One document matched: draft-ietf-rmcat-eval-criteria-02.xml
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<rfc ipr="trust200902" docName="draft-ietf-rmcat-eval-criteria-02" category="info">
<!-- What is the category field value-->
<front>
<title abbrev="Evaluating Congestion Control for RMCAT">
Evaluating Congestion Control for Interactive Real-time Media
<!--Evaluation Criteria for RTP Congestion Avoidance Techniques -->
</title>
<author fullname="Varun Singh" initials="V" surname="Singh">
<organization>Aalto University</organization>
<address>
<postal>
<street>School of Electrical Engineering</street>
<street>Otakaari 5 A</street>
<city>Espoo</city>
<region>FIN</region>
<code>02150</code>
<country>Finland</country>
</postal>
<email>varun@comnet.tkk.fi</email>
<uri>http://www.netlab.tkk.fi/~varun/</uri>
</address>
</author>
<author initials="J." surname="Ott" fullname="Joerg Ott">
<organization>Aalto University</organization>
<address>
<postal>
<street>School of Electrical Engineering</street>
<street>Otakaari 5 A</street>
<city>Espoo</city>
<region>FIN</region>
<code>02150</code>
<country>Finland</country>
</postal>
<email>jo@comnet.tkk.fi</email>
</address>
</author>
<date year="2014"/>
<area>TSV</area>
<workgroup>RMCAT WG</workgroup>
<keyword>RTP</keyword>
<keyword>RTCP</keyword>
<keyword>Congestion Control</keyword>
<abstract>
<t>The Real-time Transport Protocol (RTP) is used to transmit
media in telephony and video conferencing applications. This
document describes the guidelines to evaluate new congestion
control algorithms for interactive point-to-point real-time
media.</t>
</abstract>
</front>
<middle>
<section title="Introduction">
<t>This memo describes the guidelines to help with evaluating
new congestion control algorithms for interactive
point-to-point real time media. The requirements for the
congestion control algorithm are outlined in <xref
target="I-D.ietf-rmcat-cc-requirements" />). This document
builds upon previous work at the IETF: <xref
target="RFC5033">Specifying New Congestion Control
Algorithms</xref> and <xref target="RFC5166">Metrics for the
Evaluation of Congestion Control Algorithms</xref>.</t>
<t>The guidelines proposed in the document are intended to help
prevent a congestion collapse, promote fair capacity usage and
optimize the media flow's throughput. Furthermore, the proposed
algorithms are expected to operate within the envelope of the
circuit breakers defined in <xref
target="I-D.ietf-avtcore-rtp-circuit-breakers" />.</t>
<t>This document only provides broad-level criteria for
evaluating a new congestion control algorithm and the working
group should expect a thorough scientific study to make its
decision. The results of the evaluation are not expected to be
included within the internet-draft but should be cited in the
document.</t>
</section>
<section title="Terminology" anchor="sec-terminology">
<!--<t> 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 BCP 14, <xref target="RFC2119" /> and indicate requirement
levels for compliant implementations. </t> -->
<t> The terminology defined in <xref target="RFC3550">RTP</xref>,
<xref target="RFC3551">RTP Profile for Audio and Video Conferences
with Minimal Control</xref>, <xref target="RFC3611">RTCP Extended
Report (XR)</xref>, <xref target="RFC4585">Extended RTP Profile
for RTCP-based Feedback (RTP/AVPF)</xref> and <xref
target="RFC5506">Support for Reduced-Size RTCP</xref> apply.</t>
</section>
<section title="Metrics" anchor="cc-metrics">
<t><xref target="RFC5166" /> describes the basic metrics for
congestion control. Metrics that are of interest for interactive
multimedia are:
<list style="symbols">
<t>Throughput.</t>
<t>Minimizing oscillations in the transmission rate (stability)
when the end-to-end capacity varies slowly.</t>
<t>Delay.</t>
<t>Reactivity to transient events.</t>
<t>Packet losses and discards.</t>
<!-- <t>Users' quality of experience</t> -->
<t>Section 2.1 of <xref target="RFC5166" /> discusses the tradeoff
between throughput, delay and loss.</t>
</list></t>
<t>Each experiment is expected to log every incoming and outgoing
packet (the RTP logging format is described in <xref
target="rtp-logging" />). The logging can be done inside the
application or at the endpoints using pcap (packet capture, e.g.,
tcpdump, wireshark). The following are calculated based on the
information in the packet logs:
<list style="numbers">
<t>Sending rate, Receiver rate, Goodput</t>
<t>Packet delay</t>
<t>Packet loss</t>
<t>If using, retransmission or FEC: residual loss</t>
<t>Packets discarded from the playout or de-jitter buffer</t>
<!-- <t>[Editor's note: How to handle packet re-transmissions? loss before
retransmission, after retransmission?]</t> -->
<t>Fairness or Unfairness: Experiments testing the performance
of an RMCAT proposal against any cross-traffic must define its
expected criteria for fairness. The "unfairness" test guideline
(measured at 1s intervals) is:<vspace />
1. Does not trigger the circuit breaker.<vspace />
2. No RMCAT stream achieves more than 3 times the average throughput
of the RMCAT stream with the lowest average throughput, for a case
when the competing streams have similar RTTs.<vspace />
3. RTT should not grow by a factor of 3 for the existing flows when a
new flow is added.
