Network Working Group                                      B. Constantine
Internet-Draft                                                       JDSU
Intended status: Informational                                  G. Forget
Expires: December 18, 2010                  Bell Canada (Ext. Consultant)
                                                             L. Jorgenson
                                                        Apparent Networks
                                                         Reinhard Schrage
                                                       Schrage Consulting
                                                             June 8, 2010

                    TCP Throughput Testing Methodology


   This memo describes a methodology for measuring sustained TCP
   throughput performance in an end-to-end managed network environment.
   This memo is intended to provide a practical approach to help users
   validate the TCP layer performance of a managed network, which should
   provide a better indication of end-user application level experience.
   In the methodology, various TCP and network parameters are identified
   that should be tested as part of the network verification at the TCP

Status of this Memo

   This Internet-Draft is submitted to IETF in full conformance with the
   provisions of BCP 78 and BCP 79.

   Internet-Drafts are working documents of the Internet Engineering
   Task Force (IETF), its areas, and its working groups.  Note that
   other groups may also distribute working documents as Internet-
   Drafts.  Creation date June 8, 2010.

   Internet-Drafts are draft documents valid for a maximum of six months
   and may be updated, replaced, or obsoleted by other documents at any
   time.  It is inappropriate to use Internet-Drafts as reference
   material or to cite them other than as "work in progress."

   The list of current Internet-Drafts can be accessed at

   The list of Internet-Draft Shadow Directories can be accessed at

   This Internet-Draft will expire on December 18, 2010.

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   Copyright Notice

   Copyright (c) 2010 IETF Trust and the persons identified as the
   document authors.  All rights reserved.

   This document is subject to BCP 78 and the IETF Trust's Legal
   Provisions Relating to IETF Documents
   ( in effect on the date of
   publication of this document.  Please review these documents
   carefully, as they describe your rights and restrictions with respect
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   the Trust Legal Provisions and are provided without warranty as
   described in the BSD License.

Table of Contents

   1.  Introduction . . . . . . . . . . . . . . . . . . . . . . . . .  3
   2.  Goals of this Methodology. . . . . . . . . . . . . . . . . . .  4
     2.1   TCP Equilibrium State Throughput . . . . . . . . . . . . .  5
     2.2   Metric for TCP Throughput Tests  . . . . . . . . . . . . .  6
   3.  TCP Throughput Testing Methodology . . . . . . . . . . . . . .  7
     3.1   Determine Network Path MTU . . . . . . . . . . . . . . . .  8
     3.2.  Baseline Round-trip Delay and Bandwidth. . . . . . . . . .  9
         3.2.1  Techniques to Measure Round Trip Time . . . . . . . . 10
         3.2.2  Techniques to Measure End-end Bandwidth . . . . . . . 10
     3.3.  Single TCP Connection Throughput Tests . . . . . . . . . . 11
         3.3.1 Interpretation of the Single Connection TCP
               Throughput Results . . . . . . . . . . . . . . . . . . 14
     3.4. Traffic Management Testing  . . . . . . . . . . . . . . . . 14
         3.4.1 Multiple TCP Connections - below Link Capacity . . . . 14
         3.4.2 Multiple TCP Connections - over Link Capacity. . . . . 15
         3.4.3 Interpretation of Multiple TCP Connection Results. . . 16

   4.  Acknowledgements . . . . . . . . . . . . . . . . . . . . . . . 16
   5.  References . . . . . . . . . . . . . . . . . . . . . . . . . . 16
   Authors' Addresses . . . . . . . . . . . . . . . . . . . . . . . . 17

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1. Introduction

   Even though RFC2544 was meant to benchmark network equipment and
   used by network equipment manufacturers (NEMs), network providers
   have used it to benchmark operational networks in order to
   verify SLAs (Service Level Agreements) before turning on a service
   to their business customers.  Testing an operational network prior to
   customer activation is referred to as "turn-up" testing and the SLA
   is generally Layer 2/3 packet throughput, delay, loss and

   Network providers are coming to the realization that RFC2544 testing
   and TCP layer testing are required to more adequately ensure end-user
   satisfaction. Therefore, the network provider community desires to
   measure network throughput performance at the TCP layer. Measuring
   TCP throughput provides a meaningful measure with respect to the end
   user's application SLA (and ultimately reach some level of TCP
   testing interoperability which does not exist today).

