draft-ietf-ippm-tcp-throughput-tm-07.txt		draft-ietf-ippm-tcp-throughput-tm-08.txt
	Network Working Group B. Constantine		Network Working Group B. Constantine
	Internet-Draft JDSU		Internet-Draft JDSU
	Intended status: Informational G. Forget		Intended status: Informational G. Forget
	Expires: March 24, 2011 Bell Canada (Ext. Consultant)		Expires: May 14, 2011 Bell Canada (Ext. Consultant)
			Rudiger Geib
			Deutsche Telekom
	Reinhard Schrage		Reinhard Schrage
	Schrage Consulting		Schrage Consulting
	September 24, 2010
			November 14, 2010
	Framework for TCP Throughput Testing		Framework for TCP Throughput Testing
	draft-ietf-ippm-tcp-throughput-tm-07.txt		draft-ietf-ippm-tcp-throughput-tm-08.txt
	Abstract		Abstract
	This document describes a framework for measuring sustained TCP		This framework describes a methodology for measuring end-to-end TCP
	throughput performance in an end-to-end managed network environment.		throughput performance in a managed IP network. The intention is to
	This document is intended to provide a practical methodology to help		provide a practical methodology to validate TCP layer performance.
	users validate the TCP layer performance of a managed network, which		The goal is to provide a better indication of the user experience.
	should provide a better indication of end-user experience. In the		In this framework, various TCP and IP parameters are identified and
	framework, various TCP and network parameters are identified that		should be tested as part of a managed IP network verification.
	should be tested as part of the network verification at the TCP
	layer.
	Requirements Language		Requirements Language
	The key words "MUST", "MUST NOT", "REQUIRED", "SHALL", "SHALL NOT",		The key words "MUST", "MUST NOT", "REQUIRED", "SHALL", "SHALL NOT",
	"SHOULD", "SHOULD NOT", "RECOMMENDED", "MAY", and "OPTIONAL" in this		"SHOULD", "SHOULD NOT", "RECOMMENDED", "MAY", and "OPTIONAL" in this
	document are to be interpreted as described in RFC 2119 [RFC2119].		document are to be interpreted as described in RFC 2119 [RFC2119].
	Status of this Memo		Status of this Memo
	This Internet-Draft is submitted in full conformance with the		This Internet-Draft is submitted in full conformance with the
	skipping to change at page 1, line 45		skipping to change at page 1, line 46
	Internet-Drafts are working documents of the Internet Engineering		Internet-Drafts are working documents of the Internet Engineering
	Task Force (IETF). Note that other groups may also distribute		Task Force (IETF). Note that other groups may also distribute
	working documents as Internet-Drafts. The list of current Internet-		working documents as Internet-Drafts. The list of current Internet-
	Drafts is at http://datatracker.ietf.org/drafts/current/.		Drafts is at http://datatracker.ietf.org/drafts/current/.
	Internet-Drafts are draft documents valid for a maximum of six months		Internet-Drafts are draft documents valid for a maximum of six months
	and may be updated, replaced, or obsoleted by other documents at any		and may be updated, replaced, or obsoleted by other documents at any
	time. It is inappropriate to use Internet-Drafts as reference		time. It is inappropriate to use Internet-Drafts as reference
	material or to cite them other than as "work in progress."		material or to cite them other than as "work in progress."
	This Internet-Draft will expire on March 24, 2011.		This Internet-Draft will expire on May 14, 2011.
	Copyright Notice		Copyright Notice
	Copyright (c) 2010 IETF Trust and the persons identified as the		Copyright (c) 2010 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
	(http://trustee.ietf.org/license-info) in effect on the date of		(http://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
	skipping to change at page 2, line 25		skipping to change at page 2, line 25
	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 . . . . . . . . . . . . . . . . . . . . . . . . . 3
	1.1 Test Set-up and Terminology . . . . . . . . . . . . . . . 4		1.1 Test Set-up and Terminology . . . . . . . . . . . . . . . 4
	2. Scope and Goals of this methodology. . . . . . . . . . . . . . 5		2. Scope and Goals of this methodology. . . . . . . . . . . . . . 5
	2.1 TCP Equilibrium State Throughput . . . . . . . . . . . . . 6		2.1 TCP Equilibrium. . . . . . . . . . . . . . . . . . . . . . 6
	2.2 Metrics for TCP Throughput Tests . . . . . . . . . . . . . 7		3. TCP Throughput Testing Methodology . . . . . . . . . . . . . . 7
	3. TCP Throughput Testing Methodology . . . . . . . . . . . . . . 9		3.1 Determine Network Path MTU . . . . . . . . . . . . . . . . 9
	3.1 Determine Network Path MTU . . . . . . . . . . . . . . . . 11		3.2. Baseline Round Trip Time and Bandwidth . . . . . . . . . . 10
	3.2. Baseline Round Trip Time and Bandwidth . . . . . . . . . . 13		3.2.1 Techniques to Measure Round Trip Time . . . . . . . . 10
	3.2.1 Techniques to Measure Round Trip Time . . . . . . . . 13		3.2.2 Techniques to Measure end-to-end Bandwidth. . . . . . 11
	3.2.2 Techniques to Measure End-end Bandwidth . . . . . . . 14		3.3. TCP Throughput Tests . . . . . . . . . . . . . . . . . . . 12
	3.3. TCP Throughput Tests . . . . . . . . . . . . . . . . . . . 14		3.3.1 Calculate Ideal TCP Receive Window Size. . . . . . . . 12
	3.3.1 Calculate Optimum TCP Window Size. . . . . . . . . . . 15		3.3.2 Metrics for TCP Throughput Tests . . . . . . . . . . . 15
	3.3.2 Conducting the TCP Throughput Tests. . . . . . . . . . 17		3.3.3 Conducting the TCP Throughput Tests. . . . . . . . . . 18
	3.3.3 Single vs. Multiple TCP Connection Testing . . . . . . 18		3.3.4 Single vs. Multiple TCP Connection Testing . . . . . . 19
	3.3.4 Interpretation of the TCP Throughput Results . . . . . 19		3.3.5 Interpretation of the TCP Throughput Results . . . . . 20
	3.4. Traffic Management Tests . . . . . . . . . . . . . . . . . 19		3.4. Traffic Management Tests . . . . . . . . . . . . . . . . . 20
	3.4.1 Traffic Shaping Tests. . . . . . . . . . . . . . . . . 20		3.4.1 Traffic Shaping Tests. . . . . . . . . . . . . . . . . 21
	3.4.1.1 Interpretation of Traffic Shaping Test Results. . . 20		3.4.1.1 Interpretation of Traffic Shaping Test Results. . . 21
	3.4.2 RED Tests. . . . . . . . . . . . . . . . . . . . . . . 21		3.4.2 RED Tests. . . . . . . . . . . . . . . . . . . . . . . 22
	3.4.2.1 Interpretation of RED Results . . . . . . . . . . . 21		3.4.2.1 Interpretation of RED Results . . . . . . . . . . . 23
	4. Security Considerations . . . . . . . . . . . . . . . . . . . 22		4. Security Considerations . . . . . . . . . . . . . . . . . . . 23
	5. IANA Considerations . . . . . . . . . . . . . . . . . . . . . 22		5. IANA Considerations . . . . . . . . . . . . . . . . . . . . . 23
	5.1. Registry Specification . . . . . . . . . . . . . . . . . . 22		6. Acknowledgments . . . . . . . . . . . . . . . . . . . . . . . 23
	5.2. Registry Contents . . . . . . . . . . . . . . . . . . . . 22		7. References . . . . . . . . . . . . . . . . . . . . . . . . . . 24
	6. Acknowledgments . . . . . . . . . . . . . . . . . . . . . . . 22		7.1 Normative References . . . . . . . . . . . . . . . . . . . 24
	7. References . . . . . . . . . . . . . . . . . . . . . . . . . . 22		7.2 Informative References . . . . . . . . . . . . . . . . . . 24
	7.1 Normative References . . . . . . . . . . . . . . . . . . . 22
	7.2 Informative References . . . . . . . . . . . . . . . . . . 23
	Authors' Addresses . . . . . . . . . . . . . . . . . . . . . . . . 23		Authors' Addresses . . . . . . . . . . . . . . . . . . . . . . . . 25
	1. Introduction		1. Introduction
	Network providers are coming to the realization that Layer 2/3		Network providers are coming to the realization that Layer 2/3
	testing and TCP layer testing are required to more adequately ensure		testing is not enough to adequately ensure end-user's satisfaction.
	end-user satisfaction. Testing an operational network prior to		An SLA (Service Level Agreement) is provided to business customers
	customer activation is referred to as "turn-up" testing and the SLA		and is generally based upon Layer 2/3 criteria such as access rate,
	(Service Level Agreement) is generally based upon Layer 2/3		latency, packet loss and delay variations. On the other hand,
	information rate, packet delay, loss and delay variation. Therefore,		measuring TCP throughput provides meaningful results with respect to
	the network provider community desires to measure network throughput		user experience. Thus, the network provider community desires to
	performance at the TCP layer. Measuring TCP throughput provides a		measure IP network throughput performance at the TCP layer.
	meaningful measure with respect to the end user experience (and
	ultimately reach some level of TCP testing interoperability which
	does not exist today).
	Additionally, end-users (business enterprises) seek to conduct		Additionally, business enterprise customers seek to conduct
	repeatable TCP throughput tests between enterprise locations. Since		repeatable TCP throughput tests between locations. Since these
	these enterprises rely on the networks of the providers, a common		enterprises rely on the networks of the providers, a common test
	test methodology (and metrics) would be equally beneficial to both		methodology with predefined metrics will benefit both parties.
	parties.
	So the intent behind this TCP throughput methodology is to define		Note that the primary focus of this methodology is managed business
	a methodology for testing sustained TCP layer performance. In this		class IP networks; i.e. those Ethernet terminated services for which
	document, sustained TCP throughput is that amount of data per unit		businesses are provided an SLA from the network provider. End-users
	time that TCP transports during equilibrium (steady state), i.e.		with "best effort" access between locations can use this methodology,
	after the initial slow start phase. We refer to this state as TCP		but this framework and its metrics are intended to be used in a
	Equilibrium, and that the equilibrium throughput is the maximum		predictable managed IP service environment.
	achievable for the TCP connection(s).
	There are many variables to consider when conducting a TCP throughput		So the intent behind this document is to define a methodology for
	test and this methodology focuses on some of the most common		testing sustained TCP layer performance. In this document, the
	parameters that MUST be considered such as:		maximum achievable TCP throughput is that amount of data per unit
			time that TCP transports when trying to reach Equilibrium, i.e.
			after the initial slow start and congestion avoidance phases. We
			refer to this as the maximum achievable TCP Throughput for the TCP
			connection(s).
			TCP uses a congestion window, (TCP CWND), to determine how many
			packets it can send at one time. A larger TCP CWND permits a higher
			throughput. TCP "slow start" and "congestion avoidance" algorithms
			together determine the TCP CWND size. The Maximum TCP CWND size is
			also tributary to the buffer space allocated by the kernel for each
			socket. For each socket, there is a default buffer size that can be
			changed by the program using a system library called just before
			opening the socket. There is also a kernel enforced maximum buffer
			size. This buffer size can be adjusted at both ends of the socket
			(send and receive). In order to obtain the maximum throughput, it
			is critical to use optimal TCP Send and Receive Socket Buffer sizes
			as well as the optimal TCP Receive Window size.
