==================================================

From michawe@ifi.uio.no  Tue Mar  5 06:45:17 2013
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Subject: [rmcat] preview of -01 update to draft-welzl-rmcat-coupled-cc
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Hi all,

We (yes, now two authors) have made an update to the coupled-cc draft, =
attached, in which we have improved the example algorithm a bit.
If you have already given this document a high-level read, don't worry =
about it. If you're planning to go through the algorithm step by step =
and follow our example, it's better if you use this newer version.

Thanks,

cheers,
Michael

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RTP Media Congestion Avoidance                                  M. Welzl
Techniques (rmcat)                                              S. Islam
Internet-Draft                                        University of Oslo
Intended status: Experimental                              March 5, 2013
Expires: September 6, 2013


                Coupled congestion control for RTP media
                    draft-welzl-rmcat-coupled-cc-01

Abstract

   When multiple congestion controlled RTP sessions traverse the same
   network bottleneck, it can be beneficial to combine their controls
   such that the total on-the-wire behavior is improved.  This document
   describes such a method for flows that have the same sender, in a way
   that is as flexible and simple as possible while minimizing the
   amount of changes needed to existing RTP applications.

Status of this Memo

   This Internet-Draft is submitted in full conformance with the
   provisions of BCP 78 and BCP 79.

   Internet-Drafts are working documents of the Internet Engineering
   Task Force (IETF).  Note that other groups may also distribute
   working documents as Internet-Drafts.  The list of current Internet-
   Drafts is at http://datatracker.ietf.org/drafts/current/.

   Internet-Drafts are draft documents valid for a maximum of six months
   and may be updated, replaced, or obsoleted by other documents at any
   time.  It is inappropriate to use Internet-Drafts as reference
   material or to cite them other than as "work in progress."

   This Internet-Draft will expire on September 6, 2013.

Copyright Notice

   Copyright (c) 2013 IETF Trust and the persons identified as the
   document authors.  All rights reserved.

   This document is subject to BCP 78 and the IETF Trust's Legal
   Provisions Relating to IETF Documents
   (http://trustee.ietf.org/license-info) in effect on the date of
   publication of this document.  Please review these documents
   carefully, as they describe your rights and restrictions with respect
   to this document.  Code Components extracted from this document must
   include Simplified BSD License text as described in Section 4.e of



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   the Trust Legal Provisions and are provided without warranty as
   described in the Simplified BSD License.


Table of Contents

   1.  Introduction . . . . . . . . . . . . . . . . . . . . . . . . .  3
   2.  Definitions  . . . . . . . . . . . . . . . . . . . . . . . . .  3
   3.  Limitations  . . . . . . . . . . . . . . . . . . . . . . . . .  4
   4.  Architectural overview . . . . . . . . . . . . . . . . . . . .  5
   5.  Roles  . . . . . . . . . . . . . . . . . . . . . . . . . . . .  6
     5.1.  SBD  . . . . . . . . . . . . . . . . . . . . . . . . . . .  6
     5.2.  FSE  . . . . . . . . . . . . . . . . . . . . . . . . . . .  6
     5.3.  Flows  . . . . . . . . . . . . . . . . . . . . . . . . . .  7
       5.3.1.  Example algorithm  . . . . . . . . . . . . . . . . . .  7
       5.3.2.  Example operation  . . . . . . . . . . . . . . . . . .  9
   6.  Acknowledgements . . . . . . . . . . . . . . . . . . . . . . . 14
   7.  IANA Considerations  . . . . . . . . . . . . . . . . . . . . . 14
   8.  Security Considerations  . . . . . . . . . . . . . . . . . . . 14
   9.  References . . . . . . . . . . . . . . . . . . . . . . . . . . 15
     9.1.  Normative References . . . . . . . . . . . . . . . . . . . 15
     9.2.  Informative References . . . . . . . . . . . . . . . . . . 15
   Appendix A.  Changes from -00 to -01 . . . . . . . . . . . . . . . 16
   Authors' Addresses . . . . . . . . . . . . . . . . . . . . . . . . 16



























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1.  Introduction

   When there is enough data to send, a congestion controller must
   increase its sending rate until the path's capacity has been reached;
   depending on the controller, sometimes the rate is increased further,
   until packets are ECN-marked or dropped.  This process inevitably
   creates undesirable queuing delay -- an effect that is amplified when
   multiple congestion controlled connections traverse the same network
   bottleneck.  When such connections originate from the same host, it
   would therefore be ideal to use only one single sender-side
   congestion controller which determines the overall allowed sending
   rate, and then use a local scheduler to assign a proportion of this
   rate to each RTP session.  This way, priorities could also be
   implemented quite easily, as a function of the scheduler; honoring
   user-specified priorities is, for example, required by rtcweb
   [rtcweb-usecases].

   The Congestion Manager (CM) [RFC3124] provides a single congestion
   controller with a scheduling function just as described above.  It is
   hard to implement because it requires an additional congestion
   controller and removes all per-connection congestion control
   functionality, which is quite a significant change to existing RTP
   based applications.  This document presents a method that is easier
   to implement than the CM and also requires less significant changes
   to existing RTP based applications.  It attempts to roughly
   approximate the CM behavior by sharing information between existing
   congestion controllers, akin to "Ensemble Sharing" in [RFC2140].


2.  Definitions

   The key words "MUST", "MUST NOT", "REQUIRED", "SHALL", "SHALL NOT",
   "SHOULD", "SHOULD NOT", "RECOMMENDED", "MAY", and "OPTIONAL" in this
   document are to be interpreted as described in RFC 2119 [RFC2119].

   Available Bandwidth:
         The available bandwidth is the nominal link capacity minus the
         amount of traffic that traversed the link during a certain time
         interval, divided by that time interval.

   Bottleneck:
         The first link with the smallest available bandwidth along the
         path between a sender and receiver.

   Flow:
         A flow is the entity that congestion control is operating on.
         It could, for example, be a transport layer connection, an RTP
         session, or a subsession that is multiplexed onto a single RTP



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         session together with other subsessions.

   Flow Group Identifier (FGI):
         A unique identifier for each subset of flows that is limited by
         a common bottleneck.

   Flow State Exchange (FSE):
         The entity that maintains information that is exchanged between
         flows.

   Flow Group (FG):
         A group of flows having the same FGI.

   Shared Bottleneck Detection (SBD):
         The entity that determines which flows traverse the same
         bottleneck in the network, or the process of doing so.


3.  Limitations

   Sender-side only:
         Coupled congestion control as described here only operates
         inside a single host on the sender side.  This is because,
         irrespective of where the major decisions for congestion
         control are taken, the sender of a flow needs to eventually
         decide the transmission rate.  Additionally, the necessary
         information about how much data an application can currently
         send on a flow is typically only available at the sender side,
         making the sender an obvious choice for placement of the
         elements and mechanisms described here.  It is recognized that
         flows that have different senders but the same receiver, or
         different senders and different receivers can also share a
         bottleneck; such scenarios have been omitted for simplicity,
         and could be incorporated in future versions of this document.
         Note that limiting the flows on which coupled congestion
         control operates merely limits the benefits derived from the
         mechanism.

   Shared bottlenecks do not change quickly:
         As per the definition above, a bottleneck depends on cross
         traffic, and since such traffic can heavily fluctuate,
         bottlenecks can change at a high frequency (e.g., there can be
         oscillation between two or more links).  This means that, when
         flows are partially routed along different paths, they may
         quickly change between sharing and not sharing a bottleneck.
         For simplicity, here it is assumed that a shared bottleneck is
         valid for a time interval that is significantly longer than the
         interval at which congestion controllers operate.  Note that,



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         for the only SBD mechanism defined in this document
         (multiplexing on the same five-tuple), the notion of a shared
         bottleneck stays correct even in the presence of fast traffic
         fluctuations: since all flows that are assumed to share a
         bottleneck are routed in the same way, if the bottleneck
         changes, it will still be shared.


4.  Architectural overview

   Figure 1 shows the elements of the architecture for coupled
   congestion control: the Flow State Exchange (FSE), Shared Bottleneck
   Detection (SBD) and Flows.  The FSE is a storage element.  It is
   passive in that it does not actively initiate communication with
   flows and the SBD; its only active role is internal state maintenance
   (e.g., an implementation could use soft state to remove a flow's data
   after long periods of inactivity).  Every time a flow's congestion
   control mechanism would normally update its sending rate, the flow
   instead updates information in the FSE and performs a query on the
   FSE, leading to a sending rate that can be different from what the
   congestion controller originally determined.  Using information
   about/from the currently active flows, SBD updates the FSE with the
   correct Flow State Identifiers (FSIs).



                          -------  <---  Flow 1
                          | FSE |  <---  Flow 2 ..
                          -------  <---  .. Flow N
                             ^
                             |             |
                          -------          |
                          | SBD |  <-------|
                          -------


             Figure 1: Coupled congestion control architecture

   Since everything shown in Figure 1 is assumed to operate on a single
   host (the sender) only, this document only describes aspects that
   have an influence on the resulting on-the-wire behavior.  It does,
   for instance, not define how many bits must be used to represent
   FSIs, or in which way the entities communicate.  Implementations can
   take various forms: for instance, all the elements in the figure
   could be implemented within a single application, thereby operating
   on flows generated by that application only.  Another alternative
   could be to implement both the FSE and SBD together in a separate
   process which different applications communicate with via some form



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   of Inter-Process Communication (IPC).  Such an implementation would
   extend the scope to flows generated by multiple applications.  The
   FSE and SBD could also be included in the Operating System kernel.


5.  Roles

   This section gives an overview of the roles of the elements of
   coupled congestion control, and provides an example of how coupled
   congestion control can operate.

5.1.  SBD

   SBD uses knowledge about the flows to determine which flows belong in
   the same Flow Group (FG), and assigns FGIs accordingly.  This
   knowledge can be derived from measurements, by considering
   correlations among measured delay and loss as an indication of a
   shared bottleneck, or it can be based on the simple assumption that
   packets sharing the same five-tuple (IP source and destination
   address, protocol, and transport layer port number pair) are
   typically routed in the same way.  The latter method is the only one
   specified in this document: SBD MUST consider all flows that use the
   same five-tuple to belong to the same FG.  This classification
   applies to certain tunnels, or RTP flows that are multiplexed over
   one transport (cf. [transport-multiplex]).  In one way or another,
   such multiplexing will probably be recommended for use with rtcweb
   [rtcweb-rtp-usage].

5.2.  FSE

   The FSE contains a list of all flows that have registered with it.
   For each flow, it stores:

   o  a unique flow number to identify the flow

   o  the FGI of the FG that it belongs to (based on the definitions in
      this document, a flow has only one bottleneck, and can therefore
      be in only one FG)

   o  a priority P, which here is assumed to be represented as a
      floating point number in the range from 0.1 (unimportant) to 1
      (very important).  A negative value is used to indicate that a
      flow has terminated.

   o  The calculated rate CR, i.e. the rate that was most recently
      calculated by the flow's congestion controller.





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   o  The desired rate DR.  This can be smaller than the calculated rate
      if the application feeding into the flow has less data to send
      than the congestion controller would allow.  In case of a greedy
      flow, DR must be set to CR.

   o  S_CR, the sum of the calculated rates of all flows in the same FG
      (including the flow itself), as seen by the flow during its last
      rate update.

   The information listed here is enough to implement the sample flow
   algorithm given below.  FSE implementations could easily be extended
   to store, e.g., a flow's current sending rate for statistics
   gathering or future potential optimizations.

5.3.  Flows

   Flows register themselves with SBD and FSE when they start,
   deregister from the FSE when they stop, and carry out an UPDATE
   function call every time their congestion controller calculates a new
   sending rate.  Via UPDATE, they provide the newly calculated rate and
   the desired rate (less than the calculated rate in case of non-greedy
   flows, the same otherwise).  UPDATE returns a rate that should be
   used instead of the rate that the congestion controller has
   determined.

   Below, an example algorithm is described.  While other algorithms
   could be used instead, the same algorithm must be applied to all
   flows.  The way the algorithm is described here, the operations are
   carried out by the flows, but they are the same for all flows.  This
   means that the algorithm could, for example, be implemented in a
   library that provides registration, deregistration functions and the
   UPDATE function.  To minimize the number of changes to existing
   applications, one could, however, also embed this functionality in
   the FSE element.

5.3.1.  Example algorithm

   (1)  When a flow starts, it registers itself with SBD and the FSE.
        CR and DR are initialized with the congestion controller's
        initial rate.  SBD will assign the correct FGI.  When a flow is
        assigned an FGI, its S_CR is initialized to be the sum of the
        calculated rates of all the flows in its FG.

   (2)  When a flow stops, it sets its DR to 0 and negates P.







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   (3)  Every time the flow's congestion controller determines a new
        sending rate new_CR, assuming the flow's new desired rate new_DR
        to be "infinity" in case of a greedy flow with an unknown
        maximum rate, the flow calls UPDATE, which carries out the
        following tasks:

        (a)  For all the flows in its FG (including itself), it
             calculates the sum of all the absolute values of all
             priorities, S_P, and the sum of all the calculated rates,
             new_S_CR.

        (b)  DELTA =3D new_CR-CR.  CR is updated if DELTA is negative, =
or
             if new_S_CR is smaller or equal to the flow's stored S_CR
             value.  This restriction on updating CR ensures that only
             one flow can make S_CR increase at a time.  To further
             avoid increasing S_CR, a flow tries to "take" bandwidth
             from terminated flows when it increases its rate: if DELTA
             is positive, for all other flows i in its FG, it checks if
             P(i) is negative (i.e. the flow has terminated).  It then
             reduces the CR(i) values of terminated flows such that
             these values do not become negative and the total amount of
             subtraction does not exceed DELTA.  The entry of a flow i
             is removed from the FSE and S_P is updated if CR(i) =3D 0.

        (c)  It updates new_S_CR using its own updated CR, and updates
             S_CR with new_S_CR.

        (d)  It updates DR to min(new_DR, CR).

        (e)  It initializes the total leftover rate TLO and its fair
             share of the total leftover rate TLO_FS to 0.  If it is
             greedy (i.e.  DR =3D CR), it then takes its fair share of =
the
             leftover rate of non-greedy or terminated flows: for all
             non-greedy flows i in its FG (flows that have DR(i) <
             CR(i)), it adds the leftover rate abs(P(i))/S_P * S_CR -
             DR(i) to TLO.  It calculates TLO_FS from TLO as P/S_P(G) *
             TLO, where S_P(G) is the sum of the priorities of all the
             greedy flows in the FG.

        (f)  It calculates the new sending rate as min(new_DR, P/S_P *
             S_CR + TLO_FS).  This applies an upper bound in case of an
             application-limited flow, gives the flow the correct share
             of the bandwidth based on its priority, and adds any
             potentially leftover bandwidth from non-greedy flows.

   The goals of the flow algorithm are to achieve prioritization,
   improve network utilization in the face of non-greedy flows, and
   impose limits on the increase behavior such that the negative impact



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   of multiple flows trying to increase their rate together is
   minimized.  It does that by assigning a flow a sending rate that may
   not be what the flow's congestion controller expected.  It therefore
   builds on the assumption that no significant inefficiencies arise
   from temporary non-greedy behavior or from quickly jumping to a rate
   that is higher than the congestion controller intended.  How
   problematic these issues really are depends on the controllers in use
   and requires careful per-controller experimentation.  The coupled
   congestion control mechanism described here also does not require all
   controllers to be equal; effects of heterogeneous controllers, or
   homogeneous controllers being in different states, are also subject
   to experimentation.

   There are more potential issues with the algorithm described here.
   Rule 3 b) leads to a conservative behavior: it ensures that only one
   flow at a time can increase the overall sending rate.  This rule is
   probably appropriate for situations where minimizing delay is the
   major goal, but it may not fit for all purposes; it also does not
   incorporate the magnitude by which a flow can increase its rate.
   Notably, despite this limitation on the overall rate of all flows per
   FGI, immediate rate jumps of single flows could become problematic
   when the FSE is used in a highly asynchronous manner, e.g. when flows
   have very different RTTs.

5.3.2.  Example operation

   In order to illustrate the operation of the coupled congestion
   control algorithm, this section presents a toy example of two flows
   that use it.  Let us assume that both flows traverse a common 10
   Mbit/s bottleneck and use a simplistic congestion controller that
   starts out with 1 Mbit/s, increases its rate by 1 Mbit/s in the
   absence of congestion and decreases it by 2 Mbit/s in the presence of
   congestion.  For simplicity, flows are assumed to always operate in a
   round-robin fashion.  Rate numbers below without units are assumed to
   be in Mbit/s.  For illustration purposes, the actual sending rate is
   also shown for every flow in FSE diagrams even though it is not
   really stored in the FSE.

   Flow #1 begins.  It is greedy and considers itself to have top
   priority.  This is the FSE after the flow algorithm's step 1:


   ---------------------------------------------
   | # | FGI |  P  |  CR  |  DR  | S_CR | Rate |
   |   |     |     |      |      |      |      |
   | 1 |  1  |  1  |   1  |   1  |   1  |   1  |
   ---------------------------------------------




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   Its congestion controller gradually increases its rate.  Eventually,
   at some point, the FSE should look like this:


   ---------------------------------------------
   | # | FGI |  P  |  CR  |  DR  | S_CR | Rate |
   |   |     |     |      |      |      |      |
   | 1 |  1  |  1  |  10  |  10  |  10  |  10  |
   ---------------------------------------------


   Now another flow joins.  It is also greedy, and has a lower priority
   (0.5):


   -----------------------------------------------
   | # | FGI |   P   |  CR  |  DR  | S_CR | Rate |
   |   |     |       |      |      |      |      |
   | 1 |  1  |   1   |  10  |  10  |  10  |  10  |
   | 2 |  1  |  0.5  |   1  |   1  |  11  |   1  |
   -----------------------------------------------


   Now assume that the first flow updates its rate to 8, because the
   total sending rate of 11 exceeds the total capacity.  Let us take a
   closer look at what happens in step 3 of the flow algorithm.


   new_CR =3D 8. new_DR =3D infinity.
   3 a) S_P =3D 1.5; new_S_CR =3D 11.
   3 b) DELTA is negative, hence CR =3D 8.
   3 c) new_S_CR =3D 9; S_CR =3D 9.
   3 d) DR =3D CR =3D 8.
   3 e) TLO =3D 0; there are no other flows with DR < CR.
   3 f) new sending rate: min(infinity, 1/1.5 * 9 + 0) =3D 6.

   The resulting FSE looks as follows:
   -----------------------------------------------
   | # | FGI |   P   |  CR  |  DR  | S_CR | Rate |
   |   |     |       |      |      |      |      |
   | 1 |  1  |   1   |   8  |   8  |   9  |   6  |
   | 2 |  1  |  0.5  |   1  |   1  |  11  |   1  |
   -----------------------------------------------


   The effect is that flow #1 is sending with 6 Mbit/s instead of the 8
   Mbit/s that the congestion controller derived.  Let us now assume
   that flow #2 updates its rate.  Its congestion controller detects



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   that the network is not fully saturated (the actual total sending
   rate is 6+1=3D7) and increases its rate.


   new_CR=3D2. new_DR =3D infinity.
   3 a) S_P =3D 1.5; new_S_CR =3D 9.
   3 b) DELTA is positive but new_S_CR < S_CR, hence CR =3D 2.
   3 c) new_S_CR =3D 10; S_CR =3D 10.
   3 d) DR =3D CR =3D 2.
   3 e) TLO =3D 0; there are no other flows with DR < CR.
   3 f) new sending rate: min(infinity, 0.5/1.5 * 10 + 0) =3D 3.33.

   The resulting FSE looks as follows:
   -----------------------------------------------
   | # | FGI |   P   |  CR  |  DR  | S_CR | Rate |
   |   |     |       |      |      |      |      |
   | 1 |  1  |   1   |   8  |   8  |   9  |   6  |
   | 2 |  1  |  0.5  |   2  |   2  |  10  | 3.33 |
   -----------------------------------------------


   The effect is that flow #2 is now sending with 3.33 Mbit/s, which is
   close to half of the rate of flow #1 and leads to a total utilization
   of 6(#1) + 3.33(#2) =3D 9.33 Mbit/s.  Flow #2's congestion controller
   has increased its rate faster than the controller actually expected.
   Now, flow #1 updates its rate.  Its congestion controller detects
   that the network is not fully saturated and increases its rate.
   Additionally, the application feeding into flow #1 limits the flow's
   sending rate to at most 2 Mbit/s.






















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   new_CR=3D9. new_DR=3D2.
   3 a) S_P =3D 1.5; new_S_CR =3D 10.
   3 b) DELTA is positive and new_S_CR > S_CR, hence CR is not updated
   (since flow #2 has just increased S_CR, flow #1 cannot also
   increase it in this iteration).
   3 c) new_S_CR =3D 10; S_CR =3D 10.
   3 d) DR =3D 2.
   3 e) TLO =3D 0; there are no other flows with DR < CR.
   3 f) new sending rate: min(2, 1/1.5 * 10 + 0) =3D 2. Note that,
   without the 2 Mbit/s limitation from the application, the new
   sending rate for flow #1 would now be 6.66 Mbit/s, leading to
   perfect network saturation (6.66 + 3.33 =3D approx. 10).

   The resulting FSE looks as follows:
   -----------------------------------------------
   | # | FGI |   P   |  CR  |  DR  | S_CR | Rate |
   |   |     |       |      |      |      |      |
   | 1 |  1  |   1   |   8  |   2  |  10  |   2  |
   | 2 |  1  |  0.5  |   2  |   2  |  10  | 3.33 |
   -----------------------------------------------


   Now, the total rate of the two flows is 2 + 3.33 =3D 5.33 Mbit/s, =
i.e.
   the network is significantly underutilized due to the limitation of
   flow #1.  Flow #2 updates its rate.  Its congestion controller
   detects that the network is not fully saturated and increases its
   rate.


   new_CR=3D3. new_DR =3D infinity.
   3 a) S_P =3D 1.5; new_S_CR =3D 10.
   3 b) DELTA is positive but new_S_CR =3D S_CR, hence CR =3D 3.
   3 c) new_S_CR =3D 11; S_CR =3D 11.
   3 d) DR =3D 3.
   3 e) TLO =3D 0; flow #1 has DR < CR, hence TLO +=3D 1/1.5 * 11
   - 2 =3D 5.33, and TLO_FS =3D 0.5/0.5 * 5.33 =3D 5.33.
   3 f) new sending rate: min(infinity, 0.5/1.5*11 + 5.33) =3D 9.

   The resulting FSE looks as follows:
   -----------------------------------------------
   | # | FGI |   P   |  CR  |  DR  | S_CR | Rate |
   |   |     |       |      |      |      |      |
   | 1 |  1  |   1   |   8  |   2  |  10  |   2  |
   | 2 |  1  |  0.5  |   3  |   3  |  11  |   9  |
   -----------------------------------------------


   Now, the total rate of the two flows is 2 + 9 =3D 11 Mbit/s, =
exceeding



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   the total capacity by the amount of 1 Mbit/s by which the congestion
   controller of flow #2 has increased its rate.  Note that, had flow #1
   been greedy, the same total rate would have resulted after this
   iteration.  Finally, flow #1 terminates.  It sets P to -1 and DR to
   0.  Let us assume that it terminated late enough for flow #2 to still
   experience the network in a congested state, i.e. flow #2 decreases
   its rate in the next iteration.


   new_CR =3D 1. new_DR =3D infinity.
   3 a) S_P =3D 1.5; new_S_CR =3D 11.
   3 b) DELTA is negative hence CR =3D 1.
   3 c) new_S_CR =3D 9; S_CR =3D 9.
   3 d) DR =3D 1.
   3 e) TLO =3D 0; flow #1 has DR < CR, hence TLO +=3D 1/1.5 * 9 - 0 =3D =
6.
   TLO_FS=3D0.5/0.5 * 6 =3D 6.
   3 f) new sending rate: min(infinity, 0.5/1.5 * 9 + 6) =3D 9.

   The resulting FSE looks as follows:
   -----------------------------------------------
   | # | FGI |   P   |  CR  |  DR  | S_CR | Rate |
   |   |     |       |      |      |      |      |
   | 1 |  1  |   -1  |   8  |   0  |  10  |   2  |
   | 2 |  1  |  0.5  |   1  |   1  |   9  |   9  |
   -----------------------------------------------


   Now, flow #2 increases its rate as the network is underutilized.  The
   CR of the terminated flow is reduced so that the S_CR is adjusted.


   new_CR =3D 2. new_DR =3D infinity.
   3 a) S_P =3D 1.5; new_S_CR =3D 9.
   3 b) DELTA is positive but new_S_CR =3D S_CR, hence CR =3D 2.
   P(1) is negative, hence CR(1) =3D 8-1 =3D 7.
   3 c) new_S_CR =3D 9; S_CR =3D 9.
   3 d) DR =3D 2.
   3 e) TLO =3D 0; flow #1 has DR < CR, hence TLO +=3D 1/1.5 * 9 - 0 =3D =
6.
   TLO_FS=3D0.5/0.5 * 6 =3D 6.
   3 f) new sending rate: min(infinity, 0.5/1.5 * 9 + 6) =3D 9.

   The resulting FSE looks as follows:
   -----------------------------------------------
   | # | FGI |   P   |  CR  |  DR  | S_CR | Rate |
   |   |     |       |      |      |      |      |
   | 1 |  1  |   -1  |   7  |   0  |  10  |   0  |
   | 2 |  1  |  0.5  |   2  |   2  |   9  |   9  |
   -----------------------------------------------



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   Flow #2's congestion controller gradually increases its rate, and
   flow #1's CR is reduced to adjust the S_CR.  Eventually, at some
   point, FSE should look like the following:


   -----------------------------------------------
   | # | FGI |   P   |  CR  |  DR  | S_CR | Rate |
   |   |     |       |      |      |      |      |
   | 1 |  1  |   -1  |   1  |   0  |  10  |   0  |
   | 2 |  1  |  0.5  |   8  |   8  |   9  |   9  |
   -----------------------------------------------

   and then, for flow #2:

   new_CR =3D 9. new_DR =3D infinity.
   3 a) S_P =3D 1.5; new_S_CR =3D 9.
   3 b) DELTA is positive but new_S_CR =3D S_CR, hence CR =3D 9.
   P(1) is negative, hence CR(1) =3D 1-1 =3D 0, and therefore
   the entry of flow #1 is deleted and S_P is updated to 0.5.
   3 c) new_S_CR =3D 9; S_CR =3D 9.
   3 d) DR =3D 9.
   3 e) TLO =3D 0; there are no other flows.
   3 f) new sending rate: min(infinity, 0.5/0.5 * 9 + 0) =3D 9.

   The resulting FSE looks as follows:
   -----------------------------------------------
   | # | FGI |   P   |  CR  |  DR  | S_CR | Rate |
   |   |     |       |      |      |      |      |
   | 2 |  1  |  0.5  |   9  |   9  |   9  |   9  |
   -----------------------------------------------



6.  Acknowledgements

   This document has benefitted from discussions with and feedback from
   Stein Gjessing, David Hayes, Andreas Petlund, and David Ros (who also
   gave the FSE its name).


7.  IANA Considerations

   This memo includes no request to IANA.


8.  Security Considerations

   In scenarios where the architecture described in this document is



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   applied across applications, various cheating possibilities arise:
   e.g., supporting wrong values for the calculated rate, the desired
   rate, or the priority of a flow.  In the worst case, such cheating
   could either prevent other flows from sending or make them send at a
   rate that is unreasonably large.  The end result would be unfair
   behavior at the network bottleneck, akin to what could be achieved
   with any UDP based application.  Hence, since this is no worse than
   UDP in general, there seems to be no significant harm in using this
   in the absence of UDP rate limiters.

   In the case of a single-user system, it should also be in the
   interest of any application programmer to give the user the best
   possible experience by using reasonable flow priorities or even
   letting the user choose them.  In a multi-user system, this interest
   may not be given, and one could imagine the worst case of an "arms
   race" situation, where applications end up setting their priorities
   to the maximum value.  If all applications do this, the end result is
   a fair allocation in which the priority mechanism is implicitly
   eliminated, and no major harm is done.


9.  References

9.1.  Normative References

   [RFC2119]  Bradner, S., "Key words for use in RFCs to Indicate
              Requirement Levels", BCP 14, RFC 2119, March 1997.

   [RFC2140]  Touch, J., "TCP Control Block Interdependence", RFC 2140,
              April 1997.

   [RFC3124]  Balakrishnan, H. and S. Seshan, "The Congestion Manager",
              RFC 3124, June 2001.

9.2.  Informative References

   [rtcweb-rtp-usage]
              Perkins, C., Westerlund, M., and J. Ott, "Web Real-Time
              Communication (WebRTC): Media Transport and Use of RTP",
              draft-ietf-rtcweb-rtp-usage-05.txt (work in progress),
              October 2012.

   [rtcweb-usecases]
              Holmberg, C., Hakansson, S., and G. Eriksson, "Web Real-
              Time Communication Use-cases and Requirements",
              draft-ietf-rtcweb-use-cases-and-requirements-10.txt (work
              in progress), December 2012.




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   [transport-multiplex]
              Westerlund, M. and C. Perkins, "Multiple RTP Sessions on a
              Single Lower-Layer Transport",
              draft-westerlund-avtcore-transport-multiplexing-04.txt
              (work in progress), October 2012.


Appendix A.  Changes from -00 to -01

   Updated the example algorithm and its operation to let flows fairly
   take the leftover rate from other non-greedy and/or terminated flows.


Authors' Addresses

   Michael Welzl
   University of Oslo
   PO Box 1080 Blindern
   Oslo,   N-0316
   Norway

   Phone: +47 22 85 24 20
   Email: michawe@ifi.uio.no


   Safiqul Islam
   University of Oslo
   PO Box 1080 Blindern
   Oslo,   N-0316
   Norway

   Phone: +47 22 84 08 37
   Email: safiquli@ifi.uio.no


















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Hi Zahed,

>  * Req 8: This requirement focuses on long-lived TCP flows but I think =
it should focus on flows with known congestion control algorithm. in the =
fast phase, RMCAT will publish (hopefully) more than one experimental CC =
algorithms and I think it is important to look how one of the candidate =
is behaving when competing with other candidate.  But if we want to =
focus on the behavior against competing TCP flows then the requirement =
should say "It should attempt to avoid bandwidth 'collapse' when facing =
a saturating TCP flow or flows but it should not starve TCP flow (should =
not grab more than necessary)" or something along this theme. And I =
would also like to get rid of the prefix terms "short-lived" and =
"Long-lived" TCP flow as I think from a congestion controller point of =
view it is hardly possible to distinguish between them unless we have =
some explicit information from the shared bottleneck.

bringing up the terms "long-lived" and "short lived" triggered an =
outside the box thought on the topic.  Do you think its reasonable ro =
feasable to use numbers associated with average call/flow duration as a =
point of reference of what is considered a "typical real-time flow"?  =
Part of the reason that I bring this up is that I understand that one =
(maybe more) popular VoIP applications starts with a rather aggressive =
offered load, and then after some period of time (versus actual =
feedback) the application backs off to some target measure that takes =
into account accumulated path characteristics (eg, delay, loss).  =
Perhaps the time delay is to avoid oscilation.

One can argue that the initial stage of such an approach is not fair, or =
its excessively greedy (for lack of an alternative term), but that =
perhaps over time, or an average call/flow duration, it is fair.  I've =
seen numbers that range from 1.8 minutes to 4 minutes in average call =
duration, and so perhaps this could be a target reference point to gauge =
how nice various algorithms work with each other.

-ken




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From: Fu Jiantao <fuji246@gmail.com>
To: Michael Welzl <michawe@ifi.uio.no>
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Subject: Re: [rmcat] Fwd: New Version Notification for draft-welzl-rmcat-coupled-cc-00.txt
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Hi Michael,

   Following are my comments:

  1. multiplexing make the SBD simple, but it may take more effort for
classification in QoS. delay and loss correlations works fine for UDP
connection, but for having both TCP and UDP connections in police
network(drop overlimit packets directly), it seems no easy way to do SDB.

  2. the new_S_CR seems not updated with new_CR in step 3, from the
examples, it should be the new_S_CR in the last round, and I guess new_S_CR
is the common state variable sharing in each round, only so it can be used
to avoid only make S_CR increase at once.

  3. The example algorithm will only update flow(i)'s state when its cc
changed the calculated rate, other flow's state is leave untouched. There
are other algorithms, for example, updates all flow's state when flow(i)'s
state changed. What's the reason to prefer the first one, and I think the
second one is more reasonable and simple to implement.

Thanks

Jeromy


2013/1/19 Michael Welzl <michawe@ifi.uio.no>

> Hi all,
>
> See below the announcement of a draft I just posted, to address the
> milestone on "identifying and controlling groups of flows".
> I'm curious to hear your thoughts!
>
> Cheers,
> Michael
>
>
> Begin forwarded message:
>
> > From: internet-drafts@ietf.org
> > Subject: New Version Notification for draft-welzl-rmcat-coupled-cc-00.txt
> > Date: January 19, 2013 10:56:14 AM GMT+01:00
> > To: michawe@ifi.uio.no
> >
> >
> > A new version of I-D, draft-welzl-rmcat-coupled-cc-00.txt
> > has been successfully submitted by Michael Welzl and posted to the
> > IETF repository.
> >
> > Filename:      draft-welzl-rmcat-coupled-cc
> > Revision:      00
> > Title:                 Coupled congestion control for RTP media
> > Creation date:         2013-01-19
> > WG ID:                 Individual Submission
> > Number of pages: 14
> > URL:
> http://www.ietf.org/internet-drafts/draft-welzl-rmcat-coupled-cc-00.txt
> > Status:
> http://datatracker.ietf.org/doc/draft-welzl-rmcat-coupled-cc
> > Htmlized:
> http://tools.ietf.org/html/draft-welzl-rmcat-coupled-cc-00
> >
> >
> > Abstract:
> >   When multiple congestion controlled RTP sessions traverse the same
> >   network bottleneck, it can be beneficial to combine their controls
> >   such that the total on-the-wire behavior is improved.  This document
> >   describes such a method for flows that have the same sender, in a way
> >   that is as flexible and simple as possible while minimizing the
> >   amount of changes needed to existing RTP applications.
> >
> >
> >
> >
> > The IETF Secretariat
> >
>
>

--e89a8ff1c090ea883204d741572e
Content-Type: text/html; charset=ISO-8859-1
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<div dir=3D"ltr"><div><div><div><div><div><div><div>Hi Michael,<br><br></di=
v>=A0=A0 Following are my comments:<br><br></div>=A0 1. multiplexing make t=
he SBD simple, but it may take more effort for classification in QoS. delay=
 and loss correlations works fine for UDP connection, but for having both T=
CP and UDP connections in police network(drop overlimit packets directly), =
it seems no easy way to do SDB.<br>

<br></div>=A0 2. the new_S_CR seems not updated with new_CR in step 3, from=
 the examples, it should be the new_S_CR in the last round, and I guess new=
_S_CR is the common state variable sharing in each round, only so it can be=
 used to avoid only make S_CR increase at once.<br>
<br></div>=A0 3. The example algorithm will only update flow(i)&#39;s state=
 when its cc changed the calculated rate, other flow&#39;s state is leave u=
ntouched. There are other algorithms, for example, updates all flow&#39;s s=
tate when flow(i)&#39;s state changed. What&#39;s the reason to prefer the =
first one, and I think the second one is more reasonable and simple to impl=
ement.<br>
<br></div>Thanks<br></div><br></div>Jeromy<br></div><div class=3D"gmail_ext=
ra"><br><br><div class=3D"gmail_quote">2013/1/19 Michael Welzl <span dir=3D=
"ltr">&lt;<a href=3D"mailto:michawe@ifi.uio.no" target=3D"_blank">michawe@i=
fi.uio.no</a>&gt;</span><br>
<blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1p=
x #ccc solid;padding-left:1ex">Hi all,<br>
<br>
See below the announcement of a draft I just posted, to address the milesto=
ne on &quot;identifying and controlling groups of flows&quot;.<br>
I&#39;m curious to hear your thoughts!<br>
<br>
Cheers,<br>
Michael<br>
<br>
<br>
Begin forwarded message:<br>
<br>
&gt; From: <a href=3D"mailto:internet-drafts@ietf.org">internet-drafts@ietf=
.org</a><br>
&gt; Subject: New Version Notification for draft-welzl-rmcat-coupled-cc-00.=
txt<br>
&gt; Date: January 19, 2013 10:56:14 AM GMT+01:00<br>
&gt; To: <a href=3D"mailto:michawe@ifi.uio.no">michawe@ifi.uio.no</a><br>
&gt;<br>
&gt;<br>
&gt; A new version of I-D, draft-welzl-rmcat-coupled-cc-00.txt<br>
&gt; has been successfully submitted by Michael Welzl and posted to the<br>
&gt; IETF repository.<br>
&gt;<br>
&gt; Filename: =A0 =A0 =A0draft-welzl-rmcat-coupled-cc<br>
&gt; Revision: =A0 =A0 =A000<br>
&gt; Title: =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 Coupled congestion control for =
RTP media<br>
&gt; Creation date: =A0 =A0 =A0 =A0 2013-01-19<br>
&gt; WG ID: =A0 =A0 =A0 =A0 =A0 =A0 =A0 =A0 Individual Submission<br>
&gt; Number of pages: 14<br>
&gt; URL: =A0 =A0 =A0 =A0 =A0 =A0 <a href=3D"http://www.ietf.org/internet-d=
rafts/draft-welzl-rmcat-coupled-cc-00.txt" target=3D"_blank">http://www.iet=
f.org/internet-drafts/draft-welzl-rmcat-coupled-cc-00.txt</a><br>
&gt; Status: =A0 =A0 =A0 =A0 =A0<a href=3D"http://datatracker.ietf.org/doc/=
draft-welzl-rmcat-coupled-cc" target=3D"_blank">http://datatracker.ietf.org=
/doc/draft-welzl-rmcat-coupled-cc</a><br>
&gt; Htmlized: =A0 =A0 =A0 =A0<a href=3D"http://tools.ietf.org/html/draft-w=
elzl-rmcat-coupled-cc-00" target=3D"_blank">http://tools.ietf.org/html/draf=
t-welzl-rmcat-coupled-cc-00</a><br>
&gt;<br>
&gt;<br>
&gt; Abstract:<br>
&gt; =A0 When multiple congestion controlled RTP sessions traverse the same=
<br>
&gt; =A0 network bottleneck, it can be beneficial to combine their controls=
<br>
&gt; =A0 such that the total on-the-wire behavior is improved. =A0This docu=
ment<br>
&gt; =A0 describes such a method for flows that have the same sender, in a =
way<br>
&gt; =A0 that is as flexible and simple as possible while minimizing the<br=
>
&gt; =A0 amount of changes needed to existing RTP applications.<br>
&gt;<br>
&gt;<br>
&gt;<br>
&gt;<br>
&gt; The IETF Secretariat<br>
&gt;<br>
<br>
</blockquote></div><br></div>

--e89a8ff1c090ea883204d741572e--

From michawe@ifi.uio.no  Wed Mar  6 05:53:19 2013
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Hi Jeromy,

Thanks a whole lot for reading this and for your comments! More below -

On 6. mars 2013, at 14:12, Fu Jiantao wrote:

> Hi Michael,
>=20
>    Following are my comments:
>=20
>   1. multiplexing make the SBD simple, but it may take more effort for =
classification in QoS. delay and loss correlations works fine for UDP =
connection, but for having both TCP and UDP connections in police =
network(drop overlimit packets directly), it seems no easy way to do =
SDB.

As for delay and loss correlations: even that is not easy, but we're =
working on something there. I agree that policers may make it impossible =
to detect a SBD - but, for example, if flow 1 gets a lot of loss due to =
a policer and flow 2 doesn't, losses of the two flows won't correlate. =
Even if they share the same bottleneck, a correlation based scheme would =
then judge the flows as not sharing a bottleneck, which is a false =
negative and not a big problem: for all flows that are detected as not =
sharing a bottleneck, you have the behavior of "today".... the FSE =
scheme simply doesn't apply, you lose some extra benefit.


>   2. the new_S_CR seems not updated with new_CR in step 3, from the =
examples, it should be the new_S_CR in the last round, and I guess =
new_S_CR is the common state variable sharing in each round, only so it =
can be used to avoid only make S_CR increase at once.

No, new_S_CR is really a temporary variable. Maybe we should have called =
it tmp_S_CR...
new_S_CR is the newly calculated sum, as a result of all the current CR =
values in the FSE. S_CR however is a historical stored value per flow, =
which is updated every time the flow enters step 3. Thus, if new_S_CR is =
different from S_CR, some other flow Y has changed its CR since the last =
time flow X has gone through step 3.


>   3. The example algorithm will only update flow(i)'s state when its =
cc changed the calculated rate, other flow's state is leave untouched. =
There are other algorithms, for example, updates all flow's state when =
flow(i)'s state changed. What's the reason to prefer the first one, and =
I think the second one is more reasonable and simple to implement.