   The complexity of the network grows and the various queuing
   mechanisms in the network greatly affect TCP layer performance (i.e.
   improper default router settings for queuing, etc.) and devices such
   as firewalls, proxies, load-balancers can actively alter the TCP
   settings as a TCP session traverses the network (such as window size,
   MSS, etc.).  Network providers (and NEMs) are wrestling with end-end
   complexities of the above and there is a strong interest in the
   standardization of a test methodology to validate end-to-end TCP
   performance (as this is the precursor to acceptable end-user
   application performance).

   So the intent behind this draft TCP throughput work is to define
   a methodology for testing sustained TCP layer performance.  In this
   document, sustained TCP throughput is that amount of data per unit
   time that TCP transports during equilibrium (steady state), i.e.
   after the initial slow start phase. We refer to this state as TCP
   Equilibrium, and that the equalibrium throughput is the maximum
   achievable for the TCP connection(s).

   One other important note; the precursor to conducting the TCP tests
   test methodlogy is to perform "network stress tests" such as RFC2544
   Layer 2/3 tests or other conventional tests (OWAMP, etc.).  It is
   highly recommended to run traditional Layer 2/3 type test to verify
   the integrity of the network before conducting TCP testing.

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2. Goals of this Methodology

   Before defining the goals of this methodology, it is important to
   clearly define the areas that are not intended to be measured or
   analyzed by such a methodology.

   - The methodology is not intended to predict TCP throughput
   behavior during the transient stages of a TCP connection, such
   as initial slow start.

   - The methodology is not intended to definitively benchmark TCP
   implementations of one OS to another, although some users may find
   some value in conducting qualitative experiments

   - The methodology is not intended to provide detailed diagnosis
   of problems within end-points or the network itself as related to
   non-optimal TCP performance, although a results interpretation
   section for each test step may provide insight into potential
   issues within the network

   In contrast to the above exclusions, the goals of this methodology
   are to define a method to conduct a structured, end-to-end
   assessment of sustained TCP performance within a managed business
   class IP network.  A key goal is to establish a set of "best
   practices" that an engineer should apply when validating the
   ability of a managed network to carry end-user TCP applications.

   Some specific goals are to:

   - Provide a practical test approach that specifies the more well
   understood (and end-user configurable) TCP parameters such as Window
   size, MSS, # connections, and how these affect the outcome of TCP
   performance over a network

   - Provide specific test conditions (link speed, RTT, window size,
   etc.) and maximum achievable TCP throughput under TCP Equilbrium
   conditions.  For guideline purposes, provide examples of these test
   conditions and the maximum achievable TCP throughput during the
   equilbrium state.  Section 2.1 provides specific details concerning
   the definition of TCP Equilibrium within the context of this draft.

   - In test situations where the recommended procedure does not yield
   the maximum achievable TCP throughput result, this draft provides some
   possible areas within the end host or network that should be
   considered for investigation (although again, this draft is not
   intended to provide a detailed diagnosis of these issues)

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2.1 TCP Equilibrium State Throughput

   TCP connections have three (3) fundamental congestion window phases
   as documented in RFC2581.  These states are:

   - Slow Start, which occurs during the beginning of a TCP transmission
   or after a retransmission time out event

   - Congestion avoidance, which is the phase during which TCP ramps up
   to establish the maximum attainable throughput on an end-end network
   path.  Retransmissions are a natural by-product of the TCP congestion
   avoidance algorithm as it seeks to achieve maximum throughput on
   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 states.

            |        ssthresh
   TCP      |           |
   Through- |           |       Equilibrium
   put      |           |\      /\/\/\/\/\  Retransmit          /\/\ ...
            |           | \    /         |  Time-out           /
            |           |  \  /          |  _______          _/
            |  Slow   _/    |/           | /       | Slow  _/
            | Start _/      Congestion   |/        |Start_/   Congestion
            |     _/         Avoidance   Loss      |   _/     Avoidance
            |   _/                       Event     | _/
            | _/                                   |/

   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 would
   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.