<vspace />
For example, see the test scenarios described in
<xref target="I-D.sarker-rmcat-eval-test" />.</t>
<t>Convergence time: The time taken to reach a stable rate at startup,
after the available link capacity changes, or when new flows get added
to the bottleneck link.</t>
<!--
<t>[Open issue (2): Convergence time was discussed briefly in the
design meetings. It is defined as: the time it takes the congestion
control to reach a stable rate (at startup or after new RMCAT flows
are added). What is a stable rate?]</t>
-->
<t>Bandwidth Utilization, defined as ratio of the instantaneous
sending rate to the instantaneous bottleneck capacity. This metric is
useful when an RMCAT flow is by itself or competing with similar
cross-traffic.</t>
</list></t>
<t>From the logs the statistical measures (min, max, mean, standard
deviation and variance) for the whole duration or any specific part of
the session can be calculated. Also the metrics (sending rate,
receiver rate, goodput, latency) can be visualized in graphs as
variation over time, the measurements in the plot are at 1 second
intervals. Additionally, from the logs it is possible to plot the
histogram or CDF of packet delay.</t>
<t>[Open issue (1): Using Jain-fairness index (JFI) for measuring
self-fairness between RTP flows? measured at what intervals?
visualized as a CDF or a timeseries? Additionally: Use JFI
for comparing fairness between RTP and long TCP flows?
]</t>
<t> </t>
<!-- <t> <list style="empty">
<t>(i) Bandwidth Utilization: is the
ratio of the encoding rate to the (available) end-to-end path
capacity.
<list style="symbols">
<t>Under-utilization: is the period of time when the endpoint's
encoding rate is lower than the end-to-end capacity, i.e., the
bandwidth utilization is less than 1.</t>
<t>Overuse: is the period of time when the endpoint's encoding
rate is higher than the end-to-end capacity, i.e., the bandwidth
utilization is greater than 1.</t>
<t>Steady-state: is the period of time when the endpoint's
encoding rate is relatively stable, i.e., the bandwidth
utilization is constant.</t>
</list></t>
<t></t>
<t>(ii) Packet Loss and Discard Rate.</t> <t></t>
<t>(iii) Fair Share. </t> <t></t>
<t>[Editor's Note: This metric should match the ones defined in the
<xref target="I-D.ietf-rmcat-cc-requirements">RMCAT requirements</xref>
document.]</t>
<t></t>
<t>(iv) Quality: There are many different types of quality metrics for
audio and video. Audio quality is often expressed by a MOS ("Mean
Opinion Score") and can be calculated using an objective algorithm
(E-model/R-model). Section 4.7 of <xref target="RFC3611" /> can also
be used for VoIP metrics. Similarly, there exist several metrics to
measure video quality, for example Peak Signal to Noise Ratio (PSNR).
</t>
<t>[Editor's Note: Should the algorithm compare average PSNR of test
video sequences or what other video quality metric can be used? If
Quality is used as a metric, it should not be the only metric used to
compare rate-control schemes. Also, algorithms using different codecs
cannot be compared]. </t>
</list>
</t>
-->
<section title="RTP Log Format" anchor="rtp-logging">
<t>The log file is tab or comma separated containing the following
details:</t>
<figure><artwork><![CDATA[
Send or receive timestamp (unix)
RTP payload type
SSRC
RTP sequence no
RTP timestamp
marker bit
payload size
]]></artwork></figure>
<t>If the congestion control implements, retransmissions or FEC, the
evaluation should report both packet loss (before applying
error-resilience) and residual packet loss (after applying
error-resilience).</t>
<!-- <t>The retransmissions for post-repair loss metric be logged in a
separate file, as the repair streams have different payload type
and/or SSRC.</t> -->
</section>
</section>
<!--
<section title="Congestion control requirements" anchor="cc-require">
<t> </t>
</section>
-->
<section title="Guidelines" anchor="cc-guidelines">
<t>A congestion control algorithm should be tested in
simulation or a testbed environment, and the experiments should
be repeated multiple times to infer statistical significance.
