draft-ietf-rmcat-eval-criteria-14.txt		rfc8868.txt
	RMCAT WG V. Singh		Internet Engineering Task Force (IETF) V. Singh
	Internet-Draft callstats.io		Request for Comments: 8868 callstats.io
	Intended status: Informational J. Ott		Category: Informational J. Ott
	Expires: September 20, 2020 Technical University of Munich		ISSN: 2070-1721 Technical University of Munich
	S. Holmer		S. Holmer
	Google		Google
	March 19, 2020		January 2021
	Evaluating Congestion Control for Interactive Real-time Media		Evaluating Congestion Control for Interactive Real-Time Media
	draft-ietf-rmcat-eval-criteria-14
	Abstract		Abstract
	The Real-time Transport Protocol (RTP) is used to transmit media in		The Real-Time Transport Protocol (RTP) is used to transmit media in
	telephony and video conferencing applications. This document		telephony and video conferencing applications. This document
	describes the guidelines to evaluate new congestion control		describes the guidelines to evaluate new congestion control
	algorithms for interactive point-to-point real-time media.		algorithms for interactive point-to-point real-time media.
	Status of This Memo		Status of This Memo
	This Internet-Draft is submitted in full conformance with the		This document is not an Internet Standards Track specification; it is
	provisions of BCP 78 and BCP 79.		published for informational purposes.
	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 https://datatracker.ietf.org/drafts/current/.
	Internet-Drafts are draft documents valid for a maximum of six months		This document is a product of the Internet Engineering Task Force
	and may be updated, replaced, or obsoleted by other documents at any		(IETF). It represents the consensus of the IETF community. It has
	time. It is inappropriate to use Internet-Drafts as reference		received public review and has been approved for publication by the
	material or to cite them other than as "work in progress."		Internet Engineering Steering Group (IESG). Not all documents
			approved by the IESG are candidates for any level of Internet
			Standard; see Section 2 of RFC 7841.
	This Internet-Draft will expire on September 20, 2020.		Information about the current status of this document, any errata,
			and how to provide feedback on it may be obtained at
			https://www.rfc-editor.org/info/rfc8868.
	Copyright Notice		Copyright Notice
	Copyright (c) 2020 IETF Trust and the persons identified as the		Copyright (c) 2021 IETF Trust and the persons identified as the
	document authors. All rights reserved.		document authors. All rights reserved.
	This document is subject to BCP 78 and the IETF Trust's Legal		This document is subject to BCP 78 and the IETF Trust's Legal
	Provisions Relating to IETF Documents		Provisions Relating to IETF Documents
	(https://trustee.ietf.org/license-info) in effect on the date of		(https://trustee.ietf.org/license-info) in effect on the date of
	publication of this document. Please review these documents		publication of this document. Please review these documents
	carefully, as they describe your rights and restrictions with respect		carefully, as they describe your rights and restrictions with respect
	to this document. Code Components extracted from this document must		to this document. Code Components extracted from this document must
	include Simplified BSD License text as described in Section 4.e of		include Simplified BSD License text as described in Section 4.e of
	the Trust Legal Provisions and are provided without warranty as		the Trust Legal Provisions and are provided without warranty as
	described in the Simplified BSD License.		described in the Simplified BSD License.
	Table of Contents		Table of Contents
	1. Introduction . . . . . . . . . . . . . . . . . . . . . . . . 3		1. Introduction
	2. Terminology . . . . . . . . . . . . . . . . . . . . . . . . . 3		2. Terminology
	3. Metrics . . . . . . . . . . . . . . . . . . . . . . . . . . . 4		3. Metrics
	3.1. RTP Log Format . . . . . . . . . . . . . . . . . . . . . 5		3.1. RTP Log Format
	4. List of Network Parameters . . . . . . . . . . . . . . . . . 6		4. List of Network Parameters
	4.1. One-way Propagation Delay . . . . . . . . . . . . . . . . 6		4.1. One-Way Propagation Delay
	4.2. End-to-end Loss . . . . . . . . . . . . . . . . . . . . . 6		4.2. End-to-End Loss
	4.3. Drop Tail Router Queue Length . . . . . . . . . . . . . . 7		4.3. Drop-Tail Router Queue Length
	4.4. Loss generation model . . . . . . . . . . . . . . . . . . 7		4.4. Loss Generation Model
	4.5. Jitter models . . . . . . . . . . . . . . . . . . . . . . 7		4.5. Jitter Models
	4.5.1. Random Bounded PDV (RBPDV) . . . . . . . . . . . . . 8		4.5.1. Random Bounded PDV (RBPDV)
	4.5.2. Approximately Random Subject to No-Reordering Bounded		4.5.2. Approximately Random Subject to No-Reordering Bounded
	PDV (NR-RPVD) . . . . . . . . . . . . . . . . 9		PDV (NR-BPDV)
	4.5.3. Recommended distribution . . . . . . . . . . . . . . 10		4.5.3. Recommended Distribution
	5. Traffic Models . . . . . . . . . . . . . . . . . . . . . . . 10		5. Traffic Models
	5.1. TCP traffic model . . . . . . . . . . . . . . . . . . . . 10		5.1. TCP Traffic Model
	5.2. RTP Video model . . . . . . . . . . . . . . . . . . . . . 11		5.2. RTP Video Model
	5.3. Background UDP . . . . . . . . . . . . . . . . . . . . . 11		5.3. Background UDP
	6. Security Considerations . . . . . . . . . . . . . . . . . . . 12		6. Security Considerations
	7. IANA Considerations . . . . . . . . . . . . . . . . . . . . . 12		7. IANA Considerations
	8. Contributors . . . . . . . . . . . . . . . . . . . . . . . . 12		8. References
	9. Acknowledgments . . . . . . . . . . . . . . . . . . . . . . . 12		8.1. Normative References
	10. References . . . . . . . . . . . . . . . . . . . . . . . . . 13		8.2. Informative References
	10.1. Normative References . . . . . . . . . . . . . . . . . . 13		Contributors
	10.2. Informative References . . . . . . . . . . . . . . . . . 14		Acknowledgments
	Appendix A. Change Log . . . . . . . . . . . . . . . . . . . . . 15		Authors' Addresses
	A.1. Changes in draft-ietf-rmcat-eval-criteria-07 . . . . . . 15
	A.2. Changes in draft-ietf-rmcat-eval-criteria-06 . . . . . . 15
	A.3. Changes in draft-ietf-rmcat-eval-criteria-05 . . . . . . 15
	A.4. Changes in draft-ietf-rmcat-eval-criteria-04 . . . . . . 15
	A.5. Changes in draft-ietf-rmcat-eval-criteria-03 . . . . . . 15
	A.6. Changes in draft-ietf-rmcat-eval-criteria-02 . . . . . . 15
	A.7. Changes in draft-ietf-rmcat-eval-criteria-01 . . . . . . 16
	A.8. Changes in draft-ietf-rmcat-eval-criteria-00 . . . . . . 16
	A.9. Changes in draft-singh-rmcat-cc-eval-04 . . . . . . . . . 16
	A.10. Changes in draft-singh-rmcat-cc-eval-03 . . . . . . . . . 16
	A.11. Changes in draft-singh-rmcat-cc-eval-02 . . . . . . . . . 16
	A.12. Changes in draft-singh-rmcat-cc-eval-01 . . . . . . . . . 17
	Authors' Addresses . . . . . . . . . . . . . . . . . . . . . . . 17
	1. Introduction		1. Introduction
	This memo describes the guidelines to help with evaluating new		This memo describes the guidelines to help with evaluating new
	congestion control algorithms for interactive point-to-point real		congestion control algorithms for interactive point-to-point real-
	time media. The requirements for the congestion control algorithm		time media. The requirements for the congestion control algorithm
	are outlined in [I-D.ietf-rmcat-cc-requirements]). This document		are outlined in [RFC8836]. This document builds upon previous work
	builds upon previous work at the IETF: Specifying New Congestion		at the IETF: Specifying New Congestion Control Algorithms [RFC5033]
	Control Algorithms [RFC5033] and Metrics for the Evaluation of		and Metrics for the Evaluation of Congestion Control Algorithms
	Congestion Control Algorithms [RFC5166].		[RFC5166].
