RMCAT WG                                                        V. Singh
Internet-Draft                                              callstats.io
Intended status: Informational                                    J. Ott
Expires: August 30, 2020                  Technical University of Munich
                                                               S. Holmer
                                                       February 27, 2020

     Evaluating Congestion Control for Interactive Real-time Media


   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.

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
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   This Internet-Draft will expire on August 30, 2020.

Copyright Notice

   Copyright (c) 2020 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
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   publication of this document.  Please review these documents
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   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.  Terminology . . . . . . . . . . . . . . . . . . . . . . . . .   3
   3.  Metrics . . . . . . . . . . . . . . . . . . . . . . . . . . .   4
     3.1.  RTP Log Format  . . . . . . . . . . . . . . . . . . . . .   5
   4.  List of Network Parameters  . . . . . . . . . . . . . . . . .   6
     4.1.  One-way Propagation Delay . . . . . . . . . . . . . . . .   6
     4.2.  End-to-end Loss . . . . . . . . . . . . . . . . . . . . .   6
     4.3.  Drop Tail Router Queue Length . . . . . . . . . . . . . .   6
     4.4.  Loss generation model . . . . . . . . . . . . . . . . . .   7
     4.5.  Jitter models . . . . . . . . . . . . . . . . . . . . . .   7
       4.5.1.  Random Bounded PDV (RBPDV)  . . . . . . . . . . . . .   8
       4.5.2.  Approximately Random Subject to No-Reordering Bounded
               PDV         (NR-RPVD) . . . . . . . . . . . . . . . .   9
       4.5.3.  Recommended distribution  . . . . . . . . . . . . . .  10
   5.  Traffic Models  . . . . . . . . . . . . . . . . . . . . . . .  10
     5.1.  TCP traffic model . . . . . . . . . . . . . . . . . . . .  10
     5.2.  RTP Video model . . . . . . . . . . . . . . . . . . . . .  11
     5.3.  Background UDP  . . . . . . . . . . . . . . . . . . . . .  11
   6.  Security Considerations . . . . . . . . . . . . . . . . . . .  11
   7.  IANA Considerations . . . . . . . . . . . . . . . . . . . . .  12
   8.  Contributors  . . . . . . . . . . . . . . . . . . . . . . . .  12
   9.  Acknowledgments . . . . . . . . . . . . . . . . . . . . . . .  12
   10. References  . . . . . . . . . . . . . . . . . . . . . . . . .  12
     10.1.  Normative References . . . . . . . . . . . . . . . . . .  12
     10.2.  Informative References . . . . . . . . . . . . . . . . .  13
   Appendix A.  Change Log . . . . . . . . . . . . . . . . . . . . .  14
     A.1.  Changes in draft-ietf-rmcat-eval-criteria-07  . . . . . .  14
     A.2.  Changes in draft-ietf-rmcat-eval-criteria-06  . . . . . .  14
     A.3.  Changes in draft-ietf-rmcat-eval-criteria-05  . . . . . .  15
     A.4.  Changes in draft-ietf-rmcat-eval-criteria-04  . . . . . .  15
     A.5.  Changes in draft-ietf-rmcat-eval-criteria-03  . . . . . .  15
     A.6.  Changes in draft-ietf-rmcat-eval-criteria-02  . . . . . .  15
     A.7.  Changes in draft-ietf-rmcat-eval-criteria-01  . . . . . .  15
     A.8.  Changes in draft-ietf-rmcat-eval-criteria-00  . . . . . .  15
     A.9.  Changes in draft-singh-rmcat-cc-eval-04 . . . . . . . . .  15
     A.10. Changes in draft-singh-rmcat-cc-eval-03 . . . . . . . . .  16
     A.11. Changes in draft-singh-rmcat-cc-eval-02 . . . . . . . . .  16
     A.12. Changes in draft-singh-rmcat-cc-eval-01 . . . . . . . . .  16
   Authors' Addresses  . . . . . . . . . . . . . . . . . . . . . . .  16

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1.  Introduction

   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 [I-D.ietf-rmcat-cc-requirements]).  This document
   builds upon previous work at the IETF: Specifying New Congestion
   Control Algorithms [RFC5033] and Metrics for the Evaluation of
   Congestion Control Algorithms [RFC5166].

   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 RFC8083 [RFC8083].

