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Evaluating Congestion Control for Interactive Real-time Media
draft-ietf-rmcat-eval-criteria-03

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This is an older version of an Internet-Draft that was ultimately published as RFC 8868.
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Authors Varun Singh , Joerg Ott
Last updated 2015-09-10 (Latest revision 2015-03-09)
Replaces draft-singh-rmcat-cc-eval
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draft-ietf-rmcat-eval-criteria-03
RMCAT WG                                                        V. Singh
Internet-Draft                                                    J. Ott
Intended status: Informational                          Aalto University
Expires: September 11, 2015                               March 10, 2015

     Evaluating Congestion Control for Interactive Real-time Media
                   draft-ietf-rmcat-eval-criteria-03

Abstract

   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
   Task Force (IETF).  Note that other groups may also distribute
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   Drafts is at http://datatracker.ietf.org/drafts/current/.

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

   This Internet-Draft will expire on September 11, 2015.

Copyright Notice

   Copyright (c) 2015 IETF Trust and the persons identified as the
   document authors.  All rights reserved.

   This document is subject to BCP 78 and the IETF Trust's Legal
   Provisions Relating to IETF Documents
   (http://trustee.ietf.org/license-info) in effect on the date of
   publication of this document.  Please review these documents
   carefully, as they describe your rights and restrictions with respect
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   the Trust Legal Provisions and are provided without warranty as
   described in the Simplified BSD License.

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Table of Contents

   1.  Introduction  . . . . . . . . . . . . . . . . . . . . . . . .   3
   2.  Terminology . . . . . . . . . . . . . . . . . . . . . . . . .   3
   3.  Metrics . . . . . . . . . . . . . . . . . . . . . . . . . . .   3
     3.1.  RTP Log Format  . . . . . . . . . . . . . . . . . . . . .   5
   4.  Guidelines  . . . . . . . . . . . . . . . . . . . . . . . . .   5
     4.1.  Avoiding Congestion Collapse  . . . . . . . . . . . . . .   5
     4.2.  Stability . . . . . . . . . . . . . . . . . . . . . . . .   5
     4.3.  Media Traffic . . . . . . . . . . . . . . . . . . . . . .   5
     4.4.  Start-up Behaviour  . . . . . . . . . . . . . . . . . . .   6
     4.5.  Diverse Environments  . . . . . . . . . . . . . . . . . .   6
     4.6.  Varying Path Characteristics  . . . . . . . . . . . . . .   6
     4.7.  Reacting to Transient Events or Interruptions . . . . . .   7
     4.8.  Fairness With Similar Cross-Traffic . . . . . . . . . . .   7
     4.9.  Impact on Cross-Traffic . . . . . . . . . . . . . . . . .   7
     4.10. Extensions to RTP/RTCP  . . . . . . . . . . . . . . . . .   7
   5.  List of Network Parameters  . . . . . . . . . . . . . . . . .   7
     5.1.  One-way Propagation Delay . . . . . . . . . . . . . . . .   7
     5.2.  End-to-end Loss . . . . . . . . . . . . . . . . . . . . .   8
     5.3.  DropTail Router Queue Length  . . . . . . . . . . . . . .   8
     5.4.  Loss generation model . . . . . . . . . . . . . . . . . .   8
     5.5.  Jitter models . . . . . . . . . . . . . . . . . . . . . .   9
       5.5.1.  Random Bounded PDV (RBPDV)  . . . . . . . . . . . . .   9
       5.5.2.  Approximately Random Subject to No-Reordering Bounded
               PDV         (NR-RPVD) . . . . . . . . . . . . . . . .  10
   6.  Traffic Models  . . . . . . . . . . . . . . . . . . . . . . .  11
     6.1.  TCP taffic model  . . . . . . . . . . . . . . . . . . . .  11
     6.2.  RTP Video model . . . . . . . . . . . . . . . . . . . . .  12
   7.  Security Considerations . . . . . . . . . . . . . . . . . . .  12
   8.  IANA Considerations . . . . . . . . . . . . . . . . . . . . .  12
   9.  Contributors  . . . . . . . . . . . . . . . . . . . . . . . .  12
   10. Acknowledgements  . . . . . . . . . . . . . . . . . . . . . .  12
   11. References  . . . . . . . . . . . . . . . . . . . . . . . . .  12
     11.1.  Normative References . . . . . . . . . . . . . . . . . .  12
     11.2.  Informative References . . . . . . . . . . . . . . . . .  13
   Appendix A.  Application Trade-off  . . . . . . . . . . . . . . .  14
     A.1.  Measuring Quality . . . . . . . . . . . . . . . . . . . .  14
   Appendix B.  Change Log . . . . . . . . . . . . . . . . . . . . .  14
     B.1.  Changes in draft-ietf-rmcat-eval-criteria-02  . . . . . .  14
     B.2.  Changes in draft-ietf-rmcat-eval-criteria-02  . . . . . .  14
     B.3.  Changes in draft-ietf-rmcat-eval-criteria-01  . . . . . .  14
     B.4.  Changes in draft-ietf-rmcat-eval-criteria-00  . . . . . .  15
     B.5.  Changes in draft-singh-rmcat-cc-eval-04 . . . . . . . . .  15
     B.6.  Changes in draft-singh-rmcat-cc-eval-03 . . . . . . . . .  15
     B.7.  Changes in draft-singh-rmcat-cc-eval-02 . . . . . . . . .  15
     B.8.  Changes in draft-singh-rmcat-cc-eval-01 . . . . . . . . .  15
   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 [I-D.ietf-avtcore-rtp-circuit-breakers].