	The guidelines proposed in the document are intended to help prevent		The guidelines proposed in the document are intended to help prevent
	a congestion collapse, promote fair capacity usage and optimize the		a congestion collapse, to promote fair capacity usage, and to
	media flow's throughput. Furthermore, the proposed congestion		optimize the media flow's throughput. Furthermore, the proposed
	control algorithms are expected to operate within the envelope of the		congestion control algorithms are expected to operate within the
	circuit breakers defined in RFC8083 [RFC8083].		envelope of the circuit breakers defined in RFC 8083 [RFC8083].
	This document only provides the broad set of network parameters and		This document only provides the broad set of network parameters and
	and traffic models for evaluating a new congestion control algorithm.		traffic models for evaluating a new congestion control algorithm.
	The minimal requirements for congestion control proposals is to		The minimal requirement for congestion control proposals is to
	produce or present results for the test scenarios described in		produce or present results for the test scenarios described in
	[I-D.ietf-rmcat-eval-test] (Basic Test Cases), which also defines the		[RFC8867] (Basic Test Cases), which also defines the specifics for
	specifics for the test cases. Additionally, proponents may produce		the test cases. Additionally, proponents may produce evaluation
	evaluation results for the wireless test scenarios		results for the wireless test scenarios [RFC8869].
	[I-D.ietf-rmcat-wireless-tests].
	This document does not cover application-specific implications of		This document does not cover application-specific implications of
	congestion control algorithms and how those could be evaluated.		congestion control algorithms and how those could be evaluated.
	Therefore, no quality metrics are defined for performance evaluation;		Therefore, no quality metrics are defined for performance evaluation;
	quality metrics and algorithms to infer those vary between media		quality metrics and the algorithms to infer those vary between media
	types. Metrics and algorithms to assess, e.g., quality of experience		types. Metrics and algorithms to assess, e.g., the quality of
	evolve continuously so that determining suitable choices is left for		experience, evolve continuously so that determining suitable choices
	future work. However, there is consensus that each congestion		is left for future work. However, there is consensus that each
	control algorithm should be able to show that it is useful for		congestion control algorithm should be able to show that it is useful
	interactive video by performing analysis using a real codecs and		for interactive video by performing analysis using real codecs and
	video sequences and state-of-the-art quality metrics.		video sequences and state-of-the-art quality metrics.
	Beyond optimizing individual metrics, real-time applications may have		Beyond optimizing individual metrics, real-time applications may have
	further options to trade off performance, e.g., across multiple		further options to trade off performance, e.g., across multiple
	media; refer to the RMCAT requirements		media; refer to the RMCAT requirements [RFC8836] document. Such
	[I-D.ietf-rmcat-cc-requirements] document. Such trade-offs may be		trade-offs may be defined in the future.
	defined in the future.
	2. Terminology		2. Terminology
	The terminology defined in RTP [RFC3550], RTP Profile for Audio and		The terminology defined in RTP [RFC3550], RTP Profile for Audio and
	Video Conferences with Minimal Control [RFC3551], RTCP Extended		Video Conferences with Minimal Control [RFC3551], RTCP Extended
	Report (XR) [RFC3611], Extended RTP Profile for RTCP-based Feedback		Report (XR) [RFC3611], Extended RTP Profile for RTCP-Based Feedback
	(RTP/AVPF) [RFC4585] and Support for Reduced-Size RTCP [RFC5506]		(RTP/AVPF) [RFC4585] and Support for Reduced-Size RTCP [RFC5506]
	apply.		applies.
	3. Metrics		3. Metrics
	This document specifies testing criteria for evaluating congestion		This document specifies testing criteria for evaluating congestion
	control algorithms for RTP media flows. Proposed algorithms are to		control algorithms for RTP media flows. Proposed algorithms are to
	prove their performance by means of simulation and/or emulation		prove their performance by means of simulation and/or emulation
	experiments for all the cases described.		experiments for all the cases described.
	Each experiment is expected to log every incoming and outgoing packet		Each experiment is expected to log every incoming and outgoing packet
	(the RTP logging format is described in Section 3.1). The logging		(the RTP logging format is described in Section 3.1). The logging
	can be done inside the application or at the endpoints using PCAP		can be done inside the application or at the endpoints using PCAP
	(packet capture, e.g., tcpdump [tcpdump], wireshark [wireshark]).		(packet capture, e.g., tcpdump [tcpdump], Wireshark [wireshark]).
	The following metrics are calculated based on the information in the		The following metrics are calculated based on the information in the
	packet logs:		packet logs:
	1. Sending rate, Receiver rate, Goodput (measured at 200ms		1. Sending rate, receiver rate, goodput (measured at 200ms
	intervals)		intervals)
	2. Packets sent, Packets received		2. Packets sent, packets received
	3. Bytes sent, bytes received		3. Bytes sent, bytes received
	4. Packet delay		4. Packet delay
	5. Packets lost, Packets discarded (from the playout or de-jitter		5. Packets lost, packets discarded (from the playout or de-jitter
	buffer)		buffer)
	6. If using, retransmission or FEC: post-repair loss		6. If using retransmission or FEC: post-repair loss
	7. Self-Fairness and Fairness with respect to cross traffic:		7. Self-fairness and fairness with respect to cross traffic:
	Experiments testing a given congestion control proposal must		Experiments testing a given congestion control proposal must
	report on relative ratios of the average throughput (measured at		report on relative ratios of the average throughput (measured at
	coarser time intervals) obtained by each RTP media stream. In		coarser time intervals) obtained by each RTP media stream. In
	the presence of background cross-traffic such as TCP, the report		the presence of background cross-traffic such as TCP, the report
	must also include the relative ratio between average throughput		must also include the relative ratio between average throughput
	of RTP media streams and cross-traffic streams.		of RTP media streams and cross-traffic streams.