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2.2 Metrics for TCP Throughput Tests

   This draft focuses on a TCP throughtput methodology and also
   provides two 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 set of metrics that account for
   the myriad of variables (i.e. RTT variation, loss conditions, TCP
   implementation, etc.).  However, these two basic metrics are useful to
   compare network traffic management techniques, especially in section
   3.4 of this document (Traffic Management Tests).

   The TCP Efficiency metric is the percentage of bytes that were not
   retransmitted and is defined as:

                Transmitted Bytes - Retransmitted Bytes
                ---------------------------------------  x 100
                          Transmitted Bytes

   This metric is easy to understand and provides a comparative measure
   between various QoS mechanisms such as traffic management, congestion
   avoidance, and also various TCP implementations (i.e. Reno, Vegas,

   The second measure is also basic and is the TCP Transfer Time,
   which is simply the time is takes to transfer a block of data across
   simultaneous TCP connections.  The concept is useful to benchmark
   traffic management tests, where multiple connections are required and
   it simplifies comparing results of different approaches.  An
   example would be the bulk transfer of 10 MB upon 8 separate TCP
   connections (each connection uploading 10 MB).  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).  But by defining the Transfer Time as that of
   the successful transfer of 10MB over all 8 connections, the single
   transfer time metric is very useful means to rate various traffic
   management techniques (i.e. FiFO, WFQ queuing, WRED, etc.).

3. TCP Throughput Testing Methodology

   This section summarizes the specific test methodology to achieve the
   goals listed in Section 2.

   As stated in Section 1, it is considered best practice to verify
   the integrity of the network by conducting Layer2/3 stress tests
   such as RFC2544 or other methods of network stress tests.  If the
   network is not performing properly in terms of packet loss, jitter,
   etc. then the TCP layer testing will not be meaningful since the
   equalibrium throughput would be very difficult to achieve (in a
   "dysfunctional" network).

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   The following provides the sequential order of steps to conduct the
   TCP throughput testing methodology:

   1. Identify the Path MTU.  Packetization Layer Path MTU Discovery
   or PLPMTUD (RFC4821) should be conducted to verify the minimum network
   path MTU.  Conducting PLPMTUD establishes the upper limit for the MSS
   to be used in subsequent steps.

   2. Baseline Round-trip Delay and Bandwidth. These measurements provide
   estimates of the ideal TCP window size, which will be used in
   subsequent test steps.

   3. Single TCP Connection Throughput Tests.  With baseline measurements
   of round trip delay and bandwidth, a series of single connection TCP
   throughput tests can be conducted to baseline the performance of the
   network against expectations.

   4. Traffic Management Tests.  Various traffic management and queuing
   techniques are tested in this step, using multiple TCP connections.
   Multiple connection testing can verify that the network is configured
   properly for traffic shaping versus policing, various queuing
   implementations, and RED.

   Important to note are some of the key characteristics and
   considerations for the TCP test instrument.  The test host may be a
   standard computer or dedicated communications test instrument
   and these TCP test hosts be capable of emulating both a client and a
   server.  As a general rule of thumb, testing TCP throughput at rates
   greater than 250-500 Mbit/sec generally requires high performance
   server hardware or dedicated hardware based test tools.

   Whether the TCP test host is a standard computer or dedicated test
   instrument, the following areas should be considered when selecting
   a test host:

   - TCP implementation used by the test host OS, i.e. Linux OS kernel
   using TCP Reno, TCP options supported, etc.  This will obviously be
   more important when using custom test equipment where the TCP
   implementation may be customized or tuned to run in higher
   performance hardware

   - Most importantly, the TCP test host must be capable of generating
   and receiving stateful TCP test traffic at the full link speed of the
   network under test. This requirement is very serious and may require
   custom test equipment, especially on 1 GigE and 10 GigE networks.

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3.1. Determine Network Path MTU

   TCP implementations should use Path MTU Discovery techniques (PMTUD).
   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)
   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
   and the device sends an ICMP 'need to frag' message back to the host
   that originated the packet. The ICMP 'need to frag' message includes
   the next hop MTU which PMTUD uses to tune the TCP Maximum Segment
   Size (MSS). Unfortunately, because many network managers completely
   disable ICMP, this technique does not always prove reliable in real
   world situations.