			There are many variables to consider when conducting a TCP throughput
			test and this methodology focuses on the most common:
	- Path MTU and Maximum Segment Size (MSS)		- Path MTU and Maximum Segment Size (MSS)
	- RTT and Bottleneck BW		- RTT and Bottleneck BW
	- Ideal TCP Window (Bandwidth Delay Product)		- Ideal TCP Receive Window (including Ideal Receive Socket Buffer)
	- Single Connection and Multiple Connection testing		- Ideal Send Socket Buffer
			- TCP Congestion Window (TCP CWND)
	This methodology proposes a test which SHOULD be performed in		- Single Connection and Multiple Connections testing
	addition to traditional Layer 2/3 type tests, which are conducted to		This methodology proposes TCP testing that should be performed in
	verify the integrity of the network before conducting TCP tests.		addition to traditional Layer 2/3 type tests. Layer 2/3 tests are
	Examples include iperf (UDP mode) or manual packet layer test		required to verify the integrity of the network before conducting TCP
			test. Examples include iperf (UDP mode) or manual packet layer test
	techniques where packet throughput, loss, and delay measurements are		techniques where packet throughput, loss, and delay measurements are
	conducted. When available, standardized testing similar to RFC 2544		conducted. When available, standardized testing similar to RFC 2544
	[RFC2544] but adapted for use on operational networks may be used		[RFC2544] but adapted for use in operational networks may be used.
	(because RFC 2544 methods are not intended for use outside the lab		Note: RFC 2544 was never meant to be used outside a lab environment.
	environment).
	1.1 Test Set-up and Terminology		1.1 Test Set-up and Terminology
	This section provides a general overview of the test configuration		This section provides a general overview of the test configuration
	for this methodology. The test is intended to be conducted on an		for this methodology. The test is intended to be conducted on an
	end-end operational network, so there are multitudes of network		end-to-end operational and managed IP network. A multitude of
	architectures and topologies that can be tested. This test set-up		network architectures and topologies can be tested. The following
	diagram is very general and the main intent is to illustrate the		set-up diagram is very general and it only illustrates the
	segmentation of the end user and network provider domains.		segmentation within end user and network provider domains.
	Common terminologies used in the test methodology are:		Common terminologies used in the test methodology are:
			- Bottleneck Bandwidth (BB), lowest bandwidth along the complete
			path. Bottleneck Bandwidth and Bandwidth are used synonymously
			in this document. Most of the time the Bottleneck Bandwidth is
			in the access portion of the wide area network (CE - PE)
	- Customer Provided Equipment (CPE), refers to customer owned		- Customer Provided Equipment (CPE), refers to customer owned
	- Customer Edge (CE), refers to provider owned demarcation device		equipment (routers, switches, computers, etc.)
	- Provider Edge (PE), refers to provider located distribution		- Customer Edge (CE), refers to provider owned demarcation device.
	equipment		- End-user: The business enterprise customer. For the purposes of
	- P (Provider), refers to provider core network equipment		conducting TCP throughput tests, this may be the IT department.
	- Bottleneck Bandwidth*, lowest bandwidth along the complete network		- Network Under Test (NUT), refers to the tested IP network path.
	path		- Provider Edge (PE), refers to provider's distribution equipment.
	- Round-Trip Time (RTT), refers to Layer 4 back and forth delay		- P (Provider), refers to provider core network equipment.
	- Round-Trip Delay (RTD), refers to Layer 1 back and forth delay		- Round-Trip Time (RTT), refers to Layer 4 back and forth delay.
	- Network Under Test (NUT), refers to the tested IP network path		- Round-Trip Delay (RTD), refers to Layer 1 back and forth delay.
	- TCP Throughput Test Device (TCP TTD), refers to compliant TCP		- TCP Throughput Test Device (TCP TTD), refers to compliant TCP
	host that generates traffic and measures metrics as defined in		host that generates traffic and measures metrics as defined in
	this methodology		this methodology. i.e. a dedicated communications test instrument.
	+----+ +----+ +----+ +----+ +---+ +---+ +----+ +----+ +----+ +----+		+----+ +----+ +----+ +----+ +---+ +---+ +----+ +----+ +----+ +----+
	| | | | | | | | | | | | | | | | | | | |
	| TCP|-| CPE|-| CE |--| PE |-| P |--| P |-| PE |--| CE |-| CPE|-| TCP|		| TCP|-| CPE|-| CE |--| PE |-| P |--| P |-| PE |--| CE |-| CPE|-| TCP|
	| TD | | | | |BB| | | | | | | |BB| | | | | TD |		| TTD| | | | |BB| | | | | | | |BB| | | | | TTD|
	+----+ +----+ +----+**+----+ +---+ +---+ +----+**+----+ +----+ +----+		+----+ +----+ +----+ +----+ +---+ +---+ +----+ +----+ +----+ +----+
			<------------------------ NUT ------------------------>
	<------------------------ NUT ------------------------>		R >-----------------------------------------------------------|
			T |
	<-------------------------RTT ------------------------>		T <-----------------------------------------------------------|
	* Bottleneck Bandwidth and Bandwidth are used synonomously in this
	document.
	** Most of the time the Bottleneck Bandwidth is in the access portion
	of the wide area network (CE - PE)
	Note that the NUT may consist of a variety of devices including (and		Note that the NUT may consist of a variety of devices including but
	NOT limited to): load balancers, proxy servers, WAN acceleration		not limited to, load balancers, proxy servers or WAN acceleration
	devices. The detailed topology of the NUT MUST be considered when		devices. The detailed topology of the NUT should be well understood
	conducting the TCP throughput tests, but this methodology makes no		when conducting the TCP throughput tests, although this methodology
	attempt to characterize TCP performance related to specific network		makes no attempt to characterize specific network architectures.
	architectures.
	2. Scope and Goals of this Methodology		2. Scope and Goals of this Methodology
	Before defining the goals of this methodology, it is important to		Before defining the goals, it is important to clearly define the
	clearly define the areas that are out-of-scope for this		areas that are out-of-scope.
	methodology.
	- The methodology is not intended to predict TCP throughput		- This methodology is not intended to predict the TCP throughput
	behavior during the transient stages of a TCP connection, such		during the transient stages of a TCP connection, such as the initial
	as initial slow start.		slow start.
	- The methodology is not intended to definitively benchmark TCP		- This methodology is not intended to definitively benchmark TCP
	implementations of one OS to another, although some users MAY find		implementations of one OS to another, although some users may find
	some value in conducting qualitative experiments.		some value in conducting qualitative experiments.
	- The methodology is not intended to provide detailed diagnosis		- This methodology is not intended to provide detailed diagnosis
	of problems within end-points or the network itself as related to		of problems within end-points or within the network itself as
	non-optimal TCP performance, although a results interpretation		related to non-optimal TCP performance, although a results
	section for each test step MAY provide insight into potential		interpretation section for each test step may provide insight in
	issues within the network.		regards with potential issues.
	- The methodology does not propose a method to operate permanently		- This methodology does not propose to operate permanently with high
	with high measurement loads. TCP performance and optimization data of		measurement loads. TCP performance and optimization within
	operational networks MAY be captured and evaluated by using data of		operational networks may be captured and evaluated by using data
	the "TCP Extended Statistics MIB" [RFC4898].		from the "TCP Extended Statistics MIB" [RFC4898].
	- The methodology is not intended to measure TCP throughput as part		- This methodology is not intended to measure TCP throughput as part
	of an SLA, or to compare the TCP performance between service		of an SLA, or to compare the TCP performance between service
	providers or to compare between implementations of this methodology		providers or to compare between implementations of this methodology
	(test equipment).		in dedicated communications test instruments.
	In contrast to the above exclusions, the goals of this methodology		In contrast to the above exclusions, a primary goal is to define a
	are to define a method to conduct a structured, end-to-end		method to conduct a practical, end-to-end assessment of sustained
	assessment of sustained TCP performance within a managed business		TCP performance within a managed business class IP network. Another
	class IP network. A key goal is to establish a set of "best		key goal is to establish a set of "best practices" that a non-TCP
	practices" that an engineer SHOULD apply when validating the		expert should apply when validating the ability of a managed network
	ability of a managed network to carry end-user TCP applications.		to carry end-user TCP applications.
	The specific goals are to:		Other specific goals are to :
	- Provide a practical test approach that specifies well understood,		- Provide a practical test approach that specifies IP hosts
	end-user configurable TCP parameters such as TCP Window size, MSS		configurable TCP parameters such as TCP Receive Window size, Socket
	(Maximum Segment Size), number of connections, and how these affect		Buffer size, MSS (Maximum Segment Size), number of connections, and
	the outcome of TCP performance over a network.		how these affect the outcome of TCP performance over a network.
			See section 3.3.3.
	- Provide specific test conditions (link speed, RTT, TCP Window size,		- Provide specific test conditions like link speed, RTT, TCP Receive
	etc.) and maximum achievable TCP throughput under TCP Equilibrium		Window size, Socket Buffer size and maximum achievable TCP throughput
	conditions. For guideline purposes, provide examples of these test		when trying to reach TCP Equilibrium. For guideline purposes,
	conditions and the maximum achievable TCP throughput during the		provide examples of test conditions and their maximum achievable
	equilibrium state. Section 2.1 provides specific details concerning		TCP throughput. Section 2.1 provides specific details concerning the
	the definition of TCP Equilibrium within the context of this		definition of TCP Equilibrium within this methodology while section 3
	methodology.		provides specific test conditions with examples.
	- Define three (3) basic metrics that can be used to compare the		- Define three (3) basic metrics to compare the performance of TCP
	performance of TCP connections under various network conditions.		connections under various network conditions. See section 3.3.2.
	- In test situations where the RECOMMENDED procedure does not yield		- In test situations where the recommended procedure does not yield
	the maximum achievable TCP throughput result, this methodology		the maximum achievable TCP throughput results, this methodology
	provides some possible areas within the end host or network that		provides some possible areas within the end host or network that
	SHOULD be considered for investigation (although again, this		should be considered for investigation. Although again, this
	methodology is not intended to provide a detailed diagnosis of these		methodology is not intended to provide a detailed diagnosis on these
	issues).		issues. See section 3.3.5.
	2.1 TCP Equilibrium State Throughput		2.1 TCP Equilibrium
	TCP connections have three (3) fundamental congestion window phases		TCP connections have three (3) fundamental congestion window phases
	as documented in [RFC5681]. These phases are:		as documented in [RFC5681].
	- Slow Start, which occurs during the beginning of a TCP transmission		These 3 phases are:
	or after a retransmission time out event.		1 - The Slow Start phase, which occurs at the beginning of a TCP
			transmission or after a retransmission time out.
	- Congestion avoidance, which is the phase during which TCP ramps up		2 - The Congestion Avoidance phase, during which TCP ramps up to
	to establish the maximum attainable throughput on an end-end network		establish the maximum attainable throughput on an end-to-end network
	path. Retransmissions are a natural by-product of the TCP congestion		path. Retransmissions are a natural by-product of the TCP congestion
	avoidance algorithm as it seeks to achieve maximum throughput on		avoidance algorithm as it seeks to achieve maximum throughput.
	the network path.
	- Retransmission phase, which include Fast Retransmit (Tahoe) and
	Fast Recovery (Reno and New Reno). When a packet is lost, the
	Congestion avoidance phase transitions to a Fast Retransmission or
	Recovery Phase dependent upon the TCP implementation.
	The following diagram depicts these phases.
	| ssthresh
	TCP | |
	Through- | | Equilibrium
	put | |\ /\/\/\/\/\ Retransmit /\/\ ...
	| | \ / | Time-out /
	| | \ / | _______ _/
	| Slow _/ |/ | / | Slow _/
	| Start _/ Congestion |/ |Start_/ Congestion
	| _/ Avoidance Loss | _/ Avoidance
	| _/ Event | _/
	| _/ |/
	|/__________________________________________________________
	Time
	This TCP methodology provides guidelines to measure the equilibrium
	throughput which refers to the maximum sustained rate obtained by
	congestion avoidance before packet loss conditions occur (which MAY
	cause the state change from congestion avoidance to a retransmission
	phase). All maximum achievable throughputs specified in Section 3 are
	with respect to this equilibrium state.