	During static periods of a test (i.e., when bottleneck bandwidth		During static periods of a test (i.e., when bottleneck bandwidth
	is constant and no arrival/departure of streams), these report		is constant and no arrival/departure of streams), these reports
	on relative ratios serve as an indicator of how fair the RTP		on relative ratios serve as an indicator of how fairly the RTP
	streams share bandwidth amongst themselves and against cross-		streams share bandwidth amongst themselves and against cross-
	traffic streams. The throughput measurement interval should be		traffic streams. The throughput measurement interval should be
	set at a few values (for example, at 1s, 5s, and 20s) in order		set at a few values (for example, at 1 s, 5 s, and 20 s) in
	to measure fairness across different time scales.		order to measure fairness across different timescales.
	As a general guideline, the relative ratio between congestion
			As a general guideline, the relative ratio between congestion-
	controlled RTP flows with the same priority level and similar		controlled RTP flows with the same priority level and similar
	path RTT should be bounded between (0.333 and 3.) For example,		path RTT should be bounded between 0.333 and 3. For example,
	see the test scenarios described in [I-D.ietf-rmcat-eval-test].		see the test scenarios described in [RFC8867].
	8. Convergence time: The time taken to reach a stable rate at		8. Convergence time: The time taken to reach a stable rate at
	startup, after the available link capacity changes, or when new		startup, after the available link capacity changes, or when new
	flows get added to the bottleneck link.		flows get added to the bottleneck link.
	9. Instability or oscillation in the sending rate: The frequency or		9. Instability or oscillation in the sending rate: The frequency or
	number of instances when the sending rate oscillates between an		number of instances when the sending rate oscillates between an
	high watermark level and a low watermark level, or vice-versa in		high watermark level and a low watermark level, or vice-versa in
	a defined time window. For example, the watermarks can be set		a defined time window. For example, the watermarks can be set
	at 4x interval: 500 Kbps, 2 Mbps, and a time window of 500ms.		at 4x interval: 500 Kbps, 2 Mbps, and a time window of 500 ms.
	10. Bandwidth Utilization, defined as ratio of the instantaneous		10. Bandwidth utilization, defined as the ratio of the instantaneous
	sending rate to the instantaneous bottleneck capacity. This		sending rate to the instantaneous bottleneck capacity: This
	metric is useful only when a congestion controlled RTP flow is		metric is useful only when a congestion-controlled RTP flow is
	by itself or competing with similar cross-traffic.		by itself or is competing with similar cross-traffic.
	Note that the above metrics are all objective application-independent		Note that the above metrics are all objective application-independent
	metrics. Refer to Section 3, in [I-D.ietf-netvc-testing] for		metrics. Refer to Section 3 of [netvc-testing] for objective metrics
	objective metrics for evaluating codecs.		for evaluating codecs.
	From the logs the statistical measures (min, max, mean, standard		From the logs, the statistical measures (min, max, mean, standard
	deviation and variance) for the whole duration or any specific part		deviation, and variance) for the whole duration or any specific part
	of the session can be calculated. Also the metrics (sending rate,		of the session can be calculated. Also the metrics (sending rate,
	receiver rate, goodput, latency) can be visualized in graphs as		receiver rate, goodput, latency) can be visualized in graphs as
	variation over time, the measurements in the plot are at 1 second		variation over time; the measurements in the plot are at one-second
	intervals. Additionally, from the logs it is possible to plot the		intervals. Additionally, from the logs, it is possible to plot the
	histogram or CDF of packet delay.		histogram or cumulative distribution function (CDF) of packet delay.
	3.1. RTP Log Format		3.1. RTP Log Format
	Having a common log format simplifies running analyses across and		Having a common log format simplifies running analyses across
	comparing different measurements. The log file should be tab or		different measurement setups and comparing their results.
	comma separated containing the following details:
	Send or receive timestamp (Unix): <int>.<int> -- sec.usec decimal		Send or receive timestamp (Unix): <int>.<int> -- sec.usec decimal
	RTP payload type <int> -- decimal		RTP payload type <int> -- decimal
	SSRC <int> -- hexadecimal		SSRC <int> -- hexadecimal
	RTP sequence no <int> -- decimal		RTP sequence no <int> -- decimal
	RTP timestamp <int> -- decimal		RTP timestamp <int> -- decimal
	marker bit 0|1 -- character		marker bit 0|1 -- character
	Payload size <int> -- # bytes, decimal		Payload size <int> -- # bytes, decimal
	Each line of the log file should be terminated with CRLF, CR, or LF		Each line of the log file should be terminated with CRLF, CR, or LF
	characters. Empty lines are disregarded.		characters. Empty lines are disregarded.
	If the congestion control implements retransmissions or FEC, the		If the congestion control implements retransmissions or Forward Error
	evaluation should report both packet loss (before applying error-		Correction (FEC), the evaluation should report both packet loss
	resilience) and residual packet loss (after applying error-		(before applying error resilience) and residual packet loss (after
	resilience).		applying error resilience).
	These data should suffice to compute the media-encoding independent		These data should suffice to compute the media-encoding independent
	metrics described above. Use of a common log will allow simplified		metrics described above. Use of a common log will allow simplified
	post-processing and analysis across different implementations.		post-processing and analysis across different implementations.
	4. List of Network Parameters		4. List of Network Parameters
	The implementors initially are encouraged to choose evaluation		The implementors are encouraged to choose evaluation settings from
	settings from the following values:		the following values initially:
	4.1. One-way Propagation Delay		4.1. One-Way Propagation Delay
	Experiments are expected to verify that the congestion control is		Experiments are expected to verify that the congestion control is
	able to work across a broad range of path characteristics, also		able to work across a broad range of path characteristics, including
	including challenging situations, for example over trans-continental		challenging situations, for example, over transcontinental and/or
	and/or satellite links. Tests thus account for the following		satellite links. Tests thus account for the following different
	different latencies:		latencies:
	1. Very low latency: 0-1ms		1. Very low latency: 0-1 ms
	2. Low latency: 50ms		2. Low latency: 50 ms
	3. High latency: 150ms		3. High latency: 150 ms
	4. Extreme latency: 300ms		4. Extreme latency: 300 ms
	4.2. End-to-end Loss		4.2. End-to-End Loss
	Many paths in the Internet today are largely lossless but, with		Many paths in the Internet today are largely lossless; however, in
	wireless networks and interference, towards remote regions, or in		scenarios featuring interference in wireless networks, sending to and
	scenarios featuring high/fast mobility, media flows may exhibit		receiving from remote regions, or high/fast mobility, media flows may
	substantial packet loss. This variety needs to be reflected		exhibit substantial packet loss. This variety needs to be reflected
	appropriately by the tests.		appropriately by the tests.
