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
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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
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to this document. Code Components extracted from this document must to this document. Code Components extracted from this document must
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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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