   This document only provides the broad set of network parameters and
   and traffic models for evaluating a new congestion control algorithm.
   The minimal requirements for congestion control proposals is to
   produce or present results for the test scenarios described in
   [I-D.ietf-rmcat-eval-test] (Basic Test Cases), which also defines the
   specifics for the test cases.  Additionally, proponents may produce
   evaluation results for the wireless test scenarios

   This document does not cover application-specific implications of
   congestion control algorithms and how those could be evaluated.
   Therefore, no quality metrics are defined for performance evaluation;
   quality metrics and algorithms to infer those vary between media
   types.  Metrics and algorithms to assess, e.g., quality of experience
   evolve continuously so that determining suitable choices is left for
   future work.  However, there is consensus that each congestion
   control algorithm should be able to show that it is useful for
   interactive video by performing analysis using a real codecs and
   video sequences and state-of-the-art quality metrics.

   Beyond optimizing individual metrics, real-time applications may have
   further options to trade off performance, e.g., across multiple
   media; refer to the RMCAT requirements
   [I-D.ietf-rmcat-cc-requirements] document.  Such trade-offs may be
   defined in the future.

2.  Terminology

   The terminology defined in RTP [RFC3550], RTP Profile for Audio and
   Video Conferences with Minimal Control [RFC3551], RTCP Extended
   Report (XR) [RFC3611], Extended RTP Profile for RTCP-based Feedback

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   (RTP/AVPF) [RFC4585] and Support for Reduced-Size RTCP [RFC5506]

3.  Metrics

   This document specifies testing criteria for evaluating congestion
   control algorithms for RTP media flows.  Proposed algorithms are to
   prove their performance by means of simulation and/or emulation
   experiments for all the cases described.

   Each experiment is expected to log every incoming and outgoing packet
   (the RTP logging format is described in Section 3.1).  The logging
   can be done inside the application or at the endpoints using PCAP
   (packet capture, e.g., tcpdump, wireshark).  The following metrics
   are calculated based on the information in the packet logs:

   1.   Sending rate, Receiver rate, Goodput (measured at 200ms

   2.   Packets sent, Packets received

   3.   Bytes sent, bytes received

   4.   Packet delay

   5.   Packets lost, Packets discarded (from the playout or de-jitter

   6.   If using, retransmission or FEC: post-repair loss

   7.   Self-Fairness and Fairness with respect to cross traffic:
        Experiments testing a given congestion control proposal must
        report on relative ratios of the average throughput (measured at
        coarser time intervals) obtained by each RTP media stream.  In
        the presence of background cross-traffic such as TCP, the report
        must also include the relative ratio between average throughput
        of RTP media streams and cross-traffic streams.
        During static periods of a test (i.e., when bottleneck bandwidth
        is constant and no arrival/departure of streams), these report
        on relative ratios serve as an indicator of how fair the RTP
        streams share bandwidth amongst themselves and against cross-
        traffic streams.  The throughput measurement interval should be
        set at a few values (for example, at 1s, 5s, and 20s) in order
        to measure fairness across different time scales.
        As a general guideline, the relative ratio between congestion
        controlled RTP flows with the same priority level and similar
        path RTT should be bounded between (0.333 and 3.)  For example,
        see the test scenarios described in [I-D.ietf-rmcat-eval-test].

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   8.   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.

   9.   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.

   10.  Bandwidth Utilization, defined as ratio of the instantaneous
        sending rate to the instantaneous bottleneck capacity.  This
        metric is useful only when a congestion controlled RTP flow is
        by itself or competing with similar cross-traffic.

   Note that the above metrics are all objective application-independent
   metrics.  Refer to Section 3, in [I-D.ietf-netvc-testing] for
   objective metrics for evaluating codecs.

   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.

3.1.  RTP Log Format

   Having a common log format simplifies running analyses across and
   comparing different measurements.  The log file should be tab or
   comma separated containing the following details:

           Send or receive timestamp (unix)
           RTP payload type
           RTP sequence no
           RTP timestamp
           marker bit
           payload size

   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-

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4.  List of Network Parameters

   The implementors initially are encouraged to choose evaluation
   settings from the following values:

4.1.  One-way Propagation Delay

   Experiments are expected to verify that the congestion control is
   able to work across a broad range of path characteristics, also
   including challenging situations, for example over trans-continental
   and/or satellite links.  Tests thus account for the following
   different latencies:

   1.  Very low latency: 0-1ms

   2.  Low latency: 50ms

   3.  High latency: 150ms

   4.  Extreme latency: 300ms

4.2.  End-to-end Loss

   Many paths in the Internet today are largely lossless but, with
   wireless networks and interference, towards remote regions, or in
   scenarios featuring high/fast mobility, media flows may exhibit
   substantial packet loss.  This variety needs to be reflected
   appropriately by the tests.

   To model a wide range of lossy links, the experiments can choose one
   of the following loss rates, the fractional loss is the ratio of
   packets lost and packets sent.