	To model a wide range of lossy links, the experiments can choose one		To model a wide range of lossy links, the experiments can choose one
	of the following loss rates, the fractional loss is the ratio of		of the following loss rates; the fractional loss is the ratio of
	packets lost and packets sent.		packets lost and packets sent:
	1. no loss: 0%		1. no loss: 0%
	2. 1%		2. 1%
	3. 5%		3. 5%
	4. 10%		4. 10%
	5. 20%		5. 20%
	4.3. Drop Tail Router Queue Length		4.3. Drop-Tail Router Queue Length
	Routers should be configured to use Drop Trail queues in the		Routers should be configured to use drop-tail queues in the
	experiments due to their (still) prevalent nature. Experimentation		experiments due to their (still) prevalent nature. Experimentation
	with AQM schemes is encouraged but not mandatory.		with Active Queue Management (AQM) schemes is encouraged but not
			mandatory.
	The router queue length is measured as the time taken to drain the		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		FIFO queue. It has been noted in various discussions that the queue
	length in the current deployed Internet varies significantly. While		length in the currently deployed Internet varies significantly.
	the core backbone network has very short queue length, the home		While the core backbone network has very short queue length, the home
	gateways usually have larger queue length. Those various queue		gateways usually have larger queue length. Those various queue
	lengths can be categorized in the following way:		lengths can be categorized in the following way:
	1. QoS-aware (or short): 70ms		1. QoS-aware (or short): 70 ms
	2. Nominal: 300-500ms		2. Nominal: 300-500 ms
	3. Buffer-bloated: 1000-2000ms		3. Buffer-bloated: 1000-2000 ms
	Here the size of the queue is measured in bytes or packets and to		Here the size of the queue is measured in bytes or packets. To
	convert the queue length measured in seconds to queue length in		convert the queue length measured in seconds to queue length in
	bytes:		bytes:
	QueueSize (in bytes) = QueueSize (in sec) x Throughput (in bps)/8		QueueSize (in bytes) = QueueSize (in sec) x Throughput (in bps)/8
	4.4. Loss generation model		4.4. Loss Generation Model
	Many models for generating packet loss are available, some yield		Many models for generating packet loss are available: some generate
	correlated, others independent losses; losses can also be extracted		correlated packet losses, others generate independent packet losses.
	from packet traces. As a (simple) minimum loss model with minimal		In addition, packet losses can also be extracted from packet traces.
	parameterization (i.e., the loss rate), independent random losses		As a (simple) minimum loss model with minimal parameterization (i.e.,
	must be used in the evaluation.		the loss rate), independent random losses must be used in the
			evaluation.
	It is known that independent loss models may reflect reality poorly		It is known that independent loss models may reflect reality poorly,
	and hence more sophisticated loss models could be considered.		and hence more sophisticated loss models could be considered.
	Suitable models for correlated losses includes the Gilbert-Elliot		Suitable models for correlated losses include the Gilbert-Elliot
	model [gilbert-elliott] and losses generated by modeling a queue		model [gilbert-elliott] and models that generate losses by modeling a
	including its (different) drop behaviors.		queue with its (different) drop behaviors.
	4.5. Jitter models		4.5. Jitter Models
	This section defines jitter models for the purposes of this document.		This section defines jitter models for the purposes of this document.
	When jitter is to be applied to both the congestion controlled RTP		When jitter is to be applied to both the congestion-controlled RTP
	flow and any competing flow (such as a TCP competing flow), the		flow and any competing flow (such as a TCP competing flow), the
	competing flow will use the jitter definition below that does not		competing flow will use the jitter definition below that does not
	allow for re-ordering of packets on the competing flow (see NR-RBPDV		allow for reordering of packets on the competing flow (see NR-BPDV
	definition below).		definition below).
	Jitter is an overloaded term in communications. It is typically used		Jitter is an overloaded term in communications. It is typically used
	to refer to the variation of a metric (e.g., delay) with respect to		to refer to the variation of a metric (e.g., delay) with respect to
	some reference metric (e.g., average delay or minimum delay). For		some reference metric (e.g., average delay or minimum delay). For
	example, RFC 3550 jitter is computed as the smoothed difference in		example in RFC 3550, jitter is computed as the smoothed difference in
	packet arrival times relative to their respective expected arrival		packet arrival times relative to their respective expected arrival
	times, which is particularly meaningful if the underlying packet		times, which is particularly meaningful if the underlying packet
	delay variation was caused by a Gaussian random process.		delay variation was caused by a Gaussian random process.
	Because jitter is an overloaded term, we use the term Packet Delay		Because jitter is an overloaded term, we use the term Packet Delay
	Variation (PDV) instead to describe the variation of delay of		Variation (PDV) instead to describe the variation of delay of
	individual packets in the same sense as the IETF IPPM WG has defined		individual packets in the same sense as the IETF IP Performance
	PDV in their documents (e.g., RFC 3393) and as the ITU-T SG16 has		Metrics (IPPM) working group has defined PDV in their documents
	defined IP Packet Delay Variation (IPDV) in their documents (e.g.,		(e.g., RFC 3393) and as the ITU-T SG16 has defined IP Packet Delay
	Y.1540).		Variation (IPDV) in their documents (e.g., Y.1540).
	Most PDV distributions in packet network systems are one-sided		Most PDV distributions in packet network systems are one-sided
	distributions, the measurement of which with a finite number of		distributions, the measurement of which with a finite number of
	measurement samples results in one-sided histograms. In the usual		measurement samples results in one-sided histograms. In the usual
	packet network transport case, there is typically one packet that		packet network transport case, there is typically one packet that
	transited the network with the minimum delay; a (large) number of		transited the network with the minimum delay; a (large) number of
	packets transit the network within some (smaller) positive variation		packets transit the network within some (smaller) positive variation
	from this minimum delay, and a (small) number of the packets transit		from this minimum delay, and a (small) number of the packets transit
	the network with delays higher than the median or average transit		the network with delays higher than the median or average transit
	time (these are outliers). Although infrequent, outliers can cause		time (these are outliers). Although infrequent, outliers can cause
	significant deleterious operation in adaptive systems and should be		significant deleterious operation in adaptive systems and should be
	considered in rate adaptation designs for RTP congestion control.		considered in rate adaptation designs for RTP congestion control.
