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Versions: 00 01 02 03 04 draft-ietf-rmcat-eval-criteria

RMCAT WG                                                        V. Singh
Internet-Draft                                                    J. Ott
Intended status: Informational                          Aalto University
Expires: August 30, 2013                               February 26, 2013


     Evaluating Congestion Control for Interactive Real-time Media
                    draft-singh-rmcat-cc-eval-02.txt

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
   working documents as Internet-Drafts.  The list of current Internet-
   Drafts is at http://datatracker.ietf.org/drafts/current/.

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

   This Internet-Draft will expire on August 30, 2013.

Copyright Notice

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

   This document is subject to BCP 78 and the IETF Trust's Legal
   Provisions Relating to IETF Documents
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   include Simplified BSD License text as described in Section 4.e of
   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  . . . . . . . . . . . . . . . . . . . . . . . .   2
   2.  Terminology . . . . . . . . . . . . . . . . . . . . . . . . .   3
   3.  Metrics . . . . . . . . . . . . . . . . . . . . . . . . . . .   3
   4.  Guidelines  . . . . . . . . . . . . . . . . . . . . . . . . .   4
     4.1.  Avoiding Congestion Collapse  . . . . . . . . . . . . . .   4
     4.2.  Stability . . . . . . . . . . . . . . . . . . . . . . . .   4
     4.3.  Media Traffic . . . . . . . . . . . . . . . . . . . . . .   5
     4.4.  Diverse Environments  . . . . . . . . . . . . . . . . . .   5
     4.5.  Varying Path Characteristics  . . . . . . . . . . . . . .   5
     4.6.  Reacting to Transient Events or Interruptions . . . . . .   5
     4.7.  Fairness With Similar Cross-Traffic . . . . . . . . . . .   6
     4.8.  Impact on Cross-Traffic . . . . . . . . . . . . . . . . .   6
     4.9.  Extensions to RTP/RTCP  . . . . . . . . . . . . . . . . .   6
   5.  Minimum Requirements for Evaluation . . . . . . . . . . . . .   6
   6.  Example Evaluation Scenarios  . . . . . . . . . . . . . . . .   6
     6.1.  [S1] RTP flow on a fixed link . . . . . . . . . . . . . .   7
     6.2.  [S2] RTP flow on a variable capacity link . . . . . . . .   7
     6.3.  [S3] Fairness to RTP flows running the same congestion
           control algorithm (self-fairness) . . . . . . . . . . . .   8
     6.4.  [S4 and S5] Competing with short and long TCP flows . . .   8
   7.  Status of Proposals . . . . . . . . . . . . . . . . . . . . .   9
   8.  Security Considerations . . . . . . . . . . . . . . . . . . .   9
   9.  IANA Considerations . . . . . . . . . . . . . . . . . . . . .   9
   10. Acknowledgements  . . . . . . . . . . . . . . . . . . . . . .   9
   11. References  . . . . . . . . . . . . . . . . . . . . . . . . .   9
     11.1.  Normative References . . . . . . . . . . . . . . . . . .   9
     11.2.  Informative References . . . . . . . . . . . . . . . . .  10
   Appendix A.  Change Log . . . . . . . . . . . . . . . . . . . . .  11
     A.1.  Changes in draft-singh-rmcat-cc-eval-02 . . . . . . . . .  11
     A.2.  Changes in draft-singh-rmcat-cc-eval-01 . . . . . . . . .  11
   Authors' Addresses  . . . . . . . . . . . . . . . . . . . . . . .  11

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.jesup-rtp-congestion-reqs]).  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].








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   The guidelines proposed in the document are intended to prevent a
   congestion collapse, promote fair capacity usage and optimize the
   media flow's throughput, and quality.  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 and the working group should expect a
   thorough scientific study to make its decision.  The results of the
   evaluation are not expected to be included within the internet-draft
   but should be cited in the document.

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

   [RFC5166] describes the basic metrics for congestion control.
   Metrics that are important to interactive multimedia are:

   o  Throughput: (Sending Rate, Receiving Rate, Goodput)

   o  Minimizing oscillations in encoding rate (stability)

   o  Reactivity to transient events

   o  Packet loss and discard rate

   o  Users' quality of experience

   [Editor's Note: measurement interval and statistical measures (min,
   max, mean, median, standard deviation and variance) are yet to be
   specified.]