The following guidelines are considered for evaluation:</t>
<section title="Avoiding Congestion Collapse">
<t>The congestion control algorithm is expected to take an action,
such as reducing the sending rate, when it detects congestion.
Typically, it should intervene before the circuit breaker <xref
target="I-D.ietf-avtcore-rtp-circuit-breakers" /> is engaged. </t>
<t>Does the congestion control propose any changes to (or diverge
from) the circuit breaker conditions defined in <xref
target="I-D.ietf-avtcore-rtp-circuit-breakers" />.</t> </section>
<section title="Stability">
<t>The congestion control should be assessed for its stability
when the path characteristics do not change over time. Changing
the media encoding rate estimate too often or by too much may
adversely affect the application layer performance.</t>
</section>
<section title ="Media Traffic">
<t>The congestion control algorithm should be assessed with
different types of media behavior, i.e., the media should contain
idle and data-limited periods. For example, periods of silence for
audio, varying amount of motion for video, or bursty nature of
I-frames. </t>
<t>The evaluation may be done in two stages. In the first stage,
the endpoint generates traffic at the rate calculated by the
congestion controller. In the second stage, real codecs or models
of video codecs are used to mimic application-limited data periods
and varying video frame sizes.</t>
</section>
<section title="Start-up Behaviour">
<t>The congestion control algorithm should be assessed with
different start-rates. The main reason is to observe the behavior
of the congestion control in different test scenarios, such
as when competing with varying amount of cross-traffic or how
quickly does the congestion control algorithm achieve a stable
sending rate.</t>
<!-- <t>[Editor's note: requires a robust definition for unfriendliness
and convergence time.]</t> -->
</section>
<section title="Diverse Environments">
<t>The congestion control algorithm should be assessed in
heterogeneous environments, containing both wired and wireless
paths. Examples of wireless access technologies are: 802.11, GPRS,
HSPA, or LTE. One of the main challenges of the wireless
environments for the congestion control algorithm is to
distinguish between congestion induced loss and transmission
(bit-error) loss. Congestion control algorithms may
incorrectly identify transmission loss as congestion loss and
reduce the media encoding rate by too much, which may cause
oscillatory behavior and deteriorate the users' quality of
experience. Furthermore, packet loss may induce additional delay
in networks with wireless paths due to link-layer
retransmissions.</t>
</section>
<section title="Varying Path Characteristics">
<t>The congestion control algorithm should be evaluated for a
range of path characteristics such as, different end-to-end
capacity and latency, varying amount of cross traffic on a
bottleneck link and a router's queue length. For the moment, only
DropTail queues are used. However, if new Active Queue Management
(AQM) schemes become available, the performance of the congestion
control algorithm should be again evaluated.</t>
<t>In an experiment, if the media only flows in a single
direction, the feedback path should also be tested with varying
amounts of impairments.</t>
<t>The main motivation for the previous and current criteria is to
identify situations in which the proposed congestion control is
less performant.</t>
</section>
<section title="Reacting to Transient Events or Interruptions">
<t>The congestion control algorithm should be able to handle
changes in end-to-end capacity and latency. Latency may change
due to route updates, link failures, handovers etc. In mobile
environment the end-to-end capacity may vary due to the
interference, fading, handovers, etc. In wired networks the
end-to-end capacity may vary due to changes in resource
reservation.</t>
</section>
<section title="Fairness With Similar Cross-Traffic">
<t>The congestion control algorithm should be evaluated when
competing with other RTP flows using the same or another candidate
congestion control algorithm. The proposal should highlight the
bottleneck capacity share of each RTP flow.</t>
<!-- <t>[Editor's note: If we define Unfriendliness then that criteria
should be applied here.]</t> -->
</section>
<section title="Impact on Cross-Traffic">
<t>The congestion control algorithm should be evaluated when
competing with standard TCP. Short TCP flows may be considered
as transient events and the RTP flow may give way to the short
TCP flow to complete quickly. However, long-lived TCP flows may
starve out the RTP flow depending on router queue length. </t>
<!-- In the latter case the proposed congestion control for RTP
should be as aggressive as <xref target="RFC5681">standard
TCP</xref>.</t> -->
<t>The proposal should also measure the impact on varied number
