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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
(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
to this document. Code Components extracted from this document must
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.
Singh & Ott Expires August 30, 2013 [Page 1]
Internet-Draft Evaluating Congestion Control for RMCAT February 2013
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/
Singh & Ott Expires August 30, 2013 [Page 11]
Internet-Draft Evaluating Congestion Control for RMCAT February 2013
Joerg Ott
Aalto University
School of Electrical Engineering
Otakaari 5 A
Espoo, FIN 02150
Finland
Email: jo@comnet.tkk.fi
Singh & Ott Expires August 30, 2013 [Page 12]