   1.  no loss: 0%

   2.  1%

   3.  5%

   4.  10%

   5.  20%

4.3.  Drop Tail Router Queue Length

   Routers should be configured to use Drop Trail queues in the
   experiments due to their (still) prevalent nature.  Experimentation
   with AQM schemes is encouraged but not mandatory.

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   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:

   1.  QoS-aware (or short): 70ms

   2.  Nominal: 300-500ms

   3.  Buffer-bloated: 1000-2000ms

   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

   QueueSize (in bytes) = QueueSize (in sec) x Throughput (in bps)/8

4.4.  Loss generation model

   Many models for generating packet loss are available, some yield
   correlated, others independent losses; losses can also be extracted
   from packet traces.  As a (simple) minimum loss model with minimal
   parameterization (i.e., the loss rate), independent random losses
   must be used in the evaluation.

   It is known that independent loss models may reflect reality poorly
   and hence more sophisticated loss models could be considered.
   Suitable models for correlated losses includes the Gilbert-Elliot
   model and losses generated by modeling a queue including its
   (different) drop behaviors.

4.5.  Jitter models

   This section defines jitter models for the purposes of this document.
   When jitter is to be applied to both the congestion controlled RTP
   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).

   Jitter is an overloaded term in communications.  It is is typically
   used to refer to 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 computed as the smoothed difference in
   packet arrival times relative to their respective expected arrival

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   times, which is particularly meaningful if the underlying packet
   delay variation was caused by a Gaussian random process.

   Because jitter is an overloaded term, we use the term Packet Delay
   Variation (PDV) instead 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.,

   Most PDV distributions in packet network systems are one-sided
   distributions, the measurement of which with a finite number of
   measurement samples results in one-sided histograms.  In the usual
   packet network transport case, there is typically one packet that
   transited the network with the minimum delay; a (large) number of
   packets transit the network within some (smaller) positive variation
   from this minimum delay, and a (small) number 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 rate adaptation designs for RTP congestion control.

   In this section we define two different bounded PDV characteristics,
   1) Random Bounded PDV and 2) Approximately Random Subject to No-
   Reordering Bounded PDV.

   The former, 1) Random Bounded PDV is presented for information only,
   while the latter, 2) Approximately Random Subject to No-Reordering
   Bounded PDV, must be used in the evaluation.

4.5.1.  Random Bounded PDV (RBPDV)

   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
   defined to be two standard deviations higher than the mean.

   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.

   We assume here that s(n) is the original source time of packet n and
   the post-jitter induced emission time, j(n), for packet n is:

   j(n) = {[z(n) + x_mean] + s(n)}.

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   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.

   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.

4.5.2.  Approximately Random Subject to No-Reordering Bounded PDV (NR-

   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.

   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

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   outlier) and thus is believed to be important for adaptation
   development for congestion controlled RTP streams.

4.5.3.  Recommended distribution

   Whether Random Bounded PDV or Approximately Random Subject to No-
   Reordering Bounded PDV, it is recommended that z(n) is distributed
   according to a truncated Gaussian for the above jitter models:

   z(n) ~ |max(min(N(0, std^2), N_STD * std), -N_STD * std)|

   where N(0, std^2) is the Gaussian distribution with zero mean and
   standard deviation std.  Recommended values:

   o  std = 5 ms

   o  N_STD = 3

5.  Traffic Models

5.1.  TCP traffic model

   Long-lived TCP flows will download data throughout the session and
   are expected to have infinite amount of data to send or receive.
   This roughly applies, for example, when downloading software

   Each short TCP flow is modeled as a sequence of file downloads
   interleaved with idle periods.  Not all short TCP flows start at the
   same time, i.e., some start in the ON state while others start in the
   OFF state.

   The short TCP flows can be modeled as follows: 30 connections start
   simultaneously fetching small (30-50 KB) amounts of data, evenly
   distributed.  This covers the case where the short TCP flows are
   fetching web page resources rather than video files.

   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.

   [These values were picked based on the data available at
   http://httparchive.org/interesting.php as of October 2015].

   Many different TCP congestion control schemes are deployed today.
   Therefore, experimentation with a range of different schemes,
   especially including CUBIC, is encouraged.  Experiments must document

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   in detail which congestion control schemes they tested against and
   which parameters were used.

5.2.  RTP Video model

   [RFC8593] describes two types of video traffic models for evaluating
   candidate algorithms for RTP congestion control.  The first model
   statistically characterizes the behavior of a video encoder, whereas
   the second model uses video traces.

   Sample video test sequences are available at: [xiph-seq] and
   [HEVC-seq].  The following two video streams are the recommended
   minimum for testing: Foreman and FourPeople.

