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<!-- 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 initials="V." surname="Singh" fullname="Varun Singh">
<organization abbrev="callstats.io">
CALLSTATS I/O Oy
</organization>
<address>
<postal>
<street>Runeberginkatu 4c A 4</street>
<code>00100</code> <city>Helsinki</city>
<country>Finland</country>
</postal>
<email>varun@callstats.io</email>
<uri>
https://www.callstats.io/about
</uri>
</address>
</author>
<author initials="J." surname="Ott" fullname="Joerg Ott">
<organization>Technical University of Munich</organization>
<address>
<postal>
<street>Faculty of Informatics</street>
<street>Boltzmannstrasse 3</street>
<city>Garching bei München</city>
<region>DE</region>
<code>85748</code>
<country>Germany</country>
</postal>
<email>ott@in.tum.de</email>
</address>
</author>
<author fullname="Stefan Holmer" initials="S." surname="Holmer">
<organization abbrev="Google">Google</organization>
<address>
<postal>
<street>Kungsbron 2</street>
<code>11122</code>
<city>Stockholm</city>
<country>Sweden</country>
</postal>
<email>holmer@google.com</email>
</address>
</author>
<date year="2016"/>
<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. The minimal
requirement for RMCAT proposals is to produce or present
results for the test scenarios described in
<xref target="I-D.ietf-rmcat-eval-test" /> (Basic Test Cases).
Additionally, proponents may produce evaluation results for the
<xref target="I-D.ietf-rmcat-wireless-tests"> wireless test
scenarios</xref>.
</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 (measured at 200ms intervals)</t>
<t>Packets sent, Packets received</t>
<t>Bytes sent, bytes received</t>
<t>Packet delay</t>
<t>Packets lost, Packets discarded (from the playout or de-jitter buffer)</t>
<t>If using, retransmission or FEC: post-repair loss</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.ietf-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>Instability or oscillation in the sending rate: The frequency or
number of instances when the sending rate oscillates between an
high watermark level and a low watermark level, or vice-versa in
a defined time window. For example, the watermarks can be set at 4x
interval: 500 Kbps, 2 Mbps, and a time window of 500ms.</t>
<t>[Editor's note: Section 3, in <xref target="I-D.ietf-netvc-testing" />
contains objective Metrics for evaluating codecs.]</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 only 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>
</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>
</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>
<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 anchor="add-params" title="List of Network Parameters">
<t>The implementors initially are encouraged to choose evaluation settings
from the following values:</t>
<section anchor="scen-delay" title="One-way Propagation Delay">
<!-- -->
<t>Experiments are expected to verify that the congestion control is
able to work in challenging situations, for example over
trans-continental and/or satellite links. Typical values are: <list
style="numbers">
<t>Very low latency: 0-1ms</t>
<t>Low latency: 50ms</t>
<t>High latency: 150ms</t>
<t>Extreme latency: 300ms</t>
</list></t>
</section>
<section anchor="scen-loss" title="End-to-end Loss">
<t>To model lossy links, the experiments can choose one of the
following loss rates, the fractional loss is the ratio of packets lost
and packets sent. <list style="numbers">
<t>no loss: 0%</t>
<t>1%</t>
<t>5%</t>
<t>10%</t>
<t>20%</t>
</list></t>
</section>
<section anchor="scen-queue" title="DropTail Router Queue Length">
<t>The router queue length is measured as the time taken to drain the
FIFO queue. It has been noted in various discussions that the queue
length in the current deployed Internet varies significantly. While
the core backbone network has very short queue length, the home
gateways usually have larger queue length. Those various queue lengths
can be categorized in the following way: <list style="numbers">
<t>QoS-aware (or short): 70ms</t>
<t>Nominal: 300-500ms</t>
<t>Buffer-bloated: 1000-2000ms</t>
</list> Here the size of the queue is measured in bytes or packets
and to convert the queue length measured in seconds to queue length in
bytes:</t>
<t>QueueSize (in bytes) = QueueSize (in sec) x Throughput (in
bps)/8</t>
<!-- <t>and 2) queue length in packets:</t>
<t>QueueSize (in pkts) = QueueSize (in bytes)/MTU,
MTU=1500</t> -->
<!-- <t>[Open issue (11): Confirm the above values, do we need to
define parameters for other types of queues?]</t> -->
</section>
<section title="Loss generation model">
<t>[Open Issue: Describes the model for generating packet losses,
for example, losses can be generated using traces, or using the
Gilbert-Elliot model, or randomly (uncorrelated loss).]</t>
</section>