   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 [I-D.ietf-rmcat-eval-test] (Basic Test Cases).  The
   results of the evaluation are not expected to be included within the
   internet-draft but should be cited in the document.

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

3.  Metrics

   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 are
   calculated based on the information in the packet logs:

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

   2.   Packets sent, Packets received

   3.   Bytes sent, bytes received

   4.   Packet delay

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

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

   7.   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:
        1.  Does not trigger the circuit breaker.
        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.
        3.  RTT should not grow by a factor of 3 for the existing flows
        when a new flow is added.
        For example, see the test scenarios described in
        [I-D.ietf-rmcat-eval-test].

   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 an RMCAT flow is by itself or
        competing with similar cross-traffic.

   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.

   [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? ]

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3.1.  RTP Log Format

   The log file is tab or comma separated containing the following
   details:

           Send or receive timestamp (unix)
           RTP payload type
           SSRC
           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-
   resilience).

4.  Guidelines

   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:

4.1.  Avoiding Congestion Collapse

   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
   [I-D.ietf-avtcore-rtp-circuit-breakers] is engaged.

   Does the congestion control propose any changes to (or diverge from)
   the circuit breaker conditions defined in
   [I-D.ietf-avtcore-rtp-circuit-breakers].

4.2.  Stability

   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.

4.3.  Media Traffic

   The congestion control algorithm should be assessed with different
   types of media behavior, i.e., the media should contain idle and

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   data-limited periods.  For example, periods of silence for audio,
   varying amount of motion for video, or bursty nature of I-frames.

   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.

4.4.  Start-up Behaviour

   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.

4.5.  Diverse Environments

   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.

4.6.  Varying Path Characteristics

   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.

   In an experiment, if the media only flows in a single direction, the
   feedback path should also be tested with varying amounts of
   impairments.

   The main motivation for the previous and current criteria is to
   identify situations in which the proposed congestion control is less
   performant.

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4.7.  Reacting to Transient Events or Interruptions

   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.

4.8.  Fairness With Similar Cross-Traffic

   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.

4.9.  Impact on Cross-Traffic

   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.

   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.

4.10.  Extensions to RTP/RTCP

   The congestion control algorithm should indicate if any protocol
   extensions are required to implement it and should carefully describe
   the impact of the extension.

5.  List of Network Parameters

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

5.1.  One-way Propagation Delay

   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:

   1.  Very low latency: 0-1ms

   2.  Low latency: 50ms

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   3.  High latency: 150ms

   4.  Extreme latency: 300ms

5.2.  End-to-end Loss

   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.

   1.  no loss: 0%

   2.  1%

   3.  5%

   4.  10%

   5.  20%

5.3.  DropTail Router Queue Length

   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
   bytes:

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

5.4.  Loss generation model

   [Editor's note : 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).]

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5.5.  Jitter models

   This section defines jitter model 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).

   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.

   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).

   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 transits 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.

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

5.5.1.  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
   idefined to be two standard deviations higher than the mean.

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   Since we are typically interested in the distribution relative to the
   mean delay packet, we define the zero mean PVD 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 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.

   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.

5.5.2.  Approximately Random Subject to No-Reordering Bounded PDV (NR-
        RPVD)

   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

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   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.

   [Editor's Note: It may require to define test distributions as well.
   Example test distribution may include-

   1 - Two-sided: Uniform PDV Distribution.  Two quantities to define:
   x_min and x_max.

   2 - Two-sided: Truncated Gaussian PDV Distribution.  Four quantities
   to define: the appropriate x_min and x_max for test (e.g., +/- two
   sigma values), the standard deviation and the mean.

   3 - One Sided: TBD]

6.  Traffic Models

6.1.  TCP taffic model

   Long-lived TCP flows will download data throughout the session and
   are expected to have infinite amount of data to send or receive.

   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.

   The short TCP flows can be modelled in two ways, 1) 100s of flows
   fetching small (5-20 KB) amounts of data, or 2) 10s of flows fetching
   slightly larger (100-1000KB) amounts of data.