I'm used to congestion controllers that update their rate all the time, =
this guided my thinking. Maybe that's wrong for rmcat, where feedback is =
minimized... I also thought that this is the simplest approach, but =
maybe it's just fine to change a rate even when the controller doesn't =
do anything to it. We'll have to consider that. Anyway, the main point =
here is to come up with something simple and working. There are =
something between 7000 and 9000 possible improvements to this, I guess   =
:-)

Thanks again,

cheers,
Michael


>=20
> Thanks
>=20
> Jeromy
>=20
>=20
> 2013/1/19 Michael Welzl <michawe@ifi.uio.no>
> Hi all,
>=20
> See below the announcement of a draft I just posted, to address the =
milestone on "identifying and controlling groups of flows".
> I'm curious to hear your thoughts!
>=20
> Cheers,
> Michael
>=20
>=20
> Begin forwarded message:
>=20
> > From: internet-drafts@ietf.org
> > Subject: New Version Notification for =
draft-welzl-rmcat-coupled-cc-00.txt
> > Date: January 19, 2013 10:56:14 AM GMT+01:00
> > To: michawe@ifi.uio.no
> >
> >
> > A new version of I-D, draft-welzl-rmcat-coupled-cc-00.txt
> > has been successfully submitted by Michael Welzl and posted to the
> > IETF repository.
> >
> > Filename:      draft-welzl-rmcat-coupled-cc
> > Revision:      00
> > Title:                 Coupled congestion control for RTP media
> > Creation date:         2013-01-19
> > WG ID:                 Individual Submission
> > Number of pages: 14
> > URL:             =
http://www.ietf.org/internet-drafts/draft-welzl-rmcat-coupled-cc-00.txt
> > Status:          =
http://datatracker.ietf.org/doc/draft-welzl-rmcat-coupled-cc
> > Htmlized:        =
http://tools.ietf.org/html/draft-welzl-rmcat-coupled-cc-00
> >
> >
> > Abstract:
> >   When multiple congestion controlled RTP sessions traverse the same
> >   network bottleneck, it can be beneficial to combine their controls
> >   such that the total on-the-wire behavior is improved.  This =
document
> >   describes such a method for flows that have the same sender, in a =
way
> >   that is as flexible and simple as possible while minimizing the
> >   amount of changes needed to existing RTP applications.
> >
> >
> >
> >
> > The IETF Secretariat
> >
>=20
>=20


--Apple-Mail=_C0445B9E-F49E-4EB6-BDAF-BB1C34ACBD57
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<html><head></head><body style=3D"word-wrap: break-word; =
-webkit-nbsp-mode: space; -webkit-line-break: after-white-space; ">Hi =
Jeromy,<div><br></div><div>Thanks a whole lot for reading this and for =
your comments! More below -</div><div><br></div><div><div><div>On 6. =
mars 2013, at 14:12, Fu Jiantao wrote:</div><br =
class=3D"Apple-interchange-newline"><blockquote type=3D"cite"><div =
dir=3D"ltr"><div><div><div><div><div><div><div>Hi =
Michael,<br><br></div>&nbsp;&nbsp; Following are my =
comments:<br><br></div>&nbsp; 1. multiplexing make the SBD simple, but =
it may take more effort for classification in QoS. delay and loss =
correlations works fine for UDP connection, but for having both TCP and =
UDP connections in police network(drop overlimit packets directly), it =
seems no easy way to do =
SDB.<br></div></div></div></div></div></div></blockquote><div><br></div><d=
iv>As for delay and loss correlations: even that is not easy, but we're =
working on something there. I agree that policers may make it impossible =
to detect a SBD - but, for example, if flow 1 gets a lot of loss due to =
a policer and flow 2 doesn't, losses of the two flows won't correlate. =
Even&nbsp;if they share the same bottleneck, a correlation based scheme =
would then judge the flows as not sharing a bottleneck, which is a false =
negative and not a big problem: for all flows that are detected as not =
sharing a bottleneck, you have the behavior of "today".... the FSE =
scheme simply doesn't apply, you lose some extra =
benefit.</div><div><br></div><br><blockquote type=3D"cite"><div =
dir=3D"ltr"><div><div><div><div>&nbsp; 2. the new_S_CR seems not updated =
with new_CR in step 3, from the examples, it should be the new_S_CR in =
the last round, and I guess new_S_CR is the common state variable =
sharing in each round, only so it can be used to avoid only make S_CR =
increase at =
once.<br></div></div></div></div></div></blockquote><div><br></div>No, =
new_S_CR is really a temporary variable. Maybe we should have called it =
tmp_S_CR...</div><div>new_S_CR is the newly calculated sum, as a result =
of all the current CR values in the FSE. S_CR however is a historical =
stored value per flow, which is updated every time the flow enters step =
3. Thus, if new_S_CR is different from S_CR, some other flow Y has =
changed its CR since the last time flow X has gone through step =
3.</div><div><br></div><div><br><blockquote type=3D"cite"><div =
dir=3D"ltr"><div><div><div>&nbsp; 3. The example algorithm will only =
update flow(i)'s state when its cc changed the calculated rate, other =
flow's state is leave untouched. There are other algorithms, for =
example, updates all flow's state when flow(i)'s state changed. What's =
the reason to prefer the first one, and I think the second one is more =
reasonable and simple to =
implement.<br></div></div></div></div></blockquote><div><br></div><div>I'm=
 used to congestion controllers that update their rate all the time, =
this guided my thinking. Maybe that's wrong for rmcat, where feedback is =
minimized... I also thought that this is the simplest approach, but =
maybe it's just fine to change a rate even when the controller doesn't =
do anything to it. We'll have to consider that. Anyway, the main point =
here is to come up with something simple and working. There are =
something between 7000 and 9000 possible improvements to this, I guess =
&nbsp; :-)</div><div><br></div><div>Thanks =
again,</div><div><br></div><div>cheers,</div><div>Michael</div><div><br></=
div><br><blockquote type=3D"cite"><div dir=3D"ltr"><div><div><div>
<br></div>Thanks<br></div><br></div>Jeromy<br></div><div =
class=3D"gmail_extra"><br><br><div class=3D"gmail_quote">2013/1/19 =
Michael Welzl <span dir=3D"ltr">&lt;<a href=3D"mailto:michawe@ifi.uio.no" =
target=3D"_blank">michawe@ifi.uio.no</a>&gt;</span><br>
<blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 =
.8ex;border-left:1px #ccc solid;padding-left:1ex">Hi all,<br>
<br>
See below the announcement of a draft I just posted, to address the =
milestone on "identifying and controlling groups of flows".<br>
I'm curious to hear your thoughts!<br>
<br>
Cheers,<br>
Michael<br>
<br>
<br>
Begin forwarded message:<br>
<br>
&gt; From: <a =
href=3D"mailto:internet-drafts@ietf.org">internet-drafts@ietf.org</a><br>
&gt; Subject: New Version Notification for =
draft-welzl-rmcat-coupled-cc-00.txt<br>
&gt; Date: January 19, 2013 10:56:14 AM GMT+01:00<br>
&gt; To: <a href=3D"mailto:michawe@ifi.uio.no">michawe@ifi.uio.no</a><br>
&gt;<br>
&gt;<br>
&gt; A new version of I-D, draft-welzl-rmcat-coupled-cc-00.txt<br>
&gt; has been successfully submitted by Michael Welzl and posted to =
the<br>
&gt; IETF repository.<br>
&gt;<br>
&gt; Filename: &nbsp; &nbsp; &nbsp;draft-welzl-rmcat-coupled-cc<br>
&gt; Revision: &nbsp; &nbsp; &nbsp;00<br>
&gt; Title: &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; =
Coupled congestion control for RTP media<br>
&gt; Creation date: &nbsp; &nbsp; &nbsp; &nbsp; 2013-01-19<br>
&gt; WG ID: &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; =
Individual Submission<br>
&gt; Number of pages: 14<br>
&gt; URL: &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; <a =
href=3D"http://www.ietf.org/internet-drafts/draft-welzl-rmcat-coupled-cc-0=
0.txt" =
target=3D"_blank">http://www.ietf.org/internet-drafts/draft-welzl-rmcat-co=
upled-cc-00.txt</a><br>
&gt; Status: &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;<a =
href=3D"http://datatracker.ietf.org/doc/draft-welzl-rmcat-coupled-cc" =
target=3D"_blank">http://datatracker.ietf.org/doc/draft-welzl-rmcat-couple=
d-cc</a><br>
&gt; Htmlized: &nbsp; &nbsp; &nbsp; &nbsp;<a =
href=3D"http://tools.ietf.org/html/draft-welzl-rmcat-coupled-cc-00" =
target=3D"_blank">http://tools.ietf.org/html/draft-welzl-rmcat-coupled-cc-=
00</a><br>
&gt;<br>
&gt;<br>
&gt; Abstract:<br>
&gt; &nbsp; When multiple congestion controlled RTP sessions traverse =
the same<br>
&gt; &nbsp; network bottleneck, it can be beneficial to combine their =
controls<br>
&gt; &nbsp; such that the total on-the-wire behavior is improved. =
&nbsp;This document<br>
&gt; &nbsp; describes such a method for flows that have the same sender, =
in a way<br>
&gt; &nbsp; that is as flexible and simple as possible while minimizing =
the<br>
&gt; &nbsp; amount of changes needed to existing RTP applications.<br>
&gt;<br>
&gt;<br>
&gt;<br>
&gt;<br>
&gt; The IETF Secretariat<br>
&gt;<br>
<br>
</blockquote></div><br></div>
</blockquote></div><br></div></body></html>=

--Apple-Mail=_C0445B9E-F49E-4EB6-BDAF-BB1C34ACBD57--

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To: rmcat WG <rmcat@ietf.org>
Thread-Topic: slides, please!
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Subject: [rmcat] slides, please!
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Hi,

since RMCAT is in the first slot on Monday morning, I will need the FINAL s=
lides from all presenters by *** 6PM on Sunday *** (local time Orlando) to =
upload them.

Thanks,
Lars=

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Subject: [rmcat] Comments on eval draft (draft-singh-rmcat-cc-eval-02)
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Hi,

Now that I have finished reading the evaluation draft, I think the 
recent version look much better than the previous one however I have 
several comments, reflections, questions related to different sections 
of the draft.

Warning !!!!! long email ahead. Sorry, but I could not help it.

Metrics:

·I understand that the RFC 5166 is referred for the description of the 
metrics mentioned in this draft. However, my take is that RFC 5166 was 
written to address a wider scope than what we are trying to do in RMCAT 
hence IMHO it is best to scope the definition and description down to 
the particular scope of RMCAT- interactive real-time media over RTP. 
This means this draft should actually include proper definition of the 
metrics, methodologies to realize them within this document (or in 
additional documents).

·Barely mentioning the metrics does not really help. We need to define 
proper methodologies to use them. Some of the metrics are easy to 
visualize but some of them are not (see more below). We need to 
describe/define a common understanding of all the metrics and how to use 
them in evaluation.

·Delay: this version excludes "Delay" as metric. Could you explain why?

·Minimum oscillation in encoding rate: This is related to QoE and 
minimizing oscillationis an wanted behavior. How do we measure 
oscillation? RFC 5166 gives some ref. but how well do they represent 
interactive real-time communication?  We have variable bitrate encoder 
for video whose output will oscillate depending on scenes to encode. 
What do we do there? We perhaps will need to measure the oscillation in 
the target bitrate set bythe Congestion Control (CC) algorithm but very 
often there will be mismatch between the target rate and actual encoding 
rate.

·Reactivity to transient events: We canreact to transient event only if 
we know that it is a transient event. In testbeds, perhaps we can do 
this by setting up particular test for a particular transient event and 
very soon we will find that there is no "silver bullet" to dodge all 
favorite transient events. What can we do then? one way to handle this 
could be to setup a sequence tests of transient events and weight them 
to evaluate the reactivity to the transient events. Please let me know 
if I am missing something here.

·Packet loss and discard rate: We have three categories here- 1) loss at 
transport layer (TL), 2) loss due to late arrival (LL) and 3) residual 
loss  (RL = TL+LL).  RL will also have effects of application depended 
techniques used for countering the packet loss (for example, FEC, RTP 
retransmission). Where TL perhaps a good metric for network, RL is 
perhaps what the application care about most. We will need to define a 
relation here. Also this metric is not about how the CC algorithms 
calculate the packet loss rate (--  different algorithm may calculate 
loss event/rate differently- per RTCP Send Report/Receiver Report, 
fractional loss over a time period, per RTT etc. etc.) but about how we 
want to measure the loss, right? then, lets define it concretely.

·Bandwidth utilization: This is a metric I have hard time to grasp. 
Whatdo we actuallywant to achieve/measure with this metric? I perhaps 
understand it if I have a fixed capacity link but in real-time 
communication the input data itself can starve*//*,resulting  inless 
utilization of the available bandwidth. And for cellular networks where 
the users have different capacity view even within a cell depending on 
their radio condition, speed, distance etc. etc. ; even in a aggregate 
level (per cell measurement) will depend on what is going on other 
cells. Moreover, the view of available bandwidth changes all the time, I 
don't see how we can visualize the available bandwidth utilization. I am 
not aware of any model that actually does it. Please let me know if I 
have missed something here. The thing is if we can't really measure a 
metric in a proper/straight forward way then there is not point of 
including it.

·Convergence time: I would like add this metric in the preferred metric 
list. This is also related to fairness metric as it gives a view about 
how fast the algorithm can reach a stable bitrate when competing with 
flows with same CC algorithm.

Guideline:

·4.3 : It seems that it is necessary to have some reference cases to 
verify against.

·4.4 : Is this the plan to test against all possible access types?

·4.7 : any idea on how to "highlight the bottleneck capacity share of 
each RTP flow" for wireless/cellular networks?

·This also says "However, long-lived TCP flows may starve out the RTP 
flow depending on router queue length. In the latter case the proposed 
congestion control for RTP should be as aggressive as standard TCP"- how 
realistic is this? at some point (if there are buffer-bloats) the RMCAT 
will have to give up as it has certain delay budget to meet.

·missing guidelines

·Congested feedback channel : the CC algorithm should be evaluated 
against congested feedback channel. Could be included as separate 
guideline or a part of 4.5.

·Different start up values: as the requirement draft mentioned several 
options for start-up values, the algorithms should be tested. One can 
argue that different CC algorithm will have different strategies but 
then it should be judged against fairness and convergence time.

·Fairness to other RMCAT proposed algorithms: as part of fairness guideline.

Eval scenarios:

·This section should really list a "must be" evaluated test scenarios 
not example scenarios. Those "must be" evaluated scenarios should take 
the guides lines into account and includetherelatedmetrics so that 
people have a common view on how to evaluate their algorithms. Any kind 
of effort to write down example scenarios should be waste of time and 
energy. Of course, there could be optional test cases and people can go 
an add their scenarios and if some scenarios become very useful then 
include those in the "must-be" evaluated sets.

·Video starting rate is 128 kbps. Feels like the default behavior is 
slow start, why?

·Yes, I would like to get a value(s) for queue length when AQM is used 
along with the AQM behavior.

·I have hard time fitting a radio scenario into dumbbell topology with 
the current definition in the draft.

BR

-- 

Zahed

==================================================
ANM ZAHEDUZZAMAN SARKER


Ericsson AB
Multimedia Technologies
Laboratoriegränd 11
97128 Luleå, Sweden
Phone +46 10 717 37 43
Fax +46 920 996 21
SMS/MMS +46 76 115 37 43
zaheduzzaman.sarker@ericsson.com
www.ericsson.com

==================================================


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    <p class="MsoNormal"
      style="mso-layout-grid-align:none;text-autospace:none"><font
        face="Courier New, Courier, monospace"><span style="font-size:
          12pt;" lang="SV">Hi,<br>
          <br>
          Now that I have finished reading the evaluation draft, I think
          the recent
          version look much better than the previous one however I have
          several comments,
          reflections, questions related to different sections of the
          draft.<br>
          <br>
          Warning !!!!! long email ahead. Sorry, but I could not help
          it.<br>
          <br>
          Metrics:<o:p></o:p></span></font></p>
    <p>
    </p>
    <p class="MsoNormal"
      style="margin-top:5.0pt;margin-right:0cm;margin-bottom:5.0pt;
      margin-left:36.0pt;text-indent:-18.0pt;mso-list:l0 level1
      lfo2;mso-layout-grid-align:
      none;text-autospace:none"><!--[if !supportLists]--><font
        face="Courier New, Courier, monospace"><span style="font-size:
          12pt;" lang="SV"><span style="mso-list:Ignore">&middot;<span
              style="font-style: normal; font-variant: normal;
              font-weight: normal; font-size: 7pt; line-height: normal;
              font-size-adjust: none; font-stretch: normal;
              -moz-font-feature-settings: normal;
              -moz-font-language-override: normal;">&nbsp;&nbsp; </span></span></span><span
          style="font-size: 12pt;" lang="SV">I understand that the RFC
          5166 is referred for the description of the metrics mentioned
          in this draft. However, my take
          is that RFC 5166 was written to address a wider scope than
          what we are trying
          to do in RMCAT hence IMHO it is best to scope the definition
          and description
          down to the particular scope of RMCAT- interactive real-time
          media over RTP.
          This means this draft should actually include proper
          definition of the metrics,
          methodologies to realize them within this document (or in
          additional
          documents). <o:p></o:p></span></font><!--[endif]--></p>
    <p>
    </p>
    <p class="MsoNormal"
      style="margin-top:5.0pt;margin-right:0cm;margin-bottom:5.0pt;
      margin-left:36.0pt;text-indent:-18.0pt;mso-list:l0 level1
      lfo2;mso-layout-grid-align:
      none;text-autospace:none"><!--[if !supportLists]--><font
        face="Courier New, Courier, monospace"><span style="font-size:
          12pt;" lang="SV"><span style="mso-list:Ignore">&middot;<span
              style="font-style: normal; font-variant: normal;
              font-weight: normal; font-size: 7pt; line-height: normal;
              font-size-adjust: none; font-stretch: normal;
              -moz-font-feature-settings: normal;
              -moz-font-language-override: normal;">&nbsp;&nbsp; </span></span></span><span
          style="font-size: 12pt;" lang="SV">Barely mentioning
          the metrics does not really help. We need to define proper
          methodologies to use
          them. Some of the metrics are easy to visualize but some of
          them are not (see
          more below). We need to describe/define a common understanding
          of all the
          metrics and how to use them in evaluation.<o:p></o:p></span></font><!--[endif]--></p>
    <p>
    </p>
    <p class="MsoNormal"
      style="margin-top:5.0pt;margin-right:0cm;margin-bottom:5.0pt;
      margin-left:36.0pt;text-indent:-18.0pt;mso-list:l0 level1
      lfo2;mso-layout-grid-align:
      none;text-autospace:none"><!--[if !supportLists]--><font
        face="Courier New, Courier, monospace"><span style="font-size:
          12pt;" lang="SV"><span style="mso-list:Ignore">&middot;<span
              style="font-style: normal; font-variant: normal;
              font-weight: normal; font-size: 7pt; line-height: normal;
              font-size-adjust: none; font-stretch: normal;
              -moz-font-feature-settings: normal;
              -moz-font-language-override: normal;">&nbsp;&nbsp; </span></span></span><span
          style="font-size: 12pt;" lang="SV">Delay: this version
          excludes "Delay" as metric. Could you explain why? <o:p></o:p></span></font><!--[endif]--></p>
    <p>
    </p>
    <p class="MsoNormal"
      style="margin-top:5.0pt;margin-right:0cm;margin-bottom:5.0pt;
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      none;text-autospace:none"><!--[if !supportLists]--><font
        face="Courier New, Courier, monospace"><span style="font-size:
          10pt; color: blue;" lang="SV"></span><span style="font-size:
          12pt;" lang="SV"> <o:p></o:p></span></font></p>
    <p>
    </p>
    <p class="MsoNormal"
      style="margin-top:5.0pt;margin-right:0cm;margin-bottom:5.0pt;
      margin-left:36.0pt;text-indent:-18.0pt;mso-list:l0 level1
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        face="Courier New, Courier, monospace"><span style="font-size:
          12pt;" lang="SV"><span style="mso-list:Ignore">&middot;<span
              style="font-style: normal; font-variant: normal;
              font-weight: normal; font-size: 7pt; line-height: normal;
              font-size-adjust: none; font-stretch: normal;
              -moz-font-feature-settings: normal;
              -moz-font-language-override: normal;">&nbsp;&nbsp; </span></span></span><span
          style="font-size: 12pt;" lang="SV">Minimum oscillation
          in encoding rate: This is related to QoE and minimizing
          oscillation<span style="mso-spacerun:yes"> </span></span><span
          style="font-size: 12pt;" lang="SV">is an wanted behavior. How
          do we measure oscillation?
          RFC 5166 gives some ref. but how well do they represent
          interactive
          real-time communication?&nbsp; We have variable bitrate encoder for
          video whose
          output will oscillate depending on scenes to encode. What do
          we do there? We
          perhaps will need to measure the oscillation in the target
          bitrate set by</span><span style="font-size: 10pt; color:
          blue;" lang="SV"></span><span style="font-size: 12pt;"
          lang="SV"><span style="mso-spacerun:yes">&nbsp; </span>the
          Congestion Control (CC) algorithm but very often there will be
          mismatch between the target rate and actual encoding rate.&nbsp;</span><span
          style="font-size: 12pt;" lang="SV"><span
            style="mso-spacerun:yes">&nbsp; </span>&nbsp; &nbsp; <o:p></o:p></span></font><!--[endif]--></p>
    <p>
    </p>
    <p class="MsoNormal"
      style="margin-top:5.0pt;margin-right:0cm;margin-bottom:5.0pt;
      margin-left:36.0pt;text-indent:-18.0pt;mso-list:l0 level1
      lfo2;mso-layout-grid-align:
      none;text-autospace:none"><!--[if !supportLists]--><font
        face="Courier New, Courier, monospace"><span style="font-size:
          12pt;" lang="SV"><span style="mso-list:Ignore">&middot;<span
              style="font-style: normal; font-variant: normal;
              font-weight: normal; font-size: 7pt; line-height: normal;
              font-size-adjust: none; font-stretch: normal;
              -moz-font-feature-settings: normal;
              -moz-font-language-override: normal;">&nbsp;&nbsp; </span></span></span><span
          style="font-size: 12pt;" lang="SV">Reactivity to
          transient events: We can<span style="mso-spacerun:yes"> </span></span><span
          style="font-size: 12pt;" lang="SV">react to transient event
          only if we know that it is a
          transient event. In testbeds, perhaps we can do this by
          setting up particular test
          for a particular transient event and very soon we will find
          that there is no
          "silver bullet" to dodge all favorite transient events. What
          can we
          do then? one way to handle this could be to setup a sequence
          tests of transient
          events and weight them to evaluate the reactivity to the
          transient events.&nbsp;
          Please let me know if I am missing something here.<o:p></o:p></span></font><!--[endif]--></p>
    <p>
    </p>
    <p class="MsoNormal"
      style="margin-top:5.0pt;margin-right:0cm;margin-bottom:5.0pt;
      margin-left:36.0pt;text-indent:-18.0pt;mso-list:l0 level1
      lfo2;mso-layout-grid-align:
      none;text-autospace:none"><!--[if !supportLists]--><font
        face="Courier New, Courier, monospace"><span style="font-size:
          12pt;" lang="SV"><span style="mso-list:Ignore">&middot;<span
              style="font-style: normal; font-variant: normal;
              font-weight: normal; font-size: 7pt; line-height: normal;
              font-size-adjust: none; font-stretch: normal;
              -moz-font-feature-settings: normal;
              -moz-font-language-override: normal;">&nbsp;&nbsp; </span></span></span><span
          style="font-size: 12pt;" lang="SV">Packet loss and
          discard rate: We have three categories here- 1) loss at
          transport layer (TL),
          2) loss due to late arrival (LL) and 3) residual loss&nbsp; (RL =
          TL+LL).<span style="mso-spacerun:yes">&nbsp; RL will also have
            effects of application depended techniques used for
            countering the packet loss (for example, FEC, RTP
            retransmission). </span></span><span style="font-size:
          12pt;" lang="SV">Where TL perhaps a good metric for network,
          RL is
          perhaps what the application care about most. We will need to
          define a relation
          here. Also this metric is not about how the CC algorithms
          calculate the packet
          loss rate (--&nbsp; different algorithm may calculate loss
          event/rate
          differently- per RTCP Send Report/Receiver Report, fractional
          loss over a time
          period, per RTT etc. etc.) but about how we want to measure
          the loss, right? </span><span style="font-size: 10pt; color:
          blue;" lang="SV"></span><span style="font-size: 12pt;"
          lang="SV"><span style="mso-spacerun:yes"></span>then, lets
          define
          it concretely.<o:p></o:p></span></font><!--[endif]--></p>
    <p>
    </p>
    <p class="MsoNormal"
      style="margin-top:5.0pt;margin-right:0cm;margin-bottom:5.0pt;
      margin-left:36.0pt;text-indent:-18.0pt;mso-list:l0 level1
      lfo2;mso-layout-grid-align:
      none;text-autospace:none"><!--[if !supportLists]--><font
        face="Courier New, Courier, monospace"><span style="font-size:
          12pt;" lang="SV"><span style="mso-list:Ignore">&middot;<span
              style="font-style: normal; font-variant: normal;
              font-weight: normal; font-size: 7pt; line-height: normal;
              font-size-adjust: none; font-stretch: normal;
              -moz-font-feature-settings: normal;
              -moz-font-language-override: normal;">&nbsp;&nbsp; </span></span></span><span
          style="font-size: 12pt;" lang="SV">Bandwidth
          utilization: This is a metric I have hard time to grasp. What<span
            style="mso-spacerun:yes"> do we actually</span></span><span
          style="font-size: 12pt;" lang="SV"> want to achieve/measure
          with this metric? I perhaps
          understand it if I have a fixed capacity link but in real-time
          communication
          the input data itself can starve</span><b><i><span
              style="font-size: 10pt; color: blue;" lang="SV"></span></i></b><span
          style="font-size: 10pt; color: blue;" lang="SV"><span
            style="mso-spacerun:yes"></span>,</span><span
          style="font-size: 12pt;" lang="SV"><span
            style="mso-spacerun:yes">&nbsp; </span>resulting<span
            style="mso-spacerun:yes">&nbsp; in</span></span><span
          style="font-size: 10pt; color: blue;" lang="SV"></span><span
          style="font-size: 12pt;" lang="SV"> less utilization of the
          available bandwidth. And for
          cellular networks where the users have different capacity view
          even within a
          cell depending on their radio condition, speed, distance etc.
          etc. ; even in a
          aggregate level (per cell measurement) will depend on what is
          going on other
          cells. Moreover, the view of available bandwidth changes all
          the time, I don't see how
          we can visualize the available bandwidth utilization. I am not
          aware of any
          model that actually does it. Please let me know if I have
          missed something
          here. The thing is if we can't really measure a metric in a
          proper/straight
          forward way then there is not point of including it.<o:p></o:p></span></font><!--[endif]--></p>
    <p>
    </p>
    <p class="MsoNormal"
      style="margin-top:5.0pt;margin-right:0cm;margin-bottom:5.0pt;
      margin-left:36.0pt;text-indent:-18.0pt;mso-list:l0 level1
      lfo2;mso-layout-grid-align:
      none;text-autospace:none"><!--[if !supportLists]--><font
        face="Courier New, Courier, monospace"><span style="font-size:
          12pt;" lang="SV"><span style="mso-list:Ignore">&middot;<span
              style="font-style: normal; font-variant: normal;
              font-weight: normal; font-size: 7pt; line-height: normal;
              font-size-adjust: none; font-stretch: normal;
              -moz-font-feature-settings: normal;
              -moz-font-language-override: normal;">&nbsp;&nbsp; </span></span></span><span
          style="font-size: 12pt;" lang="SV">Convergence time: I
          would like add this metric in the preferred metric list. This
          is also related
          to fairness metric as it gives a view about how fast the
          algorithm can reach a
          stable bitrate when competing with flows with same CC
          algorithm. <o:p></o:p></span></font><!--[endif]--></p>
    <p>
    </p>
    <p class="MsoNormal"
      style="mso-layout-grid-align:none;text-autospace:none"><font
        face="Courier New, Courier, monospace"><span style="font-size:
          12pt;" lang="SV">Guideline:<o:p></o:p></span></font></p>
    <p>
    </p>
    <p class="MsoNormal"
      style="margin-top:5.0pt;margin-right:0cm;margin-bottom:5.0pt;
      margin-left:36.0pt;text-indent:-18.0pt;mso-list:l1 level1
      lfo1;tab-stops:list 18.0pt;
      mso-layout-grid-align:none;text-autospace:none"><!--[if !supportLists]--><font
        face="Courier New, Courier, monospace"><span style="font-size:
          12pt;" lang="SV"><span style="mso-list:Ignore">&middot;<span
              style="font-style: normal; font-variant: normal;
              font-weight: normal; font-size: 7pt; line-height: normal;
              font-size-adjust: none; font-stretch: normal;
              -moz-font-feature-settings: normal;
              -moz-font-language-override: normal;">&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;
            </span></span></span><span style="font-size: 12pt;"
          lang="SV">4.3 : It seems that
          it is necessary to have some reference cases to verify
          against.<o:p></o:p></span></font><!--[endif]--></p>
    <p>
    </p>
    <p class="MsoNormal"
      style="margin-top:5.0pt;margin-right:0cm;margin-bottom:5.0pt;
      margin-left:36.0pt;text-indent:-18.0pt;mso-list:l1 level1
      lfo1;tab-stops:list 18.0pt;
      mso-layout-grid-align:none;text-autospace:none"><!--[if !supportLists]--><font
        face="Courier New, Courier, monospace"><span style="font-size:
          12pt;" lang="SV"><span style="mso-list:Ignore">&middot;<span
              style="font-style: normal; font-variant: normal;
              font-weight: normal; font-size: 7pt; line-height: normal;
              font-size-adjust: none; font-stretch: normal;
              -moz-font-feature-settings: normal;
              -moz-font-language-override: normal;">&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;
            </span></span></span><span style="font-size: 12pt;"
          lang="SV">4.4 : Is this the
          plan to test against all possible access types</span><span
          style="font-size: 10pt; color: blue;" lang="SV"></span><span
          style="font-size: 12pt;" lang="SV"> ?<o:p></o:p></span></font><!--[endif]--></p>
    <p>
    </p>
    <p class="MsoNormal"
      style="margin-top:5.0pt;margin-right:0cm;margin-bottom:5.0pt;
      margin-left:36.0pt;text-indent:-18.0pt;mso-list:l1 level1
      lfo1;tab-stops:list 18.0pt;
      mso-layout-grid-align:none;text-autospace:none"><!--[if !supportLists]--><font
        face="Courier New, Courier, monospace"><span style="font-size:
          12pt;" lang="SV"><span style="mso-list:Ignore">&middot;<span
              style="font-style: normal; font-variant: normal;
              font-weight: normal; font-size: 7pt; line-height: normal;
              font-size-adjust: none; font-stretch: normal;
              -moz-font-feature-settings: normal;
              -moz-font-language-override: normal;">&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;
            </span></span></span><span style="font-size: 12pt;"
          lang="SV">4.7 : any idea on
          how to "highlight the bottleneck capacity share of each RTP
          flow" for
          wireless/cellular networks?<o:p></o:p></span></font><!--[endif]--></p>
    <p>
    </p>
    <p class="MsoNormal"
      style="margin-top:5.0pt;margin-right:0cm;margin-bottom:5.0pt;
      margin-left:72.0pt;text-indent:-18.0pt;mso-list:l1 level2
      lfo1;tab-stops:list 18.0pt;
      mso-layout-grid-align:none;text-autospace:none"><!--[if !supportLists]--><font
        face="Courier New, Courier, monospace"><span style="font-size:
          12pt;" lang="SV"><span style="mso-list:Ignore">&middot;<span
              style="font-style: normal; font-variant: normal;
              font-weight: normal; font-size: 7pt; line-height: normal;
              font-size-adjust: none; font-stretch: normal;
              -moz-font-feature-settings: normal;
              -moz-font-language-override: normal;">&nbsp;&nbsp; </span></span></span><span
          style="font-size: 12pt;" lang="SV">This also says "However,
          long-lived TCP flows may starve out the RTP flow depending on
          router queue
          length. In the latter case the proposed congestion control for
          RTP should be as
          aggressive as standard TCP"- how realistic is this? at some
          point (if
          there are buffer-bloats) the RMCAT will have to give up as it
          has certain delay
          budget to meet.<o:p></o:p></span></font><!--[endif]--></p>
    <p>
    </p>
    <p class="MsoNormal"
      style="margin-top:5.0pt;margin-right:0cm;margin-bottom:5.0pt;
      margin-left:36.0pt;text-indent:-18.0pt;mso-list:l1 level1
      lfo1;tab-stops:list 18.0pt;
      mso-layout-grid-align:none;text-autospace:none"><!--[if !supportLists]--><font
        face="Courier New, Courier, monospace"><span style="font-size:
          12pt;" lang="SV"><span style="mso-list:Ignore">&middot;<span
              style="font-style: normal; font-variant: normal;
              font-weight: normal; font-size: 7pt; line-height: normal;
              font-size-adjust: none; font-stretch: normal;
              -moz-font-feature-settings: normal;
              -moz-font-language-override: normal;">&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;
            </span></span></span><span style="font-size: 12pt;"
          lang="SV">missing guidelines<o:p></o:p></span></font><!--[endif]--></p>
    <p>
    </p>
    <p class="MsoNormal"
      style="margin-top:5.0pt;margin-right:0cm;margin-bottom:5.0pt;
      margin-left:72.0pt;text-indent:-18.0pt;mso-list:l1 level2
      lfo1;tab-stops:list 18.0pt;
      mso-layout-grid-align:none;text-autospace:none"><!--[if !supportLists]--><font
        face="Courier New, Courier, monospace"><span style="font-size:
          12pt;" lang="SV"><span style="mso-list:Ignore">&middot;<span
              style="font-style: normal; font-variant: normal;
              font-weight: normal; font-size: 7pt; line-height: normal;
              font-size-adjust: none; font-stretch: normal;
              -moz-font-feature-settings: normal;
              -moz-font-language-override: normal;">&nbsp;&nbsp; </span></span></span><span
          style="font-size: 12pt;" lang="SV">Congested feedback
          channel : the CC algorithm should be evaluated against
          congested feedback
          channel. Could be included as separate guideline or a part of
          4.5.<o:p></o:p></span></font><!--[endif]--></p>
    <p>
    </p>
    <p class="MsoNormal"
      style="margin-top:5.0pt;margin-right:0cm;margin-bottom:5.0pt;
      margin-left:72.0pt;text-indent:-18.0pt;mso-list:l1 level2
      lfo1;tab-stops:list 18.0pt;
      mso-layout-grid-align:none;text-autospace:none"><!--[if !supportLists]--><font
        face="Courier New, Courier, monospace"><span style="font-size:
          12pt;" lang="SV"><span style="mso-list:Ignore">&middot;<span
              style="font-style: normal; font-variant: normal;
              font-weight: normal; font-size: 7pt; line-height: normal;
              font-size-adjust: none; font-stretch: normal;
              -moz-font-feature-settings: normal;
              -moz-font-language-override: normal;">&nbsp;&nbsp; </span></span></span><span
          style="font-size: 12pt;" lang="SV">Different start up
          values: as the requirement draft mentioned several options for
          start-up values,
          the algorithms should be tested. One can argue that different
          CC algorithm will
          have different strategies but then it should be judged against
          fairness and
          convergence time.<o:p></o:p></span></font><!--[endif]--></p>
    <p>
    </p>
    <p class="MsoNormal"
      style="margin-top:5.0pt;margin-right:0cm;margin-bottom:5.0pt;
      margin-left:72.0pt;text-indent:-18.0pt;mso-list:l1 level2
      lfo1;tab-stops:list 18.0pt;
      mso-layout-grid-align:none;text-autospace:none"><!--[if !supportLists]--><font
        face="Courier New, Courier, monospace"><span style="font-size:
          12pt;" lang="SV"><span style="mso-list:Ignore">&middot;<span
              style="font-style: normal; font-variant: normal;
              font-weight: normal; font-size: 7pt; line-height: normal;
              font-size-adjust: none; font-stretch: normal;
              -moz-font-feature-settings: normal;
              -moz-font-language-override: normal;">&nbsp;&nbsp; </span></span></span><span
          style="font-size: 12pt;" lang="SV">Fairness to other
          RMCAT proposed algorithms: as part of fairness guideline.<o:p></o:p></span></font><!--[endif]--></p>
    <p>
    </p>
    <p class="MsoNormal"
      style="mso-layout-grid-align:none;text-autospace:none"><font
        face="Courier New, Courier, monospace"><span style="font-size:
          12pt;" lang="SV">Eval scenarios:<o:p></o:p></span></font></p>
    <p>
    </p>
    <p class="MsoNormal"
      style="margin-top:5.0pt;margin-right:0cm;margin-bottom:5.0pt;
      margin-left:36.0pt;text-indent:-18.0pt;mso-list:l0 level1
      lfo2;mso-layout-grid-align:
      none;text-autospace:none"><!--[if !supportLists]--><font
        face="Courier New, Courier, monospace"><span style="font-size:
          12pt;" lang="SV"><span style="mso-list:Ignore">&middot;<span
              style="font-style: normal; font-variant: normal;
              font-weight: normal; font-size: 7pt; line-height: normal;
              font-size-adjust: none; font-stretch: normal;
              -moz-font-feature-settings: normal;
              -moz-font-language-override: normal;">&nbsp;&nbsp; </span></span></span><span
          style="font-size: 12pt;" lang="SV">This section should
          really list a "must be" evaluated test scenarios not example
          scenarios. Those "must be" evaluated scenarios should take the
          guides
          lines into account and include<span style="mso-spacerun:yes">&nbsp;</span></span><span
          style="font-size: 10pt;" lang="SV"><span
            style="mso-spacerun:yes"></span>the</span><span
          style="font-size: 12pt;" lang="SV"> related<span
            style="mso-spacerun:yes"> </span></span><span
          style="font-size: 12pt;" lang="SV">metrics so that people have
          a
          common view on how to evaluate their algorithms. Any kind of
          effort to write
          down example scenarios should be waste of time and energy. Of
          course, there
          could be optional test cases and people can go an add their
          scenarios and if
          some scenarios become very useful then include those in the
          "must-be"
          evaluated sets.<o:p></o:p></span></font><!--[endif]--></p>
    <p>
    </p>
    <p class="MsoNormal"
      style="margin-top:5.0pt;margin-right:0cm;margin-bottom:5.0pt;
      margin-left:36.0pt;text-indent:-18.0pt;mso-list:l0 level1
      lfo2;mso-layout-grid-align:
      none;text-autospace:none"><!--[if !supportLists]--><font
        face="Courier New, Courier, monospace"><span style="font-size:
          12pt;" lang="SV"><span style="mso-list:Ignore">&middot;<span
              style="font-style: normal; font-variant: normal;
              font-weight: normal; font-size: 7pt; line-height: normal;
              font-size-adjust: none; font-stretch: normal;
              -moz-font-feature-settings: normal;
              -moz-font-language-override: normal;">&nbsp;&nbsp; </span></span></span><span
          style="font-size: 12pt;" lang="SV">Video starting rate
          is 128 kbps. Feels like the default behavior is slow start,
          why?<o:p></o:p></span></font><!--[endif]--></p>
    <p>
    </p>
    <p class="MsoNormal"
      style="margin-top:5.0pt;margin-right:0cm;margin-bottom:5.0pt;
      margin-left:36.0pt;text-indent:-18.0pt;mso-list:l0 level1
      lfo2;mso-layout-grid-align:
      none;text-autospace:none"><!--[if !supportLists]--><font
        face="Courier New, Courier, monospace"><span style="font-size:
          12pt;" lang="SV"><span style="mso-list:Ignore">&middot;<span
              style="font-style: normal; font-variant: normal;
              font-weight: normal; font-size: 7pt; line-height: normal;
              font-size-adjust: none; font-stretch: normal;
              -moz-font-feature-settings: normal;
              -moz-font-language-override: normal;">&nbsp;&nbsp; </span></span></span><span
          style="font-size: 12pt;" lang="SV">Yes, I would
          like to get a value(s) for queue length when AQM is used along
          with the AQM
          behavior.<o:p></o:p></span></font><!--[endif]--></p>
    <p>
    </p>
    <p class="MsoNormal"
      style="margin-top:5.0pt;margin-right:0cm;margin-bottom:5.0pt;
      margin-left:36.0pt;text-indent:-18.0pt;mso-list:l0 level1
      lfo2;mso-layout-grid-align:
      none;text-autospace:none"><!--[if !supportLists]--><font
        face="Courier New, Courier, monospace"><span style="font-size:
          12pt;" lang="SV"><span style="mso-list:Ignore">&middot;<span
              style="font-style: normal; font-variant: normal;
              font-weight: normal; font-size: 7pt; line-height: normal;
              font-size-adjust: none; font-stretch: normal;
              -moz-font-feature-settings: normal;
              -moz-font-language-override: normal;">&nbsp;&nbsp; </span></span></span><span
          style="font-size: 12pt;" lang="SV">I have hard time
          fitting a radio scenario into dumbbell topology with the
          current definition in
          the draft. &nbsp; <o:p></o:p></span></font><!--[endif]--></p>
    <p>
      <font face="Courier New, Courier, monospace"><span
          style="font-size: 12pt;" lang="SV">BR</span></font></p>
    <p> </p>
    <pre class="moz-signature" cols="72">-- 

Zahed

==================================================
ANM ZAHEDUZZAMAN SARKER 


Ericsson AB
Multimedia Technologies
Laboratoriegr&auml;nd 11
97128 Lule&aring;, Sweden
Phone +46 10 717 37 43
Fax +46 920 996 21
SMS/MMS +46 76 115 37 43
<a class="moz-txt-link-abbreviated" href="mailto:zaheduzzaman.sarker@ericsson.com">zaheduzzaman.sarker@ericsson.com</a>
<a class="moz-txt-link-abbreviated" href="http://www.ericsson.com">www.ericsson.com</a>

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From: Zaheduzzaman Sarker <zaheduzzaman.sarker@ericsson.com>
To: "rmcat@ietf.org" <rmcat@ietf.org>
Thread-Topic: [ipr-announce] IPR Disclosure: Telefonaktiebolaget LM Ericsson (publ)'s Statement	about IPR related to	draft-alvestrand-rtcweb-congestion-03
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Subject: [rmcat] FW: [ipr-announce] IPR Disclosure: Telefonaktiebolaget LM Ericsson	(publ)'s Statement	about IPR related to	draft-alvestrand-rtcweb-congestion-03
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Hi all,

FYI ..

BR

Zahed



-----Original Message-----
From: ipr-announce-bounces@ietf.org [mailto:ipr-announce-bounces@ietf.org] =
On Behalf Of IETF Secretariat
Sent: den 22 februari 2013 17:21
To: holmer@google.com; harald@alvestrand.no
Cc: ipr-announce@ietf.org
Subject: [ipr-announce] IPR Disclosure: Telefonaktiebolaget LM Ericsson (pu=
bl)'s Statement about IPR related to draft-alvestrand-rtcweb-congestion-03


Dear Stefan Holmer, Harald T. Alvestrand:

 An IPR disclosure that pertains to your Internet-Draft entitled "A Google =
Congestion Control Algorithm for Real-Time Communication on the World Wide =
Web"
(draft-alvestrand-rtcweb-congestion) was submitted to the IETF Secretariat =
on
2013-02-22 and has been posted on the "IETF Page of Intellectual Property R=
ights Disclosures" (https://datatracker.ietf.org/ipr/1972/). The title of t=
he IPR disclosure is "Telefonaktiebolaget LM Ericsson (publ)'s Statement ab=
out IPR related to draft-alvestrand-rtcweb-congestion-03."");

The IETF Secretariat

_______________________________________________
ipr-announce mailing list
ipr-announce@ietf.org
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From randell-ietf@jesup.org  Sun Mar 10 06:08:10 2013
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Subject: Re: [rmcat] Initial comments on draft-jesup-rmcat-reqs-01
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On 3/5/2013 10:37 AM, ken carlberg wrote:
> bringing up the terms "long-lived" and "short lived" triggered an outside the box thought on the topic.  Do you think its reasonable ro feasable to use numbers associated with average call/flow duration as a point of reference of what is considered a "typical real-time flow"?  Part of the reason that I bring this up is that I understand that one (maybe more) popular VoIP applications starts with a rather aggressive offered load, and then after some period of time (versus actual feedback) the application backs off to some target measure that takes into account accumulated path characteristics (eg, delay, loss).  Perhaps the time delay is to avoid oscilation.
>
> One can argue that the initial stage of such an approach is not fair, or its excessively greedy (for lack of an alternative term), but that perhaps over time, or an average call/flow duration, it is fair.  I've seen numbers that range from 1.8 minutes to 4 minutes in average call duration, and so perhaps this could be a target reference point to gauge how nice various algorithms work with each other.

An initial burst at a higher rate may help in some cases (force existing 
TCP flows into backoff), but can only do so by generating enough delay 
right at the start of a connection to fill the bottleneck and force 
loss.  So it won't do that - but it will allow a call to function "well" 
from the start if there is unused (typically local) bandwidth, whereas a 
classic slow-start may lead to a poor start-of-call experience.

In the past, I used bandwidth availability history (and user settings) 
to select an initial bandwidth I believed was below the likely 
available, since I wanted good startup quality with minimal chance of 
getting behind-the-ball.  A balance; as usual there's no "right" solution.

-- 
Randell Jesup
randell-ietf@jesup.org



From lars@netapp.com  Mon Mar 11 05:19:37 2013
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From: "Eggert, Lars" <lars@netapp.com>
To: "rmcat@ietf.org" <rmcat@ietf.org>
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Hi,

in the interest of avoiding that awkward pause at the beginning of the meet=
ing, where we chairs ask for a minute taker and you all stare at the floor,=
 if anyone would be willing to take minutes, please drop me a note.

Thanks,
Lars=

From randell-ietf@jesup.org  Mon Mar 11 08:02:18 2013
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Subject: [rmcat] Evaluation criteria - network parameters
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Network Parameters:

For each node we have the following possibilities:
* LAN (very low delay < 10ms)
* Low delay (<40ms ms one-way-delay)             <--- Chosen arbitrarily
* High delay (>100(?)ms OWD)
* Extreme delay (>300ms OWD) (Satellite links, etc)
Unspecified is the link capacity (and if it's asymmetric) and 
description of cross traffic.

* WiFi
** fast
** slow/poor signal           <----- Need to choose parameters for WiFi
** low-power
** loss rate
* 3G/LTE/etc

WiFi for edge nodes is a critical usecase.

For an entire network, we typically will have mixtures except in the 
simplest (same LAN/etc) cases.  In most cases, the most important 
parameter will be where the bottleneck is.  In some cases, the 
bottleneck will move from one node to another due to either varying 
cross traffic, or do due to varying link capacity (WiFi, 3G, etc).

Bottlenecks in each direction may be in different nodes. (Frequently 
with asymmetric links)

We should describe what sort of blocks we want to use in the eval 
scenarios, and then try to define specific sets of network params to 
measure each scenario.  Each scenario will need to be measured with 
multiple network configurations, though there are too many combinations 
to test them all (or even close).

Things we need to do:
Define example characteristics for each type of node we want to use.
Define which sets of nodes we want to measure for each topology.

Related decision: defining cross-traffic for each topology.

-- 
Randell Jesup
randell-ietf@jesup.org


From wes@mti-systems.com  Mon Mar 11 10:05:01 2013
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Cc: Martin Stiemerling <Martin.Stiemerling@neclab.eu>, Lars Eggert <lars.eggert@nokia.com>, Varun Singh <vsingh.ietf@gmail.com>
Subject: [rmcat] room reserved for side-meeting
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Varun is in charge, but I reserved the IESG Breakout Room
(Boca 3) for Tuesday and Thursday mornings from 8 to 9 AM
for the "design team" Lars suggested in the meeting today
to have a place to talk.

I know there was some discussion about potentially meeting
outdoors as well, so Varun should let you know where to meet;
I'm just letting you know that the room is reserved as a
possibility.  I should fit 15ish without issue, and might
have an easel/notepad to doodle on (but no projector).

-- 
Wes Eddy
MTI Systems

From vsingh.ietf@gmail.com  Mon Mar 11 11:36:58 2013
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From: Varun Singh <vsingh.ietf@gmail.com>
Date: Mon, 11 Mar 2013 14:36:36 -0400
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To: rmcat WG <rmcat@ietf.org>
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Cc: Wesley Eddy <wes@mti-systems.com>, Lars Eggert <lars.eggert@nokia.com>, Martin Stiemerling <Martin.Stiemerling@neclab.eu>
Subject: Re: [rmcat] room reserved for side-meeting
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Hi all,

We will meet in the IESG Breakout Room and if the weather permits,
we can move out into the sun.