of cross-traffic sources, i.e., few and many competing flows,
or mixing various amounts of TCP and similar cross-traffic.</t>
</section>
<section title="Extensions to RTP/RTCP">
<t>The congestion control algorithm should indicate if any
protocol extensions are required to implement it and should
carefully describe the impact of the extension.</t>
</section>
</section>
<section title="Minimum Requirements for Evaluation">
<t>The minimal requirements for RMCAT proposals is to produce or
present results for the test scenarios described in Section 5
of <xref target="I-D.sarker-rmcat-eval-test" />
(Basic Test Cases).</t>
</section>
<!--
<section title="Example Evaluation Scenarios" anchor="cc-scenario">
<t>In the scenarios listed below, all RTP flows are
bi-directional and point-to-point.</t>
<t>Unless specified, the following parameters are used in each scenario:</t>
<t><list style="symbols">
<t>Video Start Rate: 128 kbps</t>
<t>Maximum end-to-end delay: 300ms, packets arriving
after this are discarded</t>
<t>Video Frame rate: 15 </t>
<t>Audio packetization interval: 20ms </t>
<t>MTU: 1450 bytes</t>
<t>[Editor's Note: the numbers in this section are TBD]</t>
</list></t>
<t>Topology:<list style="symbols">
<t>Dumbbell, the endpoint is connected to the bottleneck
link via an access links. The bottleneck may be shared by multiple
endpoints.</t>
<t>Parking lot: there are three bottleneck links arranged horizontally,
these links are connected by access links. In this case, flows may share
different bottleneck links.</t>
</list></t>
<t>[Editor's note: Should the queue-size be specified as well?].</t>
<section title="[S1] RTP flow on a fixed link">
<t>This scenario evaluates the ramp-up to the bottleneck capacity and
the stability of the proposed congestion control algorithm.</t>
<t>This scenario uses the dumbbell topology and both the access link
can be ADSL (500kbps uplink, 256 downlink, 2ms one-way delay)
or WLAN (54Mbps, 2ms one-way delay, 2-5% packet loss rate and
link layer re-transmissions).
</t>
<t>
The bottleneck link can have one of the following capacities:
500kbps, 1Mbps, 5Mbps and link delay: 10ms, 50ms, 120ms.
</t>
<t>Each congestion control algorithm should plot the variation of the
sending rate against time, also plot the instances of packets losses.
Additionally, measure the time taken for the sending rate to reach
the end-to-end capacity (average and standard deviation over
10 simulation runs).
</t>
</section>
<section title="[S2] RTP flow on a variable capacity link">
<t>This scenario evaluates the reactivity of the proposed congestion
control algorithm to transient network events due to interference and
handovers in mobile environments.</t>
<t>This scenario uses the dumbbell topology, and both end-points use
3G/LTE access.
Sample 3G/LTE (uplink and downlink) bandwidth traces are
available at <xref target="SA4-EVAL"/>, loss patterns at
<xref target="SA4-LR"/> and the link delay: 30ms, 80ms.
The bottleneck link can have one of the following capacities:
500kbps, 5Mbps and link delay: 20ms.</t>
<t>Each congestion control algorithm should plot the variation of the
sending rate, 3G link capacity against time, also plot the
instances of packets losses.
</t>
</section>
<section title="[S3] Fairness to RTP flows running the same congestion
control algorithm (self-fairness)">
<t>This scenario shows if the proposed
algorithm can share the bottleneck link equitably, irrespective
of number of flows.</t>
<t>In this scenario there is more than one endpoint connected
to the bottleneck link.
<list style="none">
<t>(a) All the access links have the same link characteristics and
start at the same time (see [S1]). The bottleneck link can have one of
the following link capacity: 500kbpsm 5Mbpps and link delay 20ms.</t>
<t>(b) The access links have different link characteristics [See S1]
but start at the same time.</t>
<t>(c) An RTP flow is added at 10s intervals (upto 5 flows),
the late arriving flows have increasing access link
delay (0, 5, 10, 20, 50ms).
The bottleneck link can have one of the following capacities:
1Mbps, 10Mbps and link delay: 10ms, 50ms, 120ms. </t>
</list></t>
<t>[Parking lot topology simulation: TBD]</t>
</section>
<section title="[S4 and S5] Competing with short and long TCP flows">
<t>[Editor's Note: Remove these scenarios?]</t>
<t>[S4] Competing with long-lived TCP flows: In this scenario
the proposed algorithm is expected to be TCP-friendly, i.e., it
should neither starve out the competing TCP flows (causing a
congestion collapse) nor should it be starved out by TCP.</t>
<t></t>
<t>[S5] Competing with short TCP flows: Depending on the level
of statistical multiplexing on the bottleneck link, the
proposed algorithm may behave differently. If there are a few
short TCP flows then the proposed algorithm may observe these
flows as transient events and let them complete quickly.