	2.2 Metrics for TCP Throughput Tests
	This framework focuses on a TCP throughput methodology and also
	provides several basic metrics to compare results of various
	throughput tests. It is recognized that the complexity and
	unpredictability of TCP makes it impossible to develop a complete
	set of metrics that account for the myriad of variables (i.e. RTT
	variation, loss conditions, TCP implementation, etc.). However,
	these basic metrics will facilitate TCP throughput comparisons
	under varying network conditions and between network traffic
	management techniques.
	The first metric is the TCP Transfer Time, which is simply the
	measured time it takes to transfer a block of data across
	simultaneous TCP connections. The concept is useful when
	benchmarking traffic management techniques, where multiple
	connections MAY be REQUIRED.
	The TCP Transfer time MAY also be used to provide a normalized ratio
	of the actual TCP Transfer Time versus Ideal Transfer Time. This
	ratio is called the TCP Transfer Index and is defined as:
	Actual TCP Transfer Time
	-------------------------
	Ideal TCP Transfer Time
	The Ideal TCP Transfer time is derived from the network path
	bottleneck bandwidth and the various Layer 1/2/3 overheads associated
	with the network path. Additionally, the TCP Window size must be
	tuned to equal the bandwidth delay product (BDP) as described in
	Section 3.3.1.
	The following table illustrates a single connection TCP Transfer and
	the Ideal TCP Transfer time for a 100 MB file with the ideal TCP
	window size based on the BDP.
	Table 2.2: Link Speed, RTT, TCP Throughput, Ideal TCP Transfer time
	Link Maximum Achievable Ideal TCP Transfer time
	Speed RTT (ms) TCP Throughput(Mbps) Time in seconds
	--------------------------------------------------------------------
	T1 20 1.17 684.93
	T1 50 1.40 570.61
	T1 100 1.40 570.61
	T3 10 42.05 19.03
	T3 15 42.05 19.03
	T3 25 41.52 18.82
	T3(ATM) 10 36.50 21.92
	T3(ATM) 15 36.23 22.14
	T3(ATM) 25 36.27 22.05
	100M 1 91.98 8.70
	100M 2 93.44 8.56
	100M 5 93.44 8.56
	1Gig 0.1 919.82 0.87
	1Gig 0.5 934.47 0.86
	1Gig 1 934.47 0.86
	10Gig 0.05 9,344.67 0.09
	10Gig 0.3 9,344.67 0.09
	* Calculation is based on File Size in Bytes X 8 / TCP Throughput.
	** TCP Throughput is derived from Table 3.3.
	To illustrate the TCP Transfer Time Index, an example would be the
	bulk transfer of 100 MB over 5 simultaneous TCP connections (each
	connection uploading 100 MB). In this example, the Ethernet service
	provides a Committed Access Rate (CAR) of 500 Mbit/s. Each
	connection MAY achieve different throughputs during a test and the
	overall throughput rate is not always easy to determine (especially
	as the number of connections increases).
	The ideal TCP Transfer Time would be ~8 seconds, but in this example,
	the actual TCP Transfer Time was 12 seconds. The TCP Transfer Index
	would be 12/8 = 1.5, which indicates that the transfer across all
	connections took 1.5 times longer than the ideal.
	The second metric is the TCP Efficiency metric which is the
	percentage of bytes that were not retransmitted and is defined as:
	Transmitted Bytes - Retransmitted Bytes
	--------------------------------------- x 100
	Transmitted Bytes
	Transmitted bytes are the total number of TCP payload bytes to be
	transmitted which includes the original and retransmitted bytes. This
	metric provides a comparative measure between various QoS mechanisms
	such as traffic management, congestion avoidance, and also various
	TCP implementations (i.e. Reno, Vegas, etc.).
	As an example, if 100,000 bytes were sent and 2,000 had to be
	retransmitted, the TCP Efficiency SHOULD be calculated as:
	102,000 - 2,000
	---------------- x 100 = 98.03%
	102,000
	Note that the retransmitted bytes MAY have occurred more than once,
	and these multiple retransmissions are added to the Retransmitted
	Bytes count (and the Transmitted Bytes count).
	And the third metric is the Buffer Delay Percentage, which represents		3 - The Retransmission Time-out phase, which could include Fast
	the increase in RTT during a TCP throughput test from the inherent		Retransmit (Tahoe) or Fast Recovery (Reno & New Reno). When multiple
	network RTT (baseline RTT). The baseline RTT is the round-trip time		packet lost occurs, Congestion Avoidance phase transitions to Fast
	inherent to the network path under non-congested conditions.		Retransmission or Fast Recovery depending upon TCP implementations.
			If a Time-Out occurs, TCP transitions back to the Slow Start phase.
	The Buffer Delay Percentage is defined as:		The following diagram depicts these 3 phases.
	Average RTT during Transfer - Baseline RTT		| Trying to reach TCP Equilibrium >>>>>>>>>>>>>
	------------------------------------------ x 100		/\ | High ssthresh TCP CWND 3
	Baseline RTT		/\ | Loss Event * halving Retransmission
			/\ | * \ upon loss Time-Out Adjusted
			/\ | * \ /\ _______ ssthresh
			/\ | * \ / \ /M-Loss | *
			TCP | * 2 \/ \ / Events |1 *
			Through- | * Congestion\ / |Slow *
			put | 1 * Avoidance \/ |Start *
			| Slow * Half | *
			| Start * TCP CWND *
			|___*_______________________Minimum TCP CWND after Time-Out_
			Time >>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>
			Note : ssthresh = Slow Start threshold.
	As an example, the baseline RTT for the network path is 25 msec.		Through the above 3 phases, TCP is trying to reach Equilibrium, but
	During the course of a TCP transfer, the average RTT across the		since packet loss is currently its only available feedback indicator,
	entire transfer increased to 32 msec. In this example, the Buffer		TCP will never reach that goal. Although, a well tuned (and managed)
	Delay Percentage WOULD be calculated as:		IP network with well tuned IP hosts and applications should perform
			very close to TCP Equilibrium and to the BB (Bottleneck Bandwidth).
	32 - 25		This TCP methodology provides guidelines to measure the maximum
	------- x 100 = 28%		achievable TCP throughput or maximum TCP sustained rate obtained
	25		after TCP CWND has stabilized to an optimal value. All maximum
			achievable TCP throughputs specified in section 3 are with respect to
			this condition.
	Note that the TCP Transfer Time, TCP Efficiency, and Buffer Delay		It is important to clarify the interaction between the sender's Send
	metrics MUST be measured during each throughput test.		Socket Buffer and the receiver's advertised TCP Receive Window. TCP
	Poor TCP Transfer Time Indexes (TCP Transfer Time greater than Ideal		test programs such as iperf, ttcp, etc. allow the sender to control
	TCP Transfer Times) MAY be diagnosed by correlating with sub-optimal		the quantity of TCP Bytes transmitted and unacknowledged (in-flight),
	TCP Efficiency and/or Buffer Delay Percentage metrics.		commonly referred to as the Send Socket Buffer. This is done
			independently of the TCP Receive Window size advertised by the
			receiver. Implications to the capabilities of the Throughput Test
			Device (TTD) are covered at the end of section 3.
	3. TCP Throughput Testing Methodology		3. TCP Throughput Testing Methodology
	As stated in Section 1, it is considered best practice to verify		As stated earlier in section 1, it is considered best practice to
	the integrity of the network by conducting Layer2/3 stress tests		verify the integrity of the network by conducting Layer2/3 tests such
	such as [RFC2544] or other methods of network stress tests. If the		as [RFC2544] or other methods of network stress tests. Although, it
	network is not performing properly in terms of packet loss, jitter,		is important to mention here that RFC 2544 was never meant to be used
	etc. then the TCP layer testing will not be meaningful since the		outside a lab environment.
	equilibrium throughput MAY be very difficult to achieve (in a
	"dysfunctional" network).
	TCP Throughput testing MAY require cooperation between the end user		If the network is not performing properly in terms of packet loss,
			jitter, etc. then the TCP layer testing will not be meaningful. A
			dysfunctional network will not reach close enough to TCP Equilibrium
			to provide optimal TCP throughputs with the available bandwidth.
			TCP Throughput testing may require cooperation between the end user
	customer and the network provider. In a Layer 2/3 VPN architecture,		customer and the network provider. In a Layer 2/3 VPN architecture,
	the testing SHOULD be conducted on the Customer Edge (CE) router and		the testing should be conducted either on the CPE or on the CE device
	not the Provider Edge (PE) router.		and not on the PE (Provider Edge) router.
	The following represents the sequential order of steps to conduct the		The following represents the sequential order of steps for this
	TCP throughput testing methodology:		testing methodology:
	1. Identify the Path MTU. Packetization Layer Path MTU Discovery		1. Identify the Path MTU. Packetization Layer Path MTU Discovery
	or PLPMTUD, [RFC4821], MUST be conducted to verify the maximum		or PLPMTUD, [RFC4821], MUST be conducted to verify the network path
	network path MTU. Conducting PLPMTUD establishes the upper limit for		MTU. Conducting PLPMTUD establishes the upper limit for the MSS to
	the MSS to be used in subsequent steps.		be used in subsequent steps.
	2. Baseline Round Trip Time and Bandwidth. This step establishes the		2. Baseline Round Trip Time and Bandwidth. This step establishes the
	inherent, non-congested Round Trip Time (RTT) and the bottleneck		inherent, non-congested Round Trip Time (RTT) and the bottleneck
	bandwidth of the end-end network path. These measurements are used		bandwidth of the end-to-end network path. These measurements are
	to provide estimates of the ideal TCP window size, which SHOULD be		used to provide estimates of the ideal TCP Receive Window and Send
	used in subsequent test steps. These measurements reference		Socket Buffer sizes that SHOULD be used in subsequent test steps.
	[RFC2681] and [RFC4898] to measure RTD (and the associated RTT).		These measurements reference [RFC2681] and [RFC4898] to measure RTD
	Also, [RFC5136] is referenced to measure network capacity.		and the associated RTT.
	3. TCP Connection Throughput Tests. With baseline measurements		3. TCP Connection Throughput Tests. With baseline measurements
	of Round Trip Time and bottleneck bandwidth, a series of single and		of Round Trip Time and bottleneck bandwidth, single and multiple TCP
	multiple TCP connection throughput tests SHOULD be conducted to		connection throughput tests SHOULD be conducted to baseline network
	baseline the network performance expectations.		performance expectations.
	4. Traffic Management Tests. Various traffic management and queuing		4. Traffic Management Tests. Various traffic management and queuing
	techniques SHOULD be tested in this step, using multiple TCP		techniques can be tested in this step, using multiple TCP
	connections. Multiple connection testing SHOULD verify that the		connections. Multiple connections testing should verify that the
	network is configured properly for traffic shaping versus policing,		network is configured properly for traffic shaping versus policing,
	various queuing implementations, and RED.		various queuing implementations and RED.
	Important to note are some of the key characteristics and		Important to note are some of the key characteristics and
	considerations for the TCP test instrument. The test host MAY be a		considerations for the TCP test instrument. The test host may be a
	standard computer or dedicated communications test instrument		standard computer or a dedicated communications test instrument.
	and these TCP test hosts be capable of emulating both a client and a		In both cases, they must be capable of emulating both client and
	server.		server.
