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Versions: (draft-constantine-ippm-tcp-throughput-tm) 00 01 02 03 04 05 06 07 08 09 10 11 12 13 RFC 6349

Network Working Group                                     B. Constantine
Internet-Draft                                                      JDSU
Intended status: Informational                                 G. Forget
Expires: March 24, 2011                    Bell Canada (Ext. Consultant)
                                                        Reinhard Schrage
                                                      Schrage Consulting
                                                      September 24, 2010



                  Framework for TCP Throughput Testing
                draft-ietf-ippm-tcp-throughput-tm-07.txt

Abstract

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

Requirements Language

   The key words "MUST", "MUST NOT", "REQUIRED", "SHALL", "SHALL NOT",
   "SHOULD", "SHOULD NOT", "RECOMMENDED", "MAY", and "OPTIONAL" in this
   document are to be interpreted as described in RFC 2119 [RFC2119].

Status of this Memo

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

   Internet-Drafts are working documents of the Internet Engineering
   Task Force (IETF).  Note that other groups may also distribute
   working documents as Internet-Drafts.  The list of current Internet-
   Drafts is at http://datatracker.ietf.org/drafts/current/.

   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."

           This Internet-Draft will expire on March 24, 2011.







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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
   (http://trustee.ietf.org/license-info) in effect on the date of
   publication of this document.  Please review these documents
   carefully, as they describe your rights and restrictions with respect
   to this document.  Code Components extracted from this document must
   include Simplified BSD License text as described in Section 4.e of
   the Trust Legal Provisions and are provided without warranty as
   described in the Simplified BSD License.


Table of Contents

   1.  Introduction . . . . . . . . . . . . . . . . . . . . . . . . .  3
     1.1   Test Set-up and Terminology  . . . . . . . . . . . . . . .  4
   2.  Scope and Goals of this methodology. . . . . . . . . . . . . .  5
     2.1   TCP Equilibrium State Throughput . . . . . . . . . . . . .  6
     2.2   Metrics for TCP Throughput Tests . . . . . . . . . . . . .  7
   3.  TCP Throughput Testing Methodology . . . . . . . . . . . . . .  9
     3.1   Determine Network Path MTU . . . . . . . . . . . . . . . . 11
     3.2.  Baseline Round Trip Time and Bandwidth . . . . . . . . . . 13
         3.2.1  Techniques to Measure Round Trip Time . . . . . . . . 13
         3.2.2  Techniques to Measure End-end Bandwidth . . . . . . . 14
     3.3.  TCP Throughput Tests . . . . . . . . . . . . . . . . . . . 14
         3.3.1 Calculate Optimum TCP Window Size. . . . . . . . . . . 15
         3.3.2 Conducting the TCP Throughput Tests. . . . . . . . . . 17
         3.3.3 Single vs. Multiple TCP Connection Testing . . . . . . 18
         3.3.4 Interpretation of the TCP Throughput Results . . . . . 19
     3.4. Traffic Management Tests .  . . . . . . . . . . . . . . . . 19
         3.4.1 Traffic Shaping Tests. . . . . . . . . . . . . . . . . 20
          3.4.1.1 Interpretation of Traffic Shaping Test Results. . . 20
         3.4.2 RED Tests. . . . . . . . . . . . . . . . . . . . . . . 21
          3.4.2.1 Interpretation of RED Results . . . . . . . . . . . 21
   4.  Security Considerations  . . . . . . . . . . . . . . . . . . . 22
   5.  IANA Considerations  . . . . . . . . . . . . . . . . . . . . . 22
     5.1.  Registry Specification . . . . . . . . . . . . . . . . . . 22
     5.2.  Registry Contents  . . . . . . . . . . . . . . . . . . . . 22
   6.  Acknowledgments  . . . . . . . . . . . . . . . . . . . . . . . 22
   7.  References . . . . . . . . . . . . . . . . . . . . . . . . . . 22
     7.1   Normative References . . . . . . . . . . . . . . . . . . . 22
     7.2   Informative References . . . . . . . . . . . . . . . . . . 23

   Authors' Addresses . . . . . . . . . . . . . . . . . . . . . . . . 23



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1. Introduction

   Network providers are coming to the realization that Layer 2/3
   testing and TCP layer testing are required to more adequately ensure
   end-user satisfaction.  Testing an operational network prior to
   customer activation is referred to as "turn-up" testing and the SLA
   (Service Level Agreement) is generally based upon Layer 2/3
   information rate, packet delay, loss and delay variation.  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 experience (and
   ultimately reach some level of TCP testing interoperability which
   does not exist today).