	In this section we define two different bounded PDV characteristics,		In this section we define two different bounded PDV characteristics,
	1) Random Bounded PDV and 2) Approximately Random Subject to No-		1) Random Bounded PDV and 2) Approximately Random Subject to No-
	Reordering Bounded PDV.		Reordering Bounded PDV.
	The former, 1) Random Bounded PDV is presented for information only,		The former, 1) Random Bounded PDV, is presented for information only,
	while the latter, 2) Approximately Random Subject to No-Reordering		while the latter, 2) Approximately Random Subject to No-Reordering
	Bounded PDV, must be used in the evaluation.		Bounded PDV, must be used in the evaluation.
	4.5.1. Random Bounded PDV (RBPDV)		4.5.1. Random Bounded PDV (RBPDV)
	The RBPDV probability distribution function (PDF) is specified to be		The RBPDV probability distribution function (PDF) is specified to be
	of some mathematically describable function which includes some		of some mathematically describable function that includes some
	practical minimum and maximum discrete values suitable for testing.		practical minimum and maximum discrete values suitable for testing.
	For example, the minimum value, x_min, might be specified as the		For example, the minimum value, x_min, might be specified as the
	minimum transit time packet and the maximum value, x_max, might be		minimum transit time packet, and the maximum value, x_max, might be
	defined to be two standard deviations higher than the mean.		defined to be two standard deviations higher than the mean.
	Since we are typically interested in the distribution relative to the		Since we are typically interested in the distribution relative to the
	mean delay packet, we define the zero mean PDV sample, z(n), to be		mean delay packet, we define the zero mean PDV sample, z(n), to be
	z(n) = x(n) - x_mean, where x(n) is a sample of the RBPDV random		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.		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		We assume here that s(n) is the original source time of packet n and
	the post-jitter induced emission time, j(n), for packet n is:		the post-jitter induced emission time, j(n), for packet n is:
	j(n) = {[z(n) + x_mean] + s(n)}.		j(n) = {[z(n) + x_mean] + s(n)}.
	It follows that the separation in the post-jitter time of packets 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		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		always a positive quantity, we note that packet reordering at the
	receiver is possible whenever the second term is greater than the		receiver is possible whenever the second term is greater than the
	first. Said another way, whenever the difference in possible zero		first. Said another way, whenever the difference in possible zero
	mean PDV sample delays (i.e., [x_max-x_min]) exceeds the inter-		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		departure time of any two sent packets, we have the possibility of
	packet re-ordering.		packet reordering.
	There are important use cases in real networks where packets can		There are important use cases in real networks where packets can
	become re-ordered such as in load balancing topologies and during		become reordered, such as in load-balancing topologies and during
	route changes. However, for the vast majority of cases there is no		route changes. However, for the vast majority of cases, there is no
	packet re-ordering because most of the time packets follow the same		packet reordering because most of the time packets follow the same
	path. Due to this, if a packet becomes overly delayed, the packets		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		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		mobile wireless links where there are per-flow queues prior to base
	station scheduling. Owing to this important use case, we define		station scheduling. Owing to this important use case, we define
	another PDV profile similar to the above, but one that does not allow		another PDV profile similar to the above, but one that does not allow
	for re-ordering within a flow.		for reordering within a flow.
	4.5.2. Approximately Random Subject to No-Reordering Bounded PDV (NR-		4.5.2. Approximately Random Subject to No-Reordering Bounded PDV (NR-
	RPVD)		BPDV)
	No Reordering RPDV, NR-RPVD, is defined similarly to the above with		No Reordering BPDV, NR-BPDV, is defined similarly to the above with
	one important exception. Let serial(n) be defined as the		one important exception. Let serial(n) be defined as the
	serialization delay of packet n at the lowest bottleneck link rate		serialization delay of packet n at the lowest bottleneck link rate
	(or other appropriate rate) in a given test. Then we produce all the		(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		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		length of the source sequence s to be offset. The exception can be
	be stated as follows: We revisit all j(n) beginning from index n=2,		stated as follows: We revisit all j(n) beginning from index n=2, and
	and if j(n) is determined to be less than [j(n-1)+serial(n-1)], we		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		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		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		the packet n is sent immediately after packet (n-1) at the bottleneck
	link rate. Although this is generally the theoretical minimum in		link rate. Although this is generally the theoretical minimum in
	that it assumes that no other packets from other flows are in-between		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		packet n and n+1 at the bottleneck link, it is a reasonable
	assumption for per flow queuing.		assumption for per-flow queuing.
	We note that this assumption holds for some important exception		We note that this assumption holds for some important exception
	cases, such as packets immediately following outliers. There are a		cases, such as packets immediately following outliers. There are a
	multitude of software controlled elements common on end-to-end		multitude of software-controlled elements common on end-to-end
	Internet paths (such as firewalls, ALGs and other middleboxes) which		Internet paths (such as firewalls, application-layer gateways, and
	stop processing packets while servicing other functions (e.g.,		other middleboxes) that stop processing packets while servicing other
	garbage collection). Often these devices do not drop packets, but		functions (e.g., garbage collection). Often these devices do not
	rather queue them for later processing and cause many of the		drop packets, but rather queue them for later processing and cause
	outliers. Thus NR-RPVD models this particular use case (assuming		many of the outliers. Thus NR-BPDV models this particular use case
	serial(n+1) is defined appropriately for the device causing the		(assuming serial(n+1) is defined appropriately for the device causing
	outlier) and thus is believed to be important for adaptation		the outlier) and is believed to be important for adaptation
	development for congestion controlled RTP streams.		development for congestion-controlled RTP streams.
	4.5.3. Recommended distribution		4.5.3. Recommended Distribution
	Whether Random Bounded PDV or Approximately Random Subject to No-		Whether Random Bounded PDV or Approximately Random Subject to No-
	Reordering Bounded PDV, it is recommended that z(n) is distributed		Reordering Bounded PDV, it is recommended that z(n) is distributed
	according to a truncated Gaussian for the above jitter models:		according to a truncated Gaussian for the above jitter models:
	z(n) ~ |max(min(N(0, std^2), N_STD * std), -N_STD * std)|		z(n) ~ |max(min(N(0, std^(2)), N_STD * std), -N_STD * std)|
	where N(0, std^2) is the Gaussian distribution with zero mean and		where N(0, std^(2)) is the Gaussian distribution with zero mean and
	standard deviation std. Recommended values:		std is standard deviation. Recommended values:
	o std = 5 ms		std = 5 ms
	o N_STD = 3		N_STD = 3
	5. Traffic Models		5. Traffic Models
	5.1. TCP traffic model		5.1. TCP Traffic Model
	Long-lived TCP flows will download data throughout the session and		Long-lived TCP flows will download data throughout the session and
	are expected to have infinite amount of data to send or receive.		are expected to have infinite amount of data to send or receive.