	Whether the TCP test host is a standard computer or a compliant TCP		The following criteria should be considered when selecting whether
	TTD, the following areas SHOULD be considered when selecting		the TCP test host can be a standard computer or has to be a dedicated
	a test host:		communications test instrument:
	- TCP implementation used by the test host OS version, i.e. Linux OS		- TCP implementation used by the test host, OS version, i.e. Linux OS
	kernel using TCP Reno, TCP options supported, etc. This will		kernel using TCP Reno, TCP options supported, etc. These will
	obviously be more important when using custom test equipment where		obviously be more important when using dedicated communications test
	the TCP implementation MAY be customized or tuned to run in higher		instruments where the TCP implementation may be customized or tuned
	performance hardware. When a compliant TCP TTD is used, the TCP		to run in higher performance hardware. When a compliant TCP TTD is
	implementation SHOULD be identified in the test results. The		used, the TCP implementation MUST be identified in the test results.
	compliant TCP TTD SHOULD be usable for complete end-to-end testing		The compliant TCP TTD should be usable for complete end-to-end
	through network security elements and SHOULD also be usable for		testing through network security elements and should also be usable
	testing network sections.		for testing network sections.
	- Most importantly, the TCP test host must be capable of generating		- More important, the TCP test host MUST be capable to generate
	and receiving stateful TCP test traffic at the full link speed of the		and receive stateful TCP test traffic at the full link speed of the
	network under test. As a general rule of thumb, testing TCP		network under test. Stateful TCP test traffic means that the test
	throughput at rates greater than 100 Mbit/sec MAY require high		host MUST fully implement a TCP stack; this is generally a comment
			aimed at dedicated communications test equipments which sometimes
			"blast" packets with TCP headers. As a general rule of thumb, testing
			TCP throughput at rates greater than 100 Mbit/sec MAY require high
	performance server hardware or dedicated hardware based test tools.		performance server hardware or dedicated hardware based test tools.
	Thus, other devices cannot realize higher TCP throughput, and user
	expectations SHOULD be set accordingly with user manual or notes on
	the results report.
	- Measuring RTT and TCP Efficiency per connection will generally		- A compliant TCP Throughput Test Device MUST allow adjusting both
	require dedicated hardware based test tools. In the absence of		Send Socket Buffer and TCP Receive Window sizes. The Receive Socket
	dedicated hardware based test tools, these measurements MAY need to		Buffer MUST be large enough to accommodate the TCP Receive Window.
	be conducted with packet capture tools (conduct TCP throughput tests
	and analyze RTT and retransmission results with packet captures).		- Measuring RTT and retransmissions per connection will generally
	Another option MAY be to use "TCP Extended Statistics MIB" per		require a dedicated communications test instrument. In the absence of
			dedicated hardware based test tools, these measurements may need to
			be conducted with packet capture tools, i.e. conduct TCP throughput
			tests and analyze RTT and retransmission results in packet captures.
			Another option may be to use "TCP Extended Statistics MIB" per
	[RFC4898].		[RFC4898].
	- The compliant TCP TTD and its access to the network under test MUST		- The RFC4821 PLPMTUD test SHOULD be conducted with a dedicated
	NOT introduce a performance bottleneck of any kind.		tester which exposes the ability to run the PLPMTUD algorithm
			independent from the OS stack.
	3.1. Determine Network Path MTU		3.1. Determine Network Path MTU
	TCP implementations SHOULD use Path MTU Discovery techniques (PMTUD).		TCP implementations should use Path MTU Discovery techniques (PMTUD).
	PMTUD relies on ICMP 'need to frag' messages to learn the path MTU.		PMTUD relies on ICMP 'need to frag' messages to learn the path MTU.
	When a device has a packet to send which has the Don't Fragment (DF)		When a device has a packet to send which has the Don't Fragment (DF)
	bit in the IP header set and the packet is larger than the Maximum		bit in the IP header set and the packet is larger than the Maximum
	Transmission Unit (MTU) of the next hop link, the packet is dropped		Transmission Unit (MTU) of the next hop, the packet is dropped and
	and the device sends an ICMP 'need to frag' message back to the host		the device sends an ICMP 'need to frag' message back to the host that
	that originated the packet. The ICMP 'need to frag' message includes		originated the packet. The ICMP 'need to frag' message includes
	the next hop MTU which PMTUD uses to tune the TCP Maximum Segment		the next hop MTU which PMTUD uses to tune the TCP Maximum Segment
	Size (MSS). Unfortunately, because many network managers completely		Size (MSS). Unfortunately, because many network managers completely
	disable ICMP, this technique does not always prove reliable in real		disable ICMP, this technique does not always prove reliable.
	world situations.
	Packetization Layer Path MTU Discovery or PLPMTUD [RFC4821] MUST		Packetization Layer Path MTU Discovery or PLPMTUD [RFC4821] MUST then
	be conducted to verify the minimum network path MTU. PLPMTUD can		be conducted to verify the network path MTU. PLPMTUD can be used
	be used with or without ICMP. The following sections provide a		with or without ICMP. The following sections provide a summary of the
	summary of the PLPMTUD approach and an example using the TCP		PLPMTUD approach and an example using TCP. [RFC4821] specifies a
	protocol. [RFC4821] specifies a search_high and a search_low		search_high and a search_low parameter for the MTU. As specified in
	parameter for the MTU. As specified in [RFC4821], a value of 1024 is		[RFC4821], 1024 Bytes is a safe value for search_low in modern
	a generally safe value to choose for search_low in modern networks.		networks.
	It is important to determine the overhead of the links in the path,		It is important to determine the links overhead along the IP path,
	and then to select a TCP MSS size corresponding to the Layer 3 MTU.		and then to select a TCP MSS size corresponding to the Layer 3 MTU.
	For example, if the MTU is 1024 bytes and the TCP/IP headers are 40		For example, if the MTU is 1024 Bytes and the TCP/IP headers are 40
	bytes, then the MSS would be set to 984 bytes.		Bytes, then the MSS would be set to 984 Bytes.
	An example scenario is a network where the actual path MTU is 1240		An example scenario is a network where the actual path MTU is 1240
	bytes. The TCP client probe MUST be capable of setting the MSS for		Bytes. The TCP client probe MUST be capable of setting the MSS for
	the probe packets and could start at MSS = 984 (which corresponds		the probe packets and could start at MSS = 984 (which corresponds
	to an MTU size of 1024 bytes).		to an MTU size of 1024 Bytes).
	The TCP client probe would open a TCP connection and advertise the		The TCP client probe would open a TCP connection and advertise the
	MSS as 984. Note that the client probe MUST generate these packets		MSS as 984. Note that the client probe MUST generate these packets
	with the DF bit set. The TCP client probe then sends test traffic		with the DF bit set. The TCP client probe then sends test traffic
	per a nominal window size (8KB, etc.). The window size SHOULD be		per a small default Send Socket Buffer size of ~8KBytes. It should
	kept small to minimize the possibility of congesting the network,		be kept small to minimize the possibility of congesting the network,
	which MAY induce congestive loss. The duration of the test should		which may induce packet loss. The duration of the test should also
	also be short (10-30 seconds), again to minimize congestive effects		be short (10-30 seconds), again to minimize congestive effects
	during the test.		during the test.
	In the example of a 1240 byte path MTU, probing with an MSS equal to		In the example of a 1240 Bytes path MTU, probing with an MSS equal to
	984 would yield a successful probe and the test client packets would		984 would yield a successful probe and the test client packets would
	be successfully transferred to the test server.		be successfully transferred to the test server.
	Also note that the test client MUST verify that the MSS advertised		Also note that the test client MUST verify that the MSS advertised
	is indeed negotiated. Network devices with built-in Layer 4		is indeed negotiated. Network devices with built-in Layer 4
	capabilities can intercede during the connection establishment		capabilities can intercede during the connection establishment and
	process and reduce the advertised MSS to avoid fragmentation. This		reduce the advertised MSS to avoid fragmentation. This is certainly
	is certainly a desirable feature from a network perspective, but		a desirable feature from a network perspective, but it can yield
	can yield erroneous test results if the client test probe does not		erroneous test results if the client test probe does not confirm the
	confirm the negotiated MSS.		negotiated MSS.
	The next test probe would use the search_high value and this would		The next test probe would use the search_high value and this would
	be set to MSS = 1460 to correspond to a 1500 byte MTU. In this		be set to MSS = 1460 to correspond to a 1500 Bytes MTU. In this
	example, the test client MUST retransmit based upon time-outs (since		example, the test client will retransmit based upon time-outs, since
	no ACKs will be received from the test server). This test probe is		no ACKs will be received from the test server. This test probe is
	marked as a conclusive failure if none of the test packets are		marked as a conclusive failure if none of the test packets are
	ACK'ed. If any of the test packets are ACK'ed, congestive network		ACK'ed. If any of the test packets are ACK'ed, congestive network
	MAY be the cause and the test probe is not conclusive. Re-testing		may be the cause and the test probe is not conclusive. Re-testing
	at other times of the day is RECOMMENDED to further isolate.		at other times of the day is recommended to further isolate.
	The test is repeated until the desired granularity of the MTU is		The test is repeated until the desired granularity of the MTU is
	discovered. The method can yield precise results at the expense of		discovered. The method can yield precise results at the expense of
	probing time. One approach MAY be to reduce the probe size to		probing time. One approach may be to reduce the probe size to
	half between the unsuccessful search_high and successful search_low		half between the unsuccessful search_high and successful search_low
	value, and increase by increments of 1/2 when seeking the upper		value and raise it by half also when seeking the upper limit.
	limit.
	3.2. Baseline Round Trip Time and Bandwidth		3.2. Baseline Round Trip Time and Bandwidth
	Before stateful TCP testing can begin, it is important to determine		Before stateful TCP testing can begin, it is important to determine
	the baseline Round Trip Time (non-congested inherent delay) and		the baseline Round Trip Time (non-congested inherent delay) and
	bottleneck bandwidth of the end-end network to be tested. These		bottleneck bandwidth of the end-to-end network to be tested. These
	measurements are used to provide estimates of the ideal TCP window		measurements are used to provide estimates of the ideal TCP Receive
	size, which SHOULD be used in subsequent test steps. These latency		Window and Send Socket Buffer sizes that SHOULD be used in subsequent
	and bandwidth tests SHOULD be run during the time of day for which		test steps.
	the TCP throughput tests will occur.
	3.2.1 Techniques to Measure Round Trip Time		3.2.1 Techniques to Measure Round Trip Time
	Following the definitions used in the section 1.1; Round Trip Time		Following the definitions used in section 1.1, Round Trip Time (RTT)
	(RTT) is the time elapsed between the clocking in of the first bit		is the elapsed time between the clocking in of the first bit of a
	of a payload packet to the receipt of the last bit of the		payload sent packet to the receipt of the last bit of the
	corresponding Acknowledgment. Round Trip Delay (RTD) is used		corresponding Acknowledgment. Round Trip Delay (RTD) is used
	synonymously to twice the Link Latency. RTT measurements SHOULD use		synonymously to twice the Link Latency. RTT measurements SHOULD use
	techniques defined in [RFC2681] or statistics available from MIBs		techniques defined in [RFC2681] or statistics available from MIBs
	defined in [RFC4898].		defined in [RFC4898].
	The RTT SHOULD be baselined during "off-peak" hours to obtain a		The RTT SHOULD be baselined during "off-peak" hours to obtain a
	reliable figure for inherent network latency versus additional delay		reliable figure for inherent network latency versus additional delay
	caused by network buffering delays.		caused by network buffering. When sampling values of RTT over a test
			interval, the minimum value measured SHOULD be used as the baseline
	During the actual sustained TCP throughput tests, RTT MUST be		RTT since this will most closely estimate the inherent network
	measured along with TCP throughput. Buffer delay effects can be		latency. This inherent RTT is also used to determine the Buffer
	isolated if RTT is concurrently measured.		Delay Percentage metric which is defined in Section 3.3.2
			The following list is not meant to be exhaustive, although it
			summarizes some of the most common ways to determine round trip time.
			The desired resolution of the measurement (i.e. msec versus usec) may
			dictate whether the RTT measurement can be achieved with ICMP pings
			or by a dedicated communications test instrument with precision
			timers.