   Additionally, end-users (business enterprises) seek to conduct
   repeatable TCP throughput tests between enterprise locations.  Since
   these enterprises rely on the networks of the providers, a common
   test methodology (and metrics) would be equally beneficial to both
   parties.

   So the intent behind this TCP throughput methodology 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 equilibrium throughput is the maximum
   achievable for the TCP connection(s).

   There are many variables to consider when conducting a TCP throughput
   test and this methodology focuses on some of the most common
   parameters that MUST be considered such as:

   - Path MTU and Maximum Segment Size (MSS)
   - RTT and Bottleneck BW
   - Ideal TCP Window (Bandwidth Delay Product)
   - Single Connection and Multiple Connection testing

   This methodology proposes a test which SHOULD be performed in
   addition to traditional Layer 2/3 type tests, which are conducted to
   verify the integrity of the network before conducting TCP tests.
   Examples include iperf (UDP mode) or manual packet layer test
   techniques where packet throughput, loss, and delay measurements are
   conducted.  When available, standardized testing similar to RFC 2544
   [RFC2544] but adapted for use on operational networks may be used
   (because RFC 2544 methods are not intended for use outside the lab
   environment).

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1.1 Test Set-up and Terminology

   This section provides a general overview of the test configuration
   for this methodology.  The test is intended to be conducted on an
   end-end operational network, so there are multitudes of network
   architectures and topologies that can be tested.  This test set-up
   diagram is very general and the main intent is to illustrate the
   segmentation of the end user and network provider domains.

   Common terminologies used in the test methodology are:

   - Customer Provided Equipment (CPE), refers to customer owned
   - Customer Edge (CE), refers to provider owned demarcation device
   - Provider Edge (PE), refers to provider located distribution
     equipment
   - P (Provider), refers to provider core network equipment
   - Bottleneck Bandwidth*, lowest bandwidth along the complete network
     path
   - Round-Trip Time (RTT), refers to Layer 4 back and forth delay
   - Round-Trip Delay (RTD), refers to Layer 1 back and forth delay
   - Network Under Test (NUT), refers to the tested IP network path
   - TCP Throughput Test Device (TCP TTD), refers to compliant TCP
     host that generates traffic and measures metrics as defined in
     this methodology


 +----+ +----+ +----+  +----+ +---+  +---+ +----+  +----+ +----+ +----+
 |    | |    | |    |  |    | |   |  |   | |    |  |    | |    | |    |
 | TCP|-| CPE|-| CE |--| PE |-| P |--| P |-| PE |--| CE |-| CPE|-| TCP|
 | TD | |    | |    |BB|    | |   |  |   | |    |BB|    | |    | | TD |
 +----+ +----+ +----+**+----+ +---+  +---+ +----+**+----+ +----+ +----+

         <------------------------ NUT ------------------------>

         <-------------------------RTT ------------------------>


*  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
   NOT limited to): load balancers, proxy servers, WAN acceleration
   devices.  The detailed topology of the NUT MUST be considered when
   conducting the TCP throughput tests, but this methodology makes no
   attempt to characterize TCP performance related to specific network
   architectures.


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2. Scope and Goals of this Methodology

   Before defining the goals of this methodology, it is important to
   clearly define the areas that are out-of-scope for this
   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.

   - The methodology does not propose a method to operate permanently
   with high measurement loads. TCP performance and optimization data of
   operational networks MAY be captured and evaluated by using data of
   the "TCP Extended Statistics MIB" [RFC4898].

   - The methodology is not intended to measure TCP throughput as part
   of an SLA, or to compare the TCP performance between service
   providers or to compare between implementations of this methodology
   (test equipment).

   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.

   The specific goals are to:

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

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

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   - Define three (3) basic metrics that can be used to compare the
   performance of TCP connections under various network conditions.

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


2.1 TCP Equilibrium State Throughput

   TCP connections have three (3) fundamental congestion window phases
   as documented in [RFC5681].  These phases 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 phases.



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




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   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.








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   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.).