Alternatively, if there are many short flows then the proposed
algorithm may have to compete with the flows as if they were
long lived TCP flows.</t>
<t><xref target="TCP-eval-suite"/> contains examples of TCP traffic
load and scenario settings.</t>
<t>[Editor's Note: definition of many and
few short TCP flows may depend on the bottleneck link
capacity.]</t>
<t></t>
<t>[Editor's Note: clarify if media packets are generated using
a traffic generator.]</t>
</section>
</section>
-->
<section title="Status of Proposals" anchor="cc-proposal">
<t>Congestion control algorithms are expected to be published
as "Experimental" documents until they are shown to be safe to
deploy. An algorithm published as a draft should be
experimented in simulation, or a controlled environment
(testbed) to show its applicability. Every congestion control
algorithm should include a note describing the environments in
which the algorithm is tested and safe to deploy. It is
possible that an algorithm is not recommended for certain
environments or perform sub-optimally for the user.</t>
<t>[Editor's Note: Should there be a distinction between
"Informational" and "Experimental" drafts for congestion
control algorithms in RMCAT. <xref target="RFC5033" />
describes Informational proposals as algorithms that are not
safe for deployment but are proposals to experiment with in
simulation/testbeds. While Experimental algorithms are ones
that are deemed safe in some environments but require a more
thorough evaluation (from the community).]</t>
</section>
<section title="Security Considerations">
<t>Security issues have not been discussed in this memo.</t>
<!-- Congestion Collapse, Denial of Service -->
</section>
<section title="IANA Considerations">
<t>There are no IANA impacts in this memo.</t>
</section>
<section anchor="contrib" title="Contributors">
<t>The content and concepts within this document are a product of
the discussion carried out in the Design Team.</t>
<t>Michael Ramalho provided the text for a specific scenario,
which is now covered in <xref target="I-D.sarker-rmcat-eval-test" />.</t>
</section>
<section title="Acknowledgements">
<t> Much of this document is derived from previous work on
congestion control at the IETF.</t>
<t> The authors would like to thank
Harald Alvestrand,
Anna Brunstrom,
Luca De Cicco,
Wesley Eddy,
Lars Eggert,
Kevin Gross,
Vinayak Hegde,
Stefan Holmer,
Randell Jesup,
Karen Nielsen,
Piers O'Hanlon,
Colin Perkins,
Michael Ramalho,
Zaheduzzaman Sarker,
Timothy B. Terriberry,
Michael Welzl, and
Mo Zanaty
for providing valuable feedback on earlier versions of this draft.
Additionally, also thank the participants of the design team for
their comments and discussion related to the evaluation
criteria.</t>
</section>
</middle>
<back>
<references title="Normative References">
<!--&rfc2119;-->
<!-- RTP related -->
&rfc3550;
&rfc3551;
&rfc3611;
&rfc4585;
&rfc5506;
<!--RMCAT related -->
&I-D.ietf-rmcat-cc-requirements;
&I-D.ietf-avtcore-rtp-circuit-breakers;
</references>
<references title="Informative References">
&rfc5033; <!-- CC Evaluation -->
&rfc5166; <!-- CC Metrics -->
&rfc5681; <!-- Standard TCP -->
&I-D.sarker-rmcat-eval-test;
<!--
&I-D.ietf-xrblock-rtcp-xr-qoe;
<?rfc include="reference.3GPP.R1.081955"?>
-->
<reference anchor="SA4-EVAL">
<front>
<title>LTE Link Level Throughput Data for SA4 Evaluation Framework</title>
<author initials="3GPP" surname="R1-081955" fullname="3GPP R1-081955">
<organization />
</author>
<date month="5" year="2008" />
<abstract>
<t>In R1-081720, 3GPP SA4 has requested RAN1 and RAN2 for link
level throughput traces to be used in an evaluation framework
they are developing for dynamic video rate adaptation.