	This roughly applies, for example, when downloading software		This roughly applies, for example, when downloading software
	distributions.		distributions.
	Each short TCP flow is modeled as a sequence of file downloads		Each short TCP flow is modeled as a sequence of file downloads
	interleaved with idle periods. Not all short TCP flows start at the		interleaved with idle periods. Not all short TCP flows start at the
	same time, i.e., some start in the ON state while others start in the		same time, i.e., some start in the ON state while others start in the
	OFF state.		OFF state.
	The short TCP flows can be modeled as follows: 30 connections start		The short TCP flows can be modeled as follows: 30 connections start
	simultaneously fetching small (30-50 KB) amounts of data, evenly		simultaneously fetching small (30-50 KB) amounts of data, evenly
	distributed. This covers the case where the short TCP flows are		distributed. This covers the case where the short TCP flows are
	fetching web page resources rather than video files.		fetching web page resources rather than video files.
	The idle period between bursts of starting a group of TCP flows is		The idle period between bursts of starting a group of TCP flows is
	typically derived from an exponential distribution with the mean		typically derived from an exponential distribution with the mean
	value of 10 seconds.		value of 10 seconds.
	[These values were picked based on the data available at		| These values were picked based on the data available at
	http://httparchive.org/interesting.php as of October 2015].		| <https://httparchive.org/reports/state-of-the-
			| web?start=2015_10_01&end=2015_11_01&view=list> as of October
			| 2015.
	Many different TCP congestion control schemes are deployed today.		Many different TCP congestion control schemes are deployed today.
	Therefore, experimentation with a range of different schemes,		Therefore, experimentation with a range of different schemes,
	especially including CUBIC, is encouraged. Experiments must document		especially including CUBIC [RFC8312], is encouraged. Experiments
	in detail which congestion control schemes they tested against and		must document in detail which congestion control schemes they tested
	which parameters were used.		against and which parameters were used.
	5.2. RTP Video model		5.2. RTP Video Model
	[RFC8593] describes two types of video traffic models for evaluating		[RFC8593] describes two types of video traffic models for evaluating
	candidate algorithms for RTP congestion control. The first model		candidate algorithms for RTP congestion control. The first model
	statistically characterizes the behavior of a video encoder, whereas		statistically characterizes the behavior of a video encoder, whereas
	the second model uses video traces.		the second model uses video traces.
	Sample video test sequences are available at [xiph-seq]. The		Sample video test sequences are available at [xiph-seq]. The
	following two video streams are the recommended minimum for testing:		following two video streams are the recommended minimum for testing:
	Foreman (CIF sequence) and FourPeople (720p); both come as raw video		Foreman (CIF sequence) and FourPeople (720p); both come as raw video
	data to be encoded dynamically. As these video sequences are short		data to be encoded dynamically. As these video sequences are short
	(300 and 600 frames, respectively, they shall be stitched together		(300 and 600 frames, respectively), they shall be stitched together
	repeatedly until the desired length is reached.		repeatedly until the desired length is reached.
	5.3. Background UDP		5.3. Background UDP
	Background UDP flow is modeled as a constant bit rate (CBR) flow. It		Background UDP flow is modeled as a constant bit rate (CBR) flow. It
	will download data at a particular CBR rate for the complete session,		will download data at a particular CBR for the complete session, or
	or will change to particular CBR rate at predefined intervals. The		will change to particular CBR at predefined intervals. The inter-
	inter packet interval is calculated based on the CBR and the packet		packet interval is calculated based on the CBR and the packet size
	size (is typically set to the path MTU size, the default value can be		(typically set to the path MTU size, the default value can be 1500
	1500 bytes).		bytes).
	Note that new transport protocols such as QUIC may use UDP but, due		Note that new transport protocols such as QUIC may use UDP; however,
	to their congestion control algorithms, will exhibit behavior		due to their congestion control algorithms, they will exhibit
	conceptually similar in nature to TCP flows above and can thus be		behavior conceptually similar in nature to TCP flows above and can
	subsumed by the above, including the division into short- and long-		thus be subsumed by the above, including the division into short-
	lived flows. As QUIC evolves independently of TCP congestion control		lived and long-lived flows. As QUIC evolves independently of TCP
	algorithms, its future congestion control should be considered as		congestion control algorithms, its future congestion control should
	competing traffic as appropriate.		be considered as competing traffic as appropriate.
	6. Security Considerations		6. Security Considerations
	This document specifies evaluation criteria and parameters for		This document specifies evaluation criteria and parameters for
	assessing and comparing the performance of congestion control		assessing and comparing the performance of congestion control
	protocols and algorithms for real-time communication. This memo		protocols and algorithms for real-time communication. This memo
	itself is thus not subject to security considerations but the		itself is thus not subject to security considerations but the
	protocols and algorithms evaluated may be. In particular, successful		protocols and algorithms evaluated may be. In particular, successful
	operation under all tests defined in this document may suffice for a		operation under all tests defined in this document may suffice for a
	comparative evaluation but must not be interpreted that the protocol		comparative evaluation but must not be interpreted that the protocol
	skipping to change at page 12, line 27 ¶		skipping to change at line 528 ¶
	Such evaluations are expected to be carried out in controlled		Such evaluations are expected to be carried out in controlled
	environments for limited numbers of parallel flows. As such, these		environments for limited numbers of parallel flows. As such, these
	evaluations are by definition limited and will not be able to		evaluations are by definition limited and will not be able to
	systematically consider possible interactions or very large groups of		systematically consider possible interactions or very large groups of
	communicating nodes under all possible circumstances, so that careful		communicating nodes under all possible circumstances, so that careful
	protocol design is advised to avoid incidentally contributing traffic		protocol design is advised to avoid incidentally contributing traffic
	that could lead to unstable networks, e.g., (local) congestion		that could lead to unstable networks, e.g., (local) congestion
	collapse.		collapse.