   Packetization Layer Path MTU Discovery or PLPMTUD (RFC4821) should
   be conducted to verify the minimum network path MTU.  PLPMTUD can
   be used with or without ICMP. The following sections provide a
   summary of the PLPMTUD approach and an example using the TCP
   protocol. RFC4821 specifies a search_high and search_low parameter
   for the MTU.  As specified in RFC4821, a value of 1024 is a generally
   safe value to choose for search_low in modern networks.

   It is important to determine the overhead of the links in the path,
   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
   bytes, then the MSS would be set to 984 bytes.

   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
   the probe packets and could start at MSS = 984 (which corresponds
   to an MTU size of 1024 bytes).

   The TCP client probe would open a TCP connection and advertise the
   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
   per a nominal window size (8KB, etc.).  The window size should be
   kept small to minimize the possibility of congesting the network,
   which could induce congestive loss.  The duration of the test should
   also be short (10-30 seconds), again to minimize congestive effects
   during the test.

   In the example of a 1240 byte path MTU, probing with an MSS equal to
   984 would yield a successful probe and the test client packets would
   be successfully transferred to the test server.

   Also note that the test client MUST verify that the MSS advertised
   is indeed negotiated.  Network devices with built-in Layer 4
   capabilities can intercede during the connection establishment
   process and reduce the advertised MSS to avoid fragmentation.  This
   is certainly a desirable feature from a network perspective, but
   can yield erroneous test results if the client test probe does not
   confirm the negotiated MSS.

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   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
   example, the test client would retransmit based upon time-outs (since
   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
   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
   at other times of the day is recommended to further isolate.

   The test is repeated until the desired granularity of the MTU is
   discovered.  The method can yield precise results at the expense of
   probing time.  One approach would be to reduce the probe size to
   half between the unsuccessful search_high and successful search_low
   value, and increase by increments of 1/2 when seeking the upper

3.2. Baseline Round-trip Delay and Bandwidth

   Before stateful TCP testing can begin, it is important to baseline
   the round trip delay and bandwidth of the network to be tested.
   These measurements provide estimates of the ideal TCP window size,
   which will be used in subsequent test steps.  These latency and
   bandwidth tests should be run over a long enough period of time to
   characterize the performance of the network over the course of a
   meaningful time period.

   One example would be to take samples during various times of the work
   day. The goal would be to determine a representative minimum, average,
   and maximum RTD and bandwidth for the network under test.  Topology
   changes are to be avoided during this time of initial convergence
   (e.g. in crossing BGP4 boundaries).

   In some cases, baselining bandwidth may not be required, since a
   network provider's end-to-end topology may be well enough defined.

   3.2.1 Techniques to Measure Round Trip Time

   We follow in the definitions used in the references of the appendix;
   hence Round Trip Time (RTT) is the time elapsed between the clocking
   in of the first bit of a payload packet to the receipt of the last
   bit of the corresponding acknowledgement.  Round Trip Delay (RTD)
   is used synonymously to twice the Link Latency.

   In any method used to baseline round trip delay between network
   end-points, it is important to realize that network latency is the
   sum of inherent network delay and congestion.  The RTT should be
   baselined during "off-peak" hours to obtain a reliable figure for
   network latency (versus additional delay caused by congestion).

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   During the actual sustained TCP throughput tests, it is critical
   to measure RTT along with measured TCP throughput. Congestive
   effects can be isolated if RTT is concurrently measured

   This is not meant to provide an exhaustive list, but summarizes some
   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.

   - 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
   test equipment RTT measurement may be compatible with delay
   measurement protocols specified in RFC5357.

   - Conduct packet captures of TCP test applications using for example
  "iperf" or FTP, etc.  By running multiple experiments, the packet
   captures can be studied to estimate RTT based upon the SYN -> SYN-ACK
   handshakes within the TCP connection set-up.

  - ICMP Pings may also be adequate to provide round trip time
   estimations.  Some limitations of ICMP Ping are the msec resolution
   and whether the network elements respond to pings (or block them).

   3.2.2 Techniques to Measure End-end Bandwidth

   There are many well established techniques available to provide
   estimated measures of bandwidth over a network.  This measurement
   should be conducted in both directions of the network, especially for
   access networks which are inherently asymmetrical.  Some of the
   asymmetric implications to TCP performance are documented in RFC-3449
   and the results of this work will be further studied to determine
   relevance to this draft.