	This is not meant to provide an exhaustive list, but summarizes some		The objective in this section is to list several techniques
	of the more common ways to determine round trip time (RTT) through
	the network. The desired resolution of the measurement (i.e. msec
	versus usec) may dictate whether the RTT measurement can be achieved
	with standard tools such as ICMP ping techniques or whether
	specialized test equipment would be required with high precision
	timers. The objective in this section is to list several techniques
	in order of decreasing accuracy.		in order of decreasing accuracy.
	- Use test equipment on each end of the network, "looping" the		- Use test equipment on each end of the network, "looping" the
	far-end tester so that a packet stream can be measured end-end. This		far-end tester so that a packet stream can be measured back and forth
	test equipment RTT measurement MAY be compatible with delay		from end-to-end. This RTT measurement may be compatible with delay
	measurement protocols specified in [RFC5357].		measurement protocols specified in [RFC5357].
	- Conduct packet captures of TCP test applications using for example		- Conduct packet captures of TCP test sessions using "iperf" or FTP,
	"iperf" or FTP, etc. By running multiple experiments, the packet		or other TCP test applications. By running multiple experiments,
	captures can be studied to estimate RTT based upon the SYN -> SYN-ACK		packet captures can then be analyzed to estimate RTT based upon the
	handshakes within the TCP connection set-up.		SYN -> SYN-ACK from the 3 way handshake at the beginning of the TCP
			sessions. Although, note that Firewalls might slow down 3 way
			handshakes, so it might be useful to compare with measured RTT later
			on in the same capture.
	- ICMP Pings MAY also be adequate to provide round trip time		- ICMP Pings may also be adequate to provide round trip time
	estimations. Some limitations of ICMP Ping MAY include msec		estimations. Some limitations with ICMP Ping may include msec
	resolution and whether the network elements respond to pings (or		resolution and whether the network elements are responding to pings
	block them).		or not. Also, ICMP is often rate-limited and segregated into
			different buffer queues, so it is not as reliable and accurate as
			in-band measurements.
	3.2.2 Techniques to Measure End-end Bandwidth		3.2.2 Techniques to Measure end-to-end Bandwidth
	There are many well established techniques available to provide		There are many well established techniques available to provide
	estimated measures of bandwidth over a network. This measurement		estimated measures of bandwidth over a network. These measurements
	SHOULD be conducted in both directions of the network, especially for		SHOULD be conducted in both directions of the network, especially for
	access networks which MAY be asymmetrical. Measurements SHOULD use		access networks, which may be asymmetrical. Measurements SHOULD use
	network capacity techniques defined in [RFC5136].		network capacity techniques defined in [RFC5136].
	The bandwidth measurement test MUST be run with stateless IP streams		Before any TCP Throughput test can be done, a bandwidth measurement
	(not stateful TCP) in order to determine the available bandwidth in		test MUST be run with stateless IP streams(not stateful TCP) in order
	each direction. And this test SHOULD obviously be performed at		to determine the available bandwidths in each direction. This test
	various intervals throughout a business day (or even across a week).		should obviously be performed at various intervals throughout a
	Ideally, the bandwidth test SHOULD produce a log output of the		business day or even across a week. Ideally, the bandwidth test
	bandwidth achieved across the test interval.		should produce logged outputs of the achieved bandwidths across the
			test interval.
	3.3. TCP Throughput Tests		3.3. TCP Throughput Tests
	This methodology specifically defines TCP throughput techniques to		This methodology specifically defines TCP throughput techniques to
	verify sustained TCP performance in a managed business network.		verify sustained TCP performance in a managed business IP network, as
	Defined in section 2.1, the equilibrium throughput reflects the		defined in section 2.1. This section and others will define the
	maximum rate achieved by a TCP connection within the congestion		method to conduct these sustained TCP throughput tests and guidelines
	avoidance phase on an end-end network path. This section and others		for the predicted results.
	will define the method to conduct these sustained throughput tests
	and guidelines of the predicted results.
	With baseline measurements of round trip time and bandwidth		With baseline measurements of round trip time and bandwidth
	from section 3.2, a series of single and multiple TCP connection		from section 3.2, a series of single and multiple TCP connection
	throughput tests can be conducted to baseline network performance		throughput tests SHOULD be conducted to baseline network performance
	against expectations.		against expectations. The number of trials and the type of testing
			(single versus multiple connections) will vary according to the
			intention of the test. One example would be a single connection test
			in which the throughput achieved by large Send Socket Buffer and TCP
			Receive Window sizes (i.e. 256KB) is to be measured. It would be
			advisable to test performance at various times of the business day.
	It is RECOMMENDED to run the tests in each direction independently		It is RECOMMENDED to run the tests in each direction independently
	first, then run both directions simultaneously. In each case, the		first, then run both directions simultaneously. In each case,
	TCP Transfer Time, TCP Efficiency, and Buffer Delay metrics MUST be		TCP Transfer Time, TCP Efficiency, and Buffer Delay Percentage MUST
	measured in each direction.		be measured in each direction. These metrics are defined in 3.3.2.
	3.3.1 Calculate Ideal TCP Window Size		3.3.1 Calculate Ideal TCP Receive Window Size
	The ideal TCP window size can be calculated from the bandwidth		The ideal TCP Receive Window size can be calculated from the
	delay product (BDP), which is:		bandwidth delay product (BDP), which is:
	BDP (bits) = RTT (sec) x Bandwidth (bps)		BDP (bits) = RTT (sec) x Bandwidth (bps)
	By dividing the BDP by 8, the "ideal" TCP window size is calculated.		Note that the RTT is being used as the "Delay" variable in the
	An example would be a T3 link with 25 msec RTT. The BDP would equal		BDP calculations.
	~1,105,000 bits and the ideal TCP window would equal ~138,000 bytes.
	The following table provides some representative network link speeds,		Then, by dividing the BDP by 8, we obtain the "ideal" TCP Receive
	latency, BDP, and associated ideal TCP window size. Sustained		Window size in Bytes. For optimal results, the Send Socket Buffer
	TCP transfers SHOULD reach nearly 100% throughput, minus the overhead		size must be adjusted to the same value at the opposite end of the
	of Layers 1-3 and the divisor of the MSS into the TCP Window.		network path.
	For this single connection baseline test, the MSS size will effect		Ideal TCP RWIN = BDP / 8
	the achieved throughput (especially for smaller TCP Window sizes).
	Table 3.2 provides the achievable, equilibrium TCP throughput (at
	Layer 4) using 1460 byte MSS. Also in this table, the 58 byte L1-L4
	overhead including the Ethernet CRC32 is used for simplicity.
	Table 3.3: Link Speed, RTT and calculated BDP, TCP Throughput		An example would be a T3 link with 25 msec RTT. The BDP would equal
			~1,105,000 bits and the ideal TCP Receive Window would be ~138
			KBytes.
	Link Ideal TCP Maximum Achievable		The following table provides some representative network Link Speeds,
	Speed* RTT (ms) BDP (bits) Window (kBytes) TCP Throughput(Mbps)		RTT, BDP, and their associated Ideal TCP Receive Window sizes.
			Table 3.3.1: Link Speed, RTT and calculated BDP & TCP Receive Window
			Link Ideal TCP
			Speed* RTT BDP Receive Window
			(Mbps) (ms) (bits) (KBytes)
	---------------------------------------------------------------------		---------------------------------------------------------------------
	T1 20 30,720 3.84 1.17		1.536 20 30,720 3.84
	T1 50 76,800 9.60 1.40		1.536 50 76,800 9.60
	T1 100 153,600 19.20 1.40		1.536 100 153,600 19.20
	T3 10 442,100 55.26 42.05		44.21 10 442,100 55.26
	T3 15 663,150 82.89 42.05		44.21 15 663,150 82.89
	T3 25 1,105,250 138.16 41.52		44.21 25 1,105,250 138.16
	T3(ATM) 10 407,040 50.88 36.50		100 1 100,000 12.50
	T3(ATM) 15 610,560 76.32 36.23		100 2 200,000 25.00
	T3(ATM) 25 1,017,600 127.20 36.27		100 5 500,000 62.50
	100M 1 100,000 12.50 91.98		1,000 0.1 100,000 12.50
	100M 2 200,000 25.00 93.44		1,000 0.5 500,000 62.50
	100M 5 500,000 62.50 93.44		1,000 1 1,000,000 125.00
	1Gig 0.1 100,000 12.50 919.82		10,000 0.05 500,000 62.50
	1Gig 0.5 500,000 62.50 934.47		10,000 0.3 3,000,000 375.00
	1Gig 1 1,000,000 125.00 934.47
	10Gig 0.05 500,000 62.50 9,344.67
	10Gig 0.3 3,000,000 375.00 9,344.67
	* Note that link speed is the bottleneck bandwidth for the NUT		* Note that link speed is the bottleneck bandwidth for the NUT
	Also, the following link speeds (available payload bandwidth) were
	used for the WAN entries:
	- T1 = 1.536 Mbits/sec (B8ZS line encoding facility)		The following serial link speeds are used:
	- T3 = 44.21 Mbits/sec (C-Bit Framing)		- T1 = 1.536 Mbits/sec (for a B8ZS line encoding facility)
	- T3(ATM) = 36.86 Mbits/sec (C-Bit Framing & PLCP, 96000 Cells per		- T3 = 44.21 Mbits/sec (for a C-Bit Framing facility)
	second)
	The calculation method used in this document is a 3 step process :		The above table illustrates the ideal TCP Receive Window size.
			If a smaller TCP Receive Window is used, then the TCP Throughput
			is not optimal. To calculate the Ideal TCP Throughput, the following
			formula is used: TCP Throughput = TCP RWIN X 8 / RTT
	1 - Determine what SHOULD be the optimal TCP Window size value		An example could be a 100 Mbps IP path with 5 ms RTT and a TCP
	based on the optimal quantity of "in-flight" octets discovered by		Receive Window size of 16KB, then:
	the BDP calculation. We take into consideration that the TCP
	Window size has to be an exact multiple value of the MSS.
	2 - Calculate the achievable layer 2 throughput by multiplying the
	value determined in step 1 with the MSS & (MSS + L2 + L3 + L4
	Overheads) divided by the RTT.
	3 - Finally, multiply the calculated value of step 2 by the MSS
	versus (MSS + L2 + L3 + L4 Overheads) ratio.
	This provides the achievable TCP Throughput value. Sometimes, the		TCP Throughput = 16 KBytes X 8 bits / 5 ms.
	maximum achievable throughput is limited by the maximum achievable		TCP Throughput = 128,000 bits / 0.005 sec.
	quantity of Ethernet Frames per second on the physical media. Then		TCP Throughput = 25.6 Mbps.
	this value is used in step 2 instead of the calculated one.
	The following diagram compares achievable TCP throughputs on a T3 link		Another example for a T3 using the same calculation formula is
	with Windows 2000/XP TCP window sizes of 16KB versus 64KB.		illustrated on the next page:
			TCP Throughput = TCP RWIN X 8 / RTT.
			TCP Throughput = 16 KBytes X 8 bits / 10 ms.
			TCP Throughput = 128,000 bits / 0.01 sec.
			TCP Throughput = 12.8 Mbps.
			When the TCP Receive Window size exceeds the BDP (i.e. T3 link,
			64 KBytes TCP Receive Window on a 10 ms RTT path), the maximum frames
			per second limit of 3664 is reached and the calculation formula is:
			TCP Throughput = Max FPS X MSS X 8.
			TCP Throughput = 3664 FPS X 1460 Bytes X 8 bits.
			TCP Throughput = 42.8 Mbps
			The following diagram compares achievable TCP throughputs on a T3
			with Send Socket Buffer & TCP Receive Window sizes of 16KB vs. 64KB.