Varun

On Mon, Mar 11, 2013 at 1:04 PM, Wesley Eddy <wes@mti-systems.com> wrote:
> Varun is in charge, but I reserved the IESG Breakout Room
> (Boca 3) for Tuesday and Thursday mornings from 8 to 9 AM
> for the "design team" Lars suggested in the meeting today
> to have a place to talk.
>
> I know there was some discussion about potentially meeting
> outdoors as well, so Varun should let you know where to meet;
> I'm just letting you know that the room is reserved as a
> possibility.  I should fit 15ish without issue, and might
> have an easel/notepad to doodle on (but no projector).
>
> --
> Wes Eddy
> MTI Systems



-- 
http://www.netlab.tkk.fi/~varun/

From lars@netapp.com  Mon Mar 11 13:04:01 2013
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From: "Eggert, Lars" <lars@netapp.com>
To: "rmcat@ietf.org" <rmcat@ietf.org>
Thread-Topic: confirmation of WG adoption calls
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Subject: [rmcat] confirmation of WG adoption calls
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Hi,

during the meeting today, I asked for consensus calls to adopt

(1) draft-jesup-rmcat-reqs as a starting point for our cc-requirements work=
 item
(2) draft-singh-rmcat-cc-eval as a starting point for our eval-criteria wor=
k item

For (1), the room as 20-0 in favor of adoption. I'd call this strong consen=
sus and would as the author to submit the next revision as a WG item. Pleas=
e speak up if you disagree with this notion.

For (2), the room was more split, with about 10-5 in favor. I'd like to und=
erstand the reservations that kept some people from supporting the consensu=
s call. Please send a brief email to the list (or to the chairs personally)=
 and let us know if you have fundamental objections or would simply see the=
 document develop a bit further before being able to support it, or any oth=
er concerns you may have.

We don't have any alternative documents targeting our eval-criteria milesto=
ne, and I'm thus at the moment leaning to call a weak consensus for adoptio=
n. But I want to make sure we understand the concerns first.

Thanks,
Lars

PS: I'm saying "I" a lot in this email, because I haven't had a chance to s=
ync up with Mirja on this yet.


From spencer@wonderhamster.org  Mon Mar 11 21:18:22 2013
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Date: Mon, 11 Mar 2013 23:18:16 -0500
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Subject: [rmcat] My notes from today's RMCAT session
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Lars asked me to take notes at today's session; I'm sending my notes to 
the list so when I've horribly misunderstood or misquoted you, you can 
clue me in ...

Thanks,

Spencer

2013-03 IETF 86 - RMCAT
Lars is chairing this meeting solo ...
Note Well - thank you, Ericsson for disclosing IPR on a Google draft

== Chair Intro
We're running late on Requirements, we have one draft to choose from
We're running late on Evaluation Criteria and group-cc
We aren't sure we need to do rtcp-requirements and application 
interactions milestones
We're going to talk with AD about milestones - proposing 1 year slip for 
submission to IESG milestones
Nothing happened on the mailing list over winter holidays, picking up a 
little speed now

No agenda bashing

== Requirements - Randall Jesup
Draft was discussed in Atlanta, revised as -01
Reviewing issues we're discussing and haven't closed on
Definition of "fair"
- we don't have one, and there's not a single definition
- doesn't have to be 1x TCP
- matters more against longer flows
- matters most against other low-delay flows
- can't collapse, can't take too long for a new flow to become usable
- needs to be bounded if we can't define it
- needs to cover both local fairness and aggregate fairness at nodes 
deeper in the net

Other issues
- "minimum bandwidth" can be application-defined.
- need to think about oscillation and possible "mode-switching" between 
"fair" and "aggressive"

Q and A
- anything we can do to bound the problem will help
- we know what UNfairness is. Can we define tests for unfairness?
- can we describe principles for identifying unfairness?
- Lars: this is the biggest issue facing the group, need to make 
progress this week
- can we agree on a list of priorities?
- how self-friendly is existing non-real time traffic?
- how self-friendly can we be with no other traffic, dedicated nets?
- need to think about what else we're competing against -RTCWeb data 
channels
- Matt: one definition is impossible, think about what matters
- Lars: we need to form a design team this week
- is friendliness part of fairness?
- can we come up with a test plan to check for fairness?
- Lars: remember that bad congestion control makes it bad for everyone
- Colin: decide whether competing with TCP is a short-term goal
- Randall: especially competing against short-term TCP flows
- Pat: need to notice the effect of fairness constraints on other 
characteristics
Lars: this may be one of the hard problems in computer science, but 
we're going to solve it before Berlin :-)

== Evaluating Congestion Control - Varuna Singh
Now at -02
- Open issue: two metrics under discussion, other metrics seem ok
- ideally, we do a distribution and then think about what "discard rate" is
- needs to be more than one number
- we can do things during evaluation we can't do in deployments

We've identified 8 evaluation guidelines
- do we need a minimum set?
- are the identified guidelines sufficient? Have proposed 3 more
- Lars: if you are proposing a solution you need to look at proposal
- have included a proposal for various evaluation scenario parameters
- also want to limit combinatorial explosion of test scenarios
- do we use media packet generators or real video streams?
- video ends up trying to keep quality consistent, not bit rate
- Randall: have to know where the edges are - not one parameter set
- Harald: need to be able to show we aren't ruining video experience
- Gorry: what happens if your delay exceeds 300ms? Need to separate out 
the transport aspects and focus on them
- we need to start someplace - pick scenarios, parameters
- need to use simple and consistent traffic stream
- would like to see continuity from requirements draft to this draft
- need to characterize requirements for various media types
- need to look at multiple network types - wired, WiFi, cellular ...
- Magnus: video is so changeable - that's why it's important to look at 
various media sources

Evaluation Scenarios
- fixed link capacity, variable link capacity, self-awareness
- do all scenarios use the same parameters?
- Matt: assuming single queue or multiple? Need to make that explicit
- Lars: want some scenarios that do it each way

== Working Group Hums
Adopt requirements draft? 20 hands yes, no nos
Adopt evaluation draft? 10 hands yes, 4 nos
What do people want to see in the document to be "yes"? No comments from 
the mikes

== Coupled Congestion Control - Michael Wetzl
Controlling groups of flows has been tried in the past
- why didn't these see deployment?
- hard to identify flows, hard to implement

Flow State Exchange (FSE)
- Flows register with FSE when they start, stop, change rates
- could use various algorithms, but competing flows all use the same one
- not priorities, but proportional share
- Magnus: need to think about RTCWeb control channels, HTTP, etc
- need to figure out shared bottleneck detection, still working on this
- browser could use flow table to self-regulate
- Dale: if everyone builds the same state table, we can understand more
- this would have security implications
- current algorithm has two problems: flow told to use rate that's not 
what congestion manager determined, highly asynchronous flows
- can this be sender-side only? Can do better management if receiver is 
involved

(Missed question here)

== On the use of RTCP feedback for congestion control - Colin Perkins
Thinking about how existing feedback channel works (with AVPF)
- Per-packet feedback would be great, but RTCP feedback is much less 
frequent, and no mechanism to piggyback information
- per-video frame? 30 fps gives you 0.033 second RTCP control interval
- assuming compound RTCP, but not cross-reporting
- per-RTT? Longer than inter-frame interval
- Hannes: can't adjust sending rates arbitrarily with most codecs, and 
frequent quality changes cause problems with user experience anyway
- conclusion - RTCP suitable for per-frame, per-RTT, not per-packet
- Colin's presentation assumes no RTCP extensions
- remember you can lose feedback packets, too. Just need stable control 
system
- don't adjust what we do in RMCAT based on existing codec limitations
- may need to accommodate massive sending rate changes
- can we ignore the fact that this is audio/video? Video streams are 
likely to drive congestion, audio is small and you can end up sending 
more feedback than audio
- packet rate in wireless networks more significant than bandwidth, need 
piggybacking to reduce that
- what about sampling? Could give you enough feedback
- two issues - how often do you sample, latency for feedback to sender
- do we have any minimum requirements that cover feedback?
- this isn't easy stuff (and wasn't easy with TCP, either)

Colin wrote the draft to generate discussion, not planning to progress

== NADA - a unified congestion control scheme for real-time media - Rong Pan
Design goals: limit self-inflicted delay, leverage a suite of network 
feedback mechanisms (loss-based, delay-based, ECN, PCN)
Sender does linear prediction to accommodate feedback latency
Simulation of NADA with ECN shows improvement in avoiding queue build
PCN allows zero standing queue

Q and A
- PCN? Standards are out but haven't been deployed yet
- Rate-shaping buffer on frame basis? Different frame types - I-frames 
can span packets, and buffering them can degrade predictive calculations 
for receiving codec
- Do you see problems with self-interference after congestion happens? 
Depends on traffic and where you stabilize
- 3-5 percent loss rates in your simulations are very bad for video traffic
- If we have a high loss rate, we need FEC or other recovery mechanisms 
that come with overhead
- Have you looked at CoDeL? Not yet ...
- How does this compete with TCP? We're looking at that
- We don't know if ECN has benefits when you're already doing 
delay-based mechanisms
- Should evaluation criteria look at both loss and delay? Probably not, 
we're evaluating the results of the algorithms - but we need to look at 
latency and loss as part of the results
- If you cause other people 5 percent loss, that's worse than causing 
yourself 5 percent loss that you can recover from

From dan@marketsoup.com  Tue Mar 12 06:51:42 2013
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From: Dan Weber <dan@marketsoup.com>
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Subject: [rmcat] ICCRG White Paper on Managing Network Uncertainty
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--90e6ba6e817abf199d04d7ba9605
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Hi everyone,

Varun suggested this paper: End-to-End Transmission Control by Modeling
Uncertainty about the Network State (
http://conferences.sigcomm.org/hotnets/2011/papers/hotnetsX-final100.pdf)

This is worth a look when discussing our evaluation criteria.

Thanks,
Dan

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Hi everyone,<div><br></div><div>Varun suggested this paper:=C2=A0End-to-End=
 Transmission Control by Modeling Uncertainty about the Network State (<a h=
ref=3D"http://conferences.sigcomm.org/hotnets/2011/papers/hotnetsX-final100=
.pdf">http://conferences.sigcomm.org/hotnets/2011/papers/hotnetsX-final100.=
pdf</a>)</div>
<div><br></div><div>This is worth a look when discussing our evaluation cri=
teria.</div><div><br></div><div>Thanks,</div><div>Dan</div><div><br></div>

--90e6ba6e817abf199d04d7ba9605--

From vsingh.ietf@gmail.com  Tue Mar 12 07:00:00 2013
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From: Varun Singh <vsingh.ietf@gmail.com>
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Subject: [rmcat] Design Meeting 1
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Hi all,

The discussion yielded the following application scenarios
(network scenarios/link characteristics will be discussed
separately):

0. Single Flow (no fairness)
1. Competing with the same/homogeneous flows
    (i.e., all flows are interactive multimedia)
2. Competing with long-lived TCP
3. Competing with bursty TCP
4. Competing with bitorrent/LEDBAT.
5. Competing with unresponsive interactive multimedia
    (i.e., not only CBR)
---
6. Competing with data-channel
    (probably covered by bursty/long-lived tcp flows?)

These are an initial list of application scenarios, if anyone has
additional scenarios, send them to the list before the next design
meeting session on Thursday (see below). So
that we consider it for the fairness discussion.

If you missed the session today and are interested to contribute,
you are welcome to join the next session on Thursday (see below)

Thursday Agenda (14 March):
Location: (8-9am, Room: Boca III)

Recap (5mins)
Network scenarios
Flow scenarios
History of Fairness (15mins, Bob Briscoe volunteered to present)
Principles of Fairness

If required, we will have an additional session on Thursday/Friday.

Attendees today (tuesday):
Kevin Gross
Bob Briscoe
Michael Ramalho
Zahed Sarker
Colin Perkins
Randell Jesup
Dan Weber
Varun Singh
Wes Eddy

-Varun



-- 
http://www.netlab.tkk.fi/~varun/

From zaheduzzaman.sarker@ericsson.com  Tue Mar 12 07:23:17 2013
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Hi,

it is important to note that the scenarios discussed in the design team 
meeting are those where we think "fairness" matters.
These are not complete set of evaluation scenarios.

Our focus was "fairness" and we started with identifying set of 
scenarios where we want to see the "fairness". The idea is to come up 
with principles that will help achieving fairness in those scenarios.

BR

Zahed


On 2013-03-12 14:59, Varun Singh wrote:
> Hi all,
>
> The discussion yielded the following application scenarios
> (network scenarios/link characteristics will be discussed
> separately):
>
> 0. Single Flow (no fairness)
> 1. Competing with the same/homogeneous flows
>      (i.e., all flows are interactive multimedia)
> 2. Competing with long-lived TCP
> 3. Competing with bursty TCP
> 4. Competing with bitorrent/LEDBAT.
> 5. Competing with unresponsive interactive multimedia
>      (i.e., not only CBR)
> ---
> 6. Competing with data-channel
>      (probably covered by bursty/long-lived tcp flows?)
>
> These are an initial list of application scenarios, if anyone has
> additional scenarios, send them to the list before the next design
> meeting session on Thursday (see below). So
> that we consider it for the fairness discussion.
>
> If you missed the session today and are interested to contribute,
> you are welcome to join the next session on Thursday (see below)
>
> Thursday Agenda (14 March):
> Location: (8-9am, Room: Boca III)
>
> Recap (5mins)
> Network scenarios
> Flow scenarios
> History of Fairness (15mins, Bob Briscoe volunteered to present)
> Principles of Fairness
>
> If required, we will have an additional session on Thursday/Friday.
>
> Attendees today (tuesday):
> Kevin Gross
> Bob Briscoe
> Michael Ramalho
> Zahed Sarker
> Colin Perkins
> Randell Jesup
> Dan Weber
> Varun Singh
> Wes Eddy
>
> -Varun
>
>
>

-- 

Zahed

==================================================
ANM ZAHEDUZZAMAN SARKER


Ericsson AB
Multimedia Technologies
Laboratoriegränd 11
97128 Luleå, Sweden
Phone +46 10 717 37 43
Fax +46 920 996 21
SMS/MMS +46 76 115 37 43
zaheduzzaman.sarker@ericsson.com
www.ericsson.com

==================================================


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On 4 Mar 2013, at 23:41, Fu Jiantao wrote:
> Hi Colin,
>=20
>    I've got the following comments:
>=20
>   1."there is little point in providing feedback more often than the =
codec can adapt", i think there is no need to send those overlimits data =
onto the network if we've got feedback that the network is congested, so =
i think it make sense if there is traffic shaper on the sender side, =
those overlimit packets can buffered and dropped at the sender side, =
rather than sending them to the network which affects other traffics.

This draft isn't proposing a congestion control algorithm, it's =
exploring the space of possible feedback using RTCP as defined. It might =
make sense to send feedback as you suggest.=20

>   2. How about ack in a given interval, this can be treat as variant =
of per-frame ack(several frames per ack). This is another kind which has =
less load than per-frame ack.

Sure.

>   3. There can be adaptive feedback which feedback when there is need =
to do so, for example, when the given interval is not enough for the =
adaption(the queuing delay goes large very fast), the feedback can be =
accelerated in this case.

I agree that this is an option.

>   4. The feedback strategy has some relationship with =
sender-based/receiver-based, is this in the scope of this draft?

I'm not intending to progress this draft, but it's certainly an issue to =
consider when designing a congestion control algorithm.

Colin




> Thanks
>=20
> Jeromy
>=20
>=20
> 2013/2/26 Colin Perkins <csp@csperkins.org>
> I submitted the following draft that talks about the types of =
congestion control feedback that can be sent using RTCP as is currently =
specified, and their suitability for implementing congestion control for =
unicast multimedia applications. In particular, it talks about the rate =
of congestion feedback that can be supported.
>=20
> Comments are welcome. I'm also happy to present this in Orlando, if =
the chairs think it useful.
>=20
> Colin
>=20
>=20
>=20
>=20
> Begin forwarded message:
> > From: internet-drafts@ietf.org
> > Subject: I-D Action: draft-perkins-rmcat-rtp-cc-feedback-00.txt
> > Date: 18 February 2013 23:28:42 GMT
> > To: i-d-announce@ietf.org
> > Reply-To: internet-drafts@ietf.org
> >
> >
> > A New Internet-Draft is available from the on-line Internet-Drafts =
directories.
> >
> >
> >       Title           : On the Use of RTP Control Protocol (RTCP) =
Feedback for Unicast Multimedia Congestion Control
> >       Author(s)       : Colin Perkins
> >       Filename        : draft-perkins-rmcat-rtp-cc-feedback-00.txt
> >       Pages           : 8
> >       Date            : 2013-02-18
> >
> > Abstract:
> > This memo discusses the types of congestion control feedback that it
> > is possible to send using the RTP Control Protocol (RTCP), and their
> > suitability of use in implementing congestion control for unicast
> > multimedia applications.
> >
> >
> > The IETF datatracker status page for this draft is:
> > https://datatracker.ietf.org/doc/draft-perkins-rmcat-rtp-cc-feedback
> >
> > There's also a htmlized version available at:
> > http://tools.ietf.org/html/draft-perkins-rmcat-rtp-cc-feedback-00
>=20
>=20
> --
> Colin Perkins
> http://csperkins.org/
>=20
>=20
>=20
>=20



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




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<html><head></head><body style="word-wrap: break-word; -webkit-nbsp-mode: space; -webkit-line-break: after-white-space; "><div><div>On 4 Mar 2013, at 23:41, Fu Jiantao wrote:</div><blockquote type="cite"><div dir="ltr"><div><div><div><div><div>Hi Colin,<br><br></div>&nbsp;&nbsp; I've got the following comments:<br><br>&nbsp;
 1."there is little point in
   providing feedback more often than the codec can adapt", i think 
there is no need to send those overlimits data onto the network if we've
 got feedback that the network is congested, so i think it make sense if
 there is traffic shaper on the sender side, those overlimit packets can
 buffered and dropped at the sender side, rather than sending them to 
the network which affects other traffics.<br></div></div></div></div></div></blockquote><div><br></div><div>This draft isn't proposing a congestion control algorithm, it's exploring the space of possible feedback using RTCP as defined. It might make sense to send feedback as you suggest.&nbsp;</div><br><blockquote type="cite"><div dir="ltr"><div><div><div>&nbsp; 2. How about ack in a given interval, this can be treat as 
variant of per-frame ack(several frames per ack). This is another kind 
which has less load than per-frame ack.<br></div></div></div></div></blockquote><div><br></div><div>Sure.</div><br><blockquote type="cite"><div dir="ltr"><div><div>&nbsp; 3. There can be 
adaptive feedback which feedback when there is need to do so, for 
example, when the given interval is not enough for the adaption(the 
queuing delay goes large very fast), the feedback can be accelerated in 
this case.<br></div></div></div></blockquote><div><br></div>I agree that this is an option.</div><div><br><blockquote type="cite"><div dir="ltr"><div>&nbsp; 4. The feedback strategy has some relationship with sender-based/receiver-based, is this in the scope of this draft?<br></div></div></blockquote><div><br></div><div>I'm not intending to progress this draft, but it's certainly an issue to consider when designing a congestion control algorithm.</div><div><br></div><div>Colin</div><div><br></div><div><br></div><div><br></div><br><blockquote type="cite"><div dir="ltr">Thanks<br><br>Jeromy</div><div class="gmail_extra"><br><br><div class="gmail_quote">
2013/2/26 Colin Perkins <span dir="ltr">&lt;<a href="mailto:csp@csperkins.org" target="_blank">csp@csperkins.org</a>&gt;</span><br><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">
I submitted the following draft that talks about the types of congestion control feedback that can be sent using RTCP as is currently specified, and their suitability for implementing congestion control for unicast multimedia applications. In particular, it talks about the rate of congestion feedback that can be supported.<br>

<br>
Comments are welcome. I'm also happy to present this in Orlando, if the chairs think it useful.<br>
<br>
Colin<br>
<br>
<br>
<br>
<br>
Begin forwarded message:<br>
&gt; From: <a href="mailto:internet-drafts@ietf.org">internet-drafts@ietf.org</a><br>
&gt; Subject: I-D Action: draft-perkins-rmcat-rtp-cc-feedback-00.txt<br>
&gt; Date: 18 February 2013 23:28:42 GMT<br>
&gt; To: <a href="mailto:i-d-announce@ietf.org">i-d-announce@ietf.org</a><br>
&gt; Reply-To: <a href="mailto:internet-drafts@ietf.org">internet-drafts@ietf.org</a><br>
&gt;<br>
&gt;<br>
&gt; A New Internet-Draft is available from the on-line Internet-Drafts directories.<br>
&gt;<br>
&gt;<br>
&gt; &nbsp; &nbsp; &nbsp; Title &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; : On the Use of RTP Control Protocol (RTCP) Feedback for Unicast Multimedia Congestion Control<br>
&gt; &nbsp; &nbsp; &nbsp; Author(s) &nbsp; &nbsp; &nbsp; : Colin Perkins<br>
&gt; &nbsp; &nbsp; &nbsp; Filename &nbsp; &nbsp; &nbsp; &nbsp;: draft-perkins-rmcat-rtp-cc-feedback-00.txt<br>
&gt; &nbsp; &nbsp; &nbsp; Pages &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; : 8<br>
&gt; &nbsp; &nbsp; &nbsp; Date &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;: 2013-02-18<br>
&gt;<br>
&gt; Abstract:<br>
&gt; This memo discusses the types of congestion control feedback that it<br>
&gt; is possible to send using the RTP Control Protocol (RTCP), and their<br>
&gt; suitability of use in implementing congestion control for unicast<br>
&gt; multimedia applications.<br>
&gt;<br>
&gt;<br>
&gt; The IETF datatracker status page for this draft is:<br>
&gt; <a href="https://datatracker.ietf.org/doc/draft-perkins-rmcat-rtp-cc-feedback" target="_blank">https://datatracker.ietf.org/doc/draft-perkins-rmcat-rtp-cc-feedback</a><br>
&gt;<br>
&gt; There's also a htmlized version available at:<br>
&gt; <a href="http://tools.ietf.org/html/draft-perkins-rmcat-rtp-cc-feedback-00" target="_blank">http://tools.ietf.org/html/draft-perkins-rmcat-rtp-cc-feedback-00</a><br>
<span class="HOEnZb"><font color="#888888"><br>
<br>
--<br>
Colin Perkins<br>
<a href="http://csperkins.org/" target="_blank">http://csperkins.org/</a><br>
<br>
<br>
<br>
</font></span></blockquote></div><br></div>
</blockquote></div><br><div apple-content-edited="true">
<span class="Apple-style-span" style="border-collapse: separate; color: rgb(0, 0, 0); font-family: 'Lucida Sans Typewriter'; 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; font-size: medium; "><span class="Apple-style-span" style="font-size: 9px; "><div><br class="Apple-interchange-newline"><br class="khtml-block-placeholder"></div><div>--&nbsp;</div><div></div><div>Colin Perkins</div><div><a href="http://csperkins.org/">http://csperkins.org/</a></div><div><br></div></span></span><br class="Apple-interchange-newline">
</div>
<br></body></html>
--Apple-Mail=_33C6ED5A-B80A-4AD3-9A5F-8C3FED4C6ACF--

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From: Dan Weber <dan@marketsoup.com>
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Subject: [rmcat] RMCAT Evaluation Ideas
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--bcaec51d2980b1aac104d7da86f0
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Hi everyone,

While I sat on the airplane today, it became clear to me that there is a
reasonable way to do some evaluation of congestion control algorithms that
can be both indicative of real use cases (e.g. live streaming video) while
also being easy to test (e.g. fixed rate packet generators).

Three ideas here:

1. Test cases
We will never agree on what a definition of fairness is, and I think it is
a lost cause.  However, we can definitely reach consensus on expected
behavior given various test cases.

Behavioral driven design gives us a great method for defining these.  It
follows the form (given the facts (concrete verifiable constants), when
action or event occurs, then I expect it to do  XYZ.).  I will start a
separate thread kicking this off.


2. Packet Generator Model
After re-reading Varun's draft and speaking with him on the evaluation
scenarios, I think it makes enough sense to build a packet generator to
implement what he suggests.  The time involved isn't substantial, and it
can help us get a unified starting point for testing even if we decide to
change it later.  It's worth at least this much.

My proposed implementation of this will be able to simulate real video
traffic as well as fixed packet generator traffic.  For the real video
traffic, it can be fed a CSV file filled with frame sizes and RTP
timestamps.  For the fixed generator traffic, ladder steps can be used.
 I'm thinking I'll start a project on Google Code, and make any interested
participant a developer.


3. Evaluation Metrics

We need an initial set of metrics for testing any congestion control
algorithm.  To evaluate any congestion control algorithm, we'll want enough
data to build our favorite line charts.  For simplicity, all metrics should
be sampled per second (maybe per millisecond?).  This also allows us to
work with real test environments easily.

Here is my initial and non-exhaustive list of metrics (based off of the
draft):

[Received] In Kbit/s
[Sent] Out Kbit/s
Packets sent/second,
Packets received/second,
Packets lost/second [Determined by using standard RTP packet loss
calculations]
RTT/s
Discards per second [ use a range of buffer calculations, 20ms 40ms, 100ms,
200ms, 500ms]


These metrics should be built into an RTP stack that we standardize for
testing.  My two preferences are jrtplib (C++) and libjitsi (Java).  At
least with both of these, we can contact the authors and get any bugs fixed
if we need them relatively easily.

Thanks,
Dan

--bcaec51d2980b1aac104d7da86f0
Content-Type: text/html; charset=UTF-8
Content-Transfer-Encoding: quoted-printable

Hi everyone,<div><br></div><div>While I sat on the airplane today, it becam=
e clear to me that there is a reasonable way to do some evaluation of conge=
stion control algorithms that can be both indicative of real use cases (e.g=
. live streaming video) while also being easy to test (e.g. fixed rate pack=
et generators).</div>
<div><br></div><div>Three ideas here:</div><div><br></div><div>1. Test case=
s</div><div>We will never agree on what a definition of fairness is, and I =
think it is a lost cause. =C2=A0However, we can definitely reach consensus =
on expected behavior given various test cases.</div>
<div><br></div><div>Behavioral driven design gives us a great method for de=
fining these. =C2=A0It follows the form (given the facts (concrete verifiab=
le constants), when action or event occurs, then I expect it to do =C2=A0XY=
Z.). =C2=A0I will start a separate thread kicking this off.</div>
<div><br></div><div><br></div><div>2. Packet Generator Model</div><div>Afte=
r re-reading Varun&#39;s draft and speaking with him on the evaluation scen=
arios, I think it makes enough sense to build a packet generator to impleme=
nt what he suggests. =C2=A0The time involved isn&#39;t substantial, and it =
can help us get a unified starting point for testing even if we decide to c=
hange it later. =C2=A0It&#39;s worth at least this much.</div>
<div><br></div><div>My proposed implementation of this will be able to simu=
late real video traffic as well as fixed packet generator traffic. =C2=A0Fo=
r the real video traffic, it can be fed a CSV file filled with frame sizes =
and RTP timestamps. =C2=A0For the fixed generator traffic, ladder steps can=
 be used. =C2=A0I&#39;m thinking I&#39;ll start a project on Google Code, a=
nd make any interested participant a developer.</div>
<div><br></div><div><br></div><div>3. Evaluation Metrics</div><div><br></di=
v><div>We need an initial set of metrics for testing any congestion control=
 algorithm. =C2=A0To evaluate any congestion control algorithm, we&#39;ll w=
ant enough data to build our favorite line charts. =C2=A0For simplicity, al=
l metrics should be sampled per second (maybe per millisecond?). =C2=A0This=
 also allows us to work with real test environments easily.</div>
<div><br></div><div>Here is my initial and non-exhaustive list of metrics (=
based off of the draft):</div><div><br></div><div>[Received] In Kbit/s</div=
><div>[Sent] Out Kbit/s</div><div>Packets sent/second,</div><div>Packets re=
ceived/second,=C2=A0</div>
<div>Packets lost/second [Determined by using standard RTP packet loss calc=
ulations]</div><div>RTT/s</div><div>Discards per second [ use a range of bu=
ffer calculations, 20ms 40ms, 100ms, 200ms, 500ms]</div><div><br></div>
<div><br></div><div>These metrics should be built into an RTP stack that we=
 standardize for testing. =C2=A0My two preferences are jrtplib (C++) and li=
bjitsi (Java). =C2=A0At least with both of these, we can contact the author=
s and get any bugs fixed if we need them relatively easily.</div>
<div><br></div><div>Thanks,</div><div>Dan</div><div><pre class=3D"newpage" =
style=3D"font-size:1em;margin-top:0px;margin-bottom:0px"><br></pre></div>

--bcaec51d2980b1aac104d7da86f0--

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Subject: [rmcat] Congestion Control Behavior Suggestion Thread
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--14dae9340717cc4dab04d7dad8c0
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Hi there,

This is the  first and example behavior test case based on the concept I
proposed in "RMCAT Evaluation Ideas".  Please add your own following the
same format.  I seek to aggregate all of these test cases onto a wiki and
establish categories for them.


Given: A video media stream sends at 512kbit/s on a link with an upstream
capacity of 512 kbit/s
When: an FTP Transfer of 1GB file begins uploading
Then: Packet loss occurs.  Throughput should be diminished.  RTPCC should
notify application of developments.

Given: A link with capacity limited at 512 kbit/s
When: A video conference call starts sending Mulaw audio (ptime=20,
81.92kbit/s) and Video @ max kbitrate of 384 kbit/s
Then:  All packets should be received.

...

This may need more work, but it can get us going in the right direction.

Thanks,
Dan

--14dae9340717cc4dab04d7dad8c0
Content-Type: text/html; charset=UTF-8
Content-Transfer-Encoding: quoted-printable

Hi there,<div><br></div><div>This is the =C2=A0first and example behavior t=
est case based on the concept I proposed in &quot;RMCAT Evaluation Ideas&qu=
ot;. =C2=A0Please add your own following the same format. =C2=A0I seek to a=
ggregate all of these test cases onto a wiki and establish categories for t=
hem.</div>
<div><br></div><div><br></div><div>Given: A video media stream sends at 512=
kbit/s on a link with an upstream capacity of 512 kbit/s=C2=A0</div><div>Wh=
en: an FTP Transfer of 1GB file begins uploading</div><div>Then: Packet los=
s occurs. =C2=A0Throughput should be diminished. =C2=A0RTPCC should notify =
application of developments.</div>
<div><br></div><div>Given: A link with capacity limited at 512 kbit/s</div>=
<div>When: A video conference call starts sending Mulaw audio (ptime=3D20, =
81.92kbit/s) and Video @ max kbitrate of 384 kbit/s</div><div>Then: =C2=A0A=
ll packets should be received.</div>
<div><br></div><div>...</div><div><br></div><div>This may need more work, b=
ut it can get us going in the right direction.</div><div><br></div><div>Tha=
nks,</div><div>Dan</div>

--14dae9340717cc4dab04d7dad8c0--

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Hi Dan,

I made a testing tool to test my rate adaptation algorithm 
implementation in my MCU. You can find the source code here:

http://sourceforge.net/p/mcumediaserver/code/745/tree/trunk/cca/

It consist on a random packet generator (which can be parametrized to 
emulate certain video behaviors, like different sizes for I/P frames) 
and a receiever. I implementes a full RTP/RTCP stack with preliminary 
support for NACK and FEC (which I think it is essential for evaluating 
the behavior of the algorithm).

The current algorithm implemented is based on google's congestion 
control algorithm, but it could be also interesting to create a comon 
interface so we could test different algorithms and compare between them.

Best regards
Sergio

El 14/03/2013 4:57, Dan Weber escribiÃ³:
> Hi everyone,
>
> While I sat on the airplane today, it became clear to me that there is 
> a reasonable way to do some evaluation of congestion control 
> algorithms that can be both indicative of real use cases (e.g. live 
> streaming video) while also being easy to test (e.g. fixed rate packet 
> generators).
>
> Three ideas here:
>
> 1. Test cases
> We will never agree on what a definition of fairness is, and I think 
> it is a lost cause.  However, we can definitely reach consensus on 
> expected behavior given various test cases.
>
> Behavioral driven design gives us a great method for defining these. 
>  It follows the form (given the facts (concrete verifiable constants), 
> when action or event occurs, then I expect it to do  XYZ.).  I will 
> start a separate thread kicking this off.
>
>
> 2. Packet Generator Model
> After re-reading Varun's draft and speaking with him on the evaluation 
> scenarios, I think it makes enough sense to build a packet generator 
> to implement what he suggests.  The time involved isn't substantial, 
> and it can help us get a unified starting point for testing even if we 
> decide to change it later.  It's worth at least this much.
>
> My proposed implementation of this will be able to simulate real video 
> traffic as well as fixed packet generator traffic.  For the real video 
> traffic, it can be fed a CSV file filled with frame sizes and RTP 
> timestamps.  For the fixed generator traffic, ladder steps can be 
> used.  I'm thinking I'll start a project on Google Code, and make any 
> interested participant a developer.
>
>
> 3. Evaluation Metrics
>
> We need an initial set of metrics for testing any congestion control 
> algorithm.  To evaluate any congestion control algorithm, we'll want 
> enough data to build our favorite line charts.  For simplicity, all 
> metrics should be sampled per second (maybe per millisecond?).  This 
> also allows us to work with real test environments easily.
>
> Here is my initial and non-exhaustive list of metrics (based off of 
> the draft):
>
> [Received] In Kbit/s
> [Sent] Out Kbit/s
> Packets sent/second,
> Packets received/second,
> Packets lost/second [Determined by using standard RTP packet loss 
> calculations]
> RTT/s
> Discards per second [ use a range of buffer calculations, 20ms 40ms, 
> 100ms, 200ms, 500ms]
>
>
> These metrics should be built into an RTP stack that we standardize 
> for testing.  My two preferences are jrtplib (C++) and libjitsi 
> (Java).  At least with both of these, we can contact the authors and 
> get any bugs fixed if we need them relatively easily.
>
> Thanks,
> Dan


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From: Stefan Holmer <stefan@webrtc.org>
To: Sergio Garcia Murillo <sergio.garcia.murillo@gmail.com>
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Subject: Re: [rmcat] RMCAT Evaluation Ideas
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--047d7b5d2774f5516a04d7df32d1
Content-Type: text/plain; charset=ISO-8859-1
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Another option could be to develop something based on the RTP stack on
http://webrtc.googlecode.com. We have some legacy
code<https://code.google.com/p/webrtc/source/browse/#svn%2Ftrunk%2Fwebrtc%2=
Fmodules%2Frtp_rtcp%2Ftest%2FBWEStandAlone>
which
contains, among other things, various load generators, but it hasn't been
used for quite some time, so I think it would need a bit of work to get
back to a working state.

/Stefan


On Thu, Mar 14, 2013 at 10:04 AM, Sergio Garcia Murillo <
sergio.garcia.murillo@gmail.com> wrote:

> Hi Dan,
>
> I made a testing tool to test my rate adaptation algorithm implementation
> in my MCU. You can find the source code here:
>
> http://sourceforge.net/p/**mcumediaserver/code/745/tree/**trunk/cca/<http=
://sourceforge.net/p/mcumediaserver/code/745/tree/trunk/cca/>
>
> It consist on a random packet generator (which can be parametrized to
> emulate certain video behaviors, like different sizes for I/P frames) and=
 a
> receiever. I implementes a full RTP/RTCP stack with preliminary support f=
or
> NACK and FEC (which I think it is essential for evaluating the behavior o=
f
> the algorithm).
>
> The current algorithm implemented is based on google's congestion control
> algorithm, but it could be also interesting to create a comon interface s=
o
> we could test different algorithms and compare between them.
>
> Best regards
> Sergio
>
> El 14/03/2013 4:57, Dan Weber escribi=F3:
>
>  Hi everyone,
>>
>> While I sat on the airplane today, it became clear to me that there is a
>> reasonable way to do some evaluation of congestion control algorithms th=
at
>> can be both indicative of real use cases (e.g. live streaming video) whi=
le
>> also being easy to test (e.g. fixed rate packet generators).
>>
>> Three ideas here:
>>
>> 1. Test cases
>> We will never agree on what a definition of fairness is, and I think it
>> is a lost cause.  However, we can definitely reach consensus on expected
>> behavior given various test cases.
>>
>> Behavioral driven design gives us a great method for defining these.  It
>> follows the form (given the facts (concrete verifiable constants), when
>> action or event occurs, then I expect it to do  XYZ.).  I will start a
>> separate thread kicking this off.
>>
>>
>> 2. Packet Generator Model
>> After re-reading Varun's draft and speaking with him on the evaluation
>> scenarios, I think it makes enough sense to build a packet generator to
>> implement what he suggests.  The time involved isn't substantial, and it
>> can help us get a unified starting point for testing even if we decide t=
o
>> change it later.  It's worth at least this much.
>>
>> My proposed implementation of this will be able to simulate real video
>> traffic as well as fixed packet generator traffic.  For the real video
>> traffic, it can be fed a CSV file filled with frame sizes and RTP
>> timestamps.  For the fixed generator traffic, ladder steps can be used.
>>  I'm thinking I'll start a project on Google Code, and make any interest=
ed
>> participant a developer.
>>
>>
>> 3. Evaluation Metrics
>>
>> We need an initial set of metrics for testing any congestion control
>> algorithm.  To evaluate any congestion control algorithm, we'll want eno=
ugh
>> data to build our favorite line charts.  For simplicity, all metrics sho=
uld
>> be sampled per second (maybe per millisecond?).  This also allows us to
>> work with real test environments easily.
>>
>> Here is my initial and non-exhaustive list of metrics (based off of the
>> draft):
>>
>> [Received] In Kbit/s
>> [Sent] Out Kbit/s
>> Packets sent/second,
>> Packets received/second,
>> Packets lost/second [Determined by using standard RTP packet loss
>> calculations]
>> RTT/s
>> Discards per second [ use a range of buffer calculations, 20ms 40ms,
>> 100ms, 200ms, 500ms]
>>
>>
>> These metrics should be built into an RTP stack that we standardize for
>> testing.  My two preferences are jrtplib (C++) and libjitsi (Java).  At
>> least with both of these, we can contact the authors and get any bugs fi=
xed
>> if we need them relatively easily.
>>
>> Thanks,
>> Dan
>>
>
>

--047d7b5d2774f5516a04d7df32d1
Content-Type: text/html; charset=ISO-8859-1
Content-Transfer-Encoding: quoted-printable

<div dir=3D"ltr">Another option could be to develop something based on the =
RTP stack on <a href=3D"http://webrtc.googlecode.com" class=3D"cremed">http=
://webrtc.googlecode.com</a>. We have some <a href=3D"https://code.google.c=
om/p/webrtc/source/browse/#svn%2Ftrunk%2Fwebrtc%2Fmodules%2Frtp_rtcp%2Ftest=
%2FBWEStandAlone" class=3D"cremed">legacy code</a>=A0which contains, among =
other things, various load generators, but it hasn&#39;t been used for quit=
e some time, so I think it would need a bit of work to get back to a workin=
g state.<div>
<br></div><div style>/Stefan</div></div><div class=3D"gmail_extra"><br><br>=
<div class=3D"gmail_quote">On Thu, Mar 14, 2013 at 10:04 AM, Sergio Garcia =
Murillo <span dir=3D"ltr">&lt;<a href=3D"mailto:sergio.garcia.murillo@gmail=
.com" target=3D"_blank">sergio.garcia.murillo@gmail.com</a>&gt;</span> wrot=
e:<br>
<blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1p=
x #ccc solid;padding-left:1ex">Hi Dan,<br>
<br>
I made a testing tool to test my rate adaptation algorithm implementation i=
n my MCU. You can find the source code here:<br>
<br>
<a href=3D"http://sourceforge.net/p/mcumediaserver/code/745/tree/trunk/cca/=
" target=3D"_blank">http://sourceforge.net/p/<u></u>mcumediaserver/code/745=
/tree/<u></u>trunk/cca/</a><br>
<br>
It consist on a random packet generator (which can be parametrized to emula=
te certain video behaviors, like different sizes for I/P frames) and a rece=
iever. I implementes a full RTP/RTCP stack with preliminary support for NAC=
K and FEC (which I think it is essential for evaluating the behavior of the=
 algorithm).<br>

<br>
The current algorithm implemented is based on google&#39;s congestion contr=
ol algorithm, but it could be also interesting to create a comon interface =
so we could test different algorithms and compare between them.<br>
<br>
Best regards<br>
Sergio<br>
<br>
El 14/03/2013 4:57, Dan Weber escribi=F3:<div class=3D"HOEnZb"><div class=
=3D"h5"><br>
<blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1p=
x #ccc solid;padding-left:1ex">
Hi everyone,<br>
<br>
While I sat on the airplane today, it became clear to me that there is a re=
asonable way to do some evaluation of congestion control algorithms that ca=
n be both indicative of real use cases (e.g. live streaming video) while al=
so being easy to test (e.g. fixed rate packet generators).<br>

<br>
Three ideas here:<br>
<br>
1. Test cases<br>
We will never agree on what a definition of fairness is, and I think it is =
a lost cause. =A0However, we can definitely reach consensus on expected beh=
avior given various test cases.<br>
<br>
Behavioral driven design gives us a great method for defining these. =A0It =
follows the form (given the facts (concrete verifiable constants), when act=
ion or event occurs, then I expect it to do =A0XYZ.). =A0I will start a sep=
arate thread kicking this off.<br>

<br>
<br>
2. Packet Generator Model<br>
After re-reading Varun&#39;s draft and speaking with him on the evaluation =
scenarios, I think it makes enough sense to build a packet generator to imp=
lement what he suggests. =A0The time involved isn&#39;t substantial, and it=
 can help us get a unified starting point for testing even if we decide to =
change it later. =A0It&#39;s worth at least this much.<br>

<br>
My proposed implementation of this will be able to simulate real video traf=
fic as well as fixed packet generator traffic. =A0For the real video traffi=
c, it can be fed a CSV file filled with frame sizes and RTP timestamps. =A0=
For the fixed generator traffic, ladder steps can be used. =A0I&#39;m think=
ing I&#39;ll start a project on Google Code, and make any interested partic=
ipant a developer.<br>

<br>
<br>
3. Evaluation Metrics<br>
<br>
We need an initial set of metrics for testing any congestion control algori=
thm. =A0To evaluate any congestion control algorithm, we&#39;ll want enough=
 data to build our favorite line charts. =A0For simplicity, all metrics sho=
uld be sampled per second (maybe per millisecond?). =A0This also allows us =
to work with real test environments easily.<br>

<br>
Here is my initial and non-exhaustive list of metrics (based off of the dra=
ft):<br>
<br>
[Received] In Kbit/s<br>
[Sent] Out Kbit/s<br>
Packets sent/second,<br>
Packets received/second,<br>
Packets lost/second [Determined by using standard RTP packet loss calculati=
ons]<br>
RTT/s<br>
Discards per second [ use a range of buffer calculations, 20ms 40ms, 100ms,=
 200ms, 500ms]<br>
<br>
<br>
These metrics should be built into an RTP stack that we standardize for tes=
ting. =A0My two preferences are jrtplib (C++) and libjitsi (Java). =A0At le=
ast with both of these, we can contact the authors and get any bugs fixed i=
f we need them relatively easily.<br>

<br>
Thanks,<br>
Dan<br>
</blockquote>
<br>
</div></div></blockquote></div><br></div>

--047d7b5d2774f5516a04d7df32d1--

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From: "Eggert, Lars" <lars@netapp.com>
To: Dan Weber <dan@marketsoup.com>
Thread-Topic: [rmcat] Congestion Control Behavior Suggestion Thread
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Hi,

On Mar 14, 2013, at 0:20, Dan Weber <dan@marketsoup.com> wrote:
> This may need more work, but it can get us going in the right direction.

thanks for starting this! Yes, there needs to be more detail. In case folks=
 haven't see it yet, we tried to specify such scenarios for doing performan=
ce measurements for TCP extensions a few years ago: http://eggert.org/paper=
s/2008-pfldnet.pdf

The intent with that paper, and I guess the intent now, would be to nail do=
wn all constants sufficiently and describe the parameterization of each tes=
t sufficiently so that someone could set up a simulation or experiment base=
d only on the text. (For the TCP scenarios described in the paper, someone =
- from NEC? - actually did provide an ns2 script.)

Lars=

From vsingh.ietf@gmail.com  Thu Mar 14 06:23:36 2013
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From: Varun Singh <vsingh.ietf@gmail.com>
Date: Thu, 14 Mar 2013 09:23:02 -0400
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Subject: [rmcat] Design Meeting 2
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HI all,

We did a quick recap of the earlier meeting and
Bob Briscoe discussed "History of Fairness"
and discussion centered around
measuring congestion, fairness and trade-off
of throughput, delay and discards.

We will meet again tomorrow (15.03) at 8am
(if Boca 3 is available, we'll continue
the meeting there).

Bob will send a reading list of RFCs and papers.