   The idle period is typically derived from an exponential distribution
   with the mean value of 10 seconds.

   [Open issue: short-lived/bursty TCP cross-traffic parameters are
   still to be agreed upon].

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6.2.  RTP Video model

   [I-D.zhu-rmcat-video-traffic-source] 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.

7.  Security Considerations

   Security issues have not been discussed in this memo.

8.  IANA Considerations

   There are no IANA impacts in this memo.

9.  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.

10.  Acknowledgements

   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,
   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.

11.  References

11.1.  Normative References

   [RFC3550]  Schulzrinne, H., Casner, S., Frederick, R., and V.
              Jacobson, "RTP: A Transport Protocol for Real-Time
              Applications", STD 64, RFC 3550, July 2003.

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

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   [RFC3611]  Friedman, T., Caceres, R., and A. Clark, "RTP Control
              Protocol Extended Reports (RTCP XR)", RFC 3611, 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, July
              2006.

   [RFC5506]  Johansson, I. and M. Westerlund, "Support for Reduced-Size
              Real-Time Transport Control Protocol (RTCP): Opportunities
              and Consequences", RFC 5506, April 2009.

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

   [I-D.ietf-avtcore-rtp-circuit-breakers]
              Perkins, C. and V. Singh, "Multimedia Congestion Control:
              Circuit Breakers for Unicast RTP Sessions", draft-ietf-
              avtcore-rtp-circuit-breakers-09 (work in progress), March
              2015.

11.2.  Informative References

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

   [RFC5166]  Floyd, S., "Metrics for the Evaluation of Congestion
              Control Mechanisms", RFC 5166, March 2008.

   [RFC5681]  Allman, M., Paxson, V., and E. Blanton, "TCP Congestion
              Control", RFC 5681, September 2009.

   [I-D.ietf-rmcat-eval-test]
              Sarker, Z., Singh, V., Zhu, X., and M. Ramalho, "Test
              Cases for Evaluating RMCAT Proposals", draft-ietf-rmcat-
              eval-test-00 (work in progress), August 2014.

   [I-D.zhu-rmcat-video-traffic-source]
              Zhu, X., Cruz, S., and Z. Sarker, "Modeling Video Traffic
              Sources for RMCAT Evaluations", draft-zhu-rmcat-video-
              traffic-source-00 (work in progress), October 2014.

   [SA4-EVAL]
              R1-081955, 3GPP., "LTE Link Level Throughput Data for SA4
              Evaluation Framework", 3GPP R1-081955, 5 2008.

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   [SA4-LR]   S4-050560, 3GPP., "Error Patterns for MBMS Streaming over
              UTRAN and GERAN", 3GPP S4-050560, 5 2008.

   [TCP-eval-suite]
              Lachlan, A., Marcondes, C., Floyd, S., Dunn, L., Guillier,
              R., Gang, W., Eggert, L., Ha, S., and I. Rhee, "Towards a
              Common TCP Evaluation Suite", Proc. PFLDnet. 2008, August
              2008.

Appendix A.  Application Trade-off

   Application trade-off is yet to be defined. see RMCAT requirements
   [I-D.ietf-rmcat-cc-requirements] document.  Perhaps each experiment
   should define the application's expectation or trade-off.

A.1.  Measuring Quality

   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.

Appendix B.  Change Log

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

B.1.  Changes in draft-ietf-rmcat-eval-criteria-02

   o  Keep-alive version.

   o  Moved link parameters and traffic models from eval-test

B.2.  Changes in draft-ietf-rmcat-eval-criteria-02

   o  Incorporated fairness test as a working test.

   o  Updated text on mimimum evaluation requirements.

B.3.  Changes in draft-ietf-rmcat-eval-criteria-01

   o  Removed Appendix B.

   o  Removed Section on Evaluation Parameters.

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B.4.  Changes in draft-ietf-rmcat-eval-criteria-00

   o  Updated references.

   o  Resubmitted as WG draft.

B.5.  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.

   o  clarified use of DropTail and AQM.

   o  Updated text in "Minimum Requirements for Evaluation"

B.6.  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.

   o

B.7.  Changes in draft-singh-rmcat-cc-eval-02

   o  Added scenario descriptions.

B.8.  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.

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   o  Added reference to TCP evaluation suite in example evaluation
      scenarios.

Authors' Addresses

   Varun Singh
   Aalto University
   School of Electrical Engineering
   Otakaari 5 A
   Espoo, FIN  02150
   Finland

   Email: varun@comnet.tkk.fi
   URI:   http://www.netlab.tkk.fi/~varun/

   Joerg Ott
   Aalto University
   School of Electrical Engineering
   Otakaari 5 A
   Espoo, FIN  02150
   Finland

   Email: jo@comnet.tkk.fi

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