5.3.  Background UDP

   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).

   Note that new transport protocols such as QUIC may use UDP but, due
   to their congestion control algorithms, will exhibit behavior
   conceptually similar in nature to TCP flows above and can thus be
   subsumed by the above, including the division into short- and long-
   lived flows.  As QUIC evolves independently of TCP congestion control
   algorithms, its future congestion control should be considered as
   competing traffic as appropriate.

6.  Security Considerations

   This document specifies evaluation criteria and parameters for
   assessing and comparing the performance of congestion control
   protocols and algorithms for real-time communication.  This memo
   itself is thus not subject to security considerations but the
   protocols and algorithms evaluated may be.  In particular, successful
   operation under all tests defined in this document may suffice for a
   comparative evaluation but must not be interpreted that the protocol
   is free of risks when deployed on the Internet as briefly described
   in the following by example.

   Such evaluations are expected to be carried out in controlled
   environments for limited numbers of parallel flows.  As such, these
   evaluations are by definition limited and will not be able to
   systematically consider possible interactions or very large groups of
   communicating nodes under all possible circumstances, so that careful

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   protocol design is advised to avoid incidentally contributing traffic
   that could lead to unstable networks, e.g., (local) congestion

   This specification focuses on assessing the regular operation of the
   protocols and algorithms under considerations.  It does not suggest
   checks against malicious use of the protocols -- by the sender, the
   receiver, or intermediate parties, e.g., through faked, dropped,
   replicated, or modified congestion signals.  It is up to the protocol
   specifications themselves to ensure that authenticity, integrity,
   and/or plausibility of received signals are checked and the
   appropriate actions (or non-actions) are taken.

7.  IANA Considerations

   There are no IANA impacts in this memo.

8.  Contributors

   The content and concepts within this document are a product of the
   discussion carried out in the Design Team.

   Michael Ramalho provided the text for the Jitter model.

9.  Acknowledgments

   Much of this document is derived from previous work on congestion
   control at the IETF.

   The authors would like to thank Harald Alvestrand, Anna Brunstrom,
   Luca De Cicco, Wesley Eddy, Lars Eggert, Kevin Gross, Vinayak Hegde,
   Randell Jesup, Mirja Kuehlewind, Karen Nielsen, Piers O'Hanlon, Colin
   Perkins, Michael Ramalho, Zaheduzzaman Sarker, Timothy B.
   Terriberry, Michael Welzl, Mo Zanaty, and Xiaoqing Zhu 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.

10.  References

10.1.  Normative References

              Jesup, R. and Z. Sarker, "Congestion Control Requirements
              for Interactive Real-Time Media", draft-ietf-rmcat-cc-
              requirements-09 (work in progress), December 2014.

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   [RFC3550]  Schulzrinne, H., Casner, S., Frederick, R., and V.
              Jacobson, "RTP: A Transport Protocol for Real-Time
              Applications", STD 64, RFC 3550, DOI 10.17487/RFC3550,
              July 2003, <https://www.rfc-editor.org/info/rfc3550>.

   [RFC3551]  Schulzrinne, H. and S. Casner, "RTP Profile for Audio and
              Video Conferences with Minimal Control", STD 65, RFC 3551,
              DOI 10.17487/RFC3551, July 2003,

   [RFC3611]  Friedman, T., Ed., Caceres, R., Ed., and A. Clark, Ed.,
              "RTP Control Protocol Extended Reports (RTCP XR)",
              RFC 3611, DOI 10.17487/RFC3611, November 2003,

   [RFC4585]  Ott, J., Wenger, S., Sato, N., Burmeister, C., and J. Rey,
              "Extended RTP Profile for Real-time Transport Control
              Protocol (RTCP)-Based Feedback (RTP/AVPF)", RFC 4585,
              DOI 10.17487/RFC4585, July 2006,

   [RFC5506]  Johansson, I. and M. Westerlund, "Support for Reduced-Size
              Real-Time Transport Control Protocol (RTCP): Opportunities
              and Consequences", RFC 5506, DOI 10.17487/RFC5506, April
              2009, <https://www.rfc-editor.org/info/rfc5506>.

   [RFC8083]  Perkins, C. and V. Singh, "Multimedia Congestion Control:
              Circuit Breakers for Unicast RTP Sessions", RFC 8083,
              DOI 10.17487/RFC8083, March 2017,

   [RFC8593]  Zhu, X., Mena, S., and Z. Sarker, "Video Traffic Models
              for RTP Congestion Control Evaluations", RFC 8593,
              DOI 10.17487/RFC8593, May 2019,

10.2.  Informative References

              HEVC, "Test Sequences",
              http://www.netlab.tkk.fi/~varun/test_sequences/ .