Attendees:
Randell Jesup
Zahed Sarker
Michael Ramalho
Mo Zanaty
Kevin Gross
Gory Fairhurst
Bob Briscoe
Varun Singh


-- 
http://www.netlab.tkk.fi/~varun/

From vsingh.ietf@gmail.com  Thu Mar 14 06:35:09 2013
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From: Varun Singh <vsingh.ietf@gmail.com>
Date: Thu, 14 Mar 2013 09:34:44 -0400
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Subject: Re: [rmcat] Congestion Control Behavior Suggestion Thread
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Hi Dan,

On Thu, Mar 14, 2013 at 12:20 AM, Dan Weber <dan@marketsoup.com> wrote:
> Hi there,
>
> This is the  first and example behavior test case based on the concept I
> proposed in "RMCAT Evaluation Ideas".  Please add your own following the
> same format.  I seek to aggregate all of these test cases onto a wiki and
> establish categories for them.
>
>
> Given: A video media stream sends at 512kbit/s on a link with an upstream
> capacity of 512 kbit/s
> When: an FTP Transfer of 1GB file begins uploading
> Then: Packet loss occurs.  Throughput should be diminished.  RTPCC should
> notify application of developments.
>
> Given: A link with capacity limited at 512 kbit/s
> When: A video conference call starts sending Mulaw audio (ptime=20,
> 81.92kbit/s) and Video @ max kbitrate of 384 kbit/s
> Then:  All packets should be received.
>

The "Then" part may depend also on what other network characteristics
may be applied (path loss and/or delay).

> ...
>
> This may need more work, but it can get us going in the right direction.
>
> Thanks,
> Dan



-- 
http://www.netlab.tkk.fi/~varun/

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From vsingh.ietf@gmail.com  Thu Mar 14 07:09:46 2013
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From: Varun Singh <vsingh.ietf@gmail.com>
Date: Thu, 14 Mar 2013 10:09:20 -0400
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Cc: rmcat WG <rmcat@ietf.org>, Sergio Garcia Murillo <sergio.garcia.murillo@gmail.com>
Subject: Re: [rmcat] RMCAT Evaluation Ideas
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Hi Stefan, Sergio,

Is it possible for either of you to characterize the behaviour of the
packet generator?
I am mainly looking for plots related to the different video parameters.

Thanks,
Varun

On Thu, Mar 14, 2013 at 5:32 AM, Stefan Holmer <stefan@webrtc.org> wrote:
> Another option could be to develop something based on the RTP stack on
> http://webrtc.googlecode.com. We have some legacy code which contains, am=
ong
> other things, various load generators, but it hasn't been used for quite
> some time, so I think it would need a bit of work to get back to a workin=
g
> state.
>
> /Stefan
>
>
> On Thu, Mar 14, 2013 at 10:04 AM, Sergio Garcia Murillo
> <sergio.garcia.murillo@gmail.com> wrote:
>>
>> Hi Dan,
>>
>> I made a testing tool to test my rate adaptation algorithm implementatio=
n
>> in my MCU. You can find the source code here:
>>
>> http://sourceforge.net/p/mcumediaserver/code/745/tree/trunk/cca/
>>
>> It consist on a random packet generator (which can be parametrized to
>> emulate certain video behaviors, like different sizes for I/P frames) an=
d a
>> receiever. I implementes a full RTP/RTCP stack with preliminary support =
for
>> NACK and FEC (which I think it is essential for evaluating the behavior =
of
>> the algorithm).
>>
>> The current algorithm implemented is based on google's congestion contro=
l
>> algorithm, but it could be also interesting to create a comon interface =
so
>> we could test different algorithms and compare between them.
>>
>> Best regards
>> Sergio
>>
>> El 14/03/2013 4:57, Dan Weber escribi=F3:
>>
>>> Hi everyone,
>>>
>>> While I sat on the airplane today, it became clear to me that there is =
a
>>> reasonable way to do some evaluation of congestion control algorithms t=
hat
>>> can be both indicative of real use cases (e.g. live streaming video) wh=
ile
>>> also being easy to test (e.g. fixed rate packet generators).
>>>
>>> Three ideas here:
>>>
>>> 1. Test cases
>>> We will never agree on what a definition of fairness is, and I think it
>>> is a lost cause.  However, we can definitely reach consensus on expecte=
d
>>> behavior given various test cases.
>>>
>>> Behavioral driven design gives us a great method for defining these.  I=
t
>>> follows the form (given the facts (concrete verifiable constants), when
>>> action or event occurs, then I expect it to do  XYZ.).  I will start a
>>> separate thread kicking this off.
>>>
>>>
>>> 2. Packet Generator Model
>>> After re-reading Varun's draft and speaking with him on the evaluation
>>> scenarios, I think it makes enough sense to build a packet generator to
>>> implement what he suggests.  The time involved isn't substantial, and i=
t can
>>> help us get a unified starting point for testing even if we decide to c=
hange
>>> it later.  It's worth at least this much.
>>>
>>> My proposed implementation of this will be able to simulate real video
>>> traffic as well as fixed packet generator traffic.  For the real video
>>> traffic, it can be fed a CSV file filled with frame sizes and RTP
>>> timestamps.  For the fixed generator traffic, ladder steps can be used.=
  I'm
>>> thinking I'll start a project on Google Code, and make any interested
>>> participant a developer.
>>>
>>>
>>> 3. Evaluation Metrics
>>>
>>> We need an initial set of metrics for testing any congestion control
>>> algorithm.  To evaluate any congestion control algorithm, we'll want en=
ough
>>> data to build our favorite line charts.  For simplicity, all metrics sh=
ould
>>> be sampled per second (maybe per millisecond?).  This also allows us to=
 work
>>> with real test environments easily.
>>>
>>> Here is my initial and non-exhaustive list of metrics (based off of the
>>> draft):
>>>
>>> [Received] In Kbit/s
>>> [Sent] Out Kbit/s
>>> Packets sent/second,
>>> Packets received/second,
>>> Packets lost/second [Determined by using standard RTP packet loss
>>> calculations]
>>> RTT/s
>>> Discards per second [ use a range of buffer calculations, 20ms 40ms,
>>> 100ms, 200ms, 500ms]
>>>
>>>
>>> These metrics should be built into an RTP stack that we standardize for
>>> testing.  My two preferences are jrtplib (C++) and libjitsi (Java).  At
>>> least with both of these, we can contact the authors and get any bugs f=
ixed
>>> if we need them relatively easily.
>>>
>>> Thanks,
>>> Dan
>>
>>
>



--=20
http://www.netlab.tkk.fi/~varun/

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On Mar 14, 2013, at 10:07, Zaheduzzaman Sarker <zaheduzzaman.sarker@ericsso=
n.com> wrote:
> For the fixed generator traffic, ladder steps can be used.  I'm thinking =
I'll start a project on Google Code, and make any interested participant a =
developer

Suggestion: take a look at if an existing generator can be extended, rather=
 than starting from scratch? E.g. flowgrind

Lars=

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From: "Bill Ver Steeg (versteb)" <versteb@cisco.com>
To: "Eggert, Lars" <lars@netapp.com>
Thread-Topic: [rmcat] RMCAT Evaluation Ideas
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Subject: Re: [rmcat] RMCAT Evaluation Ideas
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Iperf can also generate a wide variety of traffic. I have also recently mod=
ified it  to track one-way delay, and can help with changing the code to se=
nd data patterns and track delay.=20

I will try to make tomorrow's 8am meeting, and we can compare notes on what=
 tools are available.=20

Bill VerSteeg

On Mar 14, 2013, at 10:15 AM, "Eggert, Lars" <lars@netapp.com> wrote:

> On Mar 14, 2013, at 10:07, Zaheduzzaman Sarker <zaheduzzaman.sarker@erics=
son.com> wrote:
>> For the fixed generator traffic, ladder steps can be used.  I'm thinking=
 I'll start a project on Google Code, and make any interested participant a=
 developer
>=20
> Suggestion: take a look at if an existing generator can be extended, rath=
er than starting from scratch? E.g. flowgrind
>=20
> Lars

From bob.briscoe@bt.com  Thu Mar 14 08:08:10 2013
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Subject: [rmcat] History of Fairness Discussions in the IETF
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--=====================_440193949==.ALT
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RMCAT-ers,

At the fairness design team mtg this morning, I promised to send this 
list of RFCs & docs to the list. If you want to cut to the chase, 
jump over the pre-history and read from RFC5033 onwards.

RFCs that specify best current practice, guidelines and principles 
for designers of congestion control:
- the `Gateway Congestion Control Survey' 
[<http://tools.ietf.org/html/rfc1254>RFC1254],
- recommendations on AQM [<http://tools.ietf.org/html/rfc2309>RFC2309],
- `Congestion Control Principles' 
[<http://tools.ietf.org/html/rfc2914>RFC2914],
- `General Architectural and Policy Considerations' 
[<http://tools.ietf.org/html/rfc3426>RFC3426]
- IAB Concerns Regarding Congestion Control for Voice Traffic 
[<http://tools.ietf.org/html/rfc3714>RFC3714];

RFCs that recommend how new transport protocols should interact with 
existing ones:
- recommendations on AQM [<http://tools.ietf.org/html/rfc2309>RFC2309],
- Criteria for Evaluating Reliable Multicast Transports 
[<http://tools.ietf.org/html/rfc2357>RFC2357],
- `Congestion Control Principles' 
[<http://tools.ietf.org/html/rfc2914>RFC2914] and
- Guidelines for new RTP profiles [<http://tools.ietf.org/html/rfc3550>RFC3550]
________________ VVV _____skip to here if in a hurry _____ VVV ______
- Specifying New Congestion Control Algorithms 
[<http://tools.ietf.org/html/rfc5033>RFC5033]

Individual contributions (that led to RFC5033)
- Flow Rate Fairness: Dismantling a Religion 
[<http://tools.ietf.org/internet-drafts/draft-briscoe-tsvarea-fair-02.pdf>draft-briscoe-tsvarea-fair-02.pdf]
- Problem Statement: Transport Protocols Don't Have To Do Fairness 
[<http://tools.ietf.org/html/draft-briscoe-tsvwg-relax-fairness>draft-briscoe-tsvwg-relax-fairness]
- Comments on the Usefulness of Simple Best-Effort Traffic 
[<http://tools.ietf.org/html/rfc5290>RFC5290]

Fairness Metrics
- ConEx Concepts & Use Cases [<http://tools.ietf.org/html/rfc6789>RFC6789]

Manifests of RFCs relevant to Fairness & Congestion Control:
- http://tools.ietf.org/html/draft-briscoe-tsvarea-fair-02#section-9
- Congestion Control in the RFC Series 
[<http://tools.ietf.org/html/rfc5783>RFC5783]


Bob


________________________________________________________________
Bob Briscoe,                                                  BT 
--=====================_440193949==.ALT
Content-Type: text/html; charset="us-ascii"

<html>
<body>
RMCAT-ers,<br><br>
At the fairness design team mtg this morning, I promised to send this
list of RFCs &amp; docs to the list. If you want to cut to the chase,
jump over the pre-history and read from RFC5033 onwards.<br><br>
RFCs that specify best current practice, guidelines and principles for
designers of congestion control:<br>
- the `Gateway Congestion Control Survey'
[<a href="http://tools.ietf.org/html/rfc1254">RFC1254</a>], <br>
- recommendations on AQM
[<a href="http://tools.ietf.org/html/rfc2309">RFC2309</a>], <br>
- `Congestion Control Principles'
[<a href="http://tools.ietf.org/html/rfc2914">RFC2914</a>], <br>
- `General Architectural and Policy Considerations'
[<a href="http://tools.ietf.org/html/rfc3426">RFC3426</a>] <br>
- IAB Concerns Regarding Congestion Control for Voice Traffic
[<a href="http://tools.ietf.org/html/rfc3714">RFC3714</a>]; <br><br>
RFCs that recommend how new transport protocols should interact with
existing ones: <br>
- recommendations on AQM
[<a href="http://tools.ietf.org/html/rfc2309">RFC2309</a>], <br>
- Criteria for Evaluating Reliable Multicast Transports
[<a href="http://tools.ietf.org/html/rfc2357">RFC2357</a>], <br>
- `Congestion Control Principles'
[<a href="http://tools.ietf.org/html/rfc2914">RFC2914</a>] and <br>
- Guidelines for new RTP profiles
[<a href="http://tools.ietf.org/html/rfc3550">RFC3550</a>]<br>
________________ VVV _____skip to here if in a hurry _____ VVV
______<br>
- Specifying New Congestion Control Algorithms
[<a href="http://tools.ietf.org/html/rfc5033">RFC5033</a>]<br><br>
Individual contributions (that led to RFC5033)<br>
- Flow Rate Fairness: Dismantling a Religion
[<a href="http://tools.ietf.org/internet-drafts/draft-briscoe-tsvarea-fair-02.pdf">
draft-briscoe-tsvarea-fair-02.pdf</a>]<br>
- Problem Statement: Transport Protocols Don't Have To Do Fairness
[<a href="http://tools.ietf.org/html/draft-briscoe-tsvwg-relax-fairness">
draft-briscoe-tsvwg-relax-fairness</a>]<br>
- Comments on the Usefulness of Simple Best-Effort Traffic
[<a href="http://tools.ietf.org/html/rfc5290">RFC5290</a>]<br><br>
Fairness Metrics<br>
- ConEx Concepts &amp; Use Cases
[<a href="http://tools.ietf.org/html/rfc6789">RFC6789</a>]<br><br>
Manifests of RFCs relevant to Fairness &amp; Congestion Control:<br>
-
<a href="http://tools.ietf.org/html/draft-briscoe-tsvarea-fair-02#section-9" eudora="autourl">
http://tools.ietf.org/html/draft-briscoe-tsvarea-fair-02#section-9<br>
</a>- Congestion Control in the RFC Series
[<a href="http://tools.ietf.org/html/rfc5783">RFC5783</a>]<br><br>
<br>
Bob<br><br>
<x-sigsep><p></x-sigsep>
________________________________________________________________<br>
Bob
Briscoe,&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;
BT</body>
</html>

--=====================_440193949==.ALT--

From mirja.kuehlewind@ikr.uni-stuttgart.de  Thu Mar 14 08:22:23 2013
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From: Mirja Kuehlewind <mirja.kuehlewind@ikr.uni-stuttgart.de>
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Subject: Re: [rmcat] RMCAT Evaluation Ideas
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Hi,

just to name one more option:

we have a Java-based, event-driven simulation framework which we recently=20
extended to integrate a virtual machine (using QEMU). That means we can use=
=20
traffic generated by a real operating system in simulations. Currently we=20
only have a small relay program running in the virtual machine that can be=
=20
controlled by the simulation program to send chunks of data. Thus a specifi=
c=20
generator could be implemented in simulation program. The relay program sen=
ds=20
only TCP traffic at the moment. Moreover, we can also run a real applicatio=
n=20
in the virtual machine. At the moment we are still working on the framework=
=2E=20
Before we will publish the code, we need to find some more bug fixes. But t=
he=20
simulations are working and we are using this for all our current TCP=20
simulations.

Mirja


On Thursday 14 March 2013 15:07:38 Zaheduzzaman Sarker wrote:
> 2. Packet Generator Model
> After re-reading Varun's draft and speaking with him on the evaluation
> scenarios, I think it makes enough sense to build a packet generator to
> implement what he suggests.  The time involved isn't substantial, and it
> can help us get a unified starting point for testing even if we decide to
> change it later.  It's worth at least this much.
>
> My proposed implementation of this will be able to simulate real video
> traffic as well as fixed packet generator traffic.  For the real video
> traffic, it can be fed a CSV file filled with frame sizes and RTP
> timestamps. [Zaheduzzaman Sarker] we already use some sort of packet
> generator in our simulations, right? And it might be also not that hard to
> come up with a packet generator. However, the hard part is to realize how
> representative those packet generator are for our usecases at hand? IMHO,
> we will need to verify the generate packet trace a with some reference
> trace. Also remember we might have QoE as evaluation  metric too and I he=
ar
> people are also talking about subjective quality related to fairness. It
> will be hard to achieve that with just simple packet generators. For the
> fixed generator traffic, ladder steps can be used.  I'm thinking I'll sta=
rt
> a project on Google Code, and make any interested participant a developer
>
>
>
> BR
>
> Zahed



=2D-=20
=2D------------------------------------------------------------------
Dipl.-Ing. Mirja K=FChlewind
Institute of Communication Networks and Computer Engineering (IKR)
University of Stuttgart, Germany
Pfaffenwaldring 47, D-70569 Stuttgart

tel: +49(0)711/685-67973
email: mirja.kuehlewind@ikr.uni-stuttgart.de
web: www.ikr.uni-stuttgart.de
=2D------------------------------------------------------------------

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From: "Xiaoqing Zhu (xiaoqzhu)" <xiaoqzhu@cisco.com>
To: Mirja Kuehlewind <mirja.kuehlewind@ikr.uni-stuttgart.de>
Thread-Topic: [rmcat] RMCAT Evaluation Ideas
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Date: Thu, 14 Mar 2013 15:41:58 +0000
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References: <CACrWU7MrhqdwZgKt83X_i9OK4B_RiPOfaKoTGEqZm8eim6L-0w@mail.gmail.com> <E0F7A68B07B53F4FBD12DABD61CBA90E098B03@ESESSMB307.ericsson.se> <201303141612.02292.mirja.kuehlewind@ikr.uni-stuttgart.de>
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Hi,=20

Great idea on the packet generator model. We will also be happy to contribu=
te on this piece as well.=20

In the past we have put together some some ns-2 code for via in two approac=
hes:=20

- trace-driven:  where the packet sizes and intervals are taken from h.264 =
encoding result of video sequences. one could also switch between different=
 rate versions of the same video content at GOP boundaries.=20

- statistical modeling: where the encoder rate control behavior is modeled =
statistically, with tunable parameters on burstiness, range of deviation fr=
om target rate, and lag in reaction (i.e., how long it takes the encoder to=
 react to a change in target rate).=20

Thanks,
Xiaoqing





On Mar 14, 2013, at 10:12 AM, Mirja Kuehlewind wrote:

> Hi,
>=20
> just to name one more option:
>=20
> we have a Java-based, event-driven simulation framework which we recently=
=20
> extended to integrate a virtual machine (using QEMU). That means we can u=
se=20
> traffic generated by a real operating system in simulations. Currently we=
=20
> only have a small relay program running in the virtual machine that can b=
e=20
> controlled by the simulation program to send chunks of data. Thus a speci=
fic=20
> generator could be implemented in simulation program. The relay program s=
ends=20
> only TCP traffic at the moment. Moreover, we can also run a real applicat=
ion=20
> in the virtual machine. At the moment we are still working on the framewo=
rk.=20
> Before we will publish the code, we need to find some more bug fixes. But=
 the=20
> simulations are working and we are using this for all our current TCP=20
> simulations.
>=20
> Mirja
>=20
>=20
> On Thursday 14 March 2013 15:07:38 Zaheduzzaman Sarker wrote:
>> 2. Packet Generator Model
>> After re-reading Varun's draft and speaking with him on the evaluation
>> scenarios, I think it makes enough sense to build a packet generator to
>> implement what he suggests.  The time involved isn't substantial, and it
>> can help us get a unified starting point for testing even if we decide t=
o
>> change it later.  It's worth at least this much.
>>=20
>> My proposed implementation of this will be able to simulate real video
>> traffic as well as fixed packet generator traffic.  For the real video
>> traffic, it can be fed a CSV file filled with frame sizes and RTP
>> timestamps. [Zaheduzzaman Sarker] we already use some sort of packet
>> generator in our simulations, right? And it might be also not that hard =
to
>> come up with a packet generator. However, the hard part is to realize ho=
w
>> representative those packet generator are for our usecases at hand? IMHO=
,
>> we will need to verify the generate packet trace a with some reference
>> trace. Also remember we might have QoE as evaluation  metric too and I h=
ear
>> people are also talking about subjective quality related to fairness. It
>> will be hard to achieve that with just simple packet generators. For the
>> fixed generator traffic, ladder steps can be used.  I'm thinking I'll st=
art
>> a project on Google Code, and make any interested participant a develope=
r
>>=20
>>=20
>>=20
>> BR
>>=20
>> Zahed
>=20
>=20
>=20
> --=20
> -------------------------------------------------------------------
> Dipl.-Ing. Mirja K=FChlewind
> Institute of Communication Networks and Computer Engineering (IKR)
> University of Stuttgart, Germany
> Pfaffenwaldring 47, D-70569 Stuttgart
>=20
> tel: +49(0)711/685-67973
> email: mirja.kuehlewind@ikr.uni-stuttgart.de
> web: www.ikr.uni-stuttgart.de
> -------------------------------------------------------------------


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From: "Xiaoqing Zhu (xiaoqzhu)" <xiaoqzhu@cisco.com>
To: rmcat WG <rmcat@ietf.org>
Thread-Topic: New Version Notification for draft-zhu-rmcat-nada-01.txt
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Hi,

We submitted the updated version of the NADA scheme, as presented by Rong o=
n Monday.

The major change from the last version is that all forms of congestion sign=
als from the network (loss/delay/ECN/PCN) are now integrated into a composi=
te signal (which we call "equivalent delay"), so that the sender can react =
in exactly the same manner. No more mode switching between ECN-enabled and =
ECN-disabled at the sender, as in draft-00 and criticized by Michael in pre=
vious email discussions.

We also much appreciate all the feedback and discussions on the scheme duri=
ng the WG meeting on Monday, as summarized in Spencer's notes. Will try to =
address these issues, clarify some of the doubts, and improve our design ac=
cordingly.

Thanks,
Xiaoqing


Begin forwarded message:

From: <internet-drafts@ietf.org<mailto:internet-drafts@ietf.org>>
Date: March 12, 2013 1:53:42 PM CDT
To: <xiaoqzhu@cisco.com<mailto:xiaoqzhu@cisco.com>>
Cc: <ropan@cisco.com<mailto:ropan@cisco.com>>
Subject: New Version Notification for draft-zhu-rmcat-nada-01.txt


A new version of I-D, draft-zhu-rmcat-nada-01.txt
has been successfully submitted by Xiaoqing Zhu and posted to the
IETF repository.

Filename: draft-zhu-rmcat-nada
Revision: 01
Title: NADA: A Unified Congestion Control Scheme for Real-Time Media
Creation date: 2013-03-12
Group: Individual Submission
Number of pages: 11
URL:             http://www.ietf.org/internet-drafts/draft-zhu-rmcat-nada-0=
1.txt
Status:          http://datatracker.ietf.org/doc/draft-zhu-rmcat-nada
Htmlized:        http://tools.ietf.org/html/draft-zhu-rmcat-nada-01
Diff:            http://www.ietf.org/rfcdiff?url2=3Ddraft-zhu-rmcat-nada-01

Abstract:
  This document describes a scheme named network-assisted dynamic
  adaptation (NADA), a novel congestion control approach for
  interactive real-time media applications, such as video conferencing.
  In the proposed scheme, the sender regulates its sending rate based
  on either implicit or explicit congestion signaling, in a unified
  approach. The scheme can benefit from explicit congestion
  notification (ECN) markings from network nodes. It also maintains
  consistent sender behavior in the absence of such markings, by
  reacting to queuing delays and packet losses instead.

  We present here the overall system architecture, recommended
  behaviors at the sender and the receiver, as well as expected network
  node operations. Results from extensive simulation studies of the
  proposed scheme are available upon request.




The IETF Secretariat



--_000_E7175A8E3DC14048A7D3020E0338C1FA1238B58Exmbalnx13ciscoc_
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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; ">
Hi,&nbsp;
<div><br>
</div>
<div>We submitted the updated version of the NADA scheme, as presented by R=
ong on Monday. &nbsp;</div>
<div><br>
</div>
<div>The major change from the last version is that all forms of congestion=
 signals from the network (loss/delay/ECN/PCN) are now integrated into a co=
mposite signal (which we call &quot;equivalent delay&quot;), so that the se=
nder can react in exactly the same manner.
 No more mode switching between ECN-enabled and ECN-disabled at the sender,=
 as in draft-00 and criticized by Michael in previous email discussions.&nb=
sp;</div>
<div><br>
</div>
<div>We also much appreciate all the feedback and discussions on the scheme=
 during the WG meeting on Monday, as summarized in Spencer's notes. Will tr=
y to address these issues, clarify some of the doubts, and improve our desi=
gn accordingly.&nbsp;</div>
<div><br>
</div>
<div>Thanks,</div>
<div>Xiaoqing</div>
<div><br>
</div>
<div>
<div><br>
<div>Begin forwarded message:</div>
<br class=3D"Apple-interchange-newline">
<blockquote type=3D"cite">
<div style=3D"margin-top: 0px; margin-right: 0px; margin-bottom: 0px; margi=
n-left: 0px;">
<span style=3D"font-family:'Helvetica'; font-size:medium; color:rgba(0, 0, =
0, 1);"><b>From:
</b></span><span style=3D"font-family:'Helvetica'; font-size:medium;">&lt;<=
a href=3D"mailto:internet-drafts@ietf.org">internet-drafts@ietf.org</a>&gt;=
<br>
</span></div>
<div style=3D"margin-top: 0px; margin-right: 0px; margin-bottom: 0px; margi=
n-left: 0px;">
<span style=3D"font-family:'Helvetica'; font-size:medium; color:rgba(0, 0, =
0, 1);"><b>Date:
</b></span><span style=3D"font-family:'Helvetica'; font-size:medium;">March=
 12, 2013 1:53:42 PM CDT<br>
</span></div>
<div style=3D"margin-top: 0px; margin-right: 0px; margin-bottom: 0px; margi=
n-left: 0px;">
<span style=3D"font-family:'Helvetica'; font-size:medium; color:rgba(0, 0, =
0, 1);"><b>To:
</b></span><span style=3D"font-family:'Helvetica'; font-size:medium;">&lt;<=
a href=3D"mailto:xiaoqzhu@cisco.com">xiaoqzhu@cisco.com</a>&gt;<br>
</span></div>
<div style=3D"margin-top: 0px; margin-right: 0px; margin-bottom: 0px; margi=
n-left: 0px;">
<span style=3D"font-family:'Helvetica'; font-size:medium; color:rgba(0, 0, =
0, 1);"><b>Cc:
</b></span><span style=3D"font-family:'Helvetica'; font-size:medium;">&lt;<=
a href=3D"mailto:ropan@cisco.com">ropan@cisco.com</a>&gt;<br>
</span></div>
<div style=3D"margin-top: 0px; margin-right: 0px; margin-bottom: 0px; margi=
n-left: 0px;">
<span style=3D"font-family:'Helvetica'; font-size:medium; color:rgba(0, 0, =
0, 1);"><b>Subject:
</b></span><span style=3D"font-family:'Helvetica'; font-size:medium;"><b>Ne=
w Version Notification for draft-zhu-rmcat-nada-01.txt</b><br>
</span></div>
<br>
<div><br>
A new version of I-D, draft-zhu-rmcat-nada-01.txt<br>
has been successfully submitted by Xiaoqing Zhu and posted to the<br>
IETF repository.<br>
<br>
Filename:<span class=3D"Apple-tab-span" style=3D"white-space:pre"> </span>d=
raft-zhu-rmcat-nada<br>
Revision:<span class=3D"Apple-tab-span" style=3D"white-space:pre"> </span>0=
1<br>
Title:<span class=3D"Apple-tab-span" style=3D"white-space:pre"> </span><spa=
n class=3D"Apple-tab-span" style=3D"white-space:pre"></span>NADA: A Unified=
 Congestion Control Scheme for Real-Time Media<br>
Creation date:<span class=3D"Apple-tab-span" style=3D"white-space:pre"> </s=
pan>2013-03-12<br>
Group:<span class=3D"Apple-tab-span" style=3D"white-space:pre"> </span><spa=
n class=3D"Apple-tab-span" style=3D"white-space:pre"></span>Individual Subm=
ission<br>
Number of pages: 11<br>
URL: &nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbs=
p;<a href=3D"http://www.ietf.org/internet-drafts/draft-zhu-rmcat-nada-01.tx=
t">http://www.ietf.org/internet-drafts/draft-zhu-rmcat-nada-01.txt</a><br>
Status: &nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;<a href=3D"ht=
tp://datatracker.ietf.org/doc/draft-zhu-rmcat-nada">http://datatracker.ietf=
.org/doc/draft-zhu-rmcat-nada</a><br>
Htmlized: &nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;<a href=3D"http://tools=
.ietf.org/html/draft-zhu-rmcat-nada-01">http://tools.ietf.org/html/draft-zh=
u-rmcat-nada-01</a><br>
Diff: &nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;<a =
href=3D"http://www.ietf.org/rfcdiff?url2=3Ddraft-zhu-rmcat-nada-01">http://=
www.ietf.org/rfcdiff?url2=3Ddraft-zhu-rmcat-nada-01</a><br>
<br>
Abstract:<br>
&nbsp;&nbsp;This document describes a scheme named network-assisted dynamic=
<br>
&nbsp;&nbsp;adaptation (NADA), a novel congestion control approach for<br>
&nbsp;&nbsp;interactive real-time media applications, such as video confere=
ncing.<br>
&nbsp;&nbsp;In the proposed scheme, the sender regulates its sending rate b=
ased<br>
&nbsp;&nbsp;on either implicit or explicit congestion signaling, in a unifi=
ed<br>
&nbsp;&nbsp;approach. The scheme can benefit from explicit congestion<br>
&nbsp;&nbsp;notification (ECN) markings from network nodes. It also maintai=
ns<br>
&nbsp;&nbsp;consistent sender behavior in the absence of such markings, by<=
br>
&nbsp;&nbsp;reacting to queuing delays and packet losses instead.<br>
<br>
&nbsp;&nbsp;We present here the overall system architecture, recommended<br=
>
&nbsp;&nbsp;behaviors at the sender and the receiver, as well as expected n=
etwork<br>
&nbsp;&nbsp;node operations. Results from extensive simulation studies of t=
he<br>
&nbsp;&nbsp;proposed scheme are available upon request.<br>
<br>
<br>
<br>
<br>
The IETF Secretariat<br>
<br>
</div>
</blockquote>
</div>
<br>
</div>
</body>
</html>

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Subject: [rmcat] Packet Generator Requirements
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Hi everyone,

After looking at the technology suggestions and responses to the last
thread (RMCAT Evaluation Ideas), I think it would make sense if we defined
some basic criteria for what makes an effective testing tool.

My intent is to utilize a tool in which we can get true apples to apples
comparisons while being inclusive of as many people as possible.  Likewise,
it should also be reasonably comparable to a real world implementation.

Here are some questions that come to mind:

1) Does the testing tool (packet generator) implement or use an existing
RTP stack implementation?
2) Is the tool or suggested application portable? e.g. Can we run this on
windows, linux, and mac os x?
3) Is the application framework extensible to support new flows?  e.g. What
difficulty is presented if someone wants to send an actual video stream
versus just generated packets?
4) Is it possible to implement congestion control algorithms on top of it
or its output?
5) What are all of the metrics it supports?  Does it support outputting
these metrics in common units such as seconds, packets, etc.?
6) Does it output data in a human readable or easily parseable format such
as CSV or text?
7) Does the data it outputs support the generation of line charts?

Feel free to add any questions you think may fit.

If you have an implementation that you think is worth standardizing on,
please answer these questions with your submission.

Thanks!
Dan

--14dae9340e3b32c47604d7e5d59b
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Hi everyone,<br><br>After looking at the technology suggestions and respons=
es to the last thread (RMCAT Evaluation Ideas), I think it would make sense=
 if we defined some basic criteria for what makes an effective testing tool=
.<br>
<br>My intent is to utilize a tool in which we can get true apples to apple=
s comparisons while being inclusive of as many people as possible.=C2=A0 Li=
kewise, it should also be reasonably comparable to a real world implementat=
ion.<br>
<br>Here are some questions that come to mind:<br><br>1) Does the testing t=
ool (packet generator) implement or use an existing RTP stack implementatio=
n?<br>2) Is the tool or suggested application portable? e.g. Can we run thi=
s on windows, linux, and mac os x?<br>
3) Is the application framework extensible to support new flows?=C2=A0 e.g.=
 What difficulty is presented if someone wants to send an actual video stre=
am versus just generated packets?<br>4) Is it possible to implement congest=
ion control algorithms on top of it or its output?<br>
5) What are all of the metrics it supports?=C2=A0 Does it support outputtin=
g these metrics in common units such as seconds, packets, etc.?<br>6) Does =
it output data in a human readable or easily parseable format such as CSV o=
r text?<br>
7) Does the data it outputs support the generation of line charts?<br><br>F=
eel free to add any questions you think may fit.<br><br>If you have an impl=
ementation that you think is worth standardizing on, please answer these qu=
estions with your submission.<br>
<br>Thanks!<br>Dan<br>

--14dae9340e3b32c47604d7e5d59b--

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From: Dan Weber <dan@marketsoup.com>
To: Varun Singh <vsingh.ietf@gmail.com>
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Subject: Re: [rmcat] Congestion Control Behavior Suggestion Thread
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--14dae9340eebbd92b204d7e5f38d
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Hi Varun,

On Thu, Mar 14, 2013 at 7:34 AM, Varun Singh <vsingh.ietf@gmail.com> wrote:

> Hi Dan,
>
> On Thu, Mar 14, 2013 at 12:20 AM, Dan Weber <dan@marketsoup.com> wrote:
> >
> > Given: A video media stream sends at 512kbit/s on a link with an upstream
> > capacity of 512 kbit/s
> > When: an FTP Transfer of 1GB file begins uploading
> > Then: Packet loss occurs.  Throughput should be diminished.  RTPCC should
> > notify application of developments.
> >
> > Given: A link with capacity limited at 512 kbit/s
> > When: A video conference call starts sending Mulaw audio (ptime=20,
> > 81.92kbit/s) and Video @ max kbitrate of 384 kbit/s
> > Then:  All packets should be received.
> >
>
> The "Then" part may depend also on what other network characteristics
> may be applied (path loss and/or delay).
>
>
> I concur.  Even putting together these surprisingly simple behavior cases
is difficult.  Every single one of these cases needs to be tangible and
easily implementable.

Your specific note carries relevance.  How do you define the network
environment?  Linux's Network Emulation (netem,
http://www.linuxfoundation.org/collaborate/workgroups/networking/netem)
provides a very valuable toolset for this sort of thing.

Do you think it would make sense to use its capabilities for defining our
'given' case?

I'm looking forward to reading Lars's paper to see if it may provide some
more insight into this matter.

Thanks,
Dan

--14dae9340eebbd92b204d7e5f38d
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Hi Varun,<br><br><div class=3D"gmail_quote">On Thu, Mar 14, 2013 at 7:34 AM=
, Varun Singh <span dir=3D"ltr">&lt;<a href=3D"mailto:vsingh.ietf@gmail.com=
" target=3D"_blank">vsingh.ietf@gmail.com</a>&gt;</span> wrote:<br><blockqu=
ote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1px #ccc s=
olid;padding-left:1ex">
Hi Dan,<br>
<div class=3D"im"><br>
On Thu, Mar 14, 2013 at 12:20 AM, Dan Weber &lt;<a href=3D"mailto:dan@marke=
tsoup.com">dan@marketsoup.com</a>&gt; wrote:<br>
&gt;<br>
&gt; Given: A video media stream sends at 512kbit/s on a link with an upstr=
eam<br>
&gt; capacity of 512 kbit/s<br>
&gt; When: an FTP Transfer of 1GB file begins uploading<br>
&gt; Then: Packet loss occurs. =C2=A0Throughput should be diminished. =C2=
=A0RTPCC should<br>
&gt; notify application of developments.<br>
&gt;<br>
&gt; Given: A link with capacity limited at 512 kbit/s<br>
&gt; When: A video conference call starts sending Mulaw audio (ptime=3D20,<=
br>
&gt; 81.92kbit/s) and Video @ max kbitrate of 384 kbit/s<br>
&gt; Then: =C2=A0All packets should be received.<br>
&gt;<br>
<br>
</div>The &quot;Then&quot; part may depend also on what other network chara=
cteristics<br>
may be applied (path loss and/or delay).<br>
<div class=3D"HOEnZb"><div class=3D"h5"><br>
<br></div></div></blockquote><div>I concur.=C2=A0 Even putting together the=
se surprisingly simple behavior cases is difficult.=C2=A0 Every single one =
of these cases needs to be tangible and easily implementable.=C2=A0 <br><br=
>Your specific note carries relevance.=C2=A0 How do you define the network =
environment?=C2=A0 Linux&#39;s Network Emulation (netem, <a href=3D"http://=
www.linuxfoundation.org/collaborate/workgroups/networking/netem">http://www=
.linuxfoundation.org/collaborate/workgroups/networking/netem</a>) provides =
a very valuable toolset for this sort of thing.=C2=A0 <br>
<br>Do you think it would make sense to use its capabilities for defining o=
ur &#39;given&#39; case? <br><br>I&#39;m looking forward to reading Lars&#3=
9;s paper to see if it may provide some more insight into this matter.<br>
<br>Thanks,<br>Dan<br><br></div></div>

--14dae9340eebbd92b204d7e5f38d--

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Subject: Re: [rmcat] Congestion Control Behavior Suggestion Thread
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--14dae93403918ad7c204d7e64c4b
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On 14.3.2013, at 13.35, Dan Weber <dan@marketsoup.com> wrote:

Hi Varun,

On Thu, Mar 14, 2013 at 7:34 AM, Varun Singh <vsingh.ietf@gmail.com> wrote:

> Hi Dan,
>
> On Thu, Mar 14, 2013 at 12:20 AM, Dan Weber <dan@marketsoup.com> wrote:
> >
> > Given: A video media stream sends at 512kbit/s on a link with an upstream
> > capacity of 512 kbit/s
> > When: an FTP Transfer of 1GB file begins uploading
> > Then: Packet loss occurs.  Throughput should be diminished.  RTPCC should
> > notify application of developments.
> >
> > Given: A link with capacity limited at 512 kbit/s
> > When: A video conference call starts sending Mulaw audio (ptime=20,
> > 81.92kbit/s) and Video @ max kbitrate of 384 kbit/s
> > Then:  All packets should be received.
> >
>
> The "Then" part may depend also on what other network characteristics
> may be applied (path loss and/or delay).
>
>
> I concur.  Even putting together these surprisingly simple behavior cases
is difficult.  Every single one of these cases needs to be tangible and
easily implementable.


My take for the cc-eval was to define some of these characteristics as
variables. Then the implementer can pick values for a variable from a set.

This will create a combinatorial explosion of tests for each scenario, but
I hope we can attempt to constrain the number of variables in each
scenario.

Your specific note carries relevance.  How do you define the network
environment?  Linux's Network Emulation (netem,
http://www.linuxfoundation.org/collaborate/workgroups/networking/netem)
provides a very valuable toolset for this sort of thing.


I have used both netem, and dummynet.

However there is netkit, AutoNetkit. Both of these have been recommended
 but I haven't used either for RTP.

Do you think it would make sense to use its capabilities for defining our
'given' case?



Typically the tools allow setting delay, throughput, packet loss rate,
queue length. (The list may be longer).

However, packet loss rate is random, which doesn't work well for some of
our tests.

For dummynet, I have a wrapper written up for changing some of these
params.
See
https://github.com/vr000m/Research-Scripts/tree/master/dummynet


I'm looking forward to reading Lars's paper to see if it may provide some
more insight into this matter.


Thanks,
Dan

--14dae93403918ad7c204d7e64c4b
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<html><head><meta http-equiv=3D"content-type" content=3D"text/html; charset=
=3Dutf-8"></head><body dir=3D"auto"><div><br></div><div>On 14.3.2013, at 13=
.35, Dan Weber &lt;<a href=3D"mailto:dan@marketsoup.com">dan@marketsoup.com=
</a>&gt; wrote:<br>
<br></div><blockquote type=3D"cite"><div>Hi Varun,<br><br><div class=3D"gma=
il_quote">On Thu, Mar 14, 2013 at 7:34 AM, Varun Singh <span dir=3D"ltr">&l=
t;<a href=3D"mailto:vsingh.ietf@gmail.com" target=3D"_blank">vsingh.ietf@gm=
ail.com</a>&gt;</span> wrote:<br>
<blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1p=
x #ccc solid;padding-left:1ex">
Hi Dan,<br>
<div class=3D"im"><br>
On Thu, Mar 14, 2013 at 12:20 AM, Dan Weber &lt;<a href=3D"mailto:dan@marke=
tsoup.com">dan@marketsoup.com</a>&gt; wrote:<br>
&gt;<br>
&gt; Given: A video media stream sends at 512kbit/s on a link with an upstr=
eam<br>
&gt; capacity of 512 kbit/s<br>
&gt; When: an FTP Transfer of 1GB file begins uploading<br>
&gt; Then: Packet loss occurs. =A0Throughput should be diminished. =A0RTPCC=
 should<br>
&gt; notify application of developments.<br>
&gt;<br>
&gt; Given: A link with capacity limited at 512 kbit/s<br>
&gt; When: A video conference call starts sending Mulaw audio (ptime=3D20,<=
br>
&gt; 81.92kbit/s) and Video @ max kbitrate of 384 kbit/s<br>
&gt; Then: =A0All packets should be received.<br>
&gt;<br>
<br>
</div>The &quot;Then&quot; part may depend also on what other network chara=
cteristics<br>
may be applied (path loss and/or delay).<br>
<div class=3D"HOEnZb"><div class=3D"h5"><br>
<br></div></div></blockquote><div>I concur.=A0 Even putting together these =
surprisingly simple behavior cases is difficult.=A0 Every single one of the=
se cases needs to be tangible and easily implementable.=A0 <br><br></div></=
div>
</div></blockquote><div><br></div><div>My take for the cc-eval was to defin=
e some of these characteristics as variables. Then the implementer can pick=
 values for a variable from a set.=A0</div><div><br></div><div>This will cr=
eate a combinatorial explosion of tests for each scenario, but I hope we ca=
n attempt to constrain the number of variables in each scenario.=A0</div>
<br><blockquote type=3D"cite"><div><div class=3D"gmail_quote"><div>Your spe=
cific note carries relevance.=A0 How do you define the network environment?=
=A0 Linux&#39;s Network Emulation (netem, <a href=3D"http://www.linuxfounda=
tion.org/collaborate/workgroups/networking/netem">http://www.linuxfoundatio=
n.org/collaborate/workgroups/networking/netem</a>) provides a very valuable=
 toolset for this sort of thing.=A0 <br>

<br></div></div></div></blockquote><div><br></div><div>I have used both net=
em, and dummynet.=A0</div><div><br></div><div>However there is netkit, Auto=
Netkit. Both of these have been recommended =A0but I haven&#39;t used eithe=
r for RTP.=A0</div>
<div><br></div><blockquote type=3D"cite"><div><div class=3D"gmail_quote"><d=
iv>Do you think it would make sense to use its capabilities for defining ou=
r &#39;given&#39; case? <br><br></div></div></div></blockquote><div><br></d=
iv>
<div><br></div>Typically the tools allow setting delay, throughput, packet =
loss rate, queue length. (The list may be longer).=A0<div><br></div><div>Ho=
wever, packet loss rate is random, which doesn&#39;t work well for some of =
our tests.=A0</div>
<div><br></div><div>For dummynet, I have a wrapper written up for changing =
some of these params.=A0</div><div>See=A0</div><div><a href=3D"https://gith=
ub.com/vr000m/Research-Scripts/tree/master/dummynet">https://github.com/vr0=
00m/Research-Scripts/tree/master/dummynet</a></div>
<div><br><div><div><div><br><blockquote type=3D"cite"><div><div class=3D"gm=
ail_quote"><div>I&#39;m looking forward to reading Lars&#39;s paper to see =
if it may provide some more insight into this matter.</div></div></div></bl=
ockquote>
<br><blockquote type=3D"cite"><div><div class=3D"gmail_quote"><div>Thanks,<=
br>Dan<br><br></div></div>
</div></blockquote></div></div></div></div></body></html>

--14dae93403918ad7c204d7e64c4b--

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From: "Bill Ver Steeg (versteb)" <versteb@cisco.com>
To: Dan Weber <dan@marketsoup.com>
Thread-Topic: [rmcat] Congestion Control Behavior Suggestion Thread
Thread-Index: AQHOIGtXxJthuk91K06P9e0++zMsapilhFsAgABDVID//7Q97w==
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Subject: Re: [rmcat] Congestion Control Behavior Suggestion Thread
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--_000_DE646A448F77411AB4A41740961A590Eciscocom_
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netem is a very good emulation tool, and we often use it to get close to th=
e right answer. However, I will caution that there is a wide variety of pac=
ket schedulers in use in the wild. Not all of them the exhibit flat, determ=
inistic scheduling that netem defaults to.

One (of many) example of such a packet scheduling/limiting algorithm was de=
scribed in yesterday's congestion control research group. HFC CPE (cable mo=
dems) use a leaky bucket algorithm that provides a weighted average to enfo=
rce an SLA, and this can interact with higher level control loops. In other=
 words, if a sender is slightly under the provisioned link rate for a while=
, it can then can send at slightly over the link rate for a while. If you a=
re not careful, a variable bitrate sender can infer an incorrect SLA rate b=
ased on such scheduler behavior. I do not imply that this is the definitive=
 case, just that there can be interactions. Flow control algorithms that wo=
rk very well for a given scheduler can have unexpected entanglements with a=
nother (chain of) schedulers.

Fortunately, there are ways to introduce these behaviors in test rigs.