	45|		45|
	| _____42.1M		| _______42.8M
	40| |64K|		40| |64KB |
	TCP | | |		TCP | | |
	Throughput 35| | | _____34.3M		Throughput 35| | |
	in Mbps | | | |64K|		in Mbps | | | +-----+34.1M
	30| | | | |		30| | | |64KB |
	| | | | |		| | | | |
	25| | | | |		25| | | | |
	| | | | |		| | | | |
	20| | | | | _____20.5M		20| | | | | _______20.5M
	| | | | | |64K|		| | | | | |64KB |
	15| 14.5M____| | | | | |		15| | | | | | |
	| |16K| | | | | |		|12.8M+-----| | | | | |
	10| | | | 9.6M+---+ | | |		10| |16KB | | | | | |
	| | | | |16K| | 5.8M____+ |		| | | |8.5M+-----| | | |
	5| | | | | | | |16K| |		5| | | | |16KB | |5.1M+-----| |
	|______+___+___+_______+___+___+_______+__ +___+_______		|_____|_____|_____|____|_____|_____|____|16KB |_____|_____
	10 15 25		10 15 25
	RTT in milliseconds		RTT in milliseconds
	The following diagram shows the achievable TCP throughput on a 25ms		The following diagram shows the achievable TCP throughput on a 25ms
	T3 when the TCP Window size is increased and with the [RFC1323] TCP		T3 when Send Socket Buffer & TCP Receive Window sizes are increased.
	Window scaling option.
	45|		45|
	| +-----+42.47M		|
	40| | |		40| +-----+40.9M
	TCP | | |		TCP | | |
	Throughput 35| | |		Throughput 35| | |
	in Mbps | | |		in Mbps | | |
	30| | |		30| | |
	| | |		| | |
	25| | |		25| | |
	| ______ 21.23M | |		| | |
	20| | | | |		20| +-----+20.5M | |
	| | | | |		| | | | |
	15| | | | |		15| | | | |
	| | | | |		| | | | |
	10| +----+10.62M | | | |		10| +-----+10.2M | | | |
	| _______5.31M | | | | | |		| | | | | | |
	5| | | | | | | | |		5| +-----+5.1M | | | | | |
	|__+_____+______+____+__________+____+________+_____+___		|_____|_____|______|_____|______|_____|_______|_____|_____
	16 32 64 128		16 32 64 128*
	TCP Window size in KBytes		TCP Receive Window size in KBytes
	3.3.2 Conducting the TCP Throughput Tests		* Note that 128KB requires [RFC1323] TCP Window scaling option.
	There are several TCP tools that are commonly used in the network		3.3.2 Metrics for TCP Throughput Tests
	world and one of the most common is the "iperf" tool. With this tool,
	hosts are installed at each end of the network segment; one as client
	and the other as server. The TCP Window size of both the client and
	the server can be manually set and the achieved throughput is
	measured, either uni-directionally or bi-directionally. For higher
	BDP situations in lossy networks (long fat networks or satellite
	links, etc.), TCP options such as Selective Acknowledgment SHOULD be
	considered and also become part of the window size / throughput
	characterization.
	Host hardware performance MUST be well understood before conducting		This framework focuses on a TCP throughput methodology and also
	the TCP throughput tests and other tests in the following sections.		provides several basic metrics to compare results of various
	Dedicated test equipment will generally be REQUIRED, especially for		throughput tests. It is recognized that the complexity and
	line rates of GigE and 10 GigE. A compliant TCP TTD SHOULD provide a		unpredictability of TCP makes it impossible to develop a complete
	warning message when the expected test throughput will exceed 10% of		set of metrics that accounts for the myriad of variables (i.e. RTT
	the network bandwidth capacity. If the throughput test is expected		variation, loss conditions, TCP implementation, etc.). However,
	to exceed 10% of the provider bandwidth, then the test SHOULD be		these basic metrics will facilitate TCP throughput comparisons
	coordinated with the network provider. This does not include the		under varying network conditions and between network traffic
	customer premise bandwidth, the 10% refers directly to the provider's		management techniques.
	bandwidth (Provider Edge to Provider router).
	The TCP throughput test SHOULD be run over a long enough duration		The first metric is the TCP Transfer Time, which is simply the
	to properly exercise network buffers and also characterize		measured time it takes to transfer a block of data across
	performance during different time periods of the day.		simultaneous TCP connections. This concept is useful when
			benchmarking traffic management techniques and where multiple
			TCP connections are required.
	Note that both the TCP Transfer Time, TCP Efficiency, and Buffer		TCP Transfer time may also be used to provide a normalized ratio of
	Delay metrics MUST be measured during each throughput test.		the actual TCP Transfer Time versus the Ideal Transfer Time. This
	Poor TCP Transfer Time Indexes (TCP Transfer Time greater than Ideal		ratio is called the TCP Transfer Index and is defined as:
	TCP Transfer Times) MAY be diagnosed by correlating with sub-optimal
	TCP Efficiency and/or Buffer Delay Percentage metrics.
	3.3.3 Single vs. Multiple TCP Connection Testing		Actual TCP Transfer Time
			-------------------------
			Ideal TCP Transfer Time
			The Ideal TCP Transfer time is derived from the network path
			bottleneck bandwidth and various Layer 1/2/3/4 overheads associated
			with the network path. Additionally, both the TCP Receive Window and
			the Send Socket Buffer sizes must be tuned to equal the bandwidth
			delay product (BDP) as described in section 3.3.1.
			The following table illustrates the Ideal TCP Transfer time of a
			single TCP connection when its TCP Receive Window and Send Socket
			Buffer sizes are equal to the BDP.
			Table 3.3.2: Link Speed, RTT, BDP, TCP Throughput, and
			Ideal TCP Transfer time for a 100 MB File
			Link Maximum Ideal TCP
			Speed BDP Achievable TCP Transfer time
			(Mbps) RTT (ms) (KBytes) Throughput(Mbps) (seconds)
			--------------------------------------------------------------------
			1.536 50 9.6 1.4 571
			44.21 25 138.2 42.8 18
			100 2 25.0 94.9 9
			1,000 1 125.0 949.2 1
			10,000 0.05 62.5 9,492 0.1
			Transfer times are rounded for simplicity.
			For a 100MB file(100 x 8 = 800 Mbits), the Ideal TCP Transfer Time
			is derived as follows:
			800 Mbits
			Ideal TCP Transfer Time = -----------------------------------
			Maximum Achievable TCP Throughput
			The maximum achievable layer 2 throughput on T1 and T3 Interfaces
			is based on the maximum frames per second (FPS) permitted by the
			actual layer 1 speed when the MTU is 1500 Bytes.
			The maximum FPS for a T1 is 127 and the calculation formula is:
			FPS = T1 Link Speed / ((MTU + PPP + Flags + CRC16) X 8)
			FPS = (1.536M /((1500 Bytes + 4 Bytes + 2 Bytes + 2 Bytes) X 8 )))
			FPS = (1.536M / (1508 Bytes X 8))
			FPS = 1.536 Mbps / 12064 bits
			FPS = 127
			The maximum FPS for a T3 is 3664 and the calculation formula is:
			FPS = T3 Link Speed / ((MTU + PPP + Flags + CRC16) X 8)
			FPS = (44.21M /((1500 Bytes + 4 Bytes + 2 Bytes + 2 Bytes) X 8 )))
			FPS = (44.21M / (1508 Bytes X 8))
			FPS = 44.21 Mbps / 12064 bits
			FPS = 3664
			The 1508 equates to:
			MTU + PPP + Flags + CRC16
			Where MTU is 1500 Bytes, PPP is 4 Bytes, Flags are 2 Bytes and CRC16
			is 2 Bytes.
			Then, to obtain the Maximum Achievable TCP Throughput (layer 4), we
			simply use: MSS in Bytes X 8 bits X max FPS.
			For a T3, the maximum TCP Throughput = 1460 Bytes X 8 bits X 3664 FPS
			Maximum TCP Throughput = 11680 bits X 3664 FPS
			Maximum TCP Throughput = 42.8 Mbps.
			The maximum achievable layer 2 throughput on Ethernet Interfaces is
			based on the maximum frames per second permitted by the IEEE802.3
			standard when the MTU is 1500 Bytes.
			The maximum FPS for 100M Ethernet is 8127 and the calculation is:
			FPS = (100Mbps /(1538 Bytes X 8 bits))
			The maximum FPS for GigE is 81274 and the calculation formula is:
			FPS = (1Gbps /(1538 Bytes X 8 bits))
			The maximum FPS for 10GigE is 812743 and the calculation formula is:
			FPS = (10Gbps /(1538 Bytes X 8 bits))
			The 1538 equates to:
			MTU + Eth + CRC32 + IFG + Preamble + SFD
			Where MTU is 1500 Bytes, Ethernet is 14 Bytes, CRC32 is 4 Bytes,
			IFG is 12 Bytes, Preamble is 7 Bytes and SFD is 1 Byte.
			Note that better results could be obtained with jumbo frames on
			GigE and 10 GigE.
			Then, to obtain the Maximum Achievable TCP Throughput (layer 4), we
			simply use: MSS in Bytes X 8 bits X max FPS.
			For a 100M, the maximum TCP Throughput = 1460 B X 8 bits X 8127 FPS
			Maximum TCP Throughput = 11680 bits X 8127 FPS
			Maximum TCP Throughput = 94.9 Mbps.
			To illustrate the TCP Transfer Time Index, an example would be the
			bulk transfer of 100 MB over 5 simultaneous TCP connections (each
			connection uploading 100 MB). In this example, the Ethernet service
			provides a Committed Access Rate (CAR) of 500 Mbit/s. Each
			connection may achieve different throughputs during a test and the
			overall throughput rate is not always easy to determine (especially
			as the number of connections increases).
			The ideal TCP Transfer Time would be ~8 seconds, but in this example,
			the actual TCP Transfer Time was 12 seconds. The TCP Transfer Index
			would then be 12/8 = 1.5, which indicates that the transfer across
			all connections took 1.5 times longer than the ideal.
			The second metric is TCP Efficiency, which is the percentage of Bytes
			that were not retransmitted and is defined as:
			Transmitted Bytes - Retransmitted Bytes
			--------------------------------------- x 100
			Transmitted Bytes
			Transmitted Bytes are the total number of TCP payload Bytes to be
			transmitted which includes the original and retransmitted Bytes. This
			metric provides a comparative measure between various QoS mechanisms
			like traffic management or congestion avoidance. Various TCP
			implementations like Reno, Vegas, etc. could also be compared.
			As an example, if 100,000 Bytes were sent and 2,000 had to be
			retransmitted, the TCP Efficiency should be calculated as:
			102,000 - 2,000
			---------------- x 100 = 98.03%
			102,000
			Note that the retransmitted Bytes may have occurred more than once,
			and these multiple retransmissions are added to the Retransmitted
			Bytes count (and the Transmitted Bytes count).
			The third metric is the Buffer Delay Percentage, which represents the
			increase in RTT during a TCP throughput test with respect to
			inherent or baseline network RTT. The baseline RTT is the round-trip
			time inherent to the network path under non-congested conditions.
			(See 3.2.1 for details concerning the baseline RTT measurements).
			The Buffer Delay Percentage is defined as:
			Average RTT during Transfer - Baseline RTT
			------------------------------------------ x 100
			Baseline RTT
			As an example, the baseline RTT for the network path is 25 msec.
			During the course of a TCP transfer, the average RTT across the
			entire transfer increased to 32 msec. In this example, the Buffer
			Delay Percentage would be calculated as:
			32 - 25
			------- x 100 = 28%
			25
			Note that the TCP Transfer Time, TCP Efficiency, and Buffer Delay
			Percentage MUST be measured during each throughput test. Poor TCP
			Transfer Time Indexes (TCP Transfer Time greater than Ideal TCP
			Transfer Times) may be diagnosed by correlating with sub-optimal TCP
			Efficiency and/or Buffer Delay Percentage metrics.