   Section 2.1 of [RFC5166] discusses the tradeoff between throughput,
   delay and loss.

      (i) Bandwidth Utilization: is the ratio of the encoding rate to
      the (available) end-to-end path capacity.

      *  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.



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      *  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.

      *  Steady-state: is the period of time when the endpoint's
         encoding rate is relatively stable, i.e., the bandwidth
         utilization is constant.

      (ii) Packet Loss and Discard Rate.

      (iii) Fair Share.

      [Editor's Note: This metric should match the ones defined in the
      RMCAT requirements [I-D.jesup-rtp-congestion-reqs] document.]

      (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 [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).

      [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].

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

   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 too often or by too much may adversely affect the
   users' quality of experience.



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

4.4.  Diverse Environments

   The congestion control algorithm should be assessed in heterogeneous
   environments, containing both wired and wireless paths.  Examples of
   wireless access technologies are: 802.11x, GPRS, HSPA, or LTE.  One
   of the main challenges of the wireless environments is the inability
   to distinguish congestion induced loss from transmission (bit-error)
   loss.  Congestion control algorithms may incorrectly identify
   transmission loss as congestion loss and reduce the media encoding
   rate 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.

4.5.  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 bottle-neck link and a
   router's queue length.  The main motivation for the previous and
   current criteria is to determine under which circumstances will the
   proposed congestion control algorithm break down and also determine
   the operational range of the algorithm.

   [Editor's Note: Different types of queueing mechanisms?  Random Early
   Detection or only DropTail?].

4.6.  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.









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4.7.  Fairness With Similar Cross-Traffic

   The congestion control algorithm should be evaluated when competing
   with other RTP flows using the same congestion control algorithm.
   The proposal should highlight the bottleneck capacity share of each
   RTP flow.

4.8.  Impact on Cross-Traffic

   [Editor's Note: There was discussion about removing this guideline,
   however, no decision was made [I-D.jesup-rtp-congestion-reqs].]

   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.  In the latter case the
   proposed congestion control for RTP should be as aggressive as
   standard TCP [RFC5681].

   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.

4.9.  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.  Minimum Requirements for Evaluation

   [Editor's Note: If needed, a minimum evaluation criteria can be based
   on the above guidelines]

6.  Example Evaluation Scenarios

   In the scenarios listed below, all RTP flows are bi-directional and
   point-to-point.

   Unless specified, the following parameters are used in each scenario:

   o  Video Start Rate: 128 kbps

   o  Maximum end-to-end delay: 300ms, packets arriving after this are
      discarded

   o  Video Frame rate: 15



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   o  Audio packetization interval: 20ms

   o  MTU: 1450 bytes

   o  [Editor's Note: the numbers in this section are TBD]

   Topology:

   o  Dumbbell, the endpoint is connected to the bottleneck link via an
      access links.  The bottleneck may be shared by multiple endpoints.

   o  Parking lot: there are three bottleneck links arranged
      horizontally, these links are connected by access links.  In this
      case, flows may share different bottleneck links.

   [Editor's note: Should the queue-size be specified as well?].

6.1.  [S1] RTP flow on a fixed link

   This scenario evaluates the ramp-up to the bottleneck capacity and
   the stability of the proposed congestion control algorithm.

   This scenario uses the dumbbell topology and both the access link can
   be ADSL (500kbps uplink, 256 downlink, 2ms one-way delay) or WLAN
   (54Mbps, 2ms one-way delay, 2-5% packet loss rate and link layer re-
   transmissions).

   The bottleneck link can have one of the following capacities:
   500kbps, 1Mbps, 5Mbps and link delay: 10ms, 50ms, 120ms.

   Each congestion control algorithm should plot the variation of the
   sending rate against time, also plot the instances of packets losses.
   Additionally, measure the time taken for the sending rate to reach
   the end-to-end capacity (average and standard deviation over 10
   simulation runs).

6.2.  [S2] RTP flow on a variable capacity link

   This scenario evaluates the reactivity of the proposed congestion
   control algorithm to transient network events due to interference and
   handovers in mobile environments.

   This scenario uses the dumbbell topology, and both end-points use 3G/
   LTE access.  Sample 3G/LTE (uplink and downlink) bandwidth traces are
   available at [SA4-EVAL], loss patterns at [SA4-LR] and the link
   delay: 30ms, 80ms.  The bottleneck link can have one of the following
   capacities: 500kbps, 5Mbps and link delay: 20ms.