	This specification focuses on assessing the regular operation of the		This specification focuses on assessing the regular operation of the
	protocols and algorithms under considerations. It does not suggest		protocols and algorithms under consideration. It does not suggest
	checks against malicious use of the protocols -- by the sender, the		checks against malicious use of the protocols -- by the sender, the
	receiver, or intermediate parties, e.g., through faked, dropped,		receiver, or intermediate parties, e.g., through faked, dropped,
	replicated, or modified congestion signals. It is up to the protocol		replicated, or modified congestion signals. It is up to the protocol
	specifications themselves to ensure that authenticity, integrity,		specifications themselves to ensure that authenticity, integrity,
	and/or plausibility of received signals are checked and the		and/or plausibility of received signals are checked, and the
	appropriate actions (or non-actions) are taken.		appropriate actions (or non-actions) are taken.
	7. IANA Considerations		7. IANA Considerations
	There are no IANA impacts in this memo.		This document has no IANA actions.
	8. Contributors
	The content and concepts within this document are a product of the
	discussion carried out in the Design Team.
	Michael Ramalho provided the text for the Jitter model.
	9. Acknowledgments
	Much of this document is derived from previous work on congestion
	control at the IETF.
	The authors would like to thank Harald Alvestrand, Anna Brunstrom,
	Luca De Cicco, Wesley Eddy, Lars Eggert, Kevin Gross, Vinayak Hegde,
	Randell Jesup, Mirja Kuehlewind, Karen Nielsen, Piers O'Hanlon, Colin
	Perkins, Michael Ramalho, Zaheduzzaman Sarker, Timothy B.
	Terriberry, Michael Welzl, Mo Zanaty, and Xiaoqing Zhu for providing
	valuable feedback on earlier versions of this draft. Additionally,
	also thank the participants of the design team for their comments and
	discussion related to the evaluation criteria.
	10. References
	10.1. Normative References		8. References
	[I-D.ietf-rmcat-cc-requirements]		8.1. Normative References
	Jesup, R. and Z. Sarker, "Congestion Control Requirements
	for Interactive Real-Time Media", draft-ietf-rmcat-cc-
	requirements-09 (work in progress), December 2014.
	[RFC3550] Schulzrinne, H., Casner, S., Frederick, R., and V.		[RFC3550] Schulzrinne, H., Casner, S., Frederick, R., and V.
	Jacobson, "RTP: A Transport Protocol for Real-Time		Jacobson, "RTP: A Transport Protocol for Real-Time
	Applications", STD 64, RFC 3550, DOI 10.17487/RFC3550,		Applications", STD 64, RFC 3550, DOI 10.17487/RFC3550,
	July 2003, <https://www.rfc-editor.org/info/rfc3550>.		July 2003, <https://www.rfc-editor.org/info/rfc3550>.
	[RFC3551] Schulzrinne, H. and S. Casner, "RTP Profile for Audio and		[RFC3551] Schulzrinne, H. and S. Casner, "RTP Profile for Audio and
	Video Conferences with Minimal Control", STD 65, RFC 3551,		Video Conferences with Minimal Control", STD 65, RFC 3551,
	DOI 10.17487/RFC3551, July 2003,		DOI 10.17487/RFC3551, July 2003,
	<https://www.rfc-editor.org/info/rfc3551>.		<https://www.rfc-editor.org/info/rfc3551>.
	skipping to change at page 14, line 10 ¶		skipping to change at line 580 ¶
	[RFC8083] Perkins, C. and V. Singh, "Multimedia Congestion Control:		[RFC8083] Perkins, C. and V. Singh, "Multimedia Congestion Control:
	Circuit Breakers for Unicast RTP Sessions", RFC 8083,		Circuit Breakers for Unicast RTP Sessions", RFC 8083,
	DOI 10.17487/RFC8083, March 2017,		DOI 10.17487/RFC8083, March 2017,
	<https://www.rfc-editor.org/info/rfc8083>.		<https://www.rfc-editor.org/info/rfc8083>.
	[RFC8593] Zhu, X., Mena, S., and Z. Sarker, "Video Traffic Models		[RFC8593] Zhu, X., Mena, S., and Z. Sarker, "Video Traffic Models
	for RTP Congestion Control Evaluations", RFC 8593,		for RTP Congestion Control Evaluations", RFC 8593,
	DOI 10.17487/RFC8593, May 2019,		DOI 10.17487/RFC8593, May 2019,
	<https://www.rfc-editor.org/info/rfc8593>.		<https://www.rfc-editor.org/info/rfc8593>.
	10.2. Informative References		[RFC8836] Jesup, R. and Z. Sarker, Ed., "Congestion Control
			Requirements for Interactive Real-Time Media", RFC 8836,
			DOI 10.17487/RFC8836, January 2021,
			<https://www.rfc-editor.org/info/rfc8836>.
			8.2. Informative References
	[gilbert-elliott]		[gilbert-elliott]
	Hasslinger, G. and O. Hohlfeld, "The Gilbert-Elliott Model		Hasslinger, G. and O. Hohlfeld, "The Gilbert-Elliott Model
	for Packet Loss in Real Time Services on the Internet",		for Packet Loss in Real Time Services on the Internet",
	14th GI/ITG Conference - Measurement, Modelling and		14th GI/ITG Conference - Measurement, Modelling and
	Evalutation of Computer and Communication Systems , 3		Evalutation [sic] of Computer and Communication Systems,
	2008.		March 2008,
			<https://ieeexplore.ieee.org/document/5755057>.
	[I-D.ietf-netvc-testing]		[netvc-testing]
	Daede, T., Norkin, A., and I. Brailovskiy, "Video Codec		Daede, T., Norkin, A., and I. Brailovskiy, "Video Codec
	Testing and Quality Measurement", draft-ietf-netvc-		Testing and Quality Measurement", Work in Progress,
	testing-09 (work in progress), January 2020.		Internet-Draft, draft-ietf-netvc-testing-09, 31 January
			2020,
	[I-D.ietf-rmcat-eval-test]		<https://tools.ietf.org/html/draft-ietf-netvc-testing-09>.
	Sarker, Z., Singh, V., Zhu, X., and M. Ramalho, "Test
	Cases for Evaluating RMCAT Proposals", draft-ietf-rmcat-
	eval-test-10 (work in progress), May 2019.
	[I-D.ietf-rmcat-wireless-tests]
	Sarker, Z., Zhu, X., and J. Fu, "Evaluation Test Cases for
	Interactive Real-Time Media over Wireless Networks",
	draft-ietf-rmcat-wireless-tests-11 (work in progress),
	March 2020.
	[RFC5033] Floyd, S. and M. Allman, "Specifying New Congestion		[RFC5033] Floyd, S. and M. Allman, "Specifying New Congestion
	Control Algorithms", BCP 133, RFC 5033,		Control Algorithms", BCP 133, RFC 5033,
	DOI 10.17487/RFC5033, August 2007,		DOI 10.17487/RFC5033, August 2007,
	<https://www.rfc-editor.org/info/rfc5033>.		<https://www.rfc-editor.org/info/rfc5033>.