			3.3.3 Conducting the TCP Throughput Tests
			Several TCP tools are currently used in the network world and one of
			the most common is "iperf". With this tool, hosts are installed at
			each end of the network path; one acts as client and the other as
			a server. The Send Socket Buffer and the TCP Receive Window sizes
			of both client and server can be manually set. The achieved
			throughput can then be measured, either uni-directionally or
			bi-directionally. For higher BDP situations in lossy networks
			(long fat networks or satellite links, etc.), TCP options such as
			Selective Acknowledgment SHOULD be considered and become part of
			the window size / throughput characterization.
			Host hardware performance must be well understood before conducting
			the tests described in the following sections. A dedicated
			communications test instrument will generally be required, especially
			for line rates of GigE and 10 GigE. A compliant TCP TTD SHOULD
			provide a warning message when the expected test throughput will
			exceed 10% of the network bandwidth capacity. If the throughput test
			is expected to exceed 10% of the provider bandwidth, then the test
			should be coordinated with the network provider. This does not
			include the customer premise bandwidth, the 10% refers directly to
			the provider's bandwidth (Provider Edge to Provider router).
			The TCP throughput test should be run over a long enough duration
			to properly exercise network buffers (greater than 30 seconds) and
			also characterize performance at different time periods of the day.
			3.3.4 Single vs. Multiple TCP Connection Testing
	The decision whether to conduct single or multiple TCP connection		The decision whether to conduct single or multiple TCP connection
	tests depends upon the size of the BDP in relation to the window		tests depends upon the size of the BDP in relation to the configured
	sizes configured in the end-user environment. For example, if the		TCP Receive Window sizes configured in the end-user environment.
	BDP for a long-fat pipe turns out to be 2MB, then it is probably more		For example, if the BDP for a long fat network turns out to be 2MB,
	realistic to test this pipe with multiple connections. Assuming		then it is probably more realistic to test this network path with
	typical host computer window settings of 64 KB, using 32 connections		multiple connections. Assuming typical host computer TCP Receive
	would realistically test this pipe.		Window Sizes of 64 KB, using 32 TCP connections would realistically
			test this path.
	The following table is provided to illustrate the relationship of the		The following table is provided to illustrate the relationship
	BDP, window size, and the number of connections required to utilize		between the TCP Receive Window size and the number of TCP connections
	the available capacity. For this example, the network bandwidth is		required to utilize the available capacity of a given BDP. For this
	500 Mbps, RTT is equal to 5 ms, and the BDP equates to 312 KBytes.		example, the network bandwidth is 500 Mbps and the RTT is 5 ms, then
			the BDP equates to 312.5 KBytes.
	#Connections		TCP Number of TCP Connections
	Window to Fill Link		Window to fill available bandwidth
	------------------------		-------------------------------------
	16KB 20		16KB 20
	32KB 10		32KB 10
	64KB 5		64KB 5
	128KB 3		128KB 3
	The TCP Transfer Time metric is useful for conducting multiple		The TCP Transfer Time metric is useful for conducting multiple
	connection tests. Each connection SHOULD be configured to transfer		connection tests. Each connection should be configured to transfer
	payloads of the same size (i.e. 100 MB), and the TCP Transfer time		payloads of the same size (i.e. 100 MB), and the TCP Transfer time
	SHOULD provide a simple metric to verify the actual versus expected		should provide a simple metric to verify the actual versus expected
	results.		results.
	Note that the TCP transfer time is the time for all connections to		Note that the TCP transfer time is the time for all connections to
	complete the transfer of the configured payload size. From the		complete the transfer of the configured payload size. From the
	example table listed above, the 64KB window is considered. Each of		previous table, the 64KB window is considered. Each of the 5
	the 5 connections would be configured to transfer 100MB, and each		TCP connections would be configured to transfer 100MB, and each one
	TCP should obtain a maximum of 100 Mb/sec per connection. So for		should obtain a maximum of 100 Mb/sec. So for this example, the
	this example, the 100MB payload should be transferred across the		100MB payload should be transferred across the connections in
	connections in approximately 8 seconds (which would be the ideal TCP		approximately 8 seconds (which would be the ideal TCP transfer time
	transfer time for these conditions).		under these conditions).
	Additionally, the TCP Efficiency metric SHOULD be computed for each		Additionally, the TCP Efficiency metric MUST be computed for each
	connection tested (defined in section 2.2).		connection tested as defined in section 3.3.2.
	3.3.4 Interpretation of the TCP Throughput Results		3.3.5 Interpretation of the TCP Throughput Results
	At the end of this step, the user will document the theoretical BDP		At the end of this step, the user will document the theoretical BDP
	and a set of Window size experiments with measured TCP throughput for		and a set of Window size experiments with measured TCP throughput for
	each TCP window size setting. For cases where the sustained TCP		each TCP window size. For cases where the sustained TCP throughput
	throughput does not equal the ideal value, some possible causes		does not equal the ideal value, some possible causes are:
	are listed:
	- Network congestion causing packet loss which MAY be inferred from		- Network congestion causing packet loss which MAY be inferred from
	a poor TCP Efficiency metric (100% = no loss)		a poor TCP Efficiency % (higher TCP Efficiency % = less packet
			loss)
	- Network congestion causing an increase in RTT which MAY be inferred		- Network congestion causing an increase in RTT which MAY be inferred
	from the Buffer Delay metric (0% = no increase in RTT over baseline)		from the Buffer Delay Percentage (i.e., 0% = no increase in RTT
			over baseline)
	- Intermediate network devices which actively regenerate the TCP		- Intermediate network devices which actively regenerate the TCP
	connection and can alter window size, MSS, etc.		connection and can alter TCP Receive Window size, MSS, etc.
	- Rate limiting (policing). More discussion of traffic management		- Rate limiting (policing). More details on traffic management
	tests follows in section 3.4		tests follows in section 3.4
	3.4. Traffic Management Tests		3.4. Traffic Management Tests
	In most cases, the network connection between two geographic		In most cases, the network connection between two geographic
	locations (branch offices, etc.) is lower than the network connection		locations (branch offices, etc.) is lower than the network connection
	of the host computers. An example would be LAN connectivity of GigE		to host computers. An example would be LAN connectivity of GigE
	and WAN connectivity of 100 Mbps. The WAN connectivity may be		and WAN connectivity of 100 Mbps. The WAN connectivity may be
	physically 100 Mbps or logically 100 Mbps (over a GigE WAN		physically 100 Mbps or logically 100 Mbps (over a GigE WAN
	connection). In the later case, rate limiting is used to provide the		connection). In the later case, rate limiting is used to provide the
	WAN bandwidth per the SLA.		WAN bandwidth per the SLA.
	Traffic management techniques are employed to provide various forms		Traffic management techniques are employed to provide various forms
	of QoS, the more common include:		of QoS, the more common include:
	- Traffic Shaping		- Traffic Shaping
	- Priority queuing		- Priority queuing
	- Random Early Discard (RED, etc.)		- Random Early Discard (RED)
	Configuring the end-end network with these various traffic management		Configuring the end-to-end network with these various traffic
	mechanisms is a complex under-taking. For traffic shaping and RED		management mechanisms is a complex under-taking. For traffic shaping
	techniques, the end goal is to provide better performance for bursty		and RED techniques, the end goal is to provide better performance to
	traffic such as TCP (RED is specifically intended for TCP).		bursty traffic such as TCP,(RED is specifically intended for TCP).
	This section of the methodology provides guidelines to test traffic		This section of the methodology provides guidelines to test traffic
	shaping and RED implementations. As in section 3.3, host hardware		shaping and RED implementations. As in section 3.3, host hardware
	performance MUST be well understood before conducting the traffic		performance must be well understood before conducting the traffic
	shaping and RED tests. Dedicated test equipment will generally be		shaping and RED tests. Dedicated communications test instrument will
	REQUIRED for line rates of GigE and 10 GigE. If the throughput test		generally be REQUIRED for line rates of GigE and 10 GigE. If the
	is expected to exceed 10% of the provider bandwidth, then the test		throughput test is expected to exceed 10% of the provider bandwidth,
	SHOULD be coordinated with the network provider. This does not		then the test should be coordinated with the network provider. This
	include the customer premise bandwidth, the 10% refers directly to		does not include the customer premises bandwidth, the 10% refers to
	the provider's bandwidth (Provider Edge to Provider router).		the provider's bandwidth (Provider Edge to Provider router). Note
			that GigE and 10 GigE interfaces might benefit from hold-queue
			adjustments in order to prevent the saw-tooth TCP traffic pattern.
	3.4.1 Traffic Shaping Tests		3.4.1 Traffic Shaping Tests
	For services where the available bandwidth is rate limited, there are		For services where the available bandwidth is rate limited, two (2)
	two (2) techniques used to implement rate limiting: traffic policing		techniques can be used: traffic policing or traffic shaping.
	and traffic shaping.
	Simply stated, traffic policing marks and/or drops packets which		Simply stated, traffic policing marks and/or drops packets which
	exceed the SLA bandwidth (in most cases, excess traffic is dropped).		exceed the SLA bandwidth (in most cases, excess traffic is dropped).
	Traffic shaping employs the use of queues to smooth the bursty		Traffic shaping employs the use of queues to smooth the bursty
	traffic and then send out within the SLA bandwidth limit (without		traffic and then send out within the SLA bandwidth limit (without
	dropping packets unless the traffic shaping queue is exceeded).		dropping packets unless the traffic shaping queue is exhausted).
	Traffic shaping is generally configured for TCP data services and		Traffic shaping is generally configured for TCP data services and
	can provide improved TCP performance since the retransmissions are		can provide improved TCP performance since the retransmissions are
	reduced, which in turn optimizes TCP throughput for the given		reduced, which in turn optimizes TCP throughput for the available
	available bandwidth. Through this section, the available		bandwidth. Through this section, the rate-limited bandwidth shall
	rate-limited bandwidth shall be referred to as the		be referred to as the "bottleneck bandwidth".
	"bottleneck bandwidth".
	The ability to detect proper traffic shaping is more easily diagnosed		The ability to detect proper traffic shaping is more easily diagnosed
	when conducting a multiple TCP connection test. Proper shaping will		when conducting a multiple TCP connections test. Proper shaping will
	provide a fair distribution of the available bottleneck bandwidth,		provide a fair distribution of the available bottleneck bandwidth,
	while traffic policing will not.		while traffic policing will not.
	The traffic shaping tests are built upon the concepts of multiple		The traffic shaping tests are built upon the concepts of multiple
	connection testing as defined in section 3.3.3. Calculating the BDP		connections testing as defined in section 3.3.3. Calculating the BDP
	for the bottleneck bandwidth is first REQUIRED before selecting the		for the bottleneck bandwidth is first required before selecting the
	number of connections and TCP Window size per connection.		number of connections and Send Buffer and TCP Receive Window sizes
			per connection.
	Similar to the example in section 3.3, a typical test scenario might		Similar to the example in section 3.3, a typical test scenario might
	be: GigE LAN with a 100Mbps bottleneck bandwidth (rate limited		be: GigE LAN with a 100Mbps bottleneck bandwidth (rate limited
	logical interface), and 5 msec RTT. This would require five (5) TCP		logical interface), and 5 msec RTT. This would require five (5) TCP
	connections of 64 KB window size evenly fill the bottleneck bandwidth		connections of 64 KB Send Socket Buffer and TCP Receive Window sizes
	(about 100 Mbps per connection).		to evenly fill the bottleneck bandwidth (~100 Mbps per connection).
	The traffic shaping test SHOULD be run over a long enough duration to		The traffic shaping test should be run over a long enough duration to
	properly exercise network buffers (greater than 30 seconds) and also		properly exercise network buffers (greater than 30 seconds) and also
	characterize performance during different time periods of the day.		characterize performance during different time periods of the day.