   The bandwidth measurement test must be run with stateless IP streams
   (not stateful TCP) in order to determine the available bandwidth in
   each direction.  And this test should obviously be performed at
   various intervals throughout a business day (or even across a week).
   Ideally, the bandwidth test should produce a log output of the
   bandwidth achieved across the test interval AND the round trip delay.

   And during the actual TCP level performance measurements (Sections
   3.3 - 3.5), the test tool must be able to track round trip time
   of the TCP connection(s) during the test.  Measuring round trip time
   variation (aka "jitter") provides insight into effects of congestive
   delay on the sustained throughput achieved for the TCP layer test.

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3.3. Single TCP Connection Throughput Tests

   This draft specifically defines TCP throughput techniques to verify
   sustained TCP performance in a managed business network.  Defined
   in section 2.1, the equalibrium throughput reflects the maximum
   rate achieved by a TCP connection within the congestion avoidance
   phase on a end-end network path.  This section and others 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
   from section 3.2, a series of single connection TCP throughput tests
   can be conducted to baseline the performance of the network against
   expectations.  The optimum TCP window size can be calculated from
   the bandwidth delay product (BDP), which is:

   BDP = RTT x Bandwidth

   By dividing the BDP by 8, the "ideal" TCP window size is calculated.
   An example would be a T3 link with 25 msec RTT.  The BDP would equal
   ~1,105,000 bits and the ideal TCP window would equal ~138,000 bytes.

   The following table provides some representative network link speeds,
   latency, BDP, and associated "optimum" TCP window size.  Sustained
   TCP transfers should reach nearly 100% throughput, minus the overhead
   of Layers 1-3 and the divisor of the MSS into the window.

   For this single connection baseline test, the MSS size will effect
   the achieved throughput (especially for smaller TCP window sizes).
   Table 3.2 provides the achievable, equalibrium TCP
   throughput (at Layer 4) using 1000 byte MSS.  Also in this table,
   the case of 58 byte L1-L4 overhead including the Ethernet CRC32 is
   used for simplicity.

   Table 3.2: Link Speed, RTT and calculated BDP, TCP Throughput

   Link                               Ideal TCP      Maximum Achievable
   Speed*    RTT (ms)  BDP (bits)  Window (kbytes)  TCP Throughput(Mbps)
    T1         20        30,720          3.84              1.20
    T1         50        76,800          9.60              1.44
    T1        100       153,600         19.20              1.44
    T3         10       442,100         55.26             41.60
    T3         15       663,150         82.89             41.13
    T3         25     1,105,250        138.16             41.92
    T3(ATM)    10       407,040         50.88             32.44
    T3(ATM)    15       610,560         76.32             32.44
    T3(ATM)    25     1,017,600        127.20             32.44
    100M        1       100,000         12.50             90.699
    100M        2       200,000         25.00             92.815

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   Link                               Ideal TCP      Maximum Achievable
   Speed*    RTT (ms)  BDP (bits)  Window (kbytes)  TCP Throughput (Mbps)
    100M        5       500,000         62.50             90.699
    1Gig      0.1       100,000         12.50            906.991
    1Gig      0.5       500,000         62.50            906.991
    1Gig        1     1,000,000        125.00            906.991
    10Gig     0.05      500,000         62.50          9,069.912
    10Gig     0.3     3,000,000        375.00          9,069.912

   * Note that link speed is the minimum link speed throughput a network;
   i.e. WAN with T1 link, etc.

   Also, the following link speeds (available payload bandwidth) were
   used for the WAN entries:

   - T1 = 1.536 Mbits/sec (B8ZS line encoding facility)
   - T3 = 44.21 Mbits/sec (C-Bit Framing)
   - T3(ATM) = 36.86 Mbits/sec (C-Bit Framing & PLCP, 96000 Cells per

   The calculation method used in this document is a 3 step process :

   1 - We determine what should be the optimal TCP Window size value
       based on the optimal quantity of "in-flight" octets discovered by
       the BDP calculation. We take into consideration that the TCP
       Window size has to be an exact multiple value of the MSS.
   2 - Then we 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, we multiply the calculated value of step 2 by the MSS
       versus (MSS + L2 + L3 + L4 Overheads) ratio.