	[RFC5166] Floyd, S., Ed., "Metrics for the Evaluation of Congestion		[RFC5166] Floyd, S., Ed., "Metrics for the Evaluation of Congestion
	Control Mechanisms", RFC 5166, DOI 10.17487/RFC5166, March		Control Mechanisms", RFC 5166, DOI 10.17487/RFC5166, March
	2008, <https://www.rfc-editor.org/info/rfc5166>.		2008, <https://www.rfc-editor.org/info/rfc5166>.
	[tcpdump] "Homepage of tcpdump and libpcap",		[RFC8312] Rhee, I., Xu, L., Ha, S., Zimmermann, A., Eggert, L., and
	https://www.tcpdump.org/index.html .		R. Scheffenegger, "CUBIC for Fast Long-Distance Networks",
			RFC 8312, DOI 10.17487/RFC8312, February 2018,
	[wireshark]		<https://www.rfc-editor.org/info/rfc8312>.
	"Homepage of Wireshark", https://www.wireshark.org .
	[xiph-seq]
	Daede, T., "Video Test Media Set",
	https://media.xiph.org/video/derf/ .
	Appendix A. Change Log
	Note to the RFC-Editor: please remove this section prior to
	publication as an RFC.
	A.1. Changes in draft-ietf-rmcat-eval-criteria-07
	Updated the draft according to the discussion at IETF-101.
	o Updated the discussion on fairness. Thanks to Xiaoqing Zhu for
	providing text.
	o Fixed a simple loss model and provided pointers to more
	sophisticated ones.
	o Fixed the choice of the jitter model.
	A.2. Changes in draft-ietf-rmcat-eval-criteria-06
	o Updated Jitter.
	A.3. Changes in draft-ietf-rmcat-eval-criteria-05
	o Improved text surrounding wireless tests, video sequences, and
	short-TCP model.
	A.4. Changes in draft-ietf-rmcat-eval-criteria-04
	o Removed the guidelines section, as most of the sections are now
	covered: wireless tests, video model, etc.
	o Improved Short TCP model based on the suggestion to use
	httparchive.org.
	A.5. Changes in draft-ietf-rmcat-eval-criteria-03
	o Keep-alive version.
	o Moved link parameters and traffic models from eval-test
	A.6. Changes in draft-ietf-rmcat-eval-criteria-02
	o Incorporated fairness test as a working test.
	o Updated text on mimimum evaluation requirements.
	A.7. Changes in draft-ietf-rmcat-eval-criteria-01
	o Removed Appendix B.
	o Removed Section on Evaluation Parameters.
	A.8. Changes in draft-ietf-rmcat-eval-criteria-00
	o Updated references.
	o Resubmitted as WG draft.
	A.9. Changes in draft-singh-rmcat-cc-eval-04
	o Incorporate feedback from IETF 87, Berlin.
	o Clarified metrics: convergence time, bandwidth utilization.
	o Changed fairness criteria to fairness test.
	o Added measuring pre- and post-repair loss.
	o Added open issue of measuring video quality to appendix.
	o clarified use of DropTail and AQM.
	o Updated text in "Minimum Requirements for Evaluation"
	A.10. Changes in draft-singh-rmcat-cc-eval-03
	o Incorporate the discussion within the design team.
	o Added a section on evaluation parameters, it describes the flow
	and network characteristics.
	o Added Appendix with self-fairness experiment.		[RFC8867] Sarker, Z., Singh, V., Zhu, X., and M. Ramalho, "Test
			Cases for Evaluating Congestion Control for Interactive
			Real-Time Media", RFC 8867, DOI 10.17487/RFC8867, January
			2021, <https://www.rfc-editor.org/info/rfc8867>.
	o Changed bottleneck parameters from a proposal to an example set.		[RFC8869] Sarker, Z., Zhu, X., and J. Fu, "Evaluation Test Cases for
			Interactive Real-Time Media over Wireless Networks",
			RFC 8869, DOI 10.17487/RFC8869, January 2021,
			<https://www.rfc-editor.org/info/rfc8869>.
	o		[tcpdump] "Homepage of tcpdump and libpcap",
			<https://www.tcpdump.org/index.html>.
	A.11. Changes in draft-singh-rmcat-cc-eval-02		[wireshark]
			"Homepage of Wireshark", <https://www.wireshark.org>.
	o Added scenario descriptions.		[xiph-seq] Daede, T., "Video Test Media Set",
			<https://media.xiph.org/video/derf/>.
	A.12. Changes in draft-singh-rmcat-cc-eval-01		Contributors
	o Removed QoE metrics.		The content and concepts within this document are a product of the
			discussion carried out in the Design Team.
	o Changed stability to steady-state.		Michael Ramalho provided the text for the jitter models
			(Section 4.5).
	o Added measuring impact against few and many flows.		Acknowledgments
	o Added guideline for idle and data-limited periods.		Much of this document is derived from previous work on congestion
			control at the IETF.
	o Added reference to TCP evaluation suite in example evaluation		The authors would like to thank Harald Alvestrand, Anna Brunstrom,
	scenarios.		Luca De Cicco, Wesley Eddy, Lars Eggert, Kevin Gross, Vinayak Hegde,
			Randell Jesup, Mirja Kühlewind, Karen Nielsen, Piers O'Hanlon, Colin
			Perkins, Michael Ramalho, Zaheduzzaman Sarker, Timothy B. Terriberry,
			Michael Welzl, Mo Zanaty, and Xiaoqing Zhu for providing valuable
			feedback on draft versions of this document. Additionally, thanks to
			the participants of the Design Team for their comments and discussion
			related to the evaluation criteria.
	Authors' Addresses		Authors' Addresses
	Varun Singh		Varun Singh
	CALLSTATS I/O Oy		CALLSTATS I/O Oy
	Runeberginkatu 4c A 4		Rauhankatu 11 C
	Helsinki 00100		FI-00100 Helsinki
	Finland		Finland
	Email: varun.singh@iki.fi		Email: varun.singh@iki.fi
	URI: https://www.callstats.io/about		URI: https://www.callstats.io/
	Joerg Ott		Jörg Ott
	Technical University of Munich		Technical University of Munich
	Faculty of Informatics		Department of Informatics
			Chair of Connected Mobility
	Boltzmannstrasse 3		Boltzmannstrasse 3
	Garching bei Muenchen, DE 85748		85748 Garching
	Germany		Germany
	Email: ott@in.tum.de		Email: ott@in.tum.de
	Stefan Holmer		Stefan Holmer
	Google		Google
	Kungsbron 2		Kungsbron 2
	Stockholm 11122		SE-11122 Stockholm
	Sweden		Sweden
	Email: holmer@google.com		Email: holmer@google.com
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