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
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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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