   This gives us the achievable TCP Throughput value.  Sometimes, the
   maximum achievable throughput is limited by the maximum achievable
   quantity of Ethernet Frames per second on the physical media. Then
   this value is used in step 2 instead of the calculated one.

   There are several TCP tools that are commonly used in the network
   provider 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 maunally 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.

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   The following diagram shows the achievable TCP throughput on a T3 with
   the default Windows2000/XP TCP Window size of 17520 Bytes.

TCP          |
Throughput 35|
in Mbps      |
           15|         _______ 14.48M
             |         |     |
           10|         |     |         +-----+ 9.65M
             |         |     |         |     |        _______ 5.79M
            5|         |     |         |     |        |     |
             |_________+_____+_________+_____+________+____ +___________
                          10              15             25
                                RTT in milliseconds

   The following diagram shows the achievable TCP throughput on a 25ms T3
   when the TCP Window size is increased and with the RFC1323 TCP Window
   scaling option.

             |                                              +-----+42.47M
           40|                                              |     |
TCP          |                                              |     |
Throughput 35|                                              |     |
in Mbps      |                                              |     |
           30|                                              |     |
             |                                              |     |
           25|                                              |     |
             |                                ______ 21.23M |     |
           20|                                |    |        |     |
             |                                |    |        |     |
           15|                                |    |        |     |
             |                                |    |        |     |
           10|               +----+10.62M     |    |        |     |
             |  _______5.31M |    |           |    |        |     |
            5|  |     |      |    |           |    |        |     |
                   16           32           64              128
                               TCP Window size in Kilo Bytes

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   The single connection TCP throughput test must be run over a
   a long duration and results must be logged at the desired interval.
   The test must record RTT and TCP retransmissions at each interval.

   This correlation of retransmissions and RTT over the course of the
   test will clearly identify which portions of the transfer reached
   TCP Equilbrium state and to what effect increased RTT (congestive
   effects) may have been the cause of reduced equilibrium performance.

   Host hardware performance must be well understood before conducting
   this TCP single connection test and other tests in this section.
   Dedicated test equipment may be required, especially for line rates
   of GigE and 10 GigE.

3.3.1 Interpretation of the Single Connection TCP Throughput Results

   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
   each TCP window size setting.  For cases where the sustained TCP
   throughput does not equal the predicted value, some possible causes
   are listed:

   - Network congestion causing packet loss
   - Network congestion not causing packet loss, but effectively
   increasing the size of the required TCP window during the transfer
   - Intermediate network devices which actively regenerate the TCP
   connection and can alter window size, MSS, etc.

3.4. Traffic Management Tests

   After baselining the network under test with a single TCP connection
   (Section 3.3), the nominal capacity of the network has been
   determined.  The capacity measured in section 3.3 may be a capacity
   range and it is reasonable that some level of tuning may have been
   required (i.e. router shaping techniques employed, intermediary
   proxy like devices tuned, etc.).

   Single connection TCP testing is a useful first step to measure
   expected versus actual TCP performance and as a means to diagnose
   / tune issues in the network and active elements.  However, the
   ultimate goal of this methodology is to more closely emulate customer
   traffic, which comprise many TCP connections over a network link.

3.4.1 Multiple TCP Connections - below Link Capacity

   First, the ability of the network to carry multiple TCP connections
   to full network capacity should be tested.  Prioritization and QoS
   settings are not considered during this step, since the network
   capacity is not to be exceeded by the test traffic (section 3.5.2
   covers the over capacity test case).

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   For this multiple connection TCP throughput test, the number of
   connections will more than likely be limited by the test tool (host
   vs. dedicated test equipment).  As an example, for a GigE link with
   1 msec RTT, the optimum TCP window would equal ~128 KBytes. So under
   this condition, 8 concurrent connections with window size equal to
   16KB would fill the GigE link.  For 10G, 80 connections would be
   required to accomplish the same.

   Just as in section 3.3, the end host or test tool can not be the
   processing bottleneck or the throughput measurements will not be
   valid.  The test tool must be benchmarked in ideal lab conditions to
   verify it's ability to transfer stateful TCP traffic at the given
   network line rate.

   For this test step, it should be conducted over a reasonable test
   duration and results should be logged per interval such as throughput
   per connection, RTT, and retransmissions.