<section anchor="JM" title="Jitter models">
<t>This section defines jitter models for the purposes of this
document. When jitter is to be applied to both the RMCAT flow and any
competing flow (such as a TCP competing flow), the competing flow will
use the jitter definition below that does not allow for re-ordering of
packets on the competing flow (see NR-RBPDV definition below).</t>
<t>Jitter is an overloaded term in communications. Its meaning is
typically associated with the variation of a metric (e.g., delay) with
respect to some reference metric (e.g., average delay or minimum
delay). For example, RFC 3550 jitter is a smoothed estimate of jitter
which is particularly meaningful if the underlying packet delay
variation was caused by a Gaussian random process.</t>
<t>Because jitter is an overloaded term, we instead use the term
Packet Delay Variation (PDV) to describe the variation of delay of
individual packets in the same sense as the IETF IPPM WG has defined
PDV in their documents (e.g., RFC 3393) and as the ITU-T SG16 has
defined IP Packet Delay Variation (IPDV) in their documents (e.g.,
Y.1540).</t>
<t>Most PDV distributions in packet network systems are one-sided
distributions (the measurement of which with a finite number of
measurement samples result in one-sided histograms). In the usual
packet network transport case there is typically one packet that
transited the network with the minimum delay, then a majority of
packets also transit the system within some variation from this
minimum delay, and then a minority of the packets transit the network
with delays higher than the median or average transit time (these are
outliers). Although infrequent, outliers can cause significant
deleterious operation in adaptive systems and should be considered in
RMCAT adaptation designs.</t>
<t>In this section we define two different bounded PDV
characteristics, 1) Random Bounded PDV and 2) Approximately Random
Subject to No-Reordering Bounded PDV.</t>
<t>[Open issue: which one is used in evaluations? Are both used?]</t>
<section title="Random Bounded PDV (RBPDV)">
<t>The RBPDV probability distribution function (pdf) is specified to
be of some mathematically describable function which includes some
practical minimum and maximum discrete values suitable for testing.
For example, the minimum value, x_min, might be specified as the
minimum transit time packet and the maximum value, x_max, might be
idefined to be two standard deviations higher than the mean.</t>
<t>Since we are typically interested in the distribution relative to
the mean delay packet, we define the zero mean PDV sample, z(n), to be
z(n) = x(n) - x_mean, where x(n) is a sample of the RBPDV random
variable x and x_mean is the mean of x.</t>
<t>We assume here that s(n) is the original source time of packet n
and the post-jitter induced emmission time, j(n), for packet n is j(n)
= {[z(n) + x_mean] + s(n)}. It follows that the separation in the
post-jitter time of packets n and n+1 is {[s(n+1)-s(n)] -
[z(n)-z(n+1)]}. Since the first term is always a positive quantity, we
note that packet reordering at the receiver is possible whenever the
second term is greater than the first. Said another way, whenever the
difference in possible zero mean PDV sample delays (i.e.,
[x_max-x_min]) exceeds the inter-departure time of any two sent
packets, we have the possibility of packet re-ordering.</t>
<t>There are important use cases in real networks where packets can
become re-ordered such as in load balancing topologies and during
route changes. However, for the vast majority of cases there is no
packet re-ordering because most of the time packets follow the same
path. Due to this, if a packet becomes overly delayed, the packets
after it on that flow are also delayed. This is especially true for
mobile wireless links where there are per-flow queues prior to base
station scheduling. Owing to this important use case, we define
another PDV profile similar to the above, but one that does not allow
for re-ordering within a flow.</t>
</section>
<section title="Approximately Random Subject to No-Reordering Bounded PDV
(NR-RPVD)">
<t>No Reordering RPDV, NR-RPVD, is defined similarly to the above with
one important exception. Let serial(n) be defined as the serialization
delay of packet n at the lowest bottleneck link rate (or other
appropriate rate) in a given test. Then we produce all the post-jitter
values for j(n) for n = 1, 2, ... N, where N is the length of the
source sequence s to be offset-ed. The exception can be stated as
follows: We revisit all j(n) beginning from index n=2, and if j(n) is
determined to be less than [j(n-1)+serial(n-1)], we redefine j(n) to
be equal to [j(n-1)+serial(n-1)] and continue for all remaining n
(i.e., n = 3, 4, .. N). This models the case where the packet n is
sent immediately after packet (n-1) at the bottleneck link rate.