So, if we define a reference test environment, we either need to have a few=
 scheduler variations or recognize that the test rig is only a first-order =
approximation and live testing on a variety of network topologies is requir=
ed.

Short version-  I agree that netem is a very good tool, but you need to be =
careful how you use it.

Bvs


On Mar 14, 2013, at 1:35 PM, "Dan Weber" <dan@marketsoup.com<mailto:dan@mar=
ketsoup.com>> wrote:

Hi Varun,

On Thu, Mar 14, 2013 at 7:34 AM, Varun Singh <vsingh.ietf@gmail.com<mailto:=
vsingh.ietf@gmail.com>> wrote:
Hi Dan,

On Thu, Mar 14, 2013 at 12:20 AM, Dan Weber <dan@marketsoup.com<mailto:dan@=
marketsoup.com>> wrote:
>
> Given: A video media stream sends at 512kbit/s on a link with an upstream
> capacity of 512 kbit/s
> When: an FTP Transfer of 1GB file begins uploading
> Then: Packet loss occurs.  Throughput should be diminished.  RTPCC should
> notify application of developments.
>
> Given: A link with capacity limited at 512 kbit/s
> When: A video conference call starts sending Mulaw audio (ptime=3D20,
> 81.92kbit/s) and Video @ max kbitrate of 384 kbit/s
> Then:  All packets should be received.
>

The "Then" part may depend also on what other network characteristics
may be applied (path loss and/or delay).


I concur.  Even putting together these surprisingly simple behavior cases i=
s difficult.  Every single one of these cases needs to be tangible and easi=
ly implementable.

Your specific note carries relevance.  How do you define the network enviro=
nment?  Linux's Network Emulation (netem, http://www.linuxfoundation.org/co=
llaborate/workgroups/networking/netem) provides a very valuable toolset for=
 this sort of thing.

Do you think it would make sense to use its capabilities for defining our '=
given' case?

I'm looking forward to reading Lars's paper to see if it may provide some m=
ore insight into this matter.

Thanks,
Dan


--_000_DE646A448F77411AB4A41740961A590Eciscocom_
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Content-Transfer-Encoding: quoted-printable

<html>
<head>
<meta http-equiv=3D"Content-Type" content=3D"text/html; charset=3Dus-ascii"=
>
</head>
<body dir=3D"auto">
<div>netem is a very good emulation tool, and we often use it to get close =
to the right answer. However, I will caution that there is a wide variety o=
f packet schedulers in use in the wild. Not all of them the exhibit flat, d=
eterministic scheduling that netem
 defaults to.&nbsp;</div>
<div><br>
</div>
<div>One (of many) example of such a packet scheduling/limiting algorithm w=
as described in yesterday's congestion control research group. HFC CPE (cab=
le modems) use a leaky bucket algorithm that provides a weighted average to=
 enforce an SLA, and this can interact
 with higher level control loops. In other words, if a sender is slightly u=
nder the provisioned link rate for a while, it can then can send at slightl=
y over the link rate for a while. If you are not careful, a variable bitrat=
e sender can infer an incorrect
 SLA rate based on such scheduler behavior. I do not imply that this is the=
 definitive case, just that there can be interactions. Flow control algorit=
hms that work very well for a given scheduler can have unexpected entanglem=
ents with another (chain of) schedulers.&nbsp;</div>
<div><br>
</div>
<div>Fortunately, there are ways to introduce these behaviors in test rigs.=
&nbsp;</div>
<div><br>
</div>
<div>So, if we define a reference test environment, we either need to have =
a few scheduler variations or recognize that the test rig is only a first-o=
rder approximation and live testing on a variety of network topologies is r=
equired.&nbsp;</div>
<div><br>
</div>
<div>Short version- &nbsp;I agree that netem is a very good tool, but you n=
eed to be careful how you use it.&nbsp;</div>
<div><br>
</div>
<div>Bvs</div>
<div><br>
</div>
<div><br>
On Mar 14, 2013, at 1:35 PM, &quot;Dan Weber&quot; &lt;<a href=3D"mailto:da=
n@marketsoup.com">dan@marketsoup.com</a>&gt; wrote:<br>
<br>
</div>
<div><span></span></div>
<blockquote type=3D"cite">
<div>Hi Varun,<br>
<br>
<div class=3D"gmail_quote">On Thu, Mar 14, 2013 at 7:34 AM, Varun Singh <sp=
an dir=3D"ltr">
&lt;<a href=3D"mailto:vsingh.ietf@gmail.com" target=3D"_blank">vsingh.ietf@=
gmail.com</a>&gt;</span> wrote:<br>
<blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1p=
x #ccc solid;padding-left:1ex">
Hi Dan,<br>
<div class=3D"im"><br>
On Thu, Mar 14, 2013 at 12:20 AM, Dan Weber &lt;<a href=3D"mailto:dan@marke=
tsoup.com">dan@marketsoup.com</a>&gt; wrote:<br>
&gt;<br>
&gt; Given: A video media stream sends at 512kbit/s on a link with an upstr=
eam<br>
&gt; capacity of 512 kbit/s<br>
&gt; When: an FTP Transfer of 1GB file begins uploading<br>
&gt; Then: Packet loss occurs. &nbsp;Throughput should be diminished. &nbsp=
;RTPCC should<br>
&gt; notify application of developments.<br>
&gt;<br>
&gt; Given: A link with capacity limited at 512 kbit/s<br>
&gt; When: A video conference call starts sending Mulaw audio (ptime=3D20,<=
br>
&gt; 81.92kbit/s) and Video @ max kbitrate of 384 kbit/s<br>
&gt; Then: &nbsp;All packets should be received.<br>
&gt;<br>
<br>
</div>
The &quot;Then&quot; part may depend also on what other network characteris=
tics<br>
may be applied (path loss and/or delay).<br>
<div class=3D"HOEnZb">
<div class=3D"h5"><br>
<br>
</div>
</div>
</blockquote>
<div>I concur.&nbsp; Even putting together these surprisingly simple behavi=
or cases is difficult.&nbsp; Every single one of these cases needs to be ta=
ngible and easily implementable.&nbsp;
<br>
<br>
Your specific note carries relevance.&nbsp; How do you define the network e=
nvironment?&nbsp; Linux's Network Emulation (netem,
<a href=3D"http://www.linuxfoundation.org/collaborate/workgroups/networking=
/netem">
http://www.linuxfoundation.org/collaborate/workgroups/networking/netem</a>)=
 provides a very valuable toolset for this sort of thing.&nbsp;
<br>
<br>
Do you think it would make sense to use its capabilities for defining our '=
given' case?
<br>
<br>
I'm looking forward to reading Lars's paper to see if it may provide some m=
ore insight into this matter.<br>
<br>
Thanks,<br>
Dan<br>
<br>
</div>
</div>
</div>
</blockquote>
</body>
</html>

--_000_DE646A448F77411AB4A41740961A590Eciscocom_--

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Subject: Re: [rmcat] Congestion Control Behavior Suggestion Thread
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Hi Bill,

I understand your concerns about netem.  It's no panacea.

I'm shooting to define a table of configurable parameters to go with each
test case.

E.g.
Ploss 1% Correlation (90%)
Delay 80ms Jitter 50ms
Bandwidth 2mbit/s
etc.

It seems to me that this is required regardless of the tools we use.  If we
can standardize on which parameters are adjustable, we can include them in
every test case and use a variety of tools.  As for defining a reference
test environment, does anyone have access to some systems we could open up
for the purpose of this testing?  I may be able to call around and see if
we can get one provided for us if no one else has resources available.

Thanks,
Dan

On Thu, Mar 14, 2013 at 12:04 PM, Bill Ver Steeg (versteb) <
versteb@cisco.com> wrote:

>  netem is a very good emulation tool, and we often use it to get close to
> the right answer. However, I will caution that there is a wide variety of
> packet schedulers in use in the wild. Not all of them the exhibit flat,
> deterministic scheduling that netem defaults to.
>
>  One (of many) example of such a packet scheduling/limiting algorithm was
> described in yesterday's congestion control research group. HFC CPE (cable
> modems) use a leaky bucket algorithm that provides a weighted average to
> enforce an SLA, and this can interact with higher level control loops. In
> other words, if a sender is slightly under the provisioned link rate for a
> while, it can then can send at slightly over the link rate for a while. If
> you are not careful, a variable bitrate sender can infer an incorrect SLA
> rate based on such scheduler behavior. I do not imply that this is the
> definitive case, just that there can be interactions. Flow control
> algorithms that work very well for a given scheduler can have unexpected
> entanglements with another (chain of) schedulers.
>
>  Fortunately, there are ways to introduce these behaviors in test rigs.
>
>  So, if we define a reference test environment, we either need to have a
> few scheduler variations or recognize that the test rig is only a
> first-order approximation and live testing on a variety of network
> topologies is required.
>
>  Short version-  I agree that netem is a very good tool, but you need to
> be careful how you use it.
>
>  Bvs
>
>

--14dae9340eeb072df904d7e68267
Content-Type: text/html; charset=UTF-8
Content-Transfer-Encoding: quoted-printable

Hi Bill,<br><br>I understand your concerns about netem.=C2=A0 It&#39;s no p=
anacea.<br><br>I&#39;m shooting to define a table of configurable parameter=
s to go with each test case.<br><br>E.g.<br>Ploss 1% Correlation (90%)<br>D=
elay 80ms Jitter 50ms<br>
Bandwidth 2mbit/s<br>etc.<br><br>It seems to me that this is required regar=
dless of the tools we use.=C2=A0 If we can standardize on which parameters =
are adjustable, we can include them in every test case and use a variety of=
 tools.=C2=A0 As for defining a reference test environment, does anyone hav=
e access to some systems we could open up for the purpose of this testing?=
=C2=A0 I may be able to call around and see if we can get one provided for =
us if no one else has resources available.<br>
<br>Thanks,<br>Dan<br><br><div class=3D"gmail_quote">On Thu, Mar 14, 2013 a=
t 12:04 PM, Bill Ver Steeg (versteb) <span dir=3D"ltr">&lt;<a href=3D"mailt=
o:versteb@cisco.com" target=3D"_blank">versteb@cisco.com</a>&gt;</span> wro=
te:<br>
<blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1p=
x #ccc solid;padding-left:1ex">



<div dir=3D"auto">
<div>netem is a very good emulation tool, and we often use it to get close =
to the right answer. However, I will caution that there is a wide variety o=
f packet schedulers in use in the wild. Not all of them the exhibit flat, d=
eterministic scheduling that netem
 defaults to.=C2=A0</div>
<div><br>
</div>
<div>One (of many) example of such a packet scheduling/limiting algorithm w=
as described in yesterday&#39;s congestion control research group. HFC CPE =
(cable modems) use a leaky bucket algorithm that provides a weighted averag=
e to enforce an SLA, and this can interact
 with higher level control loops. In other words, if a sender is slightly u=
nder the provisioned link rate for a while, it can then can send at slightl=
y over the link rate for a while. If you are not careful, a variable bitrat=
e sender can infer an incorrect
 SLA rate based on such scheduler behavior. I do not imply that this is the=
 definitive case, just that there can be interactions. Flow control algorit=
hms that work very well for a given scheduler can have unexpected entanglem=
ents with another (chain of) schedulers.=C2=A0</div>

<div><br>
</div>
<div>Fortunately, there are ways to introduce these behaviors in test rigs.=
=C2=A0</div>
<div><br>
</div>
<div>So, if we define a reference test environment, we either need to have =
a few scheduler variations or recognize that the test rig is only a first-o=
rder approximation and live testing on a variety of network topologies is r=
equired.=C2=A0</div>

<div><br>
</div>
<div>Short version- =C2=A0I agree that netem is a very good tool, but you n=
eed to be careful how you use it.=C2=A0</div><span class=3D"HOEnZb"><font c=
olor=3D"#888888">
<div><br>
</div>
<div>Bvs</div></font></span><br></div></blockquote></div>

--14dae9340eeb072df904d7e68267--

From lars@netapp.com  Thu Mar 14 11:21:38 2013
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From: "Eggert, Lars" <lars@netapp.com>
To: "Bill Ver Steeg (versteb)" <versteb@cisco.com>
Thread-Topic: [rmcat] Congestion Control Behavior Suggestion Thread
Thread-Index: AQHOIGtWgEUKQwH3cECqMQTKMwzdzJilpeIAgABDVICAAAgOgIAABMYA
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Cc: Dan Weber <dan@marketsoup.com>, Varun Singh <vsingh.ietf@gmail.com>, "rmcat@ietf.org" <rmcat@ietf.org>
Subject: Re: [rmcat] Congestion Control Behavior Suggestion Thread
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Hi,

one thing to keep in mind is that we don't have to describe a fully exhaust=
ive list of tests in eval-criteria, esp. initially.

If we describe a few and the folks pushing solution candidates ran these, t=
hat'd be great. They should also run other tests, and some of them in turn =
should make it into a future revision of eval-criteria.

I do (again) want to encourage the folks that are proposing one solution ca=
ndidate to also try to compare theirs against the others directly in one me=
asurement setup, instead of everyone only bringing their set of measurement=
s and then we all trust that the setups were really fully comparable. Esp. =
great would be if folks actually implemented each other's candidates instea=
d of just running the other's code. This way, we'd have way ore confidence =
that the respective specs were complete and correct.

Lars=

From dan@marketsoup.com  Thu Mar 14 11:24:28 2013
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--14dae934037b0310b404d7e6a25f
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Hi Varun,

On Thu, Mar 14, 2013 at 12:00 PM, Varun Singh <vsingh.ietf@gmail.com> wrote:

>
> On 14.3.2013, at 13.35, Dan Weber <dan@marketsoup.com> wrote:
>
> Hi Varun,
>
> On Thu, Mar 14, 2013 at 7:34 AM, Varun Singh <vsingh.ietf@gmail.com>wrote:
>
>> Hi Dan,
>>
>> On Thu, Mar 14, 2013 at 12:20 AM, Dan Weber <dan@marketsoup.com> wrote:
>> >
>> > Given: A video media stream sends at 512kbit/s on a link with an
>> upstream
>> > capacity of 512 kbit/s
>> > When: an FTP Transfer of 1GB file begins uploading
>> > Then: Packet loss occurs.  Throughput should be diminished.  RTPCC
>> should
>> > notify application of developments.
>> >
>> > Given: A link with capacity limited at 512 kbit/s
>> > When: A video conference call starts sending Mulaw audio (ptime=20,
>> > 81.92kbit/s) and Video @ max kbitrate of 384 kbit/s
>> > Then:  All packets should be received.
>> >
>>
>> The "Then" part may depend also on what other network characteristics
>> may be applied (path loss and/or delay).
>>
>>
>> I concur.  Even putting together these surprisingly simple behavior cases
> is difficult.  Every single one of these cases needs to be tangible and
> easily implementable.
>
>
> My take for the cc-eval was to define some of these characteristics as
> variables. Then the implementer can pick values for a variable from a set.
>
> This will create a combinatorial explosion of tests for each scenario, but
> I hope we can attempt to constrain the number of variables in each
> scenario.
>

I'm not sure I understand exactly what you mean by this.  I do understand
how a combinatorial explosion can occur with countless parameters.
However, this problem mirrors ones used in other fields.  I'm not sure if
you're familiar with Genichi Taguchi but he defined the methods that are
the backbone of most popular total quality management systems.  His
solution to this particular problem is the use of an orthogonal array.  You
might enjoy reading his article on their use: "Robust Quality"
http://offermaxima.com/pdfs/RobustQuality.pdf




>
> Your specific note carries relevance.  How do you define the network
> environment?  Linux's Network Emulation (netem,
> http://www.linuxfoundation.org/collaborate/workgroups/networking/netem)
> provides a very valuable toolset for this sort of thing.
>
>
> I have used both netem, and dummynet.
>
> However there is netkit, AutoNetkit. Both of these have been recommended
>  but I haven't used either for RTP.
>
> I have not used dummynet, netkit, or autonetkit, but I'm always interested
in trying new things.  Is there a basic set of parameters that apply to all
of them?

> Do you think it would make sense to use its capabilities for defining our
> 'given' case?
>
>
>
> Typically the tools allow setting delay, throughput, packet loss rate,
> queue length. (The list may be longer).
>
> However, packet loss rate is random, which doesn't work well for some of
> our tests.
>
Netem does allow for packet loss with correlation so that you can generate
burst packet loss, although the distribution is still primarily random.
However, that may be reasonably good enough if we get the rest of the
packet loss from actually limiting the link capacity.

>
> For dummynet, I have a wrapper written up for changing some of these
> params.
> See
> https://github.com/vr000m/Research-Scripts/tree/master/dummynet
>
>
> I'll check this out when I get a chance.

Thanks,
Dan

--14dae934037b0310b404d7e6a25f
Content-Type: text/html; charset=UTF-8
Content-Transfer-Encoding: quoted-printable

Hi Varun,<br><br><div class=3D"gmail_quote">On Thu, Mar 14, 2013 at 12:00 P=
M, Varun Singh <span dir=3D"ltr">&lt;<a href=3D"mailto:vsingh.ietf@gmail.co=
m" target=3D"_blank">vsingh.ietf@gmail.com</a>&gt;</span> wrote:<br><blockq=
uote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1px #ccc =
solid;padding-left:1ex">
<div dir=3D"auto"><div class=3D"im"><div><br></div><div>On 14.3.2013, at 13=
.35, Dan Weber &lt;<a href=3D"mailto:dan@marketsoup.com" target=3D"_blank">=
dan@marketsoup.com</a>&gt; wrote:<br>
<br></div><blockquote type=3D"cite"><div>Hi Varun,<br><br><div class=3D"gma=
il_quote">On Thu, Mar 14, 2013 at 7:34 AM, Varun Singh <span dir=3D"ltr">&l=
t;<a href=3D"mailto:vsingh.ietf@gmail.com" target=3D"_blank">vsingh.ietf@gm=
ail.com</a>&gt;</span> wrote:<br>

<blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1p=
x #ccc solid;padding-left:1ex">
Hi Dan,<br>
<div><br>
On Thu, Mar 14, 2013 at 12:20 AM, Dan Weber &lt;<a href=3D"mailto:dan@marke=
tsoup.com" target=3D"_blank">dan@marketsoup.com</a>&gt; wrote:<br>
&gt;<br>
&gt; Given: A video media stream sends at 512kbit/s on a link with an upstr=
eam<br>
&gt; capacity of 512 kbit/s<br>
&gt; When: an FTP Transfer of 1GB file begins uploading<br>
&gt; Then: Packet loss occurs. =C2=A0Throughput should be diminished. =C2=
=A0RTPCC should<br>
&gt; notify application of developments.<br>
&gt;<br>
&gt; Given: A link with capacity limited at 512 kbit/s<br>
&gt; When: A video conference call starts sending Mulaw audio (ptime=3D20,<=
br>
&gt; 81.92kbit/s) and Video @ max kbitrate of 384 kbit/s<br>
&gt; Then: =C2=A0All packets should be received.<br>
&gt;<br>
<br>
</div>The &quot;Then&quot; part may depend also on what other network chara=
cteristics<br>
may be applied (path loss and/or delay).<br>
<div><div><br>
<br></div></div></blockquote><div>I concur.=C2=A0 Even putting together the=
se surprisingly simple behavior cases is difficult.=C2=A0 Every single one =
of these cases needs to be tangible and easily implementable.=C2=A0 <br><br=
></div></div>

</div></blockquote><div><br></div></div><div>My take for the cc-eval was to=
 define some of these characteristics as variables. Then the implementer ca=
n pick values for a variable from a set.=C2=A0</div><div><br></div><div>Thi=
s will create a combinatorial explosion of tests for each scenario, but I h=
ope we can attempt to constrain the number of variables in each scenario.=
=C2=A0</div>
</div></blockquote><div><br>I&#39;m not sure I understand exactly what you =
mean by this.=C2=A0 I do understand how a combinatorial explosion can occur=
 with countless parameters.=C2=A0 However, this problem mirrors ones used i=
n other fields.=C2=A0 I&#39;m not sure if you&#39;re familiar with Genichi =
Taguchi but he defined the methods that are the backbone of most popular to=
tal quality management systems.=C2=A0 His solution to this particular probl=
em is the use of an orthogonal array.=C2=A0 You might enjoy reading his art=
icle on their use: &quot;Robust Quality&quot; <a href=3D"http://offermaxima=
.com/pdfs/RobustQuality.pdf">http://offermaxima.com/pdfs/RobustQuality.pdf<=
/a><br>
<br><br></div><div>=C2=A0<br></div><blockquote class=3D"gmail_quote" style=
=3D"margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex"><div dir=
=3D"auto"><div class=3D"im">
<br><blockquote type=3D"cite"><div><div class=3D"gmail_quote"><div>Your spe=
cific note carries relevance.=C2=A0 How do you define the network environme=
nt?=C2=A0 Linux&#39;s Network Emulation (netem, <a href=3D"http://www.linux=
foundation.org/collaborate/workgroups/networking/netem" target=3D"_blank">h=
ttp://www.linuxfoundation.org/collaborate/workgroups/networking/netem</a>) =
provides a very valuable toolset for this sort of thing.=C2=A0 <br>


<br></div></div></div></blockquote><div><br></div></div><div>I have used bo=
th netem, and dummynet.=C2=A0</div><div><br></div><div>However there is net=
kit, AutoNetkit. Both of these have been recommended =C2=A0but I haven&#39;=
t used either for RTP.=C2=A0</div>
<div class=3D"im">
<div><br></div></div></div></blockquote><div>I have not used dummynet, netk=
it, or autonetkit, but I&#39;m always interested in trying new things.=C2=
=A0 Is there a basic set of parameters that apply to all of them? <br></div=
><blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1=
px #ccc solid;padding-left:1ex">
<div dir=3D"auto"><div class=3D"im"><div></div><blockquote type=3D"cite"><d=
iv><div class=3D"gmail_quote"><div>Do you think it would make sense to use =
its capabilities for defining our &#39;given&#39; case? <br><br></div></div=
></div>
</blockquote><div><br></div>
<div><br></div></div>Typically the tools allow setting delay, throughput, p=
acket loss rate, queue length. (The list may be longer).=C2=A0<div><br></di=
v><div>However, packet loss rate is random, which doesn&#39;t work well for=
 some of our tests.=C2=A0</div>
</div></blockquote><div>Netem does allow for packet loss with correlation s=
o that you can generate burst packet loss, although the distribution is sti=
ll primarily random.=C2=A0 However, that may be reasonably good enough if w=
e get the rest of the packet loss from actually limiting the link capacity.=
 <br>
</div><blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-l=
eft:1px #ccc solid;padding-left:1ex"><div dir=3D"auto">
<div><br></div><div>For dummynet, I have a wrapper written up for changing =
some of these params.=C2=A0</div><div>See=C2=A0</div><div><a href=3D"https:=
//github.com/vr000m/Research-Scripts/tree/master/dummynet" target=3D"_blank=
">https://github.com/vr000m/Research-Scripts/tree/master/dummynet</a></div>
<div class=3D"im">
<div><br><div><div><div><br></div></div></div></div></div></div></blockquot=
e><div>I&#39;ll check this out when I get a chance. <br></div><br>Thanks,<b=
r>Dan<br></div>

--14dae934037b0310b404d7e6a25f--

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From: "Bill Ver Steeg (versteb)" <versteb@cisco.com>
To: Dan Weber <dan@marketsoup.com>
Thread-Topic: [rmcat] Congestion Control Behavior Suggestion Thread
Thread-Index: AQHOIGtXxJthuk91K06P9e0++zMsapilhFsAgABDVID//7Q974AAVuEA//+wZzc=
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--_000_9257B4B462844E73AA1CEED18901424Eciscocom_
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In an ns2 environment, having a physical test rig is mute. Quite a bit of p=
rogress can be made in this domain. A simple tar file of the complete ns2 t=
est rig would make for results that can be verified, tweaked and retested.

Once we move to a real device, it seems that Linux sources and sinks with n=
etem-like impairments would work. I used to have a pretty good test rig, bu=
t I loaned it to GaTech last year.......

I do think that a well defined test rig would help in this effort, and also=
 in the broader congestion work underway at IETF. Getting gear is easier th=
an staff to run it. If I can get a coop student this summer, maybe I can co=
bble some gear together and stand up a test rig somewhere that is publicly =
accessible.


Bvs

On Mar 14, 2013, at 2:15 PM, "Dan Weber" <dan@marketsoup.com<mailto:dan@mar=
ketsoup.com>> wrote:

Hi Bill,

I understand your concerns about netem.  It's no panacea.

I'm shooting to define a table of configurable parameters to go with each t=
est case.

E.g.
Ploss 1% Correlation (90%)
Delay 80ms Jitter 50ms
Bandwidth 2mbit/s
etc.

It seems to me that this is required regardless of the tools we use.  If we=
 can standardize on which parameters are adjustable, we can include them in=
 every test case and use a variety of tools.  As for defining a reference t=
est environment, does anyone have access to some systems we could open up f=
or the purpose of this testing?  I may be able to call around and see if we=
 can get one provided for us if no one else has resources available.

Thanks,
Dan

On Thu, Mar 14, 2013 at 12:04 PM, Bill Ver Steeg (versteb) <versteb@cisco.c=
om<mailto:versteb@cisco.com>> wrote:
netem is a very good emulation tool, and we often use it to get close to th=
e right answer. However, I will caution that there is a wide variety of pac=
ket schedulers in use in the wild. Not all of them the exhibit flat, determ=
inistic scheduling that netem defaults to.

One (of many) example of such a packet scheduling/limiting algorithm was de=
scribed in yesterday's congestion control research group. HFC CPE (cable mo=
dems) use a leaky bucket algorithm that provides a weighted average to enfo=
rce an SLA, and this can interact with higher level control loops. In other=
 words, if a sender is slightly under the provisioned link rate for a while=
, it can then can send at slightly over the link rate for a while. If you a=
re not careful, a variable bitrate sender can infer an incorrect SLA rate b=
ased on such scheduler behavior. I do not imply that this is the definitive=
 case, just that there can be interactions. Flow control algorithms that wo=
rk very well for a given scheduler can have unexpected entanglements with a=
nother (chain of) schedulers.

Fortunately, there are ways to introduce these behaviors in test rigs.

So, if we define a reference test environment, we either need to have a few=
 scheduler variations or recognize that the test rig is only a first-order =
approximation and live testing on a variety of network topologies is requir=
ed.

Short version-  I agree that netem is a very good tool, but you need to be =
careful how you use it.

Bvs


--_000_9257B4B462844E73AA1CEED18901424Eciscocom_
Content-Type: text/html; charset="us-ascii"
Content-Transfer-Encoding: quoted-printable

<html>
<head>
<meta http-equiv=3D"Content-Type" content=3D"text/html; charset=3Dus-ascii"=
>
</head>
<body dir=3D"auto">
<div>In an ns2 environment, having a physical test rig is mute. Quite a bit=
 of progress can be made in this domain. A simple tar file of the complete =
ns2 test rig would make for results that can be verified, tweaked and retes=
ted.&nbsp;</div>
<div><br>
</div>
<div>Once we move to a real device, it seems that Linux sources and sinks w=
ith netem-like impairments would work. I used to have a pretty good test ri=
g, but I loaned it to GaTech last year.......&nbsp;</div>
<div><br>
</div>
<div>I do think that a well defined test rig would help in this effort, and=
 also in the broader congestion work underway at IETF. Getting gear is easi=
er than staff to run it. If I can get a coop student this summer, maybe I c=
an cobble some gear together and
 stand up a test rig somewhere that is publicly accessible.&nbsp;</div>
<div><br>
</div>
<div><br>
</div>
<div>Bvs&nbsp;</div>
<div><br>
On Mar 14, 2013, at 2:15 PM, &quot;Dan Weber&quot; &lt;<a href=3D"mailto:da=
n@marketsoup.com">dan@marketsoup.com</a>&gt; wrote:<br>
<br>
</div>
<blockquote type=3D"cite">
<div>Hi Bill,<br>
<br>
I understand your concerns about netem.&nbsp; It's no panacea.<br>
<br>
I'm shooting to define a table of configurable parameters to go with each t=
est case.<br>
<br>
E.g.<br>
Ploss 1% Correlation (90%)<br>
Delay 80ms Jitter 50ms<br>
Bandwidth 2mbit/s<br>
etc.<br>
<br>
It seems to me that this is required regardless of the tools we use.&nbsp; =
If we can standardize on which parameters are adjustable, we can include th=
em in every test case and use a variety of tools.&nbsp; As for defining a r=
eference test environment, does anyone have
 access to some systems we could open up for the purpose of this testing?&n=
bsp; I may be able to call around and see if we can get one provided for us=
 if no one else has resources available.<br>
<br>
Thanks,<br>
Dan<br>
<br>
<div class=3D"gmail_quote">On Thu, Mar 14, 2013 at 12:04 PM, Bill Ver Steeg=
 (versteb)
<span dir=3D"ltr">&lt;<a href=3D"mailto:versteb@cisco.com" target=3D"_blank=
">versteb@cisco.com</a>&gt;</span> wrote:<br>
<blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1p=
x #ccc solid;padding-left:1ex">
<div dir=3D"auto">
<div>netem is a very good emulation tool, and we often use it to get close =
to the right answer. However, I will caution that there is a wide variety o=
f packet schedulers in use in the wild. Not all of them the exhibit flat, d=
eterministic scheduling that netem
 defaults to.&nbsp;</div>
<div><br>
</div>
<div>One (of many) example of such a packet scheduling/limiting algorithm w=
as described in yesterday's congestion control research group. HFC CPE (cab=
le modems) use a leaky bucket algorithm that provides a weighted average to=
 enforce an SLA, and this can interact
 with higher level control loops. In other words, if a sender is slightly u=
nder the provisioned link rate for a while, it can then can send at slightl=
y over the link rate for a while. If you are not careful, a variable bitrat=
e sender can infer an incorrect
 SLA rate based on such scheduler behavior. I do not imply that this is the=
 definitive case, just that there can be interactions. Flow control algorit=
hms that work very well for a given scheduler can have unexpected entanglem=
ents with another (chain of) schedulers.&nbsp;</div>
<div><br>
</div>
<div>Fortunately, there are ways to introduce these behaviors in test rigs.=
&nbsp;</div>
<div><br>
</div>
<div>So, if we define a reference test environment, we either need to have =
a few scheduler variations or recognize that the test rig is only a first-o=
rder approximation and live testing on a variety of network topologies is r=
equired.&nbsp;</div>
<div><br>
</div>
<div>Short version- &nbsp;I agree that netem is a very good tool, but you n=
eed to be careful how you use it.&nbsp;</div>
<span class=3D"HOEnZb"><font color=3D"#888888">
<div><br>
</div>
<div>Bvs</div>
</font></span><br>
</div>
</blockquote>
</div>
</div>
</blockquote>
</body>
</html>

--_000_9257B4B462844E73AA1CEED18901424Eciscocom_--

From dan@marketsoup.com  Thu Mar 14 11:38:55 2013
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Subject: Re: [rmcat] Congestion Control Behavior Suggestion Thread
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--14dae93406cdac446c04d7e6d5ec
Content-Type: text/plain; charset=UTF-8

Hey Bill,

I'm not familiar with ns2, but I am in favor of having a complete test rig
that everyone can download.

Does it work with existing applications?

Thanks,
Dan



On Thu, Mar 14, 2013 at 12:30 PM, Bill Ver Steeg (versteb) <
versteb@cisco.com> wrote:

>  In an ns2 environment, having a physical test rig is mute. Quite a bit
> of progress can be made in this domain. A simple tar file of the complete
> ns2 test rig would make for results that can be verified, tweaked and
> retested.
>
>  Once we move to a real device, it seems that Linux sources and sinks
> with netem-like impairments would work. I used to have a pretty good test
> rig, but I loaned it to GaTech last year.......
>
>  I do think that a well defined test rig would help in this effort, and
> also in the broader congestion work underway at IETF. Getting gear is
> easier than staff to run it. If I can get a coop student this summer, maybe
> I can cobble some gear together and stand up a test rig somewhere that is
> publicly accessible.
>
>
>  Bvs
>
> On Mar 14, 2013, at 2:15 PM, "Dan Weber" <dan@marketsoup.com> wrote:
>
>  Hi Bill,
>
> I understand your concerns about netem.  It's no panacea.
>
> I'm shooting to define a table of configurable parameters to go with each
> test case.
>
> E.g.
> Ploss 1% Correlation (90%)
> Delay 80ms Jitter 50ms
> Bandwidth 2mbit/s
> etc.
>
> It seems to me that this is required regardless of the tools we use.  If
> we can standardize on which parameters are adjustable, we can include them
> in every test case and use a variety of tools.  As for defining a reference
> test environment, does anyone have access to some systems we could open up
> for the purpose of this testing?  I may be able to call around and see if
> we can get one provided for us if no one else has resources available.
>
> Thanks,
> Dan
>
> On Thu, Mar 14, 2013 at 12:04 PM, Bill Ver Steeg (versteb) <
> versteb@cisco.com> wrote:
>
>>  netem is a very good emulation tool, and we often use it to get close
>> to the right answer. However, I will caution that there is a wide variety
>> of packet schedulers in use in the wild. Not all of them the exhibit flat,
>> deterministic scheduling that netem defaults to.
>>
>>  One (of many) example of such a packet scheduling/limiting algorithm
>> was described in yesterday's congestion control research group. HFC CPE
>> (cable modems) use a leaky bucket algorithm that provides a weighted
>> average to enforce an SLA, and this can interact with higher level control
>> loops. In other words, if a sender is slightly under the provisioned link
>> rate for a while, it can then can send at slightly over the link rate for a
>> while. If you are not careful, a variable bitrate sender can infer an
>> incorrect SLA rate based on such scheduler behavior. I do not imply that
>> this is the definitive case, just that there can be interactions. Flow
>> control algorithms that work very well for a given scheduler can have
>> unexpected entanglements with another (chain of) schedulers.
>>
>>  Fortunately, there are ways to introduce these behaviors in test rigs.
>>
>>  So, if we define a reference test environment, we either need to have a
>> few scheduler variations or recognize that the test rig is only a
>> first-order approximation and live testing on a variety of network
>> topologies is required.
>>
>>  Short version-  I agree that netem is a very good tool, but you need to
>> be careful how you use it.
>>
>>  Bvs
>>
>>

--14dae93406cdac446c04d7e6d5ec
Content-Type: text/html; charset=UTF-8
Content-Transfer-Encoding: quoted-printable

Hey Bill,<br><br>I&#39;m not familiar with ns2, but I am in favor of having=
 a complete test rig that everyone can download.<br><br>Does it work with e=
xisting applications?<br><br>Thanks,<br>Dan<br><br><br><br><div class=3D"gm=
ail_quote">
On Thu, Mar 14, 2013 at 12:30 PM, Bill Ver Steeg (versteb) <span dir=3D"ltr=
">&lt;<a href=3D"mailto:versteb@cisco.com" target=3D"_blank">versteb@cisco.=
com</a>&gt;</span> wrote:<br><blockquote class=3D"gmail_quote" style=3D"mar=
gin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">




<div dir=3D"auto">
<div>In an ns2 environment, having a physical test rig is mute. Quite a bit=
 of progress can be made in this domain. A simple tar file of the complete =
ns2 test rig would make for results that can be verified, tweaked and retes=
ted.=C2=A0</div>

<div><br>
</div>
<div>Once we move to a real device, it seems that Linux sources and sinks w=
ith netem-like impairments would work. I used to have a pretty good test ri=
g, but I loaned it to GaTech last year.......=C2=A0</div>
<div><br>
</div>
<div>I do think that a well defined test rig would help in this effort, and=
 also in the broader congestion work underway at IETF. Getting gear is easi=
er than staff to run it. If I can get a coop student this summer, maybe I c=
an cobble some gear together and
 stand up a test rig somewhere that is publicly accessible.=C2=A0</div><spa=
n class=3D"HOEnZb"><font color=3D"#888888">
<div><br>
</div>
<div><br>
</div>
<div>Bvs=C2=A0</div></font></span><div><div class=3D"h5">
<div><br>
On Mar 14, 2013, at 2:15 PM, &quot;Dan Weber&quot; &lt;<a href=3D"mailto:da=
n@marketsoup.com" target=3D"_blank">dan@marketsoup.com</a>&gt; wrote:<br>
<br>
</div>
<blockquote type=3D"cite">
<div>Hi Bill,<br>
<br>
I understand your concerns about netem.=C2=A0 It&#39;s no panacea.<br>
<br>
I&#39;m shooting to define a table of configurable parameters to go with ea=
ch test case.<br>
<br>
E.g.<br>
Ploss 1% Correlation (90%)<br>
Delay 80ms Jitter 50ms<br>
Bandwidth 2mbit/s<br>
etc.<br>
<br>
It seems to me that this is required regardless of the tools we use.=C2=A0 =
If we can standardize on which parameters are adjustable, we can include th=
em in every test case and use a variety of tools.=C2=A0 As for defining a r=
eference test environment, does anyone have
 access to some systems we could open up for the purpose of this testing?=
=C2=A0 I may be able to call around and see if we can get one provided for =
us if no one else has resources available.<br>
<br>
Thanks,<br>
Dan<br>
<br>
<div class=3D"gmail_quote">On Thu, Mar 14, 2013 at 12:04 PM, Bill Ver Steeg=
 (versteb)
<span dir=3D"ltr">&lt;<a href=3D"mailto:versteb@cisco.com" target=3D"_blank=
">versteb@cisco.com</a>&gt;</span> wrote:<br>
<blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1p=
x #ccc solid;padding-left:1ex">
<div dir=3D"auto">
<div>netem is a very good emulation tool, and we often use it to get close =
to the right answer. However, I will caution that there is a wide variety o=
f packet schedulers in use in the wild. Not all of them the exhibit flat, d=
eterministic scheduling that netem
 defaults to.=C2=A0</div>
<div><br>
</div>
<div>One (of many) example of such a packet scheduling/limiting algorithm w=
as described in yesterday&#39;s congestion control research group. HFC CPE =
(cable modems) use a leaky bucket algorithm that provides a weighted averag=
e to enforce an SLA, and this can interact
 with higher level control loops. In other words, if a sender is slightly u=
nder the provisioned link rate for a while, it can then can send at slightl=
y over the link rate for a while. If you are not careful, a variable bitrat=
e sender can infer an incorrect
 SLA rate based on such scheduler behavior. I do not imply that this is the=
 definitive case, just that there can be interactions. Flow control algorit=
hms that work very well for a given scheduler can have unexpected entanglem=
ents with another (chain of) schedulers.=C2=A0</div>

<div><br>
</div>
<div>Fortunately, there are ways to introduce these behaviors in test rigs.=
=C2=A0</div>
<div><br>
</div>
<div>So, if we define a reference test environment, we either need to have =
a few scheduler variations or recognize that the test rig is only a first-o=
rder approximation and live testing on a variety of network topologies is r=
equired.=C2=A0</div>

<div><br>
</div>
<div>Short version- =C2=A0I agree that netem is a very good tool, but you n=
eed to be careful how you use it.=C2=A0</div>
<span><font color=3D"#888888">
<div><br>
</div>
<div>Bvs</div>
</font></span><br>
</div>
</blockquote>
</div>
</div>
</blockquote>
</div></div></div>

</blockquote></div><br>

--14dae93406cdac446c04d7e6d5ec--

From dan@marketsoup.com  Thu Mar 14 11:43:15 2013
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Subject: Re: [rmcat] Congestion Control Behavior Suggestion Thread
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This paper is excellent.  Thanks for sharing.

Do you have a copy of the NS2 script we could use for inspiration?

Thanks,
Dan

On Thu, Mar 14, 2013 at 5:34 AM, Eggert, Lars <lars@netapp.com> wrote:

> Hi,
>
> On Mar 14, 2013, at 0:20, Dan Weber <dan@marketsoup.com> wrote:
> > This may need more work, but it can get us going in the right direction.
>
> thanks for starting this! Yes, there needs to be more detail. In case
> folks haven't see it yet, we tried to specify such scenarios for doing
> performance measurements for TCP extensions a few years ago:
> http://eggert.org/papers/2008-pfldnet.pdf
>
> The intent with that paper, and I guess the intent now, would be to nail
> down all constants sufficiently and describe the parameterization of each
> test sufficiently so that someone could set up a simulation or experiment
> based only on the text. (For the TCP scenarios described in the paper,
> someone - from NEC? - actually did provide an ns2 script.)
>
> Lars

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This paper is excellent.=C2=A0 Thanks for sharing.<br><br>Do you have a cop=
y of the NS2 script we could use for inspiration?<br><br>Thanks,<br>Dan<br>=
<br><div class=3D"gmail_quote">On Thu, Mar 14, 2013 at 5:34 AM, Eggert, Lar=
s <span dir=3D"ltr">&lt;<a href=3D"mailto:lars@netapp.com" target=3D"_blank=
">lars@netapp.com</a>&gt;</span> wrote:<br>
<blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1p=
x #ccc solid;padding-left:1ex">Hi,<br>
<div class=3D"im"><br>
On Mar 14, 2013, at 0:20, Dan Weber &lt;<a href=3D"mailto:dan@marketsoup.co=
m">dan@marketsoup.com</a>&gt; wrote:<br>
&gt; This may need more work, but it can get us going in the right directio=
n.<br>
<br>
</div>thanks for starting this! Yes, there needs to be more detail. In case=
 folks haven&#39;t see it yet, we tried to specify such scenarios for doing=
 performance measurements for TCP extensions a few years ago: <a href=3D"ht=
tp://eggert.org/papers/2008-pfldnet.pdf" target=3D"_blank">http://eggert.or=
g/papers/2008-pfldnet.pdf</a><br>

<br>
The intent with that paper, and I guess the intent now, would be to nail do=
wn all constants sufficiently and describe the parameterization of each tes=
t sufficiently so that someone could set up a simulation or experiment base=
d only on the text. (For the TCP scenarios described in the paper, someone =
- from NEC? - actually did provide an ns2 script.)<br>

<span class=3D"HOEnZb"><font color=3D"#888888"><br>
Lars</font></span></blockquote></div><br>

--f46d04339d402d1a5504d7e6e584--

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From: "Eggert, Lars" <lars@netapp.com>
To: Dan Weber <dan@marketsoup.com>
Thread-Topic: [rmcat] Congestion Control Behavior Suggestion Thread
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On Mar 14, 2013, at 14:43, Dan Weber <dan@marketsoup.com>
 wrote:
> This paper is excellent.  Thanks for sharing.
>=20
> Do you have a copy of the NS2 script we could use for inspiration?

http://sourceforge.net/projects/tcpeval/

Lars

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And note that netem needs to be run on a separate system.