   Since the network is not to be driven into over capacity (by nature
   of the BDP allocated evenly to each connection), this test verifies
   the ability of the network to carry multiple TCP connections up to
   the link speed of the network.

3.4.2 Multiple TCP Connections - over Link Capacity

   In this step, the network bandwidth is intentionally exceeded with
   multiple TCP connections to test expected prioritization and queuing
   within the network.

   All conditions related to Section 3.3 set-up apply, especially the
   ability of the test hosts to transfer stateful TCP traffic at network
   line rates.

   Using the same example from Section 3.3, a GigE link with 1 msec
   RTT would require a window size of 128 KB to fill the link (with
   one TCP connection).  Assuming a 16KB window, 8 concurrent
   connections would fill the GigE link capacity and values higher than
   8 would over-subscribe the network capacity.  The user would select
   values to over-subscribe the network (i.e. possibly 10 15, 20, etc.)
   to conduct experiments to verify proper prioritization and queuing
   within the network.

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3.4.3 Interpretation of Multiple TCP Connection Test Restults

   Without any prioritization in the network, the over subscribed test
   results could assist in the queuing studies.  With proper queuing,
   the bandwidth should be shared in a reasonable manner.  The author
   understands that the term "reasonable" is too wide open, and future
   draft versions of this memo would attempt to quantify this sharing
   in more tangible terms.  It is known that if a network element
   is not set for proper queuing (i.e. FIFO), then an oversubscribed
   TCP connection test will generally show a very uneven distribution of

   With prioritization in the network, different TCP connections can be
   assigned various QoS settings via the various mechanisms (i.e. per
   VLAN, DSCP, etc.), and the higher priority connections must be
   verified to achieve the expected throughput.

4.  Acknowledgements

   The author would like to thank Gilles Forget, Loki Jorgenson,
   and Reinhard Schrage for technical review and contributions to this
   draft-03 memo.

   Also thanks to Matt Mathis and Matt Zekauskas for many good comments
   through email exchange and for pointing us to great sources of
   information pertaining to past works in the TCP capacity area.

5.  References

   [RFC2581]  Allman, M., Paxson, V., Stevens W., "TCP Congestion
              Control", RFC 2581, June 1999.

   [RFC3148]  Mathis M., Allman, M., "A Framework for Defining
              Empirical Bulk Transfer Capacity Metrics", RFC 3148, July

   [RFC2544]  Bradner, S., McQuaid, J., "Benchmarking Methodology for
              Network Interconnect Devices", RFC 2544, June 1999

   [RFC3449]  Balakrishnan, H., Padmanabhan, V. N., Fairhurst, G.,
              Sooriyabandara, M., "TCP Performance Implications of
              Network Path Asymmetry", RFC 3449, December 2002

   [RFC5357]  Hedayat, K., Krzanowski, R., Morton, A., Yum, K., Babiarz,
              J., "A Two-Way Active Measurement Protocol (TWAMP)",
              RFC 5357, October 2008

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   [RFC4821]  Mathis, M., Heffner, J., "Packetization Layer Path MTU
              Discovery", RFC 4821, June 2007

              draft-ietf-ippm-btc-cap-00.txt Allman, M., "A Bulk
              Transfer Capacity Methodology for Cooperating Hosts",
              August 2001

   [MSMO]     The Macroscopic Behavior of the TCP Congestion Avoidance
              Algorithm Mathis, M.,Semke, J, Mahdavi, J, Ott, T
              July 1997 SIGCOMM Computer Communication Review,
              Volume 27 Issue 3

   [Stevens Vol1]  TCP/IP Illustrated, Vol1, The Protocols

Authors' Addresses

   Barry Constantine
   JDSU, Test and Measurement Division
   One Milesone Center Court
   Germantown, MD 20876-7100

   Phone: +1 240 404 2227

   Gilles Forget
   Independent Consultant to Bell Canada.
   308, rue de Monaco, St-Eustache
   Qc. CANADA, Postal Code : J7P-4T5

   Phone: (514) 895-8212

   Loki Jorgenson
   Apparent Networks

   Phone: (604) 433-2333 ext 105

   Reinhard Schrage
   Schrage Consulting

   Phone: +49 (0) 5137 909540

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