Although this is generally the theoretical minimum in that it assumes
that no other packets from other flows are in-between packet n and n+1
at the bottleneck link, it is a reasonable assumption for per flow
queuing.</t>
<t>We note that this assumption holds for some important exception
cases, such as packets immediately following outliers. There are a
multitude of software controlled elements common on end-to-end
Internet paths (such as firewalls, ALGs and other middleboxes) which
stop processing packets while servicing other functions (e.g., garbage
collection). Often these devices do not drop packets, but rather queue
them for later processing and cause many of the outliers. Thus NR-RPVD
models this particular use case (assuming serial(n+1) is defined
appropriately for the device causing the outlier) and thus is believed
to be important for adaptation development for RMCAT.</t>
</section>
<section title="Recommended distribution">
<t>It is recommended that z(n) is distributed according to a truncated
Gaussian:</t>
<t>z(n) ~ |max(min(N(0, std^2), N_STD * std), -N_STD * std)|</t>
<t>where N(0, std^2) is the Gaussian distribution with zero mean and
standard deviation std. Recommended values:</t>
<t><list style="symbols">
<t>std = 5 ms</t>
<t>N_STD = 3</t>
</list></t>
</section>
</section>
</section>
<section title="WiFi or Cellular Links">
<t>
<xref target="I-D.ietf-rmcat-wireless-tests" /> describes the test
cases to simulate networks with wireless links. The document
describes mechanism to simulate both cellular and WiFi networks.
</t>
</section>
<section anchor="app-additional" title="Traffic Models">
<section title="TCP taffic model">
<t>Long-lived TCP flows will download data throughout the session and
are expected to have infinite amount of data to send or receive.
For example, to </t>
<t>Each short TCP flow is modeled as a sequence of file downloads
interleaved with idle periods. Not all short TCPs start at the same
time, i.e., some start in the ON state while others start in the OFF
state.</t>
<t>The short TCP flows can be modelled as follows: 30 connections start
simultaneously fetching small (30-50 KB) amounts of data. This covers the
case where the short TCP flows are not fetching a video file.</t>
<t>The idle period between bursts of starting a group of TCP flows is
typically derived from an exponential distribution with the mean value of
10 seconds.</t>
<t>[These values were picked based on the data available at
http://httparchive.org/interesting.php as of October 2015].</t>
</section>
<section title="RTP Video model">
<t>
<xref target="I-D.ietf-rmcat-video-traffic-model" /> describes two
types of video traffic models for evaluating RMCAT candidate algorithms.
The first model statistically characterizes the behavior of a video
encoder. Whereas the second model uses video traces.
</t>
<t> For example, test sequences are available at:
<xref target="xiph-seq"></xref> and <xref target="HEVC-seq"></xref>.
The currently chosen video streams are: Foreman and FourPeople.</t>
</section>
<section title="Background UDP">
<t>Background UDP flow is modeled as a constant
bit rate (CBR) flow. It will download data at a particular CBR
rate for the complete session, or will change to particular
CBR rate at predefined intervals. The inter packet interval is
calculated based on the CBR and the packet size (is typically
set to the path MTU size, the default value can be 1500 bytes).
</t>
</section>
</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 the Jitter model.</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,
Mirja Kuehlewind,
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;
&I-D.ietf-rmcat-wireless-tests;
</references>
<references title="Informative References">
&rfc5033; <!-- CC Evaluation -->
&rfc5166; <!-- CC Metrics -->
&rfc5681; <!-- Standard TCP -->
&I-D.ietf-rmcat-eval-test;
&I-D.ietf-rmcat-video-traffic-model;
&I-D.ietf-netvc-testing;
<!-- <?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>
<reference anchor="xiph-seq">
<front>
<title>Video Test Media Set</title>
<author fullname="Daede, T." initials="T." surname="Daede"></author>
<date month="" year="" />
</front>
<seriesInfo name="https://people.xiph.org/~tdaede/sets/" value="" />
</reference>
<reference anchor="HEVC-seq">
<front>
<title>Test Sequences</title>
<author fullname="" initials="" surname="HEVC"></author>
<date month="" year="" />
</front>
<seriesInfo name="http://www.netlab.tkk.fi/~varun/test_sequences/"
value="" />
</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-06">
<t><list style="symbols">
<t>Updated Jitter.</t>
</list></t>
</section>
<section title="Changes in draft-ietf-rmcat-eval-criteria-05">
<t><list style="symbols">
<t>Improved text surrounding wireless tests, video sequences,
and short-TCP model.</t>
</list></t>
</section>
<section title="Changes in draft-ietf-rmcat-eval-criteria-04">
<t><list style="symbols">
<t>Removed the guidelines section, as most of the sections
are now covered: wireless tests, video model, etc.</t>
<t>Improved Short TCP model based on the suggestion to use
httparchive.org.</t>
</list></t>
</section>
<section title="Changes in draft-ietf-rmcat-eval-criteria-03">
<t><list style="symbols">
<t>Keep-alive version.</t>
<t>Moved link parameters and traffic models from eval-test</t>
</list></t>
</section>
<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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