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   Each congestion control algorithm should plot the variation of the
   sending rate against time, also plot the instances of packets losses.

6.3.  [S3] Fairness to RTP flows running the same congestion control
      algorithm (self-fairness)

   This scenario shows if the proposed algorithm can share the
   bottleneck link equitably, irrespective of number of flows.

   In this scenario there is more than one endpoint connected to the
   bottleneck link.

      (a) All the access links have the same link characteristics and
      start at the same time (see [S1]).  The bottleneck link can have
      one of the following link capacity: 500kbpsm 5Mbpps and link delay
      20ms.

      (b) The access links have different link characteristics [See S1]
      but start at the same time.

      (c) An RTP flow is added at 10s intervals (upto 5 flows), the late
      arriving flows have increasing access link delay (0, 5, 10, 20,
      50ms).  The bottleneck link can have one of the following
      capacities: 1Mbps, 10Mbps and link delay: 10ms, 50ms, 120ms.

   [Parking lot topology simulation: TBD]

6.4.  [S4 and S5] Competing with short and long TCP flows

   [Editor's Note: Remove these scenarios?]

   [S4] Competing with long-lived TCP flows: In this scenario the
   proposed algorithm is expected to be TCP-friendly, i.e., it should
   neither starve out the competing TCP flows (causing a congestion
   collapse) nor should it be starved out by TCP.

   [S5] Competing with short TCP flows: Depending on the level of
   statistical multiplexing on the bottleneck link, the proposed
   algorithm may behave differently.  If there are a few short TCP flows
   then the proposed algorithm may observe these flows as transient
   events and let them complete quickly.  Alternatively, if there are
   many short flows then the proposed algorithm may have to compete with
   the flows as if they were long lived TCP flows.

   [TCP-eval-suite] contains examples of TCP traffic load and scenario
   settings.





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   [Editor's Note: definition of many and few short TCP flows may depend
   on the bottleneck link capacity.]

   [Editor's Note: clarify if media packets are generated using a
   traffic generator.]

7.  Status of Proposals

   Congestion control algorithms are expected to be published as
   "Experimental" documents until they are shown to be safe to deploy.
   An algorithm published as a draft should be experimented in
   simulation, or a controlled environment (testbed) to show its
   applicability.  Every congestion control algorithm should include a
   note describing the environments in which the algorithm is tested and
   safe to deploy.  It is possible that an algorithm is not recommended
   for certain environments or perform sub-optimally for the user.

   [Editor's Note: Should there be a distinction between "Informational"
   and "Experimental" drafts for congestion control algorithms in RMCAT.
   [RFC5033] describes Informational proposals as algorithms that are
   not safe for deployment but are proposals to experiment with in
   simulation/testbeds.  While Experimental algorithms are ones that are
   deemed safe in some environments but require a more thorough
   evaluation (from the community).]

8.  Security Considerations

   Security issues have not been discussed in this memo.

9.  IANA Considerations

   There are no IANA impacts in this memo.

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, Luca De Cicco,
   Wesley Eddy, Lars Eggert, Vinayak Hegde, Stefan Holmer, Randell
   Jesup, Piers O'Hanlon, Colin Perkins, Timothy B.  Terriberry, Michael
   Welzl, and Sarker Zaheduzzaman for providing valuable feedback on
   earlier versions of this draft.

11.  References

11.1.  Normative References




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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, July 2003.

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

   [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.jesup-rtp-congestion-reqs]
              Jesup, R. and H. Alvestrand, "Congestion Control
              Requirements For Real Time Media", draft-jesup-rtp-
              congestion-reqs-00 (work in progress), March 2012.

   [I-D.ietf-avtcore-rtp-circuit-breakers]
              Perkins, C. and V. Singh, "RTP Congestion Control: Circuit
              Breakers for Unicast Sessions", draft-ietf-avtcore-rtp-
              circuit-breakers-01 (work in progress), October 2012.

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.

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

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



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   [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.  Change Log

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

A.1.  Changes in draft-singh-rmcat-cc-eval-02

   o  Added scenario descriptions.

A.2.  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
      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/












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

   Email: jo@comnet.tkk.fi










































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