	The throughput of each connection MUST be logged during the entire		The throughput of each connection MUST be logged during the entire
	test, along with the TCP Transfer Time, TCP Efficiency, and		test, along with the TCP Transfer Time, TCP Efficiency, and
	Buffer Delay metrics.		Buffer Delay Percentage.
	3.4.1.1 Interpretation of Traffic Shaping Test Results		3.4.1.1 Interpretation of Traffic Shaping Test Results
	By plotting the throughput achieved by each TCP connection, the fair		By plotting the throughput achieved by each TCP connection, the fair
	sharing of the bandwidth is generally very obvious when traffic		sharing of the bandwidth is generally very obvious when traffic
	shaping is properly configured for the bottleneck interface. For the		shaping is properly configured for the bottleneck interface. For the
	previous example of 5 connections sharing 500 Mbps, each connection		previous example of 5 connections sharing 500 Mbps, each connection
	would consume ~100 Mbps with a smooth variation. If traffic policing		would consume ~100 Mbps with a smooth variation.
	was present on the bottleneck interface, the bandwidth sharing MAY
	not be fair and the resulting throughput plot MAY reveal "spikey"		If traffic policing was present on the bottleneck interface, the
	throughput consumption of the competing TCP connections (due to the		bandwidth sharing may not be fair and the resulting throughput plot
	retransmissions).		may reveal "spikey" throughput consumption of the competing TCP
			connections (due to the TCP retransmissions).
	3.4.2 RED Tests		3.4.2 RED Tests
	Random Early Discard techniques are specifically targeted to provide		Random Early Discard techniques are specifically targeted to provide
	congestion avoidance for TCP traffic. Before the network element		congestion avoidance for TCP traffic. Before the network element
	queue "fills" and enters the tail drop state, RED drops packets at		queue "fills" and enters the tail drop state, RED drops packets at
	configurable queue depth thresholds. This action causes TCP		configurable queue depth thresholds. This action causes TCP
	connections to back-off which helps to prevent tail drop, which in		connections to back-off which helps to prevent tail drop, which in
	turn helps to prevent global TCP synchronization.		turn helps to prevent global TCP synchronization.
	Again, rate limited interfaces can benefit greatly from RED based		Again, rate limited interfaces may benefit greatly from RED based
	techniques. Without RED, TCP is generally not able to achieve the		techniques. Without RED, TCP may not be able to achieve the full
	full bandwidth of the bottleneck interface. With RED enabled, TCP		bottleneck bandwidth. With RED enabled, TCP congestion avoidance
	congestion avoidance throttles the connections on the higher speed		throttles the connections on the higher speed interface (i.e. LAN)
	interface (i.e. LAN) and can reach equilibrium with the bottleneck		and can help achieve the full bottleneck bandwidth. The burstiness
	bandwidth (achieving closer to full throughput).		of TCP traffic is a key factor in the overall effectiveness of RED
			techniques; steady state bulk transfer flows will generally not
			benefit from RED. With bulk transfer flows, network device queues
			gracefully throttle the effective throughput rates due to increased
			delays.
	The ability to detect proper RED configuration is more easily		The ability to detect proper RED configuration is more easily
	diagnosed when conducting a multiple TCP connection test. Multiple		diagnosed when conducting a multiple TCP connections test. Multiple
	TCP connections provide the multiple bursty sources that emulate the		TCP connections provide the bursty sources that emulate the
	real-world conditions for which RED was intended.		real-world conditions for which RED was intended.
	The RED tests also build upon the concepts of multiple connection		The RED tests also builds upon the concepts of multiple connections
	testing as defined in section 3.3.3. Calculating the BDP for the		testing as defined in section 3.3.3. Calculating the BDP for the
	bottleneck bandwidth is first REQUIRED before selecting the number		bottleneck bandwidth is first required before selecting the number
	of connections and TCP Window size per connection.		of connections, the Send Socket Buffer size and the TCP Receive
			Window size per connection.
	For RED testing, the desired effect is to cause the TCP connections		For RED testing, the desired effect is to cause the TCP connections
	to burst beyond the bottleneck bandwidth so that queue drops will		to burst beyond the bottleneck bandwidth so that queue drops will
	occur. Using the same example from section 3.4.1 (traffic shaping),		occur. Using the same example from section 3.4.1 (traffic shaping),
	the 500 Mbps bottleneck bandwidth requires 5 TCP connections (with		the 500 Mbps bottleneck bandwidth requires 5 TCP connections (with
	window size of 64Kb) to fill the capacity. Some experimentation is		window size of 64KB) to fill the capacity. Some experimentation is
	REQUIRED, but it is RECOMMENDED to start with double the number of		required, but it is recommended to start with double the number of
	connections to stress the network element buffers / queues. In this		connections to stress the network element buffers / queues (10
	example, 10 connections SHOULD produce TCP bursts of 64KB for each		connections for this example).
	connection. If the timing of the TCP tester permits, these TCP
	bursts SHOULD stress queue sizes in the 512KB range. Again		The TCP TTD must be configured to generate these connections as
	experimentation will be REQUIRED and the proper number of TCP		shorter (bursty) flows versus bulk transfer type flows. These TCP
	connections and TCP window size will be dictated by the size the		bursts should stress queue sizes in the 512KB range. Again
	network element queue.		experimentation will be required; the proper number of TCP
			connections, the Send Socket Buffer and TCP Receive Window sizes will
			be dictated by the size of the network element queue.
	3.4.2.1 Interpretation of RED Results		3.4.2.1 Interpretation of RED Results
	The default queuing technique for most network devices is FIFO based.		The default queuing technique for most network devices is FIFO based.
	Without RED, the FIFO based queue will cause excessive loss to all of		Without RED, the FIFO based queue may cause excessive loss to all of
	the TCP connections and in the worst case global TCP synchronization.		the TCP connections and in the worst case global TCP synchronization.
	By plotting the aggregate throughput achieved on the bottleneck		By plotting the aggregate throughput achieved on the bottleneck
	interface, proper RED operation MAY be determined if the bottleneck		interface, proper RED operation may be determined if the bottleneck
	bandwidth is fully utilized. For the previous example of 10		bandwidth is fully utilized. For the previous example of 10
	connections (window = 64 KB) sharing 500 Mbps, each connection SHOULD		connections (window = 64 KB) sharing 500 Mbps, each connection should
	consume ~50 Mbps. If RED was not properly enabled on the interface,		consume ~50 Mbps. If RED was not properly enabled on the interface,
	then the TCP connections will retransmit at a higher rate and the		then the TCP connections will retransmit at a higher rate and the
	net effect is that the bottleneck bandwidth is not fully utilized.		net effect is that the bottleneck bandwidth is not fully utilized.
	Another means to study non-RED versus RED implementation is to use		Another means to study non-RED versus RED implementation is to use
	the TCP Transfer Time metric for all of the connections. In this		the TCP Transfer Time metric for all of the connections. In this
	example, a 100 MB payload transfer SHOULD take ideally 16 seconds		example, a 100 MB payload transfer should take ideally 16 seconds
	across all 10 connections (with RED enabled). With RED not enabled,		across all 10 connections (with RED enabled). With RED not enabled,
	the throughput across the bottleneck bandwidth MAY be greatly		the throughput across the bottleneck bandwidth may be greatly
	reduced (generally 20-40%) and the actual TCP Transfer time MAY be		reduced (generally 10-20%) and the actual TCP Transfer time may be
	proportionally longer then the Ideal TCP Transfer time.		proportionally longer then the Ideal TCP Transfer time.
	Additionally, the TCP Transfer Efficiency metric is useful, since		Additionally, non-RED implementations may exhibit a lower TCP
	non-RED implementations MAY exhibit a lower TCP Transfer Efficiency.		Transfer Efficiency.
	4. Security Considerations		4. Security Considerations
	The security considerations that apply to any active measurement of		The security considerations that apply to any active measurement of
	live networks are relevant here as well. See [RFC4656] and		live networks are relevant here as well. See [RFC4656] and
	[RFC5357].		[RFC5357].
	5. IANA Considerations		5. IANA Considerations
	This document does not REQUIRE an IANA registration for ports		This document does not REQUIRE an IANA registration for ports
	dedicated to the TCP testing described in this document.		dedicated to the TCP testing described in this document.
	6. Acknowledgments		6. Acknowledgments
	Thanks to Matt Mathis, Matt Zekauskas, Al Morton, Rudi Geib, and		Thanks to Lars Eggert, Al Morton, Matt Mathis, Matt Zekauskas,
	Yaakov Stein, Loki Jorgenson for many good comments and for pointing		Yaakov Stein, and Loki Jorgenson for many good comments and for
	us to great sources of information pertaining to past works in the		pointing us to great sources of information pertaining to past works
	TCP capacity area.		in the TCP capacity area.
	7. References		7. References
	7.1 Normative References		7.1 Normative References
	[RFC2119] Bradner, S., "Key words for use in RFCs to Indicate		[RFC2119] Bradner, S., "Key words for use in RFCs to Indicate
	Requirement Levels", BCP 14, RFC 2119, March 1997.		Requirement Levels", BCP 14, RFC 2119, March 1997.
	[RFC4656] Shalunov, S., Teitelbaum, B., Karp, A., Boote, J., and M.		[RFC4656] Shalunov, S., Teitelbaum, B., Karp, A., Boote, J., and M.
	Zekauskas, "A One-way Active Measurement Protocol		Zekauskas, "A One-way Active Measurement Protocol
	(OWAMP)", RFC 4656, September 2006.		(OWAMP)", RFC 4656, September 2006.
	skipping to change at page 23, line 28		skipping to change at page 24, line 45
	[RFC4898] Mathis, M., Heffner, J., Raghunarayan, R., "TCP Extended		[RFC4898] Mathis, M., Heffner, J., Raghunarayan, R., "TCP Extended
	Statistics MIB", May 2007		Statistics MIB", May 2007
	[RFC5136] Chimento P., Ishac, J., "Defining Network Capacity",		[RFC5136] Chimento P., Ishac, J., "Defining Network Capacity",
	February 2008		February 2008
	[RFC1323] Jacobson, V., Braden, R., Borman D., "TCP Extensions for		[RFC1323] Jacobson, V., Braden, R., Borman D., "TCP Extensions for
	High Performance", May 1992		High Performance", May 1992
	7.2. Informative References		7.2. Informative References
	Authors' Addresses		Authors' Addresses
	Barry Constantine		Barry Constantine
	JDSU, Test and Measurement Division		JDSU, Test and Measurement Division
	One Milesone Center Court		One Milesone Center Court
	Germantown, MD 20876-7100		Germantown, MD 20876-7100
	USA		USA
	Phone: +1 240 404 2227		Phone: +1 240 404 2227
	barry.constantine@jdsu.com		barry.constantine@jdsu.com
	Gilles Forget		Gilles Forget
	Independent Consultant to Bell Canada.		Independent Consultant to Bell Canada.
	308, rue de Monaco, St-Eustache		308, rue de Monaco, St-Eustache
	Qc. CANADA, Postal Code : J7P-4T5		Qc. CANADA, Postal Code : J7P-4T5
	Phone: (514) 895-8212		Phone: (514) 895-8212
	gilles.forget@sympatico.ca		gilles.forget@sympatico.ca
			Rudiger Geib
			Heinrich-Hertz-Strasse (Number: 3-7)
			Darmstadt, Germany, 64295
			Phone: +49 6151 6282747
			Ruediger.Geib@telekom.de
	Reinhard Schrage		Reinhard Schrage
	Schrage Consulting		Schrage Consulting
	Phone: +49 (0) 5137 909540		Phone: +49 (0) 5137 909540
	reinhard@schrageconsult.com		reinhard@schrageconsult.com
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