              Daede, T., Norkin, A., and I. Brailovskiy, "Video Codec
              Testing and Quality Measurement", draft-ietf-netvc-
              testing-09 (work in progress), January 2020.

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              Sarker, Z., Singh, V., Zhu, X., and M. Ramalho, "Test
              Cases for Evaluating RMCAT Proposals", draft-ietf-rmcat-
              eval-test-10 (work in progress), May 2019.

              Sarker, Z., Johansson, I., Zhu, X., Fu, J., Tan, W., and
              M. Ramalho, "Evaluation Test Cases for Interactive Real-
              Time Media over Wireless Networks", draft-ietf-rmcat-
              wireless-tests-08 (work in progress), July 2019.

   [RFC5033]  Floyd, S. and M. Allman, "Specifying New Congestion
              Control Algorithms", BCP 133, RFC 5033,
              DOI 10.17487/RFC5033, August 2007,

   [RFC5166]  Floyd, S., Ed., "Metrics for the Evaluation of Congestion
              Control Mechanisms", RFC 5166, DOI 10.17487/RFC5166, March
              2008, <https://www.rfc-editor.org/info/rfc5166>.

              Daede, T., "Video Test Media Set",
              https://people.xiph.org/~tdaede/sets/ .

Appendix A.  Change Log

   Note to the RFC-Editor: please remove this section prior to
   publication as an RFC.

A.1.  Changes in draft-ietf-rmcat-eval-criteria-07

   Updated the draft according to the discussion at IETF-101.

   o  Updated the discussion on fairness.  Thanks to Xiaoqing Zhu for
      providing text.

   o  Fixed a simple loss model and provided pointers to more
      sophisticated ones.

   o  Fixed the choice of the jitter model.

A.2.  Changes in draft-ietf-rmcat-eval-criteria-06

   o  Updated Jitter.

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A.3.  Changes in draft-ietf-rmcat-eval-criteria-05

   o  Improved text surrounding wireless tests, video sequences, and
      short-TCP model.

A.4.  Changes in draft-ietf-rmcat-eval-criteria-04

   o  Removed the guidelines section, as most of the sections are now
      covered: wireless tests, video model, etc.

   o  Improved Short TCP model based on the suggestion to use

A.5.  Changes in draft-ietf-rmcat-eval-criteria-03

   o  Keep-alive version.

   o  Moved link parameters and traffic models from eval-test

A.6.  Changes in draft-ietf-rmcat-eval-criteria-02

   o  Incorporated fairness test as a working test.

   o  Updated text on mimimum evaluation requirements.

A.7.  Changes in draft-ietf-rmcat-eval-criteria-01

   o  Removed Appendix B.

   o  Removed Section on Evaluation Parameters.

A.8.  Changes in draft-ietf-rmcat-eval-criteria-00

   o  Updated references.

   o  Resubmitted as WG draft.

A.9.  Changes in draft-singh-rmcat-cc-eval-04

   o  Incorporate feedback from IETF 87, Berlin.

   o  Clarified metrics: convergence time, bandwidth utilization.

   o  Changed fairness criteria to fairness test.

   o  Added measuring pre- and post-repair loss.

   o  Added open issue of measuring video quality to appendix.

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   o  clarified use of DropTail and AQM.

   o  Updated text in "Minimum Requirements for Evaluation"

A.10.  Changes in draft-singh-rmcat-cc-eval-03

   o  Incorporate the discussion within the design team.

   o  Added a section on evaluation parameters, it describes the flow
      and network characteristics.

   o  Added Appendix with self-fairness experiment.

   o  Changed bottleneck parameters from a proposal to an example set.


A.11.  Changes in draft-singh-rmcat-cc-eval-02

   o  Added scenario descriptions.

A.12.  Changes in draft-singh-rmcat-cc-eval-01

   o  Removed QoE metrics.

   o  Changed stability to steady-state.

   o  Added measuring impact against few and many flows.

   o  Added guideline for idle and data-limited periods.

   o  Added reference to TCP evaluation suite in example evaluation

Authors' Addresses

   Varun Singh
   Runeberginkatu 4c A 4
   Helsinki  00100

   Email: varun.singh@iki.fi
   URI:   https://www.callstats.io/about

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   Joerg Ott
   Technical University of Munich
   Faculty of Informatics
   Boltzmannstrasse 3
   Garching bei Muenchen, DE  85748

   Email: ott@in.tum.de

   Stefan Holmer
   Kungsbron 2
   Stockholm  11122

   Email: holmer@google.com

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