Jim
On Mar 14, 2013 2:30 PM, "Bill Ver Steeg (versteb)" <versteb@cisco.com>
wrote:

>  In an ns2 environment, having a physical test rig is mute. Quite a bit
> of progress can be made in this domain. A simple tar file of the complete
> ns2 test rig would make for results that can be verified, tweaked and
> retested.
>
>  Once we move to a real device, it seems that Linux sources and sinks
> with netem-like impairments would work. I used to have a pretty good test
> rig, but I loaned it to GaTech last year.......
>
>  I do think that a well defined test rig would help in this effort, and
> also in the broader congestion work underway at IETF. Getting gear is
> easier than staff to run it. If I can get a coop student this summer, maybe
> I can cobble some gear together and stand up a test rig somewhere that is
> publicly accessible.
>
>
>  Bvs
>
> On Mar 14, 2013, at 2:15 PM, "Dan Weber" <dan@marketsoup.com> wrote:
>
>  Hi Bill,
>
> I understand your concerns about netem.  It's no panacea.
>
> I'm shooting to define a table of configurable parameters to go with each
> test case.
>
> E.g.
> Ploss 1% Correlation (90%)
> Delay 80ms Jitter 50ms
> Bandwidth 2mbit/s
> etc.
>
> It seems to me that this is required regardless of the tools we use.  If
> we can standardize on which parameters are adjustable, we can include them
> in every test case and use a variety of tools.  As for defining a reference
> test environment, does anyone have access to some systems we could open up
> for the purpose of this testing?  I may be able to call around and see if
> we can get one provided for us if no one else has resources available.
>
> Thanks,
> Dan
>
> On Thu, Mar 14, 2013 at 12:04 PM, Bill Ver Steeg (versteb) <
> versteb@cisco.com> wrote:
>
>>  netem is a very good emulation tool, and we often use it to get close
>> to the right answer. However, I will caution that there is a wide variety
>> of packet schedulers in use in the wild. Not all of them the exhibit flat,
>> deterministic scheduling that netem defaults to.
>>
>>  One (of many) example of such a packet scheduling/limiting algorithm
>> was described in yesterday's congestion control research group. HFC CPE
>> (cable modems) use a leaky bucket algorithm that provides a weighted
>> average to enforce an SLA, and this can interact with higher level control
>> loops. In other words, if a sender is slightly under the provisioned link
>> rate for a while, it can then can send at slightly over the link rate for a
>> while. If you are not careful, a variable bitrate sender can infer an
>> incorrect SLA rate based on such scheduler behavior. I do not imply that
>> this is the definitive case, just that there can be interactions. Flow
>> control algorithms that work very well for a given scheduler can have
>> unexpected entanglements with another (chain of) schedulers.
>>
>>  Fortunately, there are ways to introduce these behaviors in test rigs.
>>
>>  So, if we define a reference test environment, we either need to have a
>> few scheduler variations or recognize that the test rig is only a
>> first-order approximation and live testing on a variety of network
>> topologies is required.
>>
>>  Short version-  I agree that netem is a very good tool, but you need to
>> be careful how you use it.
>>
>>  Bvs
>>
>>

--e89a8f839d51c82edc04d7eb7046
Content-Type: text/html; charset=ISO-8859-1
Content-Transfer-Encoding: quoted-printable

<p dir=3D"ltr">And note that netem needs to be run on a separate system. </=
p>
<p dir=3D"ltr">Jim</p>
<div class=3D"gmail_quote">On Mar 14, 2013 2:30 PM, &quot;Bill Ver Steeg (v=
ersteb)&quot; &lt;<a href=3D"mailto:versteb@cisco.com">versteb@cisco.com</a=
>&gt; wrote:<br type=3D"attribution"><blockquote class=3D"gmail_quote" styl=
e=3D"margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">




<div dir=3D"auto">
<div>In an ns2 environment, having a physical test rig is mute. Quite a bit=
 of progress can be made in this domain. A simple tar file of the complete =
ns2 test rig would make for results that can be verified, tweaked and retes=
ted.=A0</div>

<div><br>
</div>
<div>Once we move to a real device, it seems that Linux sources and sinks w=
ith netem-like impairments would work. I used to have a pretty good test ri=
g, but I loaned it to GaTech last year.......=A0</div>
<div><br>
</div>
<div>I do think that a well defined test rig would help in this effort, and=
 also in the broader congestion work underway at IETF. Getting gear is easi=
er than staff to run it. If I can get a coop student this summer, maybe I c=
an cobble some gear together and
 stand up a test rig somewhere that is publicly accessible.=A0</div>
<div><br>
</div>
<div><br>
</div>
<div>Bvs=A0</div>
<div><br>
On Mar 14, 2013, at 2:15 PM, &quot;Dan Weber&quot; &lt;<a href=3D"mailto:da=
n@marketsoup.com" target=3D"_blank">dan@marketsoup.com</a>&gt; wrote:<br>
<br>
</div>
<blockquote type=3D"cite">
<div>Hi Bill,<br>
<br>
I understand your concerns about netem.=A0 It&#39;s no panacea.<br>
<br>
I&#39;m shooting to define a table of configurable parameters to go with ea=
ch test case.<br>
<br>
E.g.<br>
Ploss 1% Correlation (90%)<br>
Delay 80ms Jitter 50ms<br>
Bandwidth 2mbit/s<br>
etc.<br>
<br>
It seems to me that this is required regardless of the tools we use.=A0 If =
we can standardize on which parameters are adjustable, we can include them =
in every test case and use a variety of tools.=A0 As for defining a referen=
ce test environment, does anyone have
 access to some systems we could open up for the purpose of this testing?=
=A0 I may be able to call around and see if we can get one provided for us =
if no one else has resources available.<br>
<br>
Thanks,<br>
Dan<br>
<br>
<div class=3D"gmail_quote">On Thu, Mar 14, 2013 at 12:04 PM, Bill Ver Steeg=
 (versteb)
<span dir=3D"ltr">&lt;<a href=3D"mailto:versteb@cisco.com" target=3D"_blank=
">versteb@cisco.com</a>&gt;</span> wrote:<br>
<blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1p=
x #ccc solid;padding-left:1ex">
<div dir=3D"auto">
<div>netem is a very good emulation tool, and we often use it to get close =
to the right answer. However, I will caution that there is a wide variety o=
f packet schedulers in use in the wild. Not all of them the exhibit flat, d=
eterministic scheduling that netem
 defaults to.=A0</div>
<div><br>
</div>
<div>One (of many) example of such a packet scheduling/limiting algorithm w=
as described in yesterday&#39;s congestion control research group. HFC CPE =
(cable modems) use a leaky bucket algorithm that provides a weighted averag=
e to enforce an SLA, and this can interact
 with higher level control loops. In other words, if a sender is slightly u=
nder the provisioned link rate for a while, it can then can send at slightl=
y over the link rate for a while. If you are not careful, a variable bitrat=
e sender can infer an incorrect
 SLA rate based on such scheduler behavior. I do not imply that this is the=
 definitive case, just that there can be interactions. Flow control algorit=
hms that work very well for a given scheduler can have unexpected entanglem=
ents with another (chain of) schedulers.=A0</div>

<div><br>
</div>
<div>Fortunately, there are ways to introduce these behaviors in test rigs.=
=A0</div>
<div><br>
</div>
<div>So, if we define a reference test environment, we either need to have =
a few scheduler variations or recognize that the test rig is only a first-o=
rder approximation and live testing on a variety of network topologies is r=
equired.=A0</div>

<div><br>
</div>
<div>Short version- =A0I agree that netem is a very good tool, but you need=
 to be careful how you use it.=A0</div>
<span><font color=3D"#888888">
<div><br>
</div>
<div>Bvs</div>
</font></span><br>
</div>
</blockquote>
</div>
</div>
</blockquote>
</div>

</blockquote></div>

--e89a8f839d51c82edc04d7eb7046--

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We use tc extensively in our test, I think this may be helpful, see here:
http://www.slideshare.net/fuji246/network-emulator

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<div dir="ltr"><div class="gmail_extra">We use tc extensively in our test, I think this may be helpful, see here: <a href="http://www.slideshare.net/fuji246/network-emulator">http://www.slideshare.net/fuji246/network-emulator</a><br>
</div></div>

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Testcase can be the following combinations

    - typical network(bw, rtt, qdelay, loss, token_burst)

    - with/without cross traffic(CBR, TCP/UDP etc)

    - avail-bw change(capcity change or cross traffic change)

    - inherit loss network

    - route change


Jeromy


2013/3/14 Xiaoqing Zhu (xiaoqzhu) <xiaoqzhu@cisco.com>:
> Hi,
>
> Great idea on the packet generator model. We will also be happy to contri=
bute on this piece as well.
>
> In the past we have put together some some ns-2 code for via in two appro=
aches:
>
> - trace-driven:  where the packet sizes and intervals are taken from h.26=
4 encoding result of video sequences. one could also switch between differe=
nt rate versions of the same video content at GOP boundaries.
>
> - statistical modeling: where the encoder rate control behavior is modele=
d statistically, with tunable parameters on burstiness, range of deviation =
from target rate, and lag in reaction (i.e., how long it takes the encoder =
to react to a change in target rate).
>
> Thanks,
> Xiaoqing
>
>
>
>
>
> On Mar 14, 2013, at 10:12 AM, Mirja Kuehlewind wrote:
>
>> Hi,
>>
>> just to name one more option:
>>
>> we have a Java-based, event-driven simulation framework which we recentl=
y
>> extended to integrate a virtual machine (using QEMU). That means we can =
use
>> traffic generated by a real operating system in simulations. Currently w=
e
>> only have a small relay program running in the virtual machine that can =
be
>> controlled by the simulation program to send chunks of data. Thus a spec=
ific
>> generator could be implemented in simulation program. The relay program =
sends
>> only TCP traffic at the moment. Moreover, we can also run a real applica=
tion
>> in the virtual machine. At the moment we are still working on the framew=
ork.
>> Before we will publish the code, we need to find some more bug fixes. Bu=
t the
>> simulations are working and we are using this for all our current TCP
>> simulations.
>>
>> Mirja
>>
>>
>> On Thursday 14 March 2013 15:07:38 Zaheduzzaman Sarker wrote:
>>> 2. Packet Generator Model
>>> After re-reading Varun's draft and speaking with him on the evaluation
>>> scenarios, I think it makes enough sense to build a packet generator to
>>> implement what he suggests.  The time involved isn't substantial, and i=
t
>>> can help us get a unified starting point for testing even if we decide =
to
>>> change it later.  It's worth at least this much.
>>>
>>> My proposed implementation of this will be able to simulate real video
>>> traffic as well as fixed packet generator traffic.  For the real video
>>> traffic, it can be fed a CSV file filled with frame sizes and RTP
>>> timestamps. [Zaheduzzaman Sarker] we already use some sort of packet
>>> generator in our simulations, right? And it might be also not that hard=
 to
>>> come up with a packet generator. However, the hard part is to realize h=
ow
>>> representative those packet generator are for our usecases at hand? IMH=
O,
>>> we will need to verify the generate packet trace a with some reference
>>> trace. Also remember we might have QoE as evaluation  metric too and I =
hear
>>> people are also talking about subjective quality related to fairness. I=
t
>>> will be hard to achieve that with just simple packet generators. For th=
e
>>> fixed generator traffic, ladder steps can be used.  I'm thinking I'll s=
tart
>>> a project on Google Code, and make any interested participant a develop=
er
>>>
>>>
>>>
>>> BR
>>>
>>> Zahed
>>
>>
>>
>> --
>> -------------------------------------------------------------------
>> Dipl.-Ing. Mirja K=FChlewind
>> Institute of Communication Networks and Computer Engineering (IKR)
>> University of Stuttgart, Germany
>> Pfaffenwaldring 47, D-70569 Stuttgart
>>
>> tel: +49(0)711/685-67973
>> email: mirja.kuehlewind@ikr.uni-stuttgart.de
>> web: www.ikr.uni-stuttgart.de
>> -------------------------------------------------------------------
>

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Hi,

below are the raw notes from our minute taker (thanks, Michael!)

If you spoke on the mike, please make sure your points were sufficiently we=
ll captured and send corrections/additions by mail.

Lars



RTP Media Congestion Avoidance Techniques (RMCAT) at IETF 86, Orlando
Monday March 11 2013, 9:00


Presentation:
Lars Eggert (LE), Administrativa & WG Status

Harald Alvestrand (HA): You said "add a year to milestones". I would hope t=
hat we could get adoption done within the first year.
LE: Yes, this was meant for finishing the documents, not adopting them.


Presentation:
Randell Jesup (RJ), Congestion Control Requirements For RMCAT
draft-jesup-rmcat-reqs-01 (milestone cc-requirements)=20

Michael Ramalho (MR): I would challenge us to come up with a metric of fair=
ness in terms of self-fairness and types of traffic that they're competing =
against, and have it independent of whether you're in the core or access ne=
twork.

HA: I think we know how to detect blatant unfairness. We should say "here's=
 the tests. if something fails one of these tests, it's not fair". I'm happ=
y with a definition of fairness that says "we can't tell whether you're fai=
r or not, but we can tell whether you passed these tests or not"

RJ: Agreed. Still comes down to a subjective evaluation, to check if on the=
 whole .

LE: This list could get very long. It would be nice if we could find some p=
rinciples up front. e.g., "you shouldn't start a new flow persistently". Th=
en define cases for checking against those. The people who have solution ca=
ndidates must have thought about this, so these people should make proposal=
s. If we don't solve it now in the meeting we should see if we can make som=
e progress later in ths meeting.

Toerless Eckert (TE): Let's agree on a priority list. It seems to be clear =
that total blatant unfairness is on the top of the list. Second, how self-f=
riendly or unfriendly is existing real-time traffic already, just as a comp=
arison metric. I would love to see what we can achieve not only over an Int=
ernet path with other traffic, but also if there was no other traffic (self=
-fairness).

LE: It's not only us competing against TCP. rtcweb has a data channel.
Matt Mathis: TCP fairness is a very deep rathole, guaranteed not to converg=
e. I would suggest making a list of fairness criteria, evaluate based on si=
mple principles. Preventing starvation, for instance, would be very importa=
nt, and that is something you could define rigorously.

LE: We should start a design team on this.

Faisan Hasan (FH): Shall we go down to "friendliness" instead of fairness?

Dan Weber (DW): MOS scores as in audio quality or something like that might=
 be good enough.

RJ: Problematic to evaluate.

LE: It's a bit harder, if you produce poor audio quality that's only your o=
wn problem, but a bad congestion controller harms everyone.

Colin Perkins (CP): "Compete reasonably with" - TCP. But we could decide in=
itially that that's not a goal and only compete fairly with ourselves.

RJ: When competing with TCP, at some point you have to transition to loss b=
ased mode. So therefore it=20

CP: Decision, what do we want to give up?

RJ: In the short term, we definitely want to compete with TCP. Longer term,=
 either give up or transition.

??: One has to judge the importance of fairness over other characteristics.=
 Reaction time, throughput loss, might be more problematic. Don't want to l=
ose throughput because of fairness constraints.

LE: At some point we have to make a decision. If you want to actively help =
work on this stay here, we will try to form a design team.


Presentation:
Varun Singh (VS), Evaluating Congestion Control for Interactive Real-time M=
edia
draft-singh-rmcat-cc-eval-02 (milestone eval-criteria)=20

RJ: Discard rate =3D maximum beyond which we shouldn't resume and =85 if yo=
u achieved a low average delay =85 my real question is: is this relative to=
 shortest delay or is it an absolute value?

VS: Absolute value.

MR: Should have some user related metric. like =85  find out where the tail=
 is, wait a reasonable amount of time for a packet to come in. What's a rea=
sonable amount of time?

VS: Playout delay when you receive packets.

MR: small jitter buffer will consider well it's audio or video. Small jitte=
r buffer doesn't use a number.

Zaheduzzaman Sarker (ZS): get rid of delay metric. 300 ms =85   There will =
be some queuing in the bottlenecks. You have to look into this delay figure=
 and compare with RTT etc =85   CDF of queuing delay: you get an idea of va=
lues =85 I think we shouldn't have delay.

VS: Discard rate to check if trade-off make sense or not.

ZS: But then it shouldn't be a fixed value.

DW: You can have buffering somewhere, e.g. on the video card, so how does t=
his fit in?

VS: The way I use it is that you get a feedback value from the receiver to =
calculate. It's only for evaluation case, not for deploying it.

ZS: We need a minimum set of guidelines, just those you really care about f=
or real-time interactive communication.

VS: So is this list sufficient?

ZS: We have to discuss. On the mailing list I had 3 more suggestions to thi=
s one.

MR: when you say video frame rate etc., are you creating something artifici=
al or are you using real video sources?

VS: That's an open question. Do we use a packet generation or actual video =
streams?

MR: Suggest to simplify to something we can characterize.

DW: Video is highly inconsistent. Using any form of constant bitrate create=
s serious issues in terms of evaluation. I-frame can blast 100kbytes / seco=
nd.

VS: Should we remove this and say each app starts with different values?

DW: Pick five, use the same streams for all tests. We need to compress it b=
efore we even test this. Then we can focus on network related constraints.

VS: But with congestion control you change the rates on the fly. It's not a=
 fixed stream. Depends on the encoder.

RJ: We'll have to add cases to this as we go along. But must have some unde=
rstanding of edge cases. Single set of parameters will not help us determin=
e those. Another issue, I would suggest to keep the number of variables dow=
n to something reasonable. Suggest: encoder can use all bits it has availab=
le, starts with a given rate. When congestion control gives feedback, contr=
oller changes rate (and sticks with it), with some defined delay based on t=
he frame rate. Trying to characterize all the encoder behaviors will make t=
oo many problems. I-Frames etc. should be avoided anyway, I would not try t=
o model that. Simple packet generator would be better.

HA: Not try to use real video streams for the tests we do when we develop t=
hings. But we need test scenario that can show that cc. is not ruining the =
video experience.

Gorry Fairhurst (GF): Maximum end-to-end delay of 300ms stands out, what is=
 it? Application expectation? Network requirement? Is RMCAT going to break =
the circuit beyond 300ms?

VS: No it's not a circuit breaker. The packet would not be fed into the dec=
oder, would be discarded among arrival.

GF: RMCAT might be used for various things=85 let's be clear to separate au=
dio / video / transport part of it.

ZS: We're talking about the wrong things here now. We should focus on test =
scenarios, not values. Evaluation draft should take this approach, to test =
if guidelines have been followed, but fixed parameters won't work.

MR: I agree, we have to take out some uncertainty. Packet generator that sh=
ould follow some rule of the cc. you want to test. TCP tracks bits in fligh=
t which is a round trip measure. It worked long, so our algorithm is more l=
ikely if it has a measure of capacity, measure of bits in flight. We need t=
o have a mechanism that's stable, behaves well=85 let's not worry about vid=
eo MOS score yet. Not start with a real video source, but packet generator =
fit to test a cc. mechanism.

Mo Zanaty (MZ): we're looking at it from different perspectives. Applicatio=
n perspective vs. network perspective. Now we're mixing all those, not good=
. In the requirements and evaluation, let's have a section for application-=
level objective metrics. Need to start putting down, what are the real need=
s of video applications - start rate, loss tolerance, =85 then network-cent=
ric flow behavior.

RJ: Good to separate network parameters, video parameters, application para=
meters. What are the parameters of the networks that connect the nodes for =
evaluation? Critical for determining the results of the algorithm. e.g. low=
-delay network, WiFi network =85 not a single set of points, a multi-dimens=
ional surface of results.

Magnus Westerlund (MW1): Important to evaluate that algorithms are stable. =
Minimum path delay is important. Video is dynamic, this is the challenge he=
re. For evaluation, let's get as many bits across as possible is also a pos=
sible evaluation for these cases.

Slide Evaluation Scenarios 1/3
MR: is this delay the propagation delay or the queuing delay? Queuing delay=
 is rather important (buffer bloat).

Slide Evaluation Scenarios 2/3
ZS: each user will have different capacity. Where is the bottleneck? Capaci=
ty changes.

MM: single queue assumption or using different classes?

VS: single

MM: it should be stated for cases where throughput maximizing and delay-bas=
ed mechanisms competing are guaranteed to fail.

MR: We could design a topology such that we're only competing against ourse=
lves and stay in a delay-based mode. But maybe I'm wrong, I thought this gr=
oup was about competing against other traffic too.

LE: assumption is correct. I think we want to have a set of isolated scenar=
ios but also cover running on the Internet with other traffic.

MM: Have to ask: when you have to compete, and when you have separate queue=
s, and when mode switching works.

LE: 20 hands for adopting. 0 hands against. We can call that rough consensu=
s.

Evaluation criteria draft: 10 hands for adopting. 4 or 5 hands against. So =
not clear consensus.



Presentation:
Michael Welzl (MW2), Coupled congestion control for RTP media
draft-welzl-rmcat-coupled-cc-00 (milestone group-cc)=20

*** FROM ANNA

LE: Fairness is not priorities (slide 5)

MW2: Proportional share

MW: How do you handle mix of RTP, SCTP flows etc. Can you handle all these =
cases.

MW2: Current version can handle 5-tuples, beyond that need shared bottlenec=
k detection, false negatives will not be a problem. Integrating rate-based =
and window-based is a challenge. Could have multiple FSEs.

MW: what we have in rtcweb is priorities=20

LE: assume kernel impl will not change on short time-scale, application cou=
ld do more quick adaptation.

DW: Would it make sense to broadcast state table on local LAN so everyone c=
an see what goes on?

MW2: Interesting research topic, but raises too many issues for rmcat.

ZS: Do the flows using the FSE run the same cong control algorithm? Could c=
over more scenarios than rmcat

MW2: in my tests yes, but we could use different ones.

LE: Are more working on coupled cong control. No.



Presentation:
Colin Perkins (CP), On the Use of RTP Control Protocol (RTCP) Feedback for =
Unicast Multimedia Congestion Control
draft-perkins-rmcat-rtp-cc-feedback (related to milestone rtcp-requirements=
)=20

MZ: I always assumed that we would do RSize extension.

CP: I was picking pessimistic assumption.

MZ: I think it's even doable in the per-packet case.

CP: Depends on the bandwidth. But there's maybe not even a point.

Hannes Tschofenig (HT): Sending feedback quickly has its origin in cc. for =
traffic that is very quickly adaptable. In this case, there are reasons why=
 you can't change it so fast anyway. And you may not want to do it anyway b=
ecause of the user experience.

CP: Agreed. But you do need enough feedback so you can tell when problems a=
re happening, to look at change in queuing delay for example. Frame feedbac=
k seems like a reasonable compromise, with also sending feedback faster whe=
n problems are developing.

HT: Agreed. I think RTCP is in fact a suitable mechanism for cc. feedback.

MR: Draft is very well written, thank you. Per-frame or per-packet feedback=
 is a sampling problem. Ideally you do every single packet because backchan=
nel is unreliable. Feedback has to be often enough for the control system t=
o work. The fact that video encoders are limited in how fast they can chang=
e the rate is somehow irrelevant for the question of convergence, stability=
 etc. of the controller.

CP: Agreed, this is a balancing act. We need to figure out what is appropri=
ate. If something roughly in the order of an RTT is enough I think we can d=
o that with RTCP very well.

MR: My point is that it's a control theory problem wrt. what types of feedb=
ack we want in the system.

RJ: Agreed. But these are just worst case scenarios, best case scenarios ar=
e much better. You may need to change rates dramatically in certain situati=
ons, to recover appropriately and deal with bursty situations.

TE: Is there anything video specific here? Can we ignore that it's video? P=
eople talk about VBR audio codecs that can change arbitrarily. Is the feedb=
ack for a video codec not totally decorrelated from congestion feedback?

CP: With audio, you may quickly run into situations where you send as much =
RTCP feedback as data.

RJ: RTCP-fair feedback will work doesn't mean that an algorithm should use =
RTCP feedback. This feedback is important and needed for certain cases. Pig=
gybacking may be a useful technique in conjunction with RTCP.

CP: Point is, RTCP is a valid player in the space, potentially the right so=
rt of feedback.

MZ: Agree. Even if RTCP feedback could do it, from an efficiency point of v=
iew the packet rate often matters more than the total bandwidth consumption=
, especially in wireless networks. Especially piggybacking could help.

CP: Not sending any RTCP packets that you wouldn't send anyway.

MR: Feedback could be very low in terms of bandwidth, but it needs to come =
often enough, but perhaps not more often than necessary. I see it as a samp=
ling problem.

Kevin Gross (KG): How often do you sample, how often do you get that data? =
You can cause control system instability by having too much latency in feed=
back.

CP: People developing CC. algorithms must answer that for their algorithm. =
I'm trying to show that for reasonable amounts of feedback, RTCP can work. =
Don't discard it because of the assumption it's too slow.

ZS: Do we know what the minimum desired amount of feedback is? It depends o=
n how we design our controller, that is not answered anywhere yet.

MR: TCP does it via clocking. You have to take the RTT into consideration. =
I want to make that point really clear, in order to get the control loop ri=
ght, you have to think about that.

LE: What is the intended goal of the draft?

CP: Create awareness, raise discussion. I consider this as success. Not pro=
posing to progress it any further.

LE: I could see the material in an appendix of a mechanism, to explain desi=
gn choices that were made.



Presentation:
Rong Pan (RP), NADA: A Unified Congestion Control Scheme for Real-Time Medi=
a
draft-zhu-rmcat-nada=20

ZS: PCN based signaling is something new that you want to design, it's not =
there, right?

LE: PCN standards are out.

ZS: Yes, but not used yet. We're going with things for the current Internet=
.

LE: We do a mechanism for the current Internet, but good if it works better=
 in the future Internet.

ZS: I have similar results, and shown in the CC. workshop. Second thing: yo=
u have a rate-shaping buffer at the sender. Does it relate to video frames?=
 It's per packet, right?

RP: Video frames are there, buffer basically smooths into the network.

ZS: Packet shaper on the packet level that delays packets may affect the su=
bjective quality, e.g. when doing it with I-frame. Importance of packet is =
missing.

RJ: Queue is a fair number of packets beyond a certain window in your simul=
ation. Once you're overloaded, do you end up with delay even when competing=
 with itself?

RP: Depends on where your settle, on how heavily congested things are.

RJ: Loss rates on one slide were in the 3-5% range. That's a pretty high lo=
ss rate for video.

RP: If that's the way the network gives the feedback then that's how it is.=
 But algorithm could be tuned to be more sensitive to delay.

RJ: Latency is a critical parameter. Throughput includes any recovery or FE=
C or whatever is needed to make for a high loss rate. Must be factored in. =
FEC has delay issues.

KR: Have you looked at Codel?

RP: We should evaluate that.

KR: Codel works without ECN or PCN.

RP: But it does drop packets.

KR: But the way it drops packets based on delay measurements.

Jim Gettys (JG): Codel is happy to do ECN.

RP: Have to look at it. If it's ECN that's fine. Algorithm can be adopted h=
ere. As a result you'll have less losses.

KR: How does it compete with TCP?

RP: Good question. Here we look at self-fairness. If TCP is the concern, we=
 have a similar design where, we can respond with sqrt(p) when competing wi=
th loss based.

HA: Next time you update the draft, please include a reference for PCN.

TE: Agree with ZS, bringing in more codec knowledge might be used to optimi=
ze the behavior.

ZS: Encoder would have to know that this packet has been delayed or lost.

TE: Showed that it is fair =85 test should take into account whether ECN ha=
s some benefit over delay variation, especially when there is other traffic=
. Do we have clear evidence that we could ask network operators to enable i=
t? That's still the open question now.

MZ: In Nada, losses are translated into delays, in Dflow, delays are transl=
ated into losses, in Google yet different=85 what is the right way to mix t=
he signals?

LE: Evaluation criteria are more about objective quality measures of the re=
sult. Up to the mechanism to decide what's best.

MZ: There should be objective criteria on how to mix the signals - e.g. thi=
s combination of delay and loss is better than that combination of delay an=
d loss.

LE: agreed.

RJ: If you take in more loss you effectively reduce your bit rate because y=
ou have to add FEC. Subjective quality is more important in the end.

ZS: Different algorithms handle delay and loss rates in different ways. Let=
's define how many % loss rate are okay. Google's mechanism says 10% is oka=
y.

LE: It's not so easy. In a shared bottleneck you're not going to be the one=
 who sees all the losses.




From vsingh.ietf@gmail.com  Mon Mar 18 08:23:53 2013
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Date: Mon, 18 Mar 2013 17:23:27 +0200
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Subject: [rmcat] Virtual Design Meetings
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Hi all,

To continue making progress on the fairness principles,
we will have virtual design meetings every few weeks.

At the last design meeting, we discussed possible meeting times
(1hr timeslot: 1100-noon, noon-1300 eastern US time seemed
like a good fit for most people). However, I am open to expanding the
list of available times. Drop me a note and I will add it to the
doodle choice list.

Here's a doodle poll for the next one: http://www.doodle.com/2uci3a27nw37qwk4

Mo Zanaty and Michael Romalho offered to host this via webex (with recording).

Cheers,
Varun

-- 
http://www.netlab.tkk.fi/~varun/

From kevin.gross@avanw.com  Mon Mar 18 10:04:27 2013
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Subject: Re: [rmcat] Virtual Design Meetings
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Is there a plan for digesting what we accomplished in the IETF86 design
meetings? Right now we just have raw notes. At a minimum, I would expect
revision to one of our existing drafts. I am willing to help but I need to
start by reading those drafts :)

Kevin Gross
+1-303-447-0517
Media Network Consultant
AVA Networks - www.AVAnw.com <http://www.avanw.com/>, www.X192.org


On Mon, Mar 18, 2013 at 9:23 AM, Varun Singh <vsingh.ietf@gmail.com> wrote:

> Hi all,
>
> To continue making progress on the fairness principles,
> we will have virtual design meetings every few weeks.
>
> At the last design meeting, we discussed possible meeting times
> (1hr timeslot: 1100-noon, noon-1300 eastern US time seemed
> like a good fit for most people). However, I am open to expanding the
> list of available times. Drop me a note and I will add it to the
> doodle choice list.
>
> Here's a doodle poll for the next one:
> http://www.doodle.com/2uci3a27nw37qwk4
>
> Mo Zanaty and Michael Romalho offered to host this via webex (with
> recording).
>
> Cheers,
> Varun
>
> --
> http://www.netlab.tkk.fi/~varun/
>

--f46d04446b212d985504d835f537
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Is there a plan for digesting what we accomplished in the IETF86 design mee=
tings? Right now we just have raw notes. At a minimum, I would expect revis=
ion to one of our existing drafts. I am willing to help but I need to start=
 by reading those drafts :)<div>
<br clear=3D"all"><div>Kevin Gross<br><div><a href=3D"tel:%2B1-303-447-0517=
" value=3D"+13034470517" target=3D"_blank">+1-303-447-0517</a></div><div>
Media Network Consultant<br><div>AVA Networks -=A0<a href=3D"http://www.ava=
nw.com/" target=3D"_blank">www.AVAnw.com</a>,=A0<a href=3D"http://www.X192.=
org" target=3D"_blank">www.X192.org</a></div></div></div>
<br><br><div class=3D"gmail_quote">On Mon, Mar 18, 2013 at 9:23 AM, Varun S=
ingh <span dir=3D"ltr">&lt;<a href=3D"mailto:vsingh.ietf@gmail.com" target=
=3D"_blank">vsingh.ietf@gmail.com</a>&gt;</span> wrote:<br><blockquote clas=
s=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1px #ccc solid;pad=
ding-left:1ex">

Hi all,<br>
<br>
To continue making progress on the fairness principles,<br>
we will have virtual design meetings every few weeks.<br>
<br>
At the last design meeting, we discussed possible meeting times<br>
(1hr timeslot: 1100-noon, noon-1300 eastern US time seemed<br>
like a good fit for most people). However, I am open to expanding the<br>
list of available times. Drop me a note and I will add it to the<br>
doodle choice list.<br>
<br>
Here&#39;s a doodle poll for the next one: <a href=3D"http://www.doodle.com=
/2uci3a27nw37qwk4" target=3D"_blank">http://www.doodle.com/2uci3a27nw37qwk4=
</a><br>
<br>
Mo Zanaty and Michael Romalho offered to host this via webex (with recordin=
g).<br>
<br>
Cheers,<br>
Varun<br>
<span><font color=3D"#888888"><br>
--<br>
<a href=3D"http://www.netlab.tkk.fi/~varun/" target=3D"_blank">http://www.n=
etlab.tkk.fi/~varun/</a><br>
</font></span></blockquote></div><br></div>

--f46d04446b212d985504d835f537--

From vsingh.ietf@gmail.com  Fri Mar 22 02:48:23 2013
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From: Varun Singh <vsingh.ietf@gmail.com>
Date: Fri, 22 Mar 2013 11:47:58 +0200
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To: Kevin Gross <kevin.gross@avanw.com>
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Subject: Re: [rmcat] Virtual Design Meetings
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Hi Kevin,

I was under the impression there was inclination to have a virtual meeting
before Easter? and then every 3 weeks.

>From the doodle poll, I am leaning towards timeslots on Wednesday/Thursday.
I'll leave the poll open until Saturday and inform the chosen time.

-Varun

On Mon, Mar 18, 2013 at 7:02 PM, Kevin Gross <kevin.gross@avanw.com> wrote:
> Is there a plan for digesting what we accomplished in the IETF86 design
> meetings? Right now we just have raw notes. At a minimum, I would expect
> revision to one of our existing drafts. I am willing to help but I need to
> start by reading those drafts :)
>
> Kevin Gross
> +1-303-447-0517
> Media Network Consultant
> AVA Networks - www.AVAnw.com, www.X192.org
>
>
> On Mon, Mar 18, 2013 at 9:23 AM, Varun Singh <vsingh.ietf@gmail.com> wrote:
>>
>> Hi all,
>>
>> To continue making progress on the fairness principles,
>> we will have virtual design meetings every few weeks.
>>
>> At the last design meeting, we discussed possible meeting times
>> (1hr timeslot: 1100-noon, noon-1300 eastern US time seemed
>> like a good fit for most people). However, I am open to expanding the
>> list of available times. Drop me a note and I will add it to the
>> doodle choice list.
>>
>> Here's a doodle poll for the next one:
>> http://www.doodle.com/2uci3a27nw37qwk4
>>
>> Mo Zanaty and Michael Romalho offered to host this via webex (with
>> recording).
>>
>> Cheers,
>> Varun
>>
>> --
>> http://www.netlab.tkk.fi/~varun/
>
>



-- 
http://www.netlab.tkk.fi/~varun/

From zaheduzzaman.sarker@ericsson.com  Fri Mar 22 04:31:05 2013
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Hi Varun,

Was there any meeting on 15th March on site?

Could you also share the updated meeting notes from on site meeting?

BR

Zahed

On 2013-03-18 16:23, Varun Singh wrote:
> Hi all,
>
> To continue making progress on the fairness principles,
> we will have virtual design meetings every few weeks.
>
> At the last design meeting, we discussed possible meeting times
> (1hr timeslot: 1100-noon, noon-1300 eastern US time seemed
> like a good fit for most people). However, I am open to expanding the
> list of available times. Drop me a note and I will add it to the
> doodle choice list.
>
> Here's a doodle poll for the next one: http://www.doodle.com/2uci3a27nw37qwk4
>
> Mo Zanaty and Michael Romalho offered to host this via webex (with recording).
>
> Cheers,
> Varun
>

-- 

Zahed

==================================================
ANM ZAHEDUZZAMAN SARKER


Ericsson AB
Multimedia Technologies
Laboratoriegränd 11
97128 Luleå, Sweden
Phone +46 10 717 37 43
Fax +46 920 996 21
SMS/MMS +46 76 115 37 43
zaheduzzaman.sarker@ericsson.com
www.ericsson.com

==================================================


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From: "Mo Zanaty (mzanaty)" <mzanaty@cisco.com>
To: Varun Singh <vsingh.ietf@gmail.com>, Kevin Gross <kevin.gross@avanw.com>
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Warning: Doodle appears to have a timezone bug. In the timezone selection b=
ox, it thinks the US is still on standard/winter time, but we already switc=
hed to daylight/summer time. After selecting a timezone, for example EST (G=
MT-5), the times it shows as EST (GMT-5) are incorrect and are actually EDT=
 (GMT-4).

I assume the meeting timeslots are GMT 15:00-16:00 and 16:00-17:00. That wo=
uld be EDT (GMT-4) 11:00-12:00 and 12:00-13:00. But Doodle shows EST (GMT-5=
) 11:00-1200 and 12:00-13:00. So if you believed its timezone label and man=
ually added an hour to get daylight/summer time, you get the wrong meeting =
times. If you ignore the timezone label and just look at the meeting times,=
 you get the right times.

Mo

-----Original Message-----
From: rmcat-bounces@ietf.org [mailto:rmcat-bounces@ietf.org] On Behalf Of V=
arun Singh
Sent: Friday, March 22, 2013 5:48 AM
To: Kevin Gross
Cc: rmcat WG
Subject: Re: [rmcat] Virtual Design Meetings

Hi Kevin,

I was under the impression there was inclination to have a virtual meeting
before Easter? and then every 3 weeks.

>From the doodle poll, I am leaning towards timeslots on Wednesday/Thursday.
I'll leave the poll open until Saturday and inform the chosen time.

-Varun

On Mon, Mar 18, 2013 at 7:02 PM, Kevin Gross <kevin.gross@avanw.com> wrote:
> Is there a plan for digesting what we accomplished in the IETF86 design
> meetings? Right now we just have raw notes. At a minimum, I would expect
> revision to one of our existing drafts. I am willing to help but I need t=
o
> start by reading those drafts :)
>
> Kevin Gross
> +1-303-447-0517
> Media Network Consultant
> AVA Networks - www.AVAnw.com, www.X192.org
>
>
> On Mon, Mar 18, 2013 at 9:23 AM, Varun Singh <vsingh.ietf@gmail.com> wrot=
e:
>>
>> Hi all,
>>
>> To continue making progress on the fairness principles,
>> we will have virtual design meetings every few weeks.
>>
>> At the last design meeting, we discussed possible meeting times
>> (1hr timeslot: 1100-noon, noon-1300 eastern US time seemed
>> like a good fit for most people). However, I am open to expanding the
>> list of available times. Drop me a note and I will add it to the
>> doodle choice list.
>>
>> Here's a doodle poll for the next one:
>> http://www.doodle.com/2uci3a27nw37qwk4
>>
>> Mo Zanaty and Michael Romalho offered to host this via webex (with
>> recording).
>>
>> Cheers,
>> Varun
>>
>> --
>> http://www.netlab.tkk.fi/~varun/
>
>



--=20
http://www.netlab.tkk.fi/~varun/

From vsingh.ietf@gmail.com  Fri Mar 22 11:52:38 2013
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From: Varun Singh <vsingh.ietf@gmail.com>
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Date: Fri, 22 Mar 2013 20:52:23 +0200
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To: "Mo Zanaty (mzanaty)" <mzanaty@cisco.com>
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Cc: rmcat WG <rmcat@ietf.org>, Kevin Gross <kevin.gross@avanw.com>
Subject: Re: [rmcat] Virtual Design Meetings
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Maybe there are more timezone bugs because I didn't take into account
Europe moving into DST this coming weekend.

Without taking DST into account, the GMT times were correct.

Regards,
Varun

----
http://www.netlab.tkk.fi/~varun

On 22.3.2013, at 20.14, "Mo Zanaty (mzanaty)" <mzanaty@cisco.com> wrote:

> Warning: Doodle appears to have a timezone bug. In the timezone selection=
 box, it thinks the US is still on standard/winter time, but we already swi=
tched to daylight/summer time. After selecting a timezone, for example EST =
(GMT-5), the times it shows as EST (GMT-5) are incorrect and are actually E=
DT (GMT-4).
>
> I assume the meeting timeslots are GMT 15:00-16:00 and 16:00-17:00. That =
would be EDT (GMT-4) 11:00-12:00 and 12:00-13:00. But Doodle shows EST (GMT=
-5) 11:00-1200 and 12:00-13:00. So if you believed its timezone label and m=
anually added an hour to get daylight/summer time, you get the wrong meetin=
g times. If you ignore the timezone label and just look at the meeting time=
s, you get the right times.
>
> Mo
>
> -----Original Message-----
> From: rmcat-bounces@ietf.org [mailto:rmcat-bounces@ietf.org] On Behalf Of=
 Varun Singh
> Sent: Friday, March 22, 2013 5:48 AM
> To: Kevin Gross
> Cc: rmcat WG
> Subject: Re: [rmcat] Virtual Design Meetings
>
> Hi Kevin,
>
> I was under the impression there was inclination to have a virtual meetin=
g
> before Easter? and then every 3 weeks.
>
> From the doodle poll, I am leaning towards timeslots on Wednesday/Thursda=
y.
> I'll leave the poll open until Saturday and inform the chosen time.
>
> -Varun
>
> On Mon, Mar 18, 2013 at 7:02 PM, Kevin Gross <kevin.gross@avanw.com> wrot=
e:
>> Is there a plan for digesting what we accomplished in the IETF86 design
>> meetings? Right now we just have raw notes. At a minimum, I would expect
>> revision to one of our existing drafts. I am willing to help but I need =
to
>> start by reading those drafts :)
>>
>> Kevin Gross
>> +1-303-447-0517
>> Media Network Consultant
>> AVA Networks - www.AVAnw.com, www.X192.org
>>
>>
>> On Mon, Mar 18, 2013 at 9:23 AM, Varun Singh <vsingh.ietf@gmail.com> wro=
te:
>>>
>>> Hi all,
>>>
>>> To continue making progress on the fairness principles,
>>> we will have virtual design meetings every few weeks.
>>>
>>> At the last design meeting, we discussed possible meeting times
>>> (1hr timeslot: 1100-noon, noon-1300 eastern US time seemed
>>> like a good fit for most people). However, I am open to expanding the
>>> list of available times. Drop me a note and I will add it to the
>>> doodle choice list.
>>>
>>> Here's a doodle poll for the next one:
>>> http://www.doodle.com/2uci3a27nw37qwk4
>>>
>>> Mo Zanaty and Michael Romalho offered to host this via webex (with
>>> recording).
>>>
>>> Cheers,
>>> Varun
>>>
>>> --
>>> http://www.netlab.tkk.fi/~varun/
>>
>>
>
>
>
> --
> http://www.netlab.tkk.fi/~varun/

From vsingh.ietf@gmail.com  Mon Mar 25 08:55:53 2013
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From: Varun Singh <vsingh.ietf@gmail.com>
Date: Mon, 25 Mar 2013 17:55:28 +0200
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Hi all,

I chose the second of the two sessions on Thursday,
it had the best participation.
Of the tentatives: I will make it, Mirja what about you?


Cheers,
Varun

On Fri, Mar 22, 2013 at 8:13 PM, Mo Zanaty (mzanaty) <mzanaty@cisco.com> wr=
ote:
> Warning: Doodle appears to have a timezone bug. In the timezone selection=
 box, it thinks the US is still on standard/winter time, but we already swi=
tched to daylight/summer time. After selecting a timezone, for example EST =
(GMT-5), the times it shows as EST (GMT-5) are incorrect and are actually E=
DT (GMT-4).
>
> I assume the meeting timeslots are GMT 15:00-16:00 and 16:00-17:00. That =
would be EDT (GMT-4) 11:00-12:00 and 12:00-13:00. But Doodle shows EST (GMT=
-5) 11:00-1200 and 12:00-13:00. So if you believed its timezone label and m=
anually added an hour to get daylight/summer time, you get the wrong meetin=
g times. If you ignore the timezone label and just look at the meeting time=
s, you get the right times.
>
> Mo
>
> -----Original Message-----
> From: rmcat-bounces@ietf.org [mailto:rmcat-bounces@ietf.org] On Behalf Of=
 Varun Singh
> Sent: Friday, March 22, 2013 5:48 AM
> To: Kevin Gross
> Cc: rmcat WG
> Subject: Re: [rmcat] Virtual Design Meetings
>
> Hi Kevin,
>
> I was under the impression there was inclination to have a virtual meetin=
g
> before Easter? and then every 3 weeks.
>
> From the doodle poll, I am leaning towards timeslots on Wednesday/Thursda=
y.
> I'll leave the poll open until Saturday and inform the chosen time.
>
> -Varun
>
> On Mon, Mar 18, 2013 at 7:02 PM, Kevin Gross <kevin.gross@avanw.com> wrot=
e:
>> Is there a plan for digesting what we accomplished in the IETF86 design
>> meetings? Right now we just have raw notes. At a minimum, I would expect
>> revision to one of our existing drafts. I am willing to help but I need =
to
>> start by reading those drafts :)
>>
>> Kevin Gross
>> +1-303-447-0517
>> Media Network Consultant
>> AVA Networks - www.AVAnw.com, www.X192.org
>>
>>
>> On Mon, Mar 18, 2013 at 9:23 AM, Varun Singh <vsingh.ietf@gmail.com> wro=
te:
>>>
>>> Hi all,
>>>
>>> To continue making progress on the fairness principles,
>>> we will have virtual design meetings every few weeks.
>>>
>>> At the last design meeting, we discussed possible meeting times
>>> (1hr timeslot: 1100-noon, noon-1300 eastern US time seemed
>>> like a good fit for most people). However, I am open to expanding the
>>> list of available times. Drop me a note and I will add it to the
>>> doodle choice list.
>>>
>>> Here's a doodle poll for the next one:
>>> http://www.doodle.com/2uci3a27nw37qwk4
>>>
>>> Mo Zanaty and Michael Romalho offered to host this via webex (with
>>> recording).
>>>
>>> Cheers,
>>> Varun
>>>
>>> --
>>> http://www.netlab.tkk.fi/~varun/
>>
>>
>
>
>
> --
> http://www.netlab.tkk.fi/~varun/



--=20
http://www.netlab.tkk.fi/~varun/

From dan@marketsoup.com  Mon Mar 25 08:58:21 2013
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Subject: Re: [rmcat] Virtual Design Meetings
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Content-Type: text/plain; charset=UTF-8

Can you send out a meeting invite for the time that is selected?