</t></abstract>
</front>
<seriesInfo name="3GPP" value="R1-081955" />
<format type='ZIP' octets='3459875' target='http://www.3gpp.net/ftp/tsg_ran/WG1_RL1/TSGR1_53/Docs/R1-081955.zip' />
</reference>
<reference anchor="SA4-LR">
<front>
<title>Error Patterns for MBMS Streaming over UTRAN and GERAN</title>
<author initials="3GPP" surname="S4-050560" fullname="3GPP S4-050560">
<organization />
</author>
<date month="5" year="2008" />
</front>
<seriesInfo name="3GPP" value="S4-050560" />
<format type='ZIP' octets='335322' target='http://www.3gpp.org/FTP/tsg_sa/WG4_CODEC/TSGS4_36/Docs/S4-050560.zip' />
</reference>
<reference anchor="TCP-eval-suite">
<front>
<title>Towards a Common TCP Evaluation Suite</title>
<author initials="A." surname="Lachlan" fullname="Andrew Lachlan"/>
<author initials="C." surname="Marcondes" fullname="Cesar Marcondes"/>
<author initials="S." surname="Floyd" fullname="Sally Floyd"/>
<author initials="L." surname="Dunn" fullname="Lawrence Dunn"/>
<author initials="R." surname="Guillier" fullname="Romeric Guillier"/>
<author initials="W." surname="Gang" fullname="Wang Gang"/>
<author initials="L." surname="Eggert" fullname="Lars Eggert"/>
<author initials="S." surname="Ha" fullname="Sangtae Ha"/>
<author initials="I." surname="Rhee" fullname="Injong Rhee"/>
<date month="August" year="2008"/>
</front>
<seriesInfo name="Proc. PFLDnet." value="2008"/>
</reference>
</references>
<section anchor="misc" title="Application Trade-off">
<t>Application trade-off is yet to be defined. see <xref
target="I-D.ietf-rmcat-cc-requirements">RMCAT requirements</xref>
document. Perhaps each experiment should define the application's
expectation or trade-off.</t>
<section anchor="misc-2" title="Measuring Quality">
<t>No quality metric is defined for performance evaluation, it is
currently an open issue. However, there is consensus that
congestion control algorithm should be able to show that it is
useful for interactive video by performing analysis using a real
codec and video sequences. </t>
</section>
</section>
<section anchor="App-cl" title="Change Log">
<t>Note to the RFC-Editor: please remove this section prior to
publication as an RFC.</t>
<section title="Changes in draft-ietf-rmcat-eval-criteria-02">
<t><list style="symbols">
<t>Incorporated fairness test as a working test.</t>
<t>Updated text on mimimum evaluation requirements.</t>
</list></t>
</section>
<section title="Changes in draft-ietf-rmcat-eval-criteria-01">
<t><list style="symbols">
<t>Removed Appendix B.</t>
<t>Removed Section on Evaluation Parameters.</t>
</list></t>
</section>
<section title="Changes in draft-ietf-rmcat-eval-criteria-00">
<t><list style="symbols">
<t>Updated references.</t>
<t>Resubmitted as WG draft.</t>
</list></t>
</section>
<section title="Changes in draft-singh-rmcat-cc-eval-04">
<t><list style="symbols">
<t>Incorporate feedback from IETF 87, Berlin.</t>
<t>Clarified metrics: convergence time, bandwidth
utilization.</t>
<t>Changed fairness criteria to fairness test.</t>
<t>Added measuring pre- and post-repair loss.</t>
<t>Added open issue of measuring video quality to
appendix.</t>
<t>clarified use of DropTail and AQM.</t>
<t>Updated text in "Minimum Requirements for Evaluation"</t>
</list></t>
</section>
<section title="Changes in draft-singh-rmcat-cc-eval-03">
<t><list style="symbols">
<t>Incorporate the discussion within the design team.</t>
<t>Added a section on evaluation parameters, it describes the
flow and network characteristics.</t>
<t>Added Appendix with self-fairness experiment.</t>
<t>Changed bottleneck parameters from a proposal to an example
set.</t>
<t></t>
</list></t>
</section>
<section title="Changes in draft-singh-rmcat-cc-eval-02">
<t><list style="symbols">
<t>Added scenario descriptions.</t>
</list></t>
</section>
<section title="Changes in draft-singh-rmcat-cc-eval-01">
<t><list style="symbols">
<t>Removed QoE metrics.</t>
<t>Changed stability to steady-state.</t>
<t>Added measuring impact against few and many
flows.</t>
<t>Added guideline for idle and data-limited periods.</t>
<t>Added reference to TCP evaluation suite in example
evaluation scenarios.</t>
</list></t>
</section>
</section>
</back>
</rfc>
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