Thanks,
Dan

On Mon, Mar 25, 2013 at 9:55 AM, Varun Singh <vsingh.ietf@gmail.com> wrote:

> Hi all,
>
> I chose the second of the two sessions on Thursday,
> it had the best participation.
> Of the tentatives: I will make it, Mirja what about you?
>
>
> Cheers,
> Varun
>
> On Fri, Mar 22, 2013 at 8:13 PM, Mo Zanaty (mzanaty) <mzanaty@cisco.com>
> wrote:
> > Warning: Doodle appears to have a timezone bug. In the timezone
> selection box, it thinks the US is still on standard/winter time, but we
> already switched to daylight/summer time. After selecting a timezone, for
> example EST (GMT-5), the times it shows as EST (GMT-5) are incorrect and
> are actually EDT (GMT-4).
> >
> > I assume the meeting timeslots are GMT 15:00-16:00 and 16:00-17:00. That
> would be EDT (GMT-4) 11:00-12:00 and 12:00-13:00. But Doodle shows EST
> (GMT-5) 11:00-1200 and 12:00-13:00. So if you believed its timezone label
> and manually added an hour to get daylight/summer time, you get the wrong
> meeting times. If you ignore the timezone label and just look at the
> meeting times, you get the right times.
> >
> > Mo
> >
> > -----Original Message-----
> > From: rmcat-bounces@ietf.org [mailto:rmcat-bounces@ietf.org] On Behalf
> Of Varun Singh
> > Sent: Friday, March 22, 2013 5:48 AM
> > To: Kevin Gross
> > Cc: rmcat WG
> > Subject: Re: [rmcat] Virtual Design Meetings
> >
> > Hi Kevin,
> >
> > I was under the impression there was inclination to have a virtual
> meeting
> > before Easter? and then every 3 weeks.
> >
> > From the doodle poll, I am leaning towards timeslots on
> Wednesday/Thursday.
> > I'll leave the poll open until Saturday and inform the chosen time.
> >
> > -Varun
> >
> > On Mon, Mar 18, 2013 at 7:02 PM, Kevin Gross <kevin.gross@avanw.com>
> wrote:
> >> Is there a plan for digesting what we accomplished in the IETF86 design
> >> meetings? Right now we just have raw notes. At a minimum, I would expect
> >> revision to one of our existing drafts. I am willing to help but I need
> to
> >> start by reading those drafts :)
> >>
> >> Kevin Gross
> >> +1-303-447-0517
> >> Media Network Consultant
> >> AVA Networks - www.AVAnw.com, www.X192.org
> >>
> >>
> >> On Mon, Mar 18, 2013 at 9:23 AM, Varun Singh <vsingh.ietf@gmail.com>
> wrote:
> >>>
> >>> Hi all,
> >>>
> >>> To continue making progress on the fairness principles,
> >>> we will have virtual design meetings every few weeks.
> >>>
> >>> At the last design meeting, we discussed possible meeting times
> >>> (1hr timeslot: 1100-noon, noon-1300 eastern US time seemed
> >>> like a good fit for most people). However, I am open to expanding the
> >>> list of available times. Drop me a note and I will add it to the
> >>> doodle choice list.
> >>>
> >>> Here's a doodle poll for the next one:
> >>> http://www.doodle.com/2uci3a27nw37qwk4
> >>>
> >>> Mo Zanaty and Michael Romalho offered to host this via webex (with
> >>> recording).
> >>>
> >>> Cheers,
> >>> Varun
> >>>
> >>> --
> >>> http://www.netlab.tkk.fi/~varun/
> >>
> >>
> >
> >
> >
> > --
> > http://www.netlab.tkk.fi/~varun/
>
>
>
> --
> http://www.netlab.tkk.fi/~varun/
>

--14dae934036fb0aaea04d8c1dff9
Content-Type: text/html; charset=UTF-8
Content-Transfer-Encoding: quoted-printable

Can you send out a meeting invite for the time that is selected?<div><br></=
div><div>Thanks,</div><div>Dan<br><br><div class=3D"gmail_quote">On Mon, Ma=
r 25, 2013 at 9:55 AM, Varun Singh <span dir=3D"ltr">&lt;<a href=3D"mailto:=
vsingh.ietf@gmail.com" target=3D"_blank">vsingh.ietf@gmail.com</a>&gt;</spa=
n> wrote:<br>
<blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1p=
x #ccc solid;padding-left:1ex">Hi all,<br>
<br>
I chose the second of the two sessions on Thursday,<br>
it had the best participation.<br>
Of the tentatives: I will make it, Mirja what about you?<br>
<br>
<br>
Cheers,<br>
Varun<br>
<div class=3D"HOEnZb"><div class=3D"h5"><br>
On Fri, Mar 22, 2013 at 8:13 PM, Mo Zanaty (mzanaty) &lt;<a href=3D"mailto:=
mzanaty@cisco.com">mzanaty@cisco.com</a>&gt; wrote:<br>
&gt; Warning: Doodle appears to have a timezone bug. In the timezone select=
ion box, it thinks the US is still on standard/winter time, but we already =
switched to daylight/summer time. After selecting a timezone, for example E=
ST (GMT-5), the times it shows as EST (GMT-5) are incorrect and are actuall=
y EDT (GMT-4).<br>

&gt;<br>
&gt; I assume the meeting timeslots are GMT 15:00-16:00 and 16:00-17:00. Th=
at would be EDT (GMT-4) 11:00-12:00 and 12:00-13:00. But Doodle shows EST (=
GMT-5) 11:00-1200 and 12:00-13:00. So if you believed its timezone label an=
d manually added an hour to get daylight/summer time, you get the wrong mee=
ting times. If you ignore the timezone label and just look at the meeting t=
imes, you get the right times.<br>

&gt;<br>
&gt; Mo<br>
&gt;<br>
&gt; -----Original Message-----<br>
&gt; From: <a href=3D"mailto:rmcat-bounces@ietf.org">rmcat-bounces@ietf.org=
</a> [mailto:<a href=3D"mailto:rmcat-bounces@ietf.org">rmcat-bounces@ietf.o=
rg</a>] On Behalf Of Varun Singh<br>
&gt; Sent: Friday, March 22, 2013 5:48 AM<br>
&gt; To: Kevin Gross<br>
&gt; Cc: rmcat WG<br>
&gt; Subject: Re: [rmcat] Virtual Design Meetings<br>
&gt;<br>
&gt; Hi Kevin,<br>
&gt;<br>
&gt; I was under the impression there was inclination to have a virtual mee=
ting<br>
&gt; before Easter? and then every 3 weeks.<br>
&gt;<br>
&gt; From the doodle poll, I am leaning towards timeslots on Wednesday/Thur=
sday.<br>
&gt; I&#39;ll leave the poll open until Saturday and inform the chosen time=
.<br>
&gt;<br>
&gt; -Varun<br>
&gt;<br>
&gt; On Mon, Mar 18, 2013 at 7:02 PM, Kevin Gross &lt;<a href=3D"mailto:kev=
in.gross@avanw.com">kevin.gross@avanw.com</a>&gt; wrote:<br>
&gt;&gt; Is there a plan for digesting what we accomplished in the IETF86 d=
esign<br>
&gt;&gt; meetings? Right now we just have raw notes. At a minimum, I would =
expect<br>
&gt;&gt; revision to one of our existing drafts. I am willing to help but I=
 need to<br>
&gt;&gt; start by reading those drafts :)<br>
&gt;&gt;<br>
&gt;&gt; Kevin Gross<br>
&gt;&gt; <a href=3D"tel:%2B1-303-447-0517" value=3D"+13034470517">+1-303-44=
7-0517</a><br>
&gt;&gt; Media Network Consultant<br>
&gt;&gt; AVA Networks - <a href=3D"http://www.AVAnw.com" target=3D"_blank">=
www.AVAnw.com</a>, <a href=3D"http://www.X192.org" target=3D"_blank">www.X1=
92.org</a><br>
&gt;&gt;<br>
&gt;&gt;<br>
&gt;&gt; On Mon, Mar 18, 2013 at 9:23 AM, Varun Singh &lt;<a href=3D"mailto=
:vsingh.ietf@gmail.com">vsingh.ietf@gmail.com</a>&gt; wrote:<br>
&gt;&gt;&gt;<br>
&gt;&gt;&gt; Hi all,<br>
&gt;&gt;&gt;<br>
&gt;&gt;&gt; To continue making progress on the fairness principles,<br>
&gt;&gt;&gt; we will have virtual design meetings every few weeks.<br>
&gt;&gt;&gt;<br>
&gt;&gt;&gt; At the last design meeting, we discussed possible meeting time=
s<br>
&gt;&gt;&gt; (1hr timeslot: 1100-noon, noon-1300 eastern US time seemed<br>
&gt;&gt;&gt; like a good fit for most people). However, I am open to expand=
ing the<br>
&gt;&gt;&gt; list of available times. Drop me a note and I will add it to t=
he<br>
&gt;&gt;&gt; doodle choice list.<br>
&gt;&gt;&gt;<br>
&gt;&gt;&gt; Here&#39;s a doodle poll for the next one:<br>
&gt;&gt;&gt; <a href=3D"http://www.doodle.com/2uci3a27nw37qwk4" target=3D"_=
blank">http://www.doodle.com/2uci3a27nw37qwk4</a><br>
&gt;&gt;&gt;<br>
&gt;&gt;&gt; Mo Zanaty and Michael Romalho offered to host this via webex (=
with<br>
&gt;&gt;&gt; recording).<br>
&gt;&gt;&gt;<br>
&gt;&gt;&gt; Cheers,<br>
&gt;&gt;&gt; Varun<br>
&gt;&gt;&gt;<br>
&gt;&gt;&gt; --<br>
&gt;&gt;&gt; <a href=3D"http://www.netlab.tkk.fi/~varun/" target=3D"_blank"=
>http://www.netlab.tkk.fi/~varun/</a><br>
&gt;&gt;<br>
&gt;&gt;<br>
&gt;<br>
&gt;<br>
&gt;<br>
&gt; --<br>
&gt; <a href=3D"http://www.netlab.tkk.fi/~varun/" target=3D"_blank">http://=
www.netlab.tkk.fi/~varun/</a><br>
<br>
<br>
<br>
</div></div><span class=3D"HOEnZb"><font color=3D"#888888">--<br>
<a href=3D"http://www.netlab.tkk.fi/~varun/" target=3D"_blank">http://www.n=
etlab.tkk.fi/~varun/</a><br>
</font></span></blockquote></div><br></div>

--14dae934036fb0aaea04d8c1dff9--

From mirja.kuehlewind@ikr.uni-stuttgart.de  Mon Mar 25 10:58:09 2013
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That's GMT 16:00 - 17:00 on Thursday? That would actually be fine for me...

Mirja


On Monday 25 March 2013 16:55:28 Varun Singh wrote:
> Hi all,
>
> I chose the second of the two sessions on Thursday,
> it had the best participation.
> Of the tentatives: I will make it, Mirja what about you?
>
>
> Cheers,
> Varun
>
> On Fri, Mar 22, 2013 at 8:13 PM, Mo Zanaty (mzanaty) <mzanaty@cisco.com>=
=20
wrote:
> > Warning: Doodle appears to have a timezone bug. In the timezone selecti=
on
> > box, it thinks the US is still on standard/winter time, but we already
> > switched to daylight/summer time. After selecting a timezone, for examp=
le
> > EST (GMT-5), the times it shows as EST (GMT-5) are incorrect and are
> > actually EDT (GMT-4).
> >
> > I assume the meeting timeslots are GMT 15:00-16:00 and 16:00-17:00. That
> > would be EDT (GMT-4) 11:00-12:00 and 12:00-13:00. But Doodle shows EST
> > (GMT-5) 11:00-1200 and 12:00-13:00. So if you believed its timezone lab=
el
> > and manually added an hour to get daylight/summer time, you get the wro=
ng
> > meeting times. If you ignore the timezone label and just look at the
> > meeting times, you get the right times.
> >
> > Mo
> >
> > -----Original Message-----
> > From: rmcat-bounces@ietf.org [mailto:rmcat-bounces@ietf.org] On Behalf =
Of
> > Varun Singh Sent: Friday, March 22, 2013 5:48 AM
> > To: Kevin Gross
> > Cc: rmcat WG
> > Subject: Re: [rmcat] Virtual Design Meetings
> >
> > Hi Kevin,
> >
> > I was under the impression there was inclination to have a virtual
> > meeting before Easter? and then every 3 weeks.
> >
> > From the doodle poll, I am leaning towards timeslots on
> > Wednesday/Thursday. I'll leave the poll open until Saturday and inform
> > the chosen time.
> >
> > -Varun
> >
> > On Mon, Mar 18, 2013 at 7:02 PM, Kevin Gross <kevin.gross@avanw.com>=20
wrote:
> >> Is there a plan for digesting what we accomplished in the IETF86 design
> >> meetings? Right now we just have raw notes. At a minimum, I would expe=
ct
> >> revision to one of our existing drafts. I am willing to help but I need
> >> to start by reading those drafts :)
> >>
> >> Kevin Gross
> >> +1-303-447-0517
> >> Media Network Consultant
> >> AVA Networks - www.AVAnw.com, www.X192.org
> >>
> >> On Mon, Mar 18, 2013 at 9:23 AM, Varun Singh <vsingh.ietf@gmail.com>=20
wrote:
> >>> Hi all,
> >>>
> >>> To continue making progress on the fairness principles,
> >>> we will have virtual design meetings every few weeks.
> >>>
> >>> At the last design meeting, we discussed possible meeting times
> >>> (1hr timeslot: 1100-noon, noon-1300 eastern US time seemed
> >>> like a good fit for most people). However, I am open to expanding the
> >>> list of available times. Drop me a note and I will add it to the
> >>> doodle choice list.
> >>>
> >>> Here's a doodle poll for the next one:
> >>> http://www.doodle.com/2uci3a27nw37qwk4
> >>>
> >>> Mo Zanaty and Michael Romalho offered to host this via webex (with
> >>> recording).
> >>>
> >>> Cheers,
> >>> Varun
> >>>
> >>> --
> >>> http://www.netlab.tkk.fi/~varun/
> >
> > --
> > http://www.netlab.tkk.fi/~varun/



=2D-=20
=2D------------------------------------------------------------------
Dipl.-Ing. Mirja K=FChlewind
Institute of Communication Networks and Computer Engineering (IKR)
University of Stuttgart, Germany
Pfaffenwaldring 47, D-70569 Stuttgart

tel: +49(0)711/685-67973
email: mirja.kuehlewind@ikr.uni-stuttgart.de
web: www.ikr.uni-stuttgart.de
=2D------------------------------------------------------------------

From vsingh.ietf@gmail.com  Mon Mar 25 10:58:09 2013
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From: Varun Singh <vsingh.ietf@gmail.com>
Date: Mon, 25 Mar 2013 19:57:43 +0200
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To: Dan Weber <dan@marketsoup.com>, rmcat WG <rmcat@ietf.org>
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Cc: "Mo Zanaty \(mzanaty\)" <mzanaty@cisco.com>
Subject: Re: [rmcat] Virtual Design Meetings
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Hi Dan,

You can export it from doodle:
http://www.doodle.com/export/ics?optionIndex=9&locale=en&timeZone=Europe%2FHelsinki&adminKey=&pollId=2uci3a27nw37qwk4&participantKey=

Additionally, Mo Zanaty may send and invite and instructions to access
the meeting via Webex.

Cheers,
Varun

On Mon, Mar 25, 2013 at 5:58 PM, Dan Weber <dan@marketsoup.com> wrote:
> Can you send out a meeting invite for the time that is selected?
>
> Thanks,
> Dan
>
>
> On Mon, Mar 25, 2013 at 9:55 AM, Varun Singh <vsingh.ietf@gmail.com> wrote:
>>
>> Hi all,
>>
>> I chose the second of the two sessions on Thursday,
>> it had the best participation.
>> Of the tentatives: I will make it, Mirja what about you?
>>
>>
>> Cheers,
>> Varun
>>
>> On Fri, Mar 22, 2013 at 8:13 PM, Mo Zanaty (mzanaty) <mzanaty@cisco.com>
>> wrote:
>> > Warning: Doodle appears to have a timezone bug. In the timezone
>> > selection box, it thinks the US is still on standard/winter time, but we
>> > already switched to daylight/summer time. After selecting a timezone, for
>> > example EST (GMT-5), the times it shows as EST (GMT-5) are incorrect and are
>> > actually EDT (GMT-4).
>> >
>> > I assume the meeting timeslots are GMT 15:00-16:00 and 16:00-17:00. That
>> > would be EDT (GMT-4) 11:00-12:00 and 12:00-13:00. But Doodle shows EST
>> > (GMT-5) 11:00-1200 and 12:00-13:00. So if you believed its timezone label
>> > and manually added an hour to get daylight/summer time, you get the wrong
>> > meeting times. If you ignore the timezone label and just look at the meeting
>> > times, you get the right times.
>> >
>> > Mo
>> >
>> > -----Original Message-----
>> > From: rmcat-bounces@ietf.org [mailto:rmcat-bounces@ietf.org] On Behalf
>> > Of Varun Singh
>> > Sent: Friday, March 22, 2013 5:48 AM
>> > To: Kevin Gross
>> > Cc: rmcat WG
>> > Subject: Re: [rmcat] Virtual Design Meetings
>> >
>> > Hi Kevin,
>> >
>> > I was under the impression there was inclination to have a virtual
>> > meeting
>> > before Easter? and then every 3 weeks.
>> >
>> > From the doodle poll, I am leaning towards timeslots on
>> > Wednesday/Thursday.
>> > I'll leave the poll open until Saturday and inform the chosen time.
>> >
>> > -Varun
>> >
>> > On Mon, Mar 18, 2013 at 7:02 PM, Kevin Gross <kevin.gross@avanw.com>
>> > wrote:
>> >> Is there a plan for digesting what we accomplished in the IETF86 design
>> >> meetings? Right now we just have raw notes. At a minimum, I would
>> >> expect
>> >> revision to one of our existing drafts. I am willing to help but I need
>> >> to
>> >> start by reading those drafts :)
>> >>
>> >> Kevin Gross
>> >> +1-303-447-0517
>> >> Media Network Consultant
>> >> AVA Networks - www.AVAnw.com, www.X192.org
>> >>
>> >>
>> >> On Mon, Mar 18, 2013 at 9:23 AM, Varun Singh <vsingh.ietf@gmail.com>
>> >> wrote:
>> >>>
>> >>> Hi all,
>> >>>
>> >>> To continue making progress on the fairness principles,
>> >>> we will have virtual design meetings every few weeks.
>> >>>
>> >>> At the last design meeting, we discussed possible meeting times
>> >>> (1hr timeslot: 1100-noon, noon-1300 eastern US time seemed
>> >>> like a good fit for most people). However, I am open to expanding the
>> >>> list of available times. Drop me a note and I will add it to the
>> >>> doodle choice list.
>> >>>
>> >>> Here's a doodle poll for the next one:
>> >>> http://www.doodle.com/2uci3a27nw37qwk4
>> >>>
>> >>> Mo Zanaty and Michael Romalho offered to host this via webex (with
>> >>> recording).
>> >>>
>> >>> Cheers,
>> >>> Varun
>> >>>
>> >>> --
>> >>> http://www.netlab.tkk.fi/~varun/
>> >>
>> >>
>> >
>> >
>> >
>> > --
>> > http://www.netlab.tkk.fi/~varun/
>>
>>
>>
>> --
>> http://www.netlab.tkk.fi/~varun/
>
>



-- 
http://www.netlab.tkk.fi/~varun/

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Topic: RMCAT Eval Design Team
Date: Thursday, March 28, 2013
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-------------------------------------------------------
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<div>&nbsp;</div>
<div><font face=3D"Tahoma" size=3D"2"><span style=3D"font-size:10pt;">Topic=
: RMCAT Eval Design Team
<br>

Date: Thursday, March 28, 2013 <br>

Time: 4:00 pm, GMT Time (London, GMT) <br>

Meeting Number: 203 451 310 <br>

Password: rmcat </span></font></div>
<div><font face=3D"Tahoma" size=3D"2"><span style=3D"font-size:10pt;">Host =
Key: 412564 (use this to reclaim host privileges)
<br>

<br>

------------------------------------------------------- <br>

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------------------------------------------------------- <br>

1. Go to <a href=3D"https://cisco.webex.com/ciscosales/j.php?ED=3D221585972=
&amp;UID=3D484666302&amp;PW=3DNNmNhZTJjYTI1&amp;RT=3DMiMyMQ%3D%3D"><font co=
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3. If a password is required, enter the meeting password: rmcat <br>

4. Click &quot;Join&quot;. <br>

5. If the meeting includes a teleconference, follow the instructions that a=
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<br>

<br>

------------------------------------------------------- <br>

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------------------------------------------------------- <br>

To receive a call back, provide your phone number when you join the meeting=
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<br>

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Call-in toll number (US/Canada): &#43;1-408-525-6800 <br>

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<br>

Toll-free dialing restrictions: <a href=3D"http://www.webex.com/pdf/tollfre=
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<br>

Access code:203 451 310 <br>

CCP:&#43;14085256800x203451310# <br>

</span></font></div>
<div>&nbsp;</div>
</span></font>
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From: "Mo Zanaty (mzanaty)" <mzanaty@cisco.com>
To: Varun Singh <vsingh.ietf@gmail.com>, Dan Weber <dan@marketsoup.com>, "rmcat WG" <rmcat@ietf.org>
Thread-Topic: [rmcat] Virtual Design Meetings
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Date: Tue, 26 Mar 2013 20:41:17 +0000
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Subject: Re: [rmcat] Virtual Design Meetings
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I sent a calendar invite for this Thursday at GMT 16:00 using this webex:

https://cisco.webex.com/ciscosales/j.php?ED=3D221585972&UID=3D484666302&PW=
=3DNNmNhZTJjYTI1&RT=3DMiMyMQ%3D%3D



The webex is actually recurring every Thursday if we want to reuse the link=
 for any future meetings (if the timeslot works again for enough folks). It=
 will be recorded (if everyone agrees), and the link to the recording will =
be sent to the list shortly after the meeting. (It will be a different link=
 than above.)



I assume Varun will run the agenda, since it's his draft we're "designing".



Mo



-----Original Message-----
From: rmcat-bounces@ietf.org [mailto:rmcat-bounces@ietf.org] On Behalf Of V=
arun Singh
Sent: Monday, March 25, 2013 1:58 PM
To: Dan Weber; rmcat WG
Cc: Mo Zanaty (mzanaty)
Subject: Re: [rmcat] Virtual Design Meetings



Hi Dan,



You can export it from doodle:

http://www.doodle.com/export/ics?optionIndex=3D9&locale=3Den&timeZone=3DEur=
ope%2FHelsinki&adminKey=3D&pollId=3D2uci3a27nw37qwk4&participantKey=3D



Additionally, Mo Zanaty may send and invite and instructions to access

the meeting via Webex.



Cheers,

Varun



On Mon, Mar 25, 2013 at 5:58 PM, Dan Weber <dan@marketsoup.com<mailto:dan@m=
arketsoup.com>> wrote:

> Can you send out a meeting invite for the time that is selected?

>

> Thanks,

> Dan

>

>

> On Mon, Mar 25, 2013 at 9:55 AM, Varun Singh <vsingh.ietf@gmail.com<mailt=
o:vsingh.ietf@gmail.com>> wrote:

>>

>> Hi all,

>>

>> I chose the second of the two sessions on Thursday,

>> it had the best participation.

>> Of the tentatives: I will make it, Mirja what about you?

>>

>>

>> Cheers,

>> Varun

>>

>> On Fri, Mar 22, 2013 at 8:13 PM, Mo Zanaty (mzanaty) <mzanaty@cisco.com<=
mailto:mzanaty@cisco.com>>

>> wrote:

>> > Warning: Doodle appears to have a timezone bug. In the timezone

>> > selection box, it thinks the US is still on standard/winter time, but =
we

>> > already switched to daylight/summer time. After selecting a timezone, =
for

>> > example EST (GMT-5), the times it shows as EST (GMT-5) are incorrect a=
nd are

>> > actually EDT (GMT-4).

>> >

>> > I assume the meeting timeslots are GMT 15:00-16:00 and 16:00-17:00. Th=
at

>> > would be EDT (GMT-4) 11:00-12:00 and 12:00-13:00. But Doodle shows EST

>> > (GMT-5) 11:00-1200 and 12:00-13:00. So if you believed its timezone la=
bel

>> > and manually added an hour to get daylight/summer time, you get the wr=
ong

>> > meeting times. If you ignore the timezone label and just look at the m=
eeting

>> > times, you get the right times.

>> >

>> > Mo

>> >

>> > -----Original Message-----

>> > From: rmcat-bounces@ietf.org [mailto:rmcat-bounces@ietf.org] On Behalf

>> > Of Varun Singh

>> > Sent: Friday, March 22, 2013 5:48 AM

>> > To: Kevin Gross

>> > Cc: rmcat WG

>> > Subject: Re: [rmcat] Virtual Design Meetings

>> >

>> > Hi Kevin,

>> >

>> > I was under the impression there was inclination to have a virtual

>> > meeting

>> > before Easter? and then every 3 weeks.

>> >

>> > From the doodle poll, I am leaning towards timeslots on

>> > Wednesday/Thursday.

>> > I'll leave the poll open until Saturday and inform the chosen time.

>> >

>> > -Varun

>> >

>> > On Mon, Mar 18, 2013 at 7:02 PM, Kevin Gross <kevin.gross@avanw.com<ma=
ilto:kevin.gross@avanw.com>>

>> > wrote:

>> >> Is there a plan for digesting what we accomplished in the IETF86 desi=
gn

>> >> meetings? Right now we just have raw notes. At a minimum, I would

>> >> expect

>> >> revision to one of our existing drafts. I am willing to help but I ne=
ed

>> >> to

>> >> start by reading those drafts :)

>> >>

>> >> Kevin Gross

>> >> +1-303-447-0517

>> >> Media Network Consultant

>> >> AVA Networks - www.AVAnw.com<http://www.AVAnw.com>, www.X192.org<http=
://www.X192.org>

>> >>

>> >>

>> >> On Mon, Mar 18, 2013 at 9:23 AM, Varun Singh <vsingh.ietf@gmail.com<m=
ailto:vsingh.ietf@gmail.com>>

>> >> wrote:

>> >>>

>> >>> Hi all,

>> >>>

>> >>> To continue making progress on the fairness principles,

>> >>> we will have virtual design meetings every few weeks.

>> >>>

>> >>> At the last design meeting, we discussed possible meeting times

>> >>> (1hr timeslot: 1100-noon, noon-1300 eastern US time seemed

>> >>> like a good fit for most people). However, I am open to expanding th=
e

>> >>> list of available times. Drop me a note and I will add it to the

>> >>> doodle choice list.

>> >>>

>> >>> Here's a doodle poll for the next one:

>> >>> http://www.doodle.com/2uci3a27nw37qwk4

>> >>>

>> >>> Mo Zanaty and Michael Romalho offered to host this via webex (with

>> >>> recording).

>> >>>

>> >>> Cheers,

>> >>> Varun

>> >>>

>> >>> --

>> >>> http://www.netlab.tkk.fi/~varun/

>> >>

>> >>

>> >

>> >

>> >

>> > --

>> > http://www.netlab.tkk.fi/~varun/

>>

>>

>>

>> --

>> http://www.netlab.tkk.fi/~varun/

>

>







--

http://www.netlab.tkk.fi/~varun/

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<div class=3D"WordSection1">
<p class=3D"MsoPlainText">I sent a calendar invite for this Thursday at GMT=
 16:00 using this webex:<o:p></o:p></p>
<p class=3D"MsoPlainText"><a href=3D"https://cisco.webex.com/ciscosales/j.p=
hp?ED=3D221585972&amp;UID=3D484666302&amp;PW=3DNNmNhZTJjYTI1&amp;RT=3DMiMyM=
Q%3D%3D">https://cisco.webex.com/ciscosales/j.php?ED=3D221585972&amp;UID=3D=
484666302&amp;PW=3DNNmNhZTJjYTI1&amp;RT=3DMiMyMQ%3D%3D</a><o:p></o:p></p>
<p class=3D"MsoPlainText"><o:p>&nbsp;</o:p></p>
<p class=3D"MsoPlainText">The webex is actually recurring every Thursday if=
 we want to reuse the link for any future meetings (if the timeslot works a=
gain for enough folks). It will be recorded (if everyone agrees), and the l=
ink to the recording will be sent
 to the list shortly after the meeting. (It will be a different link than a=
bove.)<o:p></o:p></p>
<p class=3D"MsoPlainText"><o:p>&nbsp;</o:p></p>
<p class=3D"MsoPlainText">I assume Varun will run the agenda, since it&#821=
7;s his draft we&#8217;re &#8220;designing&#8221;.<o:p></o:p></p>
<p class=3D"MsoPlainText"><o:p>&nbsp;</o:p></p>
<p class=3D"MsoPlainText">Mo<o:p></o:p></p>
<p class=3D"MsoPlainText"><o:p>&nbsp;</o:p></p>
<p class=3D"MsoPlainText">-----Original Message-----<br>
From: rmcat-bounces@ietf.org [mailto:rmcat-bounces@ietf.org] On Behalf Of V=
arun Singh<br>
Sent: Monday, March 25, 2013 1:58 PM<br>
To: Dan Weber; rmcat WG<br>
Cc: Mo Zanaty (mzanaty)<br>
Subject: Re: [rmcat] Virtual Design Meetings</p>
<p class=3D"MsoPlainText"><o:p>&nbsp;</o:p></p>
<p class=3D"MsoPlainText">Hi Dan,<o:p></o:p></p>
<p class=3D"MsoPlainText"><o:p>&nbsp;</o:p></p>
<p class=3D"MsoPlainText">You can export it from doodle:<o:p></o:p></p>
<p class=3D"MsoPlainText"><a href=3D"http://www.doodle.com/export/ics?optio=
nIndex=3D9&amp;locale=3Den&amp;timeZone=3DEurope%2FHelsinki&amp;adminKey=3D=
&amp;pollId=3D2uci3a27nw37qwk4&amp;participantKey"><span style=3D"color:win=
dowtext;text-decoration:none">http://www.doodle.com/export/ics?optionIndex=
=3D9&amp;locale=3Den&amp;timeZone=3DEurope%2FHelsinki&amp;adminKey=3D&amp;p=
ollId=3D2uci3a27nw37qwk4&amp;participantKey</span></a>=3D<o:p></o:p></p>
<p class=3D"MsoPlainText"><o:p>&nbsp;</o:p></p>
<p class=3D"MsoPlainText">Additionally, Mo Zanaty may send and invite and i=
nstructions to access<o:p></o:p></p>
<p class=3D"MsoPlainText">the meeting via Webex.<o:p></o:p></p>
<p class=3D"MsoPlainText"><o:p>&nbsp;</o:p></p>
<p class=3D"MsoPlainText">Cheers,<o:p></o:p></p>
<p class=3D"MsoPlainText">Varun<o:p></o:p></p>
<p class=3D"MsoPlainText"><o:p>&nbsp;</o:p></p>
<p class=3D"MsoPlainText">On Mon, Mar 25, 2013 at 5:58 PM, Dan Weber &lt;<a=
 href=3D"mailto:dan@marketsoup.com"><span style=3D"color:windowtext;text-de=
coration:none">dan@marketsoup.com</span></a>&gt; wrote:<o:p></o:p></p>
<p class=3D"MsoPlainText">&gt; Can you send out a meeting invite for the ti=
me that is selected?<o:p></o:p></p>
<p class=3D"MsoPlainText">&gt;<o:p>&nbsp;</o:p></p>
<p class=3D"MsoPlainText">&gt; Thanks,<o:p></o:p></p>
<p class=3D"MsoPlainText">&gt; Dan<o:p></o:p></p>
<p class=3D"MsoPlainText">&gt;<o:p>&nbsp;</o:p></p>
<p class=3D"MsoPlainText">&gt;<o:p>&nbsp;</o:p></p>
<p class=3D"MsoPlainText">&gt; On Mon, Mar 25, 2013 at 9:55 AM, Varun Singh=
 &lt;<a href=3D"mailto:vsingh.ietf@gmail.com"><span style=3D"color:windowte=
xt;text-decoration:none">vsingh.ietf@gmail.com</span></a>&gt; wrote:<o:p></=
o:p></p>
<p class=3D"MsoPlainText">&gt;&gt;<o:p>&nbsp;</o:p></p>
<p class=3D"MsoPlainText">&gt;&gt; Hi all,<o:p></o:p></p>
<p class=3D"MsoPlainText">&gt;&gt;<o:p>&nbsp;</o:p></p>
<p class=3D"MsoPlainText">&gt;&gt; I chose the second of the two sessions o=
n Thursday,<o:p></o:p></p>
<p class=3D"MsoPlainText">&gt;&gt; it had the best participation.<o:p></o:p=
></p>
<p class=3D"MsoPlainText">&gt;&gt; Of the tentatives: I will make it, Mirja=
 what about you?<o:p></o:p></p>
<p class=3D"MsoPlainText">&gt;&gt;<o:p>&nbsp;</o:p></p>
<p class=3D"MsoPlainText">&gt;&gt;<o:p>&nbsp;</o:p></p>
<p class=3D"MsoPlainText">&gt;&gt; Cheers,<o:p></o:p></p>
<p class=3D"MsoPlainText">&gt;&gt; Varun<o:p></o:p></p>
<p class=3D"MsoPlainText">&gt;&gt;<o:p>&nbsp;</o:p></p>
<p class=3D"MsoPlainText">&gt;&gt; On Fri, Mar 22, 2013 at 8:13 PM, Mo Zana=
ty (mzanaty) &lt;<a href=3D"mailto:mzanaty@cisco.com"><span style=3D"color:=
windowtext;text-decoration:none">mzanaty@cisco.com</span></a>&gt;<o:p></o:p=
></p>
<p class=3D"MsoPlainText">&gt;&gt; wrote:<o:p></o:p></p>
<p class=3D"MsoPlainText">&gt;&gt; &gt; Warning: Doodle appears to have a t=
imezone bug. In the timezone<o:p></o:p></p>
<p class=3D"MsoPlainText">&gt;&gt; &gt; selection box, it thinks the US is =
still on standard/winter time, but we<o:p></o:p></p>
<p class=3D"MsoPlainText">&gt;&gt; &gt; already switched to daylight/summer=
 time. After selecting a timezone, for<o:p></o:p></p>
<p class=3D"MsoPlainText">&gt;&gt; &gt; example EST (GMT-5), the times it s=
hows as EST (GMT-5) are incorrect and are<o:p></o:p></p>
<p class=3D"MsoPlainText">&gt;&gt; &gt; actually EDT (GMT-4).<o:p></o:p></p=
>
<p class=3D"MsoPlainText">&gt;&gt; &gt;<o:p></o:p></p>
<p class=3D"MsoPlainText">&gt;&gt; &gt; I assume the meeting timeslots are =
GMT 15:00-16:00 and 16:00-17:00. That<o:p></o:p></p>
<p class=3D"MsoPlainText">&gt;&gt; &gt; would be EDT (GMT-4) 11:00-12:00 an=
d 12:00-13:00. But Doodle shows EST<o:p></o:p></p>
<p class=3D"MsoPlainText">&gt;&gt; &gt; (GMT-5) 11:00-1200 and 12:00-13:00.=
 So if you believed its timezone label<o:p></o:p></p>
<p class=3D"MsoPlainText">&gt;&gt; &gt; and manually added an hour to get d=
aylight/summer time, you get the wrong<o:p></o:p></p>
<p class=3D"MsoPlainText">&gt;&gt; &gt; meeting times. If you ignore the ti=
mezone label and just look at the meeting<o:p></o:p></p>
<p class=3D"MsoPlainText">&gt;&gt; &gt; times, you get the right times.<o:p=
></o:p></p>
<p class=3D"MsoPlainText">&gt;&gt; &gt;<o:p></o:p></p>
<p class=3D"MsoPlainText">&gt;&gt; &gt; Mo<o:p></o:p></p>
<p class=3D"MsoPlainText">&gt;&gt; &gt;<o:p></o:p></p>
<p class=3D"MsoPlainText">&gt;&gt; &gt; -----Original Message-----<o:p></o:=
p></p>
<p class=3D"MsoPlainText">&gt;&gt; &gt; From: rmcat-bounces@ietf.org [mailt=
o:rmcat-bounces@ietf.org] On Behalf<o:p></o:p></p>
<p class=3D"MsoPlainText">&gt;&gt; &gt; Of Varun Singh<o:p></o:p></p>
<p class=3D"MsoPlainText">&gt;&gt; &gt; Sent: Friday, March 22, 2013 5:48 A=
M<o:p></o:p></p>
<p class=3D"MsoPlainText">&gt;&gt; &gt; To: Kevin Gross<o:p></o:p></p>
<p class=3D"MsoPlainText">&gt;&gt; &gt; Cc: rmcat WG<o:p></o:p></p>
<p class=3D"MsoPlainText">&gt;&gt; &gt; Subject: Re: [rmcat] Virtual Design=
 Meetings<o:p></o:p></p>
<p class=3D"MsoPlainText">&gt;&gt; &gt;<o:p></o:p></p>
<p class=3D"MsoPlainText">&gt;&gt; &gt; Hi Kevin,<o:p></o:p></p>
<p class=3D"MsoPlainText">&gt;&gt; &gt;<o:p></o:p></p>
<p class=3D"MsoPlainText">&gt;&gt; &gt; I was under the impression there wa=
s inclination to have a virtual<o:p></o:p></p>
<p class=3D"MsoPlainText">&gt;&gt; &gt; meeting<o:p></o:p></p>
<p class=3D"MsoPlainText">&gt;&gt; &gt; before Easter? and then every 3 wee=
ks.<o:p></o:p></p>
<p class=3D"MsoPlainText">&gt;&gt; &gt;<o:p></o:p></p>
<p class=3D"MsoPlainText">&gt;&gt; &gt; From the doodle poll, I am leaning =
towards timeslots on<o:p></o:p></p>
<p class=3D"MsoPlainText">&gt;&gt; &gt; Wednesday/Thursday.<o:p></o:p></p>
<p class=3D"MsoPlainText">&gt;&gt; &gt; I'll leave the poll open until Satu=
rday and inform the chosen time.<o:p></o:p></p>
<p class=3D"MsoPlainText">&gt;&gt; &gt;<o:p></o:p></p>
<p class=3D"MsoPlainText">&gt;&gt; &gt; -Varun<o:p></o:p></p>
<p class=3D"MsoPlainText">&gt;&gt; &gt;<o:p></o:p></p>
<p class=3D"MsoPlainText">&gt;&gt; &gt; On Mon, Mar 18, 2013 at 7:02 PM, Ke=
vin Gross &lt;<a href=3D"mailto:kevin.gross@avanw.com"><span style=3D"color=
:windowtext;text-decoration:none">kevin.gross@avanw.com</span></a>&gt;<o:p>=
</o:p></p>
<p class=3D"MsoPlainText">&gt;&gt; &gt; wrote:<o:p></o:p></p>
<p class=3D"MsoPlainText">&gt;&gt; &gt;&gt; Is there a plan for digesting w=
hat we accomplished in the IETF86 design<o:p></o:p></p>
<p class=3D"MsoPlainText">&gt;&gt; &gt;&gt; meetings? Right now we just hav=
e raw notes. At a minimum, I would<o:p></o:p></p>
<p class=3D"MsoPlainText">&gt;&gt; &gt;&gt; expect<o:p></o:p></p>
<p class=3D"MsoPlainText">&gt;&gt; &gt;&gt; revision to one of our existing=
 drafts. I am willing to help but I need<o:p></o:p></p>
<p class=3D"MsoPlainText">&gt;&gt; &gt;&gt; to<o:p></o:p></p>
<p class=3D"MsoPlainText">&gt;&gt; &gt;&gt; start by reading those drafts :=
)<o:p></o:p></p>
<p class=3D"MsoPlainText">&gt;&gt; &gt;&gt;<o:p></o:p></p>
<p class=3D"MsoPlainText">&gt;&gt; &gt;&gt; Kevin Gross<o:p></o:p></p>
<p class=3D"MsoPlainText">&gt;&gt; &gt;&gt; &#43;1-303-447-0517<o:p></o:p><=
/p>
<p class=3D"MsoPlainText">&gt;&gt; &gt;&gt; Media Network Consultant<o:p></=
o:p></p>
<p class=3D"MsoPlainText">&gt;&gt; &gt;&gt; AVA Networks - <a href=3D"http:=
//www.AVAnw.com"><span style=3D"color:windowtext;text-decoration:none">www.=
AVAnw.com</span></a>,
<a href=3D"http://www.X192.org"><span style=3D"color:windowtext;text-decora=
tion:none">www.X192.org</span></a><o:p></o:p></p>
<p class=3D"MsoPlainText">&gt;&gt; &gt;&gt;<o:p></o:p></p>
<p class=3D"MsoPlainText">&gt;&gt; &gt;&gt;<o:p></o:p></p>
<p class=3D"MsoPlainText">&gt;&gt; &gt;&gt; On Mon, Mar 18, 2013 at 9:23 AM=
, Varun Singh &lt;<a href=3D"mailto:vsingh.ietf@gmail.com"><span style=3D"c=
olor:windowtext;text-decoration:none">vsingh.ietf@gmail.com</span></a>&gt;<=
o:p></o:p></p>
<p class=3D"MsoPlainText">&gt;&gt; &gt;&gt; wrote:<o:p></o:p></p>
<p class=3D"MsoPlainText">&gt;&gt; &gt;&gt;&gt;<o:p></o:p></p>
<p class=3D"MsoPlainText">&gt;&gt; &gt;&gt;&gt; Hi all,<o:p></o:p></p>
<p class=3D"MsoPlainText">&gt;&gt; &gt;&gt;&gt;<o:p></o:p></p>
<p class=3D"MsoPlainText">&gt;&gt; &gt;&gt;&gt; To continue making progress=
 on the fairness principles,<o:p></o:p></p>
<p class=3D"MsoPlainText">&gt;&gt; &gt;&gt;&gt; we will have virtual design=
 meetings every few weeks.<o:p></o:p></p>
<p class=3D"MsoPlainText">&gt;&gt; &gt;&gt;&gt;<o:p></o:p></p>
<p class=3D"MsoPlainText">&gt;&gt; &gt;&gt;&gt; At the last design meeting,=
 we discussed possible meeting times<o:p></o:p></p>
<p class=3D"MsoPlainText">&gt;&gt; &gt;&gt;&gt; (1hr timeslot: 1100-noon, n=
oon-1300 eastern US time seemed<o:p></o:p></p>
<p class=3D"MsoPlainText">&gt;&gt; &gt;&gt;&gt; like a good fit for most pe=
ople). However, I am open to expanding the<o:p></o:p></p>
<p class=3D"MsoPlainText">&gt;&gt; &gt;&gt;&gt; list of available times. Dr=
op me a note and I will add it to the<o:p></o:p></p>
<p class=3D"MsoPlainText">&gt;&gt; &gt;&gt;&gt; doodle choice list.<o:p></o=
:p></p>
<p class=3D"MsoPlainText">&gt;&gt; &gt;&gt;&gt;<o:p></o:p></p>
<p class=3D"MsoPlainText">&gt;&gt; &gt;&gt;&gt; Here's a doodle poll for th=
e next one:<o:p></o:p></p>
<p class=3D"MsoPlainText">&gt;&gt; &gt;&gt;&gt; <a href=3D"http://www.doodl=
e.com/2uci3a27nw37qwk4"><span style=3D"color:windowtext;text-decoration:non=
e">http://www.doodle.com/2uci3a27nw37qwk4</span></a><o:p></o:p></p>
<p class=3D"MsoPlainText">&gt;&gt; &gt;&gt;&gt;<o:p></o:p></p>
<p class=3D"MsoPlainText">&gt;&gt; &gt;&gt;&gt; Mo Zanaty and Michael Romal=
ho offered to host this via webex (with<o:p></o:p></p>
<p class=3D"MsoPlainText">&gt;&gt; &gt;&gt;&gt; recording).<o:p></o:p></p>
<p class=3D"MsoPlainText">&gt;&gt; &gt;&gt;&gt;<o:p></o:p></p>
<p class=3D"MsoPlainText">&gt;&gt; &gt;&gt;&gt; Cheers,<o:p></o:p></p>
<p class=3D"MsoPlainText">&gt;&gt; &gt;&gt;&gt; Varun<o:p></o:p></p>
<p class=3D"MsoPlainText">&gt;&gt; &gt;&gt;&gt;<o:p></o:p></p>
<p class=3D"MsoPlainText">&gt;&gt; &gt;&gt;&gt; --<o:p></o:p></p>
<p class=3D"MsoPlainText">&gt;&gt; &gt;&gt;&gt; <a href=3D"http://www.netla=
b.tkk.fi/~varun/"><span style=3D"color:windowtext;text-decoration:none">htt=
p://www.netlab.tkk.fi/~varun/</span></a><o:p></o:p></p>
<p class=3D"MsoPlainText">&gt;&gt; &gt;&gt;<o:p></o:p></p>
<p class=3D"MsoPlainText">&gt;&gt; &gt;&gt;<o:p></o:p></p>
<p class=3D"MsoPlainText">&gt;&gt; &gt;<o:p></o:p></p>
<p class=3D"MsoPlainText">&gt;&gt; &gt;<o:p></o:p></p>
<p class=3D"MsoPlainText">&gt;&gt; &gt;<o:p></o:p></p>
<p class=3D"MsoPlainText">&gt;&gt; &gt; --<o:p></o:p></p>
<p class=3D"MsoPlainText">&gt;&gt; &gt; <a href=3D"http://www.netlab.tkk.fi=
/~varun/"><span style=3D"color:windowtext;text-decoration:none">http://www.=
netlab.tkk.fi/~varun/</span></a><o:p></o:p></p>
<p class=3D"MsoPlainText">&gt;&gt;<o:p>&nbsp;</o:p></p>
<p class=3D"MsoPlainText">&gt;&gt;<o:p>&nbsp;</o:p></p>
<p class=3D"MsoPlainText">&gt;&gt;<o:p>&nbsp;</o:p></p>
<p class=3D"MsoPlainText">&gt;&gt; --<o:p></o:p></p>
<p class=3D"MsoPlainText">&gt;&gt; <a href=3D"http://www.netlab.tkk.fi/~var=
un/"><span style=3D"color:windowtext;text-decoration:none">http://www.netla=
b.tkk.fi/~varun/</span></a><o:p></o:p></p>
<p class=3D"MsoPlainText">&gt;<o:p>&nbsp;</o:p></p>
<p class=3D"MsoPlainText">&gt;<o:p>&nbsp;</o:p></p>
<p class=3D"MsoPlainText"><o:p>&nbsp;</o:p></p>
<p class=3D"MsoPlainText"><o:p>&nbsp;</o:p></p>
<p class=3D"MsoPlainText"><o:p>&nbsp;</o:p></p>
<p class=3D"MsoPlainText">-- <o:p></o:p></p>
<p class=3D"MsoPlainText"><a href=3D"http://www.netlab.tkk.fi/~varun/"><spa=
n style=3D"color:windowtext;text-decoration:none">http://www.netlab.tkk.fi/=
~varun/</span></a><o:p></o:p></p>
</div>
</body>
</html>

--_000_3879D71E758A7E4AA99A35DD8D41D3D90F69733Cxmbrcdx14ciscoc_--

From lars@netapp.com  Wed Mar 27 03:50:25 2013
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From: "Eggert, Lars" <lars@netapp.com>
To: Spencer Dawkins <spencer@wonderhamster.org>
Thread-Topic: [rmcat] My notes from today's RMCAT session
Thread-Index: AQHOHtjBZzXXPJdWkkOvDuGjSgLYDpi56WaA
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Subject: Re: [rmcat] My notes from today's RMCAT session
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Hi,

THANK YOU!

Note that I've also gotten two more sets of notes. While this is awesome, m=
erging them is a pain. In the future, could I ask that any additional minut=
e takers sync up with the main stuckee to e.g. collaboratively minute in et=
herpad, so we end up with one set of complete notes instead of three sets o=
f rough ones?

Lars

On Mar 12, 2013, at 5:18, Spencer Dawkins <spencer@wonderhamster.org> wrote=
:

> Lars asked me to take notes at today's session; I'm sending my notes to t=
he list so when I've horribly misunderstood or misquoted you, you can clue =
me in ...
>=20
> Thanks,
>=20
> Spencer
>=20
> 2013-03 IETF 86 - RMCAT
> Lars is chairing this meeting solo ...
> Note Well - thank you, Ericsson for disclosing IPR on a Google draft
>=20
> =3D=3D Chair Intro
> We're running late on Requirements, we have one draft to choose from
> We're running late on Evaluation Criteria and group-cc
> We aren't sure we need to do rtcp-requirements and application interactio=
ns milestones
> We're going to talk with AD about milestones - proposing 1 year slip for =
submission to IESG milestones
> Nothing happened on the mailing list over winter holidays, picking up a l=
ittle speed now
>=20
> No agenda bashing
>=20
> =3D=3D Requirements - Randall Jesup
> Draft was discussed in Atlanta, revised as -01
> Reviewing issues we're discussing and haven't closed on
> Definition of "fair"
> - we don't have one, and there's not a single definition
> - doesn't have to be 1x TCP
> - matters more against longer flows
> - matters most against other low-delay flows
> - can't collapse, can't take too long for a new flow to become usable
> - needs to be bounded if we can't define it
> - needs to cover both local fairness and aggregate fairness at nodes deep=
er in the net
>=20
> Other issues
> - "minimum bandwidth" can be application-defined.
> - need to think about oscillation and possible "mode-switching" between "=
fair" and "aggressive"
>=20
> Q and A
> - anything we can do to bound the problem will help
> - we know what UNfairness is. Can we define tests for unfairness?
> - can we describe principles for identifying unfairness?
> - Lars: this is the biggest issue facing the group, need to make progress=
 this week
> - can we agree on a list of priorities?
> - how self-friendly is existing non-real time traffic?
> - how self-friendly can we be with no other traffic, dedicated nets?
> - need to think about what else we're competing against -RTCWeb data chan=
nels
> - Matt: one definition is impossible, think about what matters
> - Lars: we need to form a design team this week
> - is friendliness part of fairness?
> - can we come up with a test plan to check for fairness?
> - Lars: remember that bad congestion control makes it bad for everyone
> - Colin: decide whether competing with TCP is a short-term goal
> - Randall: especially competing against short-term TCP flows
> - Pat: need to notice the effect of fairness constraints on other charact=
eristics
> Lars: this may be one of the hard problems in computer science, but we're=
 going to solve it before Berlin :-)
>=20
> =3D=3D Evaluating Congestion Control - Varuna Singh
> Now at -02
> - Open issue: two metrics under discussion, other metrics seem ok
> - ideally, we do a distribution and then think about what "discard rate" =
is
> - needs to be more than one number
> - we can do things during evaluation we can't do in deployments
>=20
> We've identified 8 evaluation guidelines
> - do we need a minimum set?
> - are the identified guidelines sufficient? Have proposed 3 more
> - Lars: if you are proposing a solution you need to look at proposal
> - have included a proposal for various evaluation scenario parameters
> - also want to limit combinatorial explosion of test scenarios
> - do we use media packet generators or real video streams?
> - video ends up trying to keep quality consistent, not bit rate
> - Randall: have to know where the edges are - not one parameter set
> - Harald: need to be able to show we aren't ruining video experience
> - Gorry: what happens if your delay exceeds 300ms? Need to separate out t=
he transport aspects and focus on them
> - we need to start someplace - pick scenarios, parameters
> - need to use simple and consistent traffic stream
> - would like to see continuity from requirements draft to this draft
> - need to characterize requirements for various media types
> - need to look at multiple network types - wired, WiFi, cellular ...
> - Magnus: video is so changeable - that's why it's important to look at v=
arious media sources
>=20
> Evaluation Scenarios
> - fixed link capacity, variable link capacity, self-awareness
> - do all scenarios use the same parameters?
> - Matt: assuming single queue or multiple? Need to make that explicit
> - Lars: want some scenarios that do it each way
>=20
> =3D=3D Working Group Hums
> Adopt requirements draft? 20 hands yes, no nos
> Adopt evaluation draft? 10 hands yes, 4 nos
> What do people want to see in the document to be "yes"? No comments from =
the mikes
>=20
> =3D=3D Coupled Congestion Control - Michael Wetzl
> Controlling groups of flows has been tried in the past
> - why didn't these see deployment?
> - hard to identify flows, hard to implement
>=20
> Flow State Exchange (FSE)
> - Flows register with FSE when they start, stop, change rates
> - could use various algorithms, but competing flows all use the same one
> - not priorities, but proportional share
> - Magnus: need to think about RTCWeb control channels, HTTP, etc
> - need to figure out shared bottleneck detection, still working on this
> - browser could use flow table to self-regulate
> - Dale: if everyone builds the same state table, we can understand more
> - this would have security implications
> - current algorithm has two problems: flow told to use rate that's not wh=
at congestion manager determined, highly asynchronous flows
> - can this be sender-side only? Can do better management if receiver is i=
nvolved
>=20
> (Missed question here)
>=20
> =3D=3D On the use of RTCP feedback for congestion control - Colin Perkins
> Thinking about how existing feedback channel works (with AVPF)
> - Per-packet feedback would be great, but RTCP feedback is much less freq=
uent, and no mechanism to piggyback information
> - per-video frame? 30 fps gives you 0.033 second RTCP control interval
> - assuming compound RTCP, but not cross-reporting
> - per-RTT? Longer than inter-frame interval
> - Hannes: can't adjust sending rates arbitrarily with most codecs, and fr=
equent quality changes cause problems with user experience anyway
> - conclusion - RTCP suitable for per-frame, per-RTT, not per-packet
> - Colin's presentation assumes no RTCP extensions
> - remember you can lose feedback packets, too. Just need stable control s=
ystem
> - don't adjust what we do in RMCAT based on existing codec limitations
> - may need to accommodate massive sending rate changes
> - can we ignore the fact that this is audio/video? Video streams are like=
ly to drive congestion, audio is small and you can end up sending more feed=
back than audio
> - packet rate in wireless networks more significant than bandwidth, need =
piggybacking to reduce that
> - what about sampling? Could give you enough feedback
> - two issues - how often do you sample, latency for feedback to sender
> - do we have any minimum requirements that cover feedback?
> - this isn't easy stuff (and wasn't easy with TCP, either)
>=20
> Colin wrote the draft to generate discussion, not planning to progress
>=20
> =3D=3D NADA - a unified congestion control scheme for real-time media - R=
ong Pan
> Design goals: limit self-inflicted delay, leverage a suite of network fee=
dback mechanisms (loss-based, delay-based, ECN, PCN)
> Sender does linear prediction to accommodate feedback latency
> Simulation of NADA with ECN shows improvement in avoiding queue build
> PCN allows zero standing queue
>=20
> Q and A
> - PCN? Standards are out but haven't been deployed yet
> - Rate-shaping buffer on frame basis? Different frame types - I-frames ca=
n span packets, and buffering them can degrade predictive calculations for =
receiving codec
> - Do you see problems with self-interference after congestion happens? De=
pends on traffic and where you stabilize
> - 3-5 percent loss rates in your simulations are very bad for video traff=
ic
> - If we have a high loss rate, we need FEC or other recovery mechanisms t=
hat come with overhead
> - Have you looked at CoDeL? Not yet ...
> - How does this compete with TCP? We're looking at that
> - We don't know if ECN has benefits when you're already doing delay-based=
 mechanisms
> - Should evaluation criteria look at both loss and delay? Probably not, w=
e're evaluating the results of the algorithms - but we need to look at late=
ncy and loss as part of the results
> - If you cause other people 5 percent loss, that's worse than causing you=
rself 5 percent loss that you can recover from


From mirja.kuehlewind@ikr.uni-stuttgart.de  Thu Mar 28 06:50:21 2013
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From: Mirja =?iso-8859-1?q?K=FChlewind?= <mirja.kuehlewind@ikr.uni-stuttgart.de>
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Cc: "Eggert, Lars" <lars@netapp.com>
Subject: [rmcat] minutes IETF86
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Hi,

as we received three different minutes (thanks!), I tried to merge them:

http://etherpad.tools.ietf.org:9007/p/notes-ietf-86-rmcat?useMonospaceFont=true

If you have said something on the mic, please check the minutes and change if 
there is anything wrong!

Mirja


From dan@marketsoup.com  Thu Mar 28 08:06:18 2013
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Subject: Re: [rmcat] minutes IETF86
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--20cf301b643111d69f04d8fd7f5d
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This is really good.

Thanks for putting this together.

Dan

On Thu, Mar 28, 2013 at 7:50 AM, Mirja K=C3=BChlewind <
mirja.kuehlewind@ikr.uni-stuttgart.de> wrote:

> Hi,
>
> as we received three different minutes (thanks!), I tried to merge them:
>
>
> http://etherpad.tools.ietf.org:9007/p/notes-ietf-86-rmcat?useMonospaceFon=
t=3Dtrue
>
> If you have said something on the mic, please check the minutes and chang=
e
> if
> there is anything wrong!
>
> Mirja
>
>

--20cf301b643111d69f04d8fd7f5d
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This is really good.<div><br></div><div>Thanks for putting this together.</=
div><div><br></div><div>Dan<br><br><div class=3D"gmail_quote">On Thu, Mar 2=
8, 2013 at 7:50 AM, Mirja K=C3=BChlewind <span dir=3D"ltr">&lt;<a href=3D"m=
ailto:mirja.kuehlewind@ikr.uni-stuttgart.de" target=3D"_blank">mirja.kuehle=
wind@ikr.uni-stuttgart.de</a>&gt;</span> wrote:<br>
<blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1p=
x #ccc solid;padding-left:1ex">Hi,<br>
<br>
as we received three different minutes (thanks!), I tried to merge them:<br=
>
<br>
<a href=3D"http://etherpad.tools.ietf.org:9007/p/notes-ietf-86-rmcat?useMon=
ospaceFont=3Dtrue" target=3D"_blank">http://etherpad.tools.ietf.org:9007/p/=
notes-ietf-86-rmcat?useMonospaceFont=3Dtrue</a><br>
<br>
If you have said something on the mic, please check the minutes and change =
if<br>
there is anything wrong!<br>
<span class=3D"HOEnZb"><font color=3D"#888888"><br>
Mirja<br>
<br>
</font></span></blockquote></div><br></div>

--20cf301b643111d69f04d8fd7f5d--

From vsingh.ietf@gmail.com  Thu Mar 28 08:34:27 2013
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From: Varun Singh <vsingh.ietf@gmail.com>
Date: Thu, 28 Mar 2013 17:34:00 +0200
Message-ID: <CAEbPqrwTcrHB6MAgFt0H=bNad2c2uDise07=a66-XbK07Cyayg@mail.gmail.com>
To: rmcat WG <rmcat@ietf.org>
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Subject: Re: [rmcat] Virtual Design Meetings
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Hi,

Agenda:
5 mins: Recap of meeting
Rest: How to proceed with the Fairness criteria

At the IETF we discussed
- application scenarios for fairness
- discussion on fairness
- packet logs for flows

What is missing:
- network scenarios
- link characteristics


On Tue, Mar 26, 2013 at 10:41 PM, Mo Zanaty (mzanaty) <mzanaty@cisco.com> w=
rote:
> I sent a calendar invite for this Thursday at GMT 16:00 using this webex:
>
> https://cisco.webex.com/ciscosales/j.php?ED=3D221585972&UID=3D484666302&P=
W=3DNNmNhZTJjYTI1&RT=3DMiMyMQ%3D%3D
>
>
>
> The webex is actually recurring every Thursday if we want to reuse the li=
nk
> for any future meetings (if the timeslot works again for enough folks). I=
t
> will be recorded (if everyone agrees), and the link to the recording will=
 be
> sent to the list shortly after the meeting. (It will be a different link
> than above.)
>
>
>
> I assume Varun will run the agenda, since it=92s his draft we=92re =93des=
igning=94.
>
>
>
> Mo
>
>
>
> -----Original Message-----
> From: rmcat-bounces@ietf.org [mailto:rmcat-bounces@ietf.org] On Behalf Of
> Varun Singh
> Sent: Monday, March 25, 2013 1:58 PM
> To: Dan Weber; rmcat WG
> Cc: Mo Zanaty (mzanaty)
> Subject: Re: [rmcat] Virtual Design Meetings
>
>
>
> Hi Dan,
>
>
>
> You can export it from doodle:
>
> http://www.doodle.com/export/ics?optionIndex=3D9&locale=3Den&timeZone=3DE=
urope%2FHelsinki&adminKey=3D&pollId=3D2uci3a27nw37qwk4&participantKey=3D
>
>
>
> Additionally, Mo Zanaty may send and invite and instructions to access
>
> the meeting via Webex.
>
>
>
> Cheers,
>
> Varun
>
>
>
> On Mon, Mar 25, 2013 at 5:58 PM, Dan Weber <dan@marketsoup.com> wrote:
>
>> Can you send out a meeting invite for the time that is selected?
>
>>
>
>> Thanks,
>
>> Dan
>
>>
>
>>
>
>> On Mon, Mar 25, 2013 at 9:55 AM, Varun Singh <vsingh.ietf@gmail.com>
>> wrote:
>
>>>
>
>>> Hi all,
>
>>>
>
>>> I chose the second of the two sessions on Thursday,
>
>>> it had the best participation.
>
>>> Of the tentatives: I will make it, Mirja what about you?
>
>>>
>
>>>
>
>>> Cheers,
>
>>> Varun
>
>>>
>
>>> On Fri, Mar 22, 2013 at 8:13 PM, Mo Zanaty (mzanaty) <mzanaty@cisco.com=
>
>
>>> wrote:
>
>>> > Warning: Doodle appears to have a timezone bug. In the timezone
>
>>> > selection box, it thinks the US is still on standard/winter time, but
>>> > we
>
>>> > already switched to daylight/summer time. After selecting a timezone,
>>> > for
>
>>> > example EST (GMT-5), the times it shows as EST (GMT-5) are incorrect
>>> > and are
>
>>> > actually EDT (GMT-4).
>
>>> >
>
>>> > I assume the meeting timeslots are GMT 15:00-16:00 and 16:00-17:00.
>>> > That
>
>>> > would be EDT (GMT-4) 11:00-12:00 and 12:00-13:00. But Doodle shows ES=
T
>
>>> > (GMT-5) 11:00-1200 and 12:00-13:00. So if you believed its timezone
>>> > label
>
>>> > and manually added an hour to get daylight/summer time, you get the
>>> > wrong
>
>>> > meeting times. If you ignore the timezone label and just look at the
>>> > meeting
>
>>> > times, you get the right times.
>
>>> >
>
>>> > Mo
>
>>> >
>
>>> > -----Original Message-----
>
>>> > From: rmcat-bounces@ietf.org [mailto:rmcat-bounces@ietf.org] On Behal=
f
>
>>> > Of Varun Singh
>
>>> > Sent: Friday, March 22, 2013 5:48 AM
>
>>> > To: Kevin Gross
>
>>> > Cc: rmcat WG
>
>>> > Subject: Re: [rmcat] Virtual Design Meetings
>
>>> >
>
>>> > Hi Kevin,
>
>>> >
>
>>> > I was under the impression there was inclination to have a virtual
>
>>> > meeting
>
>>> > before Easter? and then every 3 weeks.
>
>>> >
>
>>> > From the doodle poll, I am leaning towards timeslots on
>
>>> > Wednesday/Thursday.
>
>>> > I'll leave the poll open until Saturday and inform the chosen time.
>
>>> >
>
>>> > -Varun
>
>>> >
>
>>> > On Mon, Mar 18, 2013 at 7:02 PM, Kevin Gross <kevin.gross@avanw.com>
>
>>> > wrote:
>
>>> >> Is there a plan for digesting what we accomplished in the IETF86
>>> >> design
>
>>> >> meetings? Right now we just have raw notes. At a minimum, I would
>
>>> >> expect
>
>>> >> revision to one of our existing drafts. I am willing to help but I
>>> >> need
>
>>> >> to
>
>>> >> start by reading those drafts :)
>
>>> >>
>
>>> >> Kevin Gross
>
>>> >> +1-303-447-0517
>
>>> >> Media Network Consultant
>
>>> >> AVA Networks - www.AVAnw.com, www.X192.org
>
>>> >>
>
>>> >>
>
>>> >> On Mon, Mar 18, 2013 at 9:23 AM, Varun Singh <vsingh.ietf@gmail.com>
>
>>> >> wrote:
>
>>> >>>
>
>>> >>> Hi all,
>
>>> >>>
>
>>> >>> To continue making progress on the fairness principles,
>
>>> >>> we will have virtual design meetings every few weeks.
>
>>> >>>
>
>>> >>> At the last design meeting, we discussed possible meeting times
>
>>> >>> (1hr timeslot: 1100-noon, noon-1300 eastern US time seemed
>
>>> >>> like a good fit for most people). However, I am open to expanding t=
he
>
>>> >>> list of available times. Drop me a note and I will add it to the
>
>>> >>> doodle choice list.
>
>>> >>>
>
>>> >>> Here's a doodle poll for the next one:
>
>>> >>> http://www.doodle.com/2uci3a27nw37qwk4
>
>>> >>>
>
>>> >>> Mo Zanaty and Michael Romalho offered to host this via webex (with
>
>>> >>> recording).
>
>>> >>>
>
>>> >>> Cheers,
>
>>> >>> Varun
>
>>> >>>
>
>>> >>> --
>
>>> >>> http://www.netlab.tkk.fi/~varun/
>
>>> >>
>
>>> >>
>
>>> >
>
>>> >
>
>>> >
>
>>> > --
>
>>> > http://www.netlab.tkk.fi/~varun/
>
>>>
>
>>>
>
>>>
>
>>> --
>
>>> http://www.netlab.tkk.fi/~varun/
>
>>
>
>>
>
>
>
>
>
>
>
> --
>
> http://www.netlab.tkk.fi/~varun/



--=20
http://www.netlab.tkk.fi/~varun/

From lars@netapp.com  Thu Mar 28 10:04:10 2013
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From: "Eggert, Lars" <lars@netapp.com>
To: RMCAT Mailing List <rmcat@ietf.org>
Thread-Topic: WG wiki
Thread-Index: AQHOK9UpSYcM+VWUk0K+ZR96+PXSCQ==
Date: Thu, 28 Mar 2013 16:56:16 +0000
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Hi,

since this just came up on the design team call: the WG wiki is at http://t=
rac.tools.ietf.org/wg/rmcat/trac/wiki

You can get a tools.ietf.org password here: https://tools.ietf.org/newlogin

Lars=

From dan@marketsoup.com  Fri Mar 29 13:33:20 2013
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From: Dan Weber <dan@marketsoup.com>
To: rmcat@ietf.org
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Subject: [rmcat] The Definition of Fairness
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As we all are aware, defining fairness presents challenges.  Michael
Ramhalko gives the best metaphor for describing it we know today, "I know
it when I see it".

To even more elaborate on his point, we can't say what is fair by seeing
it, we can only identify what is unfair.  When is something unfair?  Two
situations come to mind.

"That's not fair!"  These are the infamous words children say everyday when
they don't get their way.  Often times it could be some toy that is in use,
or maybe someone gets preferential treatment.  The child *"harmed'" *is
limited from participating in the same way as the other.  A similar case
could be waiting on a list at a busy restaurant. If you've been waiting on
the list for an hour, and someone walks in the door only to know the host
personally and jump ahead to the next available table.

Based on these examples, defining *unfairness* might be "when a participant
seeks out its own self interest at the expense of everyone else".  Because
when that man cuts you in line, he probably does so without acknowledging
your existence.  Merely assuming there are other participants can allow one
to adjust for their presence whether or not if their existence is clear.

Then, the antonym of my unfairness definition or the fairness definition is
the golden rule: "Treat others as you wish to be treated"  If the man who
cut in line then was subject to his own behavior elsewhere, he may become
upset with it as well and change his own behavior.  This is one of the
major issues with bittorrent.  Since bittorrent is so aggressive, everyone
else finds themselves in positions trying to protect their own applications
from its behavior, including competing bittorrents.

In light of this, we come to the question, "Is seeking RTP fairness with
itself enough?"  I say yes.

RTP in its current form exhibits the following behaviors:

   - It can be be short lived (short video clip)
   - It can be long lived (long telephone calls)
   - It can be bursty (i-frames go across the network)
   - It can be constant rate (constant bit rate audio codecs like mulaw)
   - It can be variable rate (video streams in general targeting consistent
   quality with changes in motion or picture)
   - It can be used in ways to maximize throughput (e.g. streaming movies
   at the highest quality a client can receive acceptably -- Hulu or
   equivalent)
   - It can be unresponsive to changes in the network (without any form of
   congestion control, RTP continues to send oblivious to other participants)
   - It competes with itself in the average case (audio + video streams
   from the same sender, multiple video streams for multi party conferencing)
   - It is time sensitive (as in real time media needs to be played back
   without prior to changes in human perception)
   - It assumes that network behavior is unpredictable (existence of RTCP +
   use of things like adaptive jitter buffers)

Let's say we agree on all of these behaviors.  It is safe to assume then
that competing with RTP is a superset of behaviors then it is to compete
with TCP, LEDBAT, or a web browser alone is it not?

Thanks,
Dan

--047d7b3a9ad2152a2804d9162eff
Content-Type: text/html; charset=UTF-8
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As we all are aware, defining fairness presents challenges. =C2=A0Michael R=
amhalko gives the best metaphor for describing it we know today, &quot;I kn=
ow it when I see it&quot;.<div><br></div><div>To even more elaborate on his=
 point, we can&#39;t say what is fair by seeing it, we can only identify wh=
at is unfair. =C2=A0When is something unfair? =C2=A0Two situations come to =
mind.</div>
<div><br></div><div>&quot;That&#39;s not fair!&quot; =C2=A0These are the in=
famous words children say everyday when they don&#39;t get their way. =C2=
=A0Often times it could be some toy that is in use, or maybe someone gets p=
referential treatment. =C2=A0The child <i>&quot;harmed&#39;&quot;=C2=A0</i>=
is limited from participating in the same way as the other. =C2=A0A similar=
 case could be waiting on a list at a busy restaurant. If you&#39;ve been w=
aiting on the list for an hour, and someone walks in the door only to know =
the host personally and jump ahead to the next available table.</div>
<div><br></div><div>Based on these examples, defining <i>unfairness</i>=C2=
=A0might be &quot;when a participant seeks out its own self interest at the=
 expense of everyone else&quot;. =C2=A0Because when that man cuts you in li=
ne, he probably does so without acknowledging your existence. =C2=A0Merely =
assuming there are other participants can allow one to adjust for their pre=
sence whether or not if their existence is clear. =C2=A0</div>
<div><br></div><div>Then, the antonym of my unfairness definition or the fa=
irness definition is the golden rule: &quot;Treat others as you wish to be =
treated&quot; =C2=A0If the man who cut in line then was subject to his own =
behavior elsewhere, he may become upset with it as well and change his own =
behavior. =C2=A0This is one of the major issues with bittorrent. =C2=A0Sinc=
e bittorrent is so aggressive, everyone else finds themselves in positions =
trying to protect their own applications from its behavior, including compe=
ting bittorrents.</div>
<div><br></div><div>In light of this, we come to the question, &quot;Is see=
king RTP fairness with itself enough?&quot; =C2=A0I say yes.</div><div><br>=
</div><div>RTP in its current form exhibits the following behaviors:</div><=
div>
<ul><li>It can be be short lived (short video clip)</li><li>It can be long =
lived (long telephone calls)</li><li>It can be bursty (i-frames go across t=
he network)</li><li>It can be constant rate (constant bit rate audio codecs=
 like mulaw)</li>
<li>It can be variable rate (video streams in general=C2=A0targeting=C2=A0c=
onsistent quality with changes in motion or picture)</li><li>It can be used=
 in ways to maximize throughput (e.g. streaming movies at the highest quali=
ty a client can receive acceptably -- Hulu or equivalent)</li>
<li>It can be unresponsive to changes in the network (without any form of c=
ongestion control, RTP continues to send oblivious to other participants)</=
li><li>It competes with itself in the average case (audio + video streams f=
rom the same sender, multiple video streams for multi party conferencing)</=
li>
<li>It is time sensitive (as in real time media needs to be played back wit=
hout prior to changes in human perception)</li><li>It assumes that network =
behavior is unpredictable (existence of RTCP + use of things like adaptive =
jitter buffers)</li>
</ul></div><div>Let&#39;s say we agree on all of these behaviors. =C2=A0It =
is safe to assume then that competing with RTP is a superset of behaviors t=
hen it is to compete with TCP, LEDBAT, or a web browser alone is it not?</d=
iv>
<div><br></div><div>Thanks,</div><div>Dan</div><div>=C2=A0</div>

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From: "Bill Ver Steeg (versteb)" <versteb@cisco.com>
To: Dan Weber <dan@marketsoup.com>, "rmcat@ietf.org" <rmcat@ietf.org>
Thread-Topic: [rmcat] The Definition of Fairness
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--_000_AE7F97DB5FEE054088D82E836BD15BE91EC0FC8Fxmbalnx05ciscoc_--

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From: Dan Weber <dan@marketsoup.com>
To: "Bill Ver Steeg (versteb)" <versteb@cisco.com>
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Subject: Re: [rmcat] The Definition of Fairness
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Hi Bill,

The prisoner's dilemma is a good example if you want to use the court of
public opinion as a moral compass.  My suggestion is to have merely give
RTP a conscience for its own behavior and not others.  It seeks to treat
itself and others by its own standards.  In a world where land is
plentiful, water flows cleanly, and sunlight is abundant, enough food can
be grown to support everyone's needs.  When battling plague or handling
severe drought, everyone needs the bare minimum for survival.   If that
bare minimum can not be attained while seeking to apply each's own general
principles first, than that man shall perish by the survival of the
fittest.

We have to run on the assumption that everyone will play fair and
reasonably, otherwise, we run the risk of a never ending game.  The US
court system exists to implement fairness understands these problems and
has settled on the premise that everyone acts in good faith unless proven
otherwise.

Thanks,
Dan

On Fri, Mar 29, 2013 at 3:08 PM, Bill Ver Steeg (versteb) <versteb@cisco.co=
m
> wrote:

>  At the risk of starting a philosophical debate, I think we should
> consider some lessons from game theory.****
>
> ** **
>
> The =E2=80=9CPrisoner=E2=80=99s Dilemma=E2=80=9D comes to mind. If an app=
lication senses that the
> programs that it is sharing a link with are =E2=80=9Cfair=E2=80=9D (where=
 fair boils down
> to =E2=80=9Cgive me a reasonable amount of BW and do not excessively dela=
y my
> traffic=E2=80=9D), the logical thing to is to also be fair. When an appli=
cation
> senses that another program is taking up all of the available link BW
> and/or causing delay for its data, that program would be foolish to allow
> the more aggressive competitor to seize all of the link BW. It would, in
> turn, be aggressive in its behavior. There are a few solutions to this
> problem, and I encourage the group to look into this a bit. The winning
> game theory strategies boil down to variations on =E2=80=9Ctit for tat=E2=
=80=9D, in which
> participant starts out being cooperative, but becomes uncooperative when
> peers are seen to be uncooperative. ****
>
> ** **
>
> The =E2=80=9Ctragedy of the commons=E2=80=9D is a similar example.****
>
> ** **
>
> In a perfect world, the network would have a way to influence application
> behaviors towards the desired behavior. Misbehaving applications would
> suffer from the delay penalty, but well behaved applications would not. *=
*
> **
>
> ** **
>
> There are some theoretical solutions that fit the bill. For instance, the
> middlebox could segregate the traffic into a (set of) sub-queue(s) on a p=
er
> QOS bucket basis. Theses sub queues would be designed such that
>  cooperative applications competed with other cooperative applications in=
 a
> (set of) sub queue(s) and aggressive/misbehaving applications competed wi=
th
> other similar traffic. So, if you tend to respond to congestion signals,
> you will be in a (set of) sub-queue(s) with good delay characteristics an=
d
> will get your =E2=80=9Cfair=E2=80=9D share. If you tend to not respond to=
 congestion
> signals, you will be in a (set of) sub-queue(s) with poor delay
> characteristics and the network will bias the BW towards the well behaved
> queue(s).****
>
> ** **
>
> Unfortunately, this behavior is easier to visualize than to implement. It
> is fraught with corner cases, lots of state and tough decisions. I return
> you to  the networking discussions=E2=80=A6=E2=80=A6..****
>
> ** **
>
> ** **
>
> Bvs****
>
> ** **
>
> ** **
>
> *From:* rmcat-bounces@ietf.org [mailto:rmcat-bounces@ietf.org] *On Behalf
> Of *Dan Weber
> *Sent:* Friday, March 29, 2013 4:33 PM
> *To:* rmcat@ietf.org
> *Subject:* [rmcat] The Definition of Fairness****
>
> ** **
>
> As we all are aware, defining fairness presents challenges.  Michael
> Ramhalko gives the best metaphor for describing it we know today, "I know
> it when I see it".****
>
> ** **
>
> To even more elaborate on his point, we can't say what is fair by seeing
> it, we can only identify what is unfair.  When is something unfair?  Two
> situations come to mind.****
>
> ** **
>
> "That's not fair!"  These are the infamous words children say everyday
> when they don't get their way.  Often times it could be some toy that is =
in
> use, or maybe someone gets preferential treatment.  The child *"harmed'" =
*is
> limited from participating in the same way as the other.  A similar case
> could be waiting on a list at a busy restaurant. If you've been waiting o=
n
> the list for an hour, and someone walks in the door only to know the host
> personally and jump ahead to the next available table.****
>
> ** **
>
> Based on these examples, defining *unfairness* might be "when a
> participant seeks out its own self interest at the expense of everyone
> else".  Because when that man cuts you in line, he probably does so witho=
ut
> acknowledging your existence.  Merely assuming there are other participan=
ts
> can allow one to adjust for their presence whether or not if their
> existence is clear.  ****
>
> ** **
>
> Then, the antonym of my unfairness definition or the fairness definition
> is the golden rule: "Treat others as you wish to be treated"  If the man
> who cut in line then was subject to his own behavior elsewhere, he may
> become upset with it as well and change his own behavior.  This is one of
> the major issues with bittorrent.  Since bittorrent is so aggressive,
> everyone else finds themselves in positions trying to protect their own
> applications from its behavior, including competing bittorrents.****
>
> ** **
>
> In light of this, we come to the question, "Is seeking RTP fairness with
> itself enough?"  I say yes.****
>
> ** **
>
> RTP in its current form exhibits the following behaviors:****
>
>    - It can be be short lived (short video clip)****
>    - It can be long lived (long telephone calls)****
>    - It can be bursty (i-frames go across the network)****
>    - It can be constant rate (constant bit rate audio codecs like mulaw)*=
*
>    **
>    - It can be variable rate (video streams in
>    general targeting consistent quality with changes in motion or picture=
)
>    ****
>    - It can be used in ways to maximize throughput (e.g. streaming movies
>    at the highest quality a client can receive acceptably -- Hulu or
>    equivalent)****
>    - It can be unresponsive to changes in the network (without any form
>    of congestion control, RTP continues to send oblivious to other
>    participants)****
>    - It competes with itself in the average case (audio + video streams
>    from the same sender, multiple video streams for multi party conferenc=
ing)
>    ****
>    - It is time sensitive (as in real time media needs to be played back
>    without prior to changes in human perception)****
>    - It assumes that network behavior is unpredictable (existence of RTCP
>    + use of things like adaptive jitter buffers)****
>
>  Let's say we agree on all of these behaviors.  It is safe to assume then
> that competing with RTP is a superset of behaviors then it is to compete
> with TCP, LEDBAT, or a web browser alone is it not?****
>
> ** **
>
> Thanks,****
>
> Dan****
>
>  ****
>

--047d7bd74a227743d404d9170207
Content-Type: text/html; charset=UTF-8
Content-Transfer-Encoding: quoted-printable

Hi Bill,<div><br></div><div>The prisoner&#39;s dilemma is a good example if=
 you want to use the court of public opinion as a moral compass. =C2=A0My s=
uggestion is to have merely give RTP a conscience for its own behavior and =
not others. =C2=A0It seeks to treat itself and others by its own standards.=
 =C2=A0In a world where land is plentiful, water flows cleanly, and sunligh=
t is abundant, enough food can be grown to support everyone&#39;s needs. =
=C2=A0When battling plague or handling severe drought, everyone needs the b=
are minimum for survival. =C2=A0 If that bare minimum can not be attained w=
hile seeking to apply each&#39;s own general principles first, than that ma=
n shall perish by the survival of the fittest. =C2=A0</div>
<div><br></div><div>We have to run on the assumption that everyone will pla=
y fair and reasonably, otherwise, we run the risk of a never ending game. =
=C2=A0The US court system exists to implement fairness understands these pr=
oblems and has settled on the premise that everyone acts in good faith unle=
ss proven otherwise.</div>
<div><br></div><div>Thanks,</div><div>Dan</div><div><br><div class=3D"gmail=
_quote">On Fri, Mar 29, 2013 at 3:08 PM, Bill Ver Steeg (versteb) <span dir=
=3D"ltr">&lt;<a href=3D"mailto:versteb@cisco.com" target=3D"_blank">versteb=
@cisco.com</a>&gt;</span> wrote:<br>
<blockquote class=3D"gmail_quote" style=3D"margin:0 0 0 .8ex;border-left:1p=
x #ccc solid;padding-left:1ex">





<div lang=3D"EN-US" link=3D"blue" vlink=3D"purple">
<div>
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;;color:#1f497d">At the risk of starting a=
 philosophical debate, I think we should consider some lessons from game th=
eory.<u></u><u></u></span></p>

<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;;color:#1f497d"><u></u>=C2=A0<u></u></spa=
n></p>
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;;color:#1f497d">The =E2=80=9CPrisoner=E2=
=80=99s Dilemma=E2=80=9D comes to mind. If an application senses that the p=
rograms that it is sharing a link with are =E2=80=9Cfair=E2=80=9D (where fa=
ir boils down to =E2=80=9Cgive
 me a reasonable amount of BW and do not excessively delay my traffic=E2=80=
=9D), the logical thing to is to also be fair. When an application senses t=
hat another program is taking up all of the available link BW and/or causin=
g delay for its data, that program would
 be foolish to allow the more aggressive competitor to seize all of the lin=
k BW. It would, in turn, be aggressive in its behavior. There are a few sol=
utions to this problem, and I encourage the group to look into this a bit. =
The winning game theory strategies
 boil down to variations on =E2=80=9Ctit for tat=E2=80=9D, in which partici=
pant starts out being cooperative, but becomes uncooperative when peers are=
 seen to be uncooperative.
<u></u><u></u></span></p>
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;;color:#1f497d"><u></u>=C2=A0<u></u></spa=
n></p>
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;;color:#1f497d">The =E2=80=9Ctragedy of t=
he commons=E2=80=9D is a similar example.<u></u><u></u></span></p>
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;;color:#1f497d"><u></u>=C2=A0<u></u></spa=
n></p>
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;;color:#1f497d">In a perfect world, the n=
etwork would have a way to influence application behaviors towards the desi=
red behavior. Misbehaving applications would suffer from
 the delay penalty, but well behaved applications would not. <u></u><u></u>=
</span></p>
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;;color:#1f497d"><u></u>=C2=A0<u></u></spa=
n></p>
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;;color:#1f497d">There are some theoretica=
l solutions that fit the bill. For instance, the middlebox could segregate =
the traffic into a (set of) sub-queue(s) on a per QOS bucket
 basis. Theses sub queues would be designed such that =C2=A0cooperative app=
lications competed with other cooperative applications in a (set of) sub qu=
eue(s) and aggressive/misbehaving applications competed with other similar =
traffic. So, if you tend to respond to
 congestion signals, you will be in a (set of) sub-queue(s) with good delay=
 characteristics and will get your =E2=80=9Cfair=E2=80=9D share. If you ten=
d to not respond to congestion signals, you will be in a (set of) sub-queue=
(s) with poor delay characteristics and the network
 will bias the BW towards the well behaved queue(s).<u></u><u></u></span></=
p>
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;;color:#1f497d"><u></u>=C2=A0<u></u></spa=
n></p>
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;;color:#1f497d">Unfortunately, this behav=
ior is easier to visualize than to implement. It is fraught with corner cas=
es, lots of state and tough decisions. I return you to =C2=A0the
 networking discussions=E2=80=A6=E2=80=A6..<u></u><u></u></span></p>
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;;color:#1f497d"><u></u>=C2=A0<u></u></spa=
n></p>
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;;color:#1f497d"><u></u>=C2=A0<u></u></spa=
n></p>
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;;color:#1f497d">Bvs<u></u><u></u></span><=
/p>
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;;color:#1f497d"><u></u>=C2=A0<u></u></spa=
n></p>
<p class=3D"MsoNormal"><span style=3D"font-size:11.0pt;font-family:&quot;Ca=
libri&quot;,&quot;sans-serif&quot;;color:#1f497d"><u></u>=C2=A0<u></u></spa=
n></p>
<p class=3D"MsoNormal"><b><span style=3D"font-size:10.0pt;font-family:&quot=
;Tahoma&quot;,&quot;sans-serif&quot;">From:</span></b><span style=3D"font-s=
ize:10.0pt;font-family:&quot;Tahoma&quot;,&quot;sans-serif&quot;"> <a href=
=3D"mailto:rmcat-bounces@ietf.org" target=3D"_blank">rmcat-bounces@ietf.org=
</a> [mailto:<a href=3D"mailto:rmcat-bounces@ietf.org" target=3D"_blank">rm=
cat-bounces@ietf.org</a>]
<b>On Behalf Of </b>Dan Weber<br>
<b>Sent:</b> Friday, March 29, 2013 4:33 PM<br>
<b>To:</b> <a href=3D"mailto:rmcat@ietf.org" target=3D"_blank">rmcat@ietf.o=
rg</a><br>
<b>Subject:</b> [rmcat] The Definition of Fairness<u></u><u></u></span></p>=
<div><div class=3D"h5">
<p class=3D"MsoNormal"><u></u>=C2=A0<u></u></p>
<p class=3D"MsoNormal">As we all are aware, defining fairness presents chal=
lenges. =C2=A0Michael Ramhalko gives the best metaphor for describing it we=
 know today, &quot;I know it when I see it&quot;.<u></u><u></u></p>
<div>
<p class=3D"MsoNormal"><u></u>=C2=A0<u></u></p>
</div>
<div>
<p class=3D"MsoNormal">To even more elaborate on his point, we can&#39;t sa=
y what is fair by seeing it, we can only identify what is unfair. =C2=A0Whe=
n is something unfair? =C2=A0Two situations come to mind.<u></u><u></u></p>
</div>
<div>
<p class=3D"MsoNormal"><u></u>=C2=A0<u></u></p>
</div>
<div>
<p class=3D"MsoNormal">&quot;That&#39;s not fair!&quot; =C2=A0These are the=
 infamous words children say everyday when they don&#39;t get their way. =
=C2=A0Often times it could be some toy that is in use, or maybe someone get=
s preferential treatment. =C2=A0The child
<i>&quot;harmed&#39;&quot;=C2=A0</i>is limited from participating in the sa=
me way as the other. =C2=A0A similar case could be waiting on a list at a b=
usy restaurant. If you&#39;ve been waiting on the list for an hour, and som=
eone walks in the door only to know the host personally and
 jump ahead to the next available table.<u></u><u></u></p>
</div>
<div>
<p class=3D"MsoNormal"><u></u>=C2=A0<u></u></p>
</div>
<div>
<p class=3D"MsoNormal">Based on these examples, defining <i>unfairness</i>=
=C2=A0might be &quot;when a participant seeks out its own self interest at =
the expense of everyone else&quot;. =C2=A0Because when that man cuts you in=
 line, he probably does so without acknowledging your existence.
 =C2=A0Merely assuming there are other participants can allow one to adjust=
 for their presence whether or not if their existence is clear. =C2=A0<u></=
u><u></u></p>
</div>
<div>
<p class=3D"MsoNormal"><u></u>=C2=A0<u></u></p>
</div>
<div>
<p class=3D"MsoNormal">Then, the antonym of my unfairness definition or the=
 fairness definition is the golden rule: &quot;Treat others as you wish to =
be treated&quot; =C2=A0If the man who cut in line then was subject to his o=
wn behavior elsewhere, he may become upset with it
 as well and change his own behavior. =C2=A0This is one of the major issues=
 with bittorrent. =C2=A0Since bittorrent is so aggressive, everyone else fi=
nds themselves in positions trying to protect their own applications from i=
ts behavior, including competing bittorrents.<u></u><u></u></p>

</div>
<div>
<p class=3D"MsoNormal"><u></u>=C2=A0<u></u></p>
</div>
<div>
<p class=3D"MsoNormal">In light of this, we come to the question, &quot;Is =
seeking RTP fairness with itself enough?&quot; =C2=A0I say yes.<u></u><u></=
u></p>
</div>
<div>
<p class=3D"MsoNormal"><u></u>=C2=A0<u></u></p>
</div>
<div>
<p class=3D"MsoNormal">RTP in its current form exhibits the following behav=
iors:<u></u><u></u></p>
</div>
<div>
<ul type=3D"disc">
<li class=3D"MsoNormal">
It can be be short lived (short video clip)<u></u><u></u></li><li class=3D"=
MsoNormal">
It can be long lived (long telephone calls)<u></u><u></u></li><li class=3D"=
MsoNormal">
It can be bursty (i-frames go across the network)<u></u><u></u></li><li cla=
ss=3D"MsoNormal">
It can be constant rate (constant bit rate audio codecs like mulaw)<u></u><=
u></u></li><li class=3D"MsoNormal">
It can be variable rate (video streams in general=C2=A0targeting=C2=A0consi=
stent quality with changes in motion or picture)<u></u><u></u></li><li clas=
s=3D"MsoNormal">
It can be used in ways to maximize throughput (e.g. streaming movies at the=
 highest quality a client can receive acceptably -- Hulu or equivalent)<u><=
/u><u></u></li><li class=3D"MsoNormal">
It can be unresponsive to changes in the network (without any form of conge=
stion control, RTP continues to send oblivious to other participants)<u></u=
><u></u></li><li class=3D"MsoNormal">
It competes with itself in the average case (audio + video streams from the=
 same sender, multiple video streams for multi party conferencing)<u></u><u=
></u></li><li class=3D"MsoNormal">
It is time sensitive (as in real time media needs to be played back without=
 prior to changes in human perception)<u></u><u></u></li><li class=3D"MsoNo=
rmal">
It assumes that network behavior is unpredictable (existence of RTCP + use =
of things like adaptive jitter buffers)<u></u><u></u></li></ul>
</div>
<div>
<p class=3D"MsoNormal">Let&#39;s say we agree on all of these behaviors. =
=C2=A0It is safe to assume then that competing with RTP is a superset of be=
haviors then it is to compete with TCP, LEDBAT, or a web browser alone is i=
t not?<u></u><u></u></p>

</div>
<div>
<p class=3D"MsoNormal"><u></u>=C2=A0<u></u></p>
</div>
<div>
<p class=3D"MsoNormal">Thanks,<u></u><u></u></p>
</div>
<div>
<p class=3D"MsoNormal">Dan<u></u><u></u></p>
</div>
<div>
<p class=3D"MsoNormal">=C2=A0<u></u><u></u></p>
</div>
</div></div></div>
</div>

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

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