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   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
   the increase in RTT during a TCP throughput test from the inherent
   network RTT (baseline RTT).  The baseline RTT is the round-trip time
   inherent to the network path under non-congested conditions.

   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
   metrics 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. TCP Throughput Testing Methodology

   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
   equilibrium throughput MAY be very difficult to achieve (in a
   "dysfunctional" network).




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   TCP Throughput testing MAY require cooperation between the end user
   customer and the network provider.  In a Layer 2/3 VPN architecture,
   the testing SHOULD be conducted on the Customer Edge (CE) router and
   not the Provider Edge (PE) router.

   The following represents 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], MUST be conducted to verify the maximum
   network path MTU.  Conducting PLPMTUD establishes the upper limit for
   the MSS to be used in subsequent steps.

   2. Baseline Round Trip Time and Bandwidth. This step establishes the
   inherent, non-congested Round Trip Time (RTT) and the bottleneck
   bandwidth of the end-end network path.  These measurements are used
   to provide estimates of the ideal TCP window size, which SHOULD be
   used in subsequent test steps.  These measurements reference
   [RFC2681] and [RFC4898] to measure RTD (and the associated RTT).
   Also, [RFC5136] is referenced to measure network capacity.

   3. TCP Connection Throughput Tests.  With baseline measurements
   of Round Trip Time and bottleneck bandwidth, a series of single and
   multiple TCP connection throughput tests SHOULD be conducted to
   baseline the network performance expectations.

   4. Traffic Management Tests.  Various traffic management and queuing
   techniques SHOULD be tested in this step, using multiple TCP
   connections.  Multiple connection testing SHOULD 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.

   Whether the TCP test host is a standard computer or a compliant TCP
   TTD, the following areas SHOULD be considered when selecting
   a test host:

   - TCP implementation used by the test host OS version, 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.  When a compliant TCP TTD is used, the TCP
   implementation SHOULD be identified in the test results. The
   compliant TCP TTD SHOULD be usable for complete end-to-end testing
   through network security elements and SHOULD also be usable for
   testing network sections.

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   - 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. 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.
   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
   require dedicated hardware based test tools. In the absence of
   dedicated hardware based test tools, these measurements MAY need to
   be conducted with packet capture tools (conduct TCP throughput tests
   and analyze RTT and retransmission results with packet captures).
   Another option MAY be to use "TCP Extended Statistics MIB" per
   [RFC4898].

  - The compliant TCP TTD and its access to the network under test MUST
    NOT introduce a performance bottleneck of any kind.


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] MUST
   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 a 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.




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   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 MAY 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.

   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 MUST 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 MAY 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
   limit.








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3.2. Baseline Round Trip Time and Bandwidth

   Before stateful TCP testing can begin, it is important to determine
   the baseline Round Trip Time (non-congested inherent delay) and
   bottleneck bandwidth of the end-end network to be tested. These
   measurements are used to provide estimates of the ideal TCP window
   size, which SHOULD be used in subsequent test steps.  These latency
   and bandwidth tests SHOULD be run during the time of day for which
   the TCP throughput tests will occur.

   3.2.1 Techniques to Measure Round Trip Time

   Following the definitions used in the section 1.1; 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 Acknowledgment.  Round Trip Delay (RTD) is used
   synonymously to twice the Link Latency.  RTT measurements SHOULD use
   techniques defined in [RFC2681] or statistics available from MIBs
   defined in [RFC4898].

   The RTT SHOULD be baselined during "off-peak" hours to obtain a
   reliable figure for inherent network latency versus additional delay
   caused by network buffering delays.

   During the actual sustained TCP throughput tests, RTT MUST be
   measured along with TCP throughput. Buffer delay 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.





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  - ICMP Pings MAY also be adequate to provide round trip time
   estimations.  Some limitations of ICMP Ping MAY include 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 MAY be asymmetrical. Measurements SHOULD use
   network capacity techniques defined in [RFC5136].

   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.


3.3. TCP Throughput Tests

   This methodology specifically defines TCP throughput techniques to
   verify sustained TCP performance in a managed business network.
   Defined in section 2.1, the equilibrium throughput reflects the
   maximum rate achieved by a TCP connection within the congestion
   avoidance phase on an 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 and multiple TCP connection
   throughput tests can be conducted to baseline network performance
   against expectations.

   It is RECOMMENDED to run the tests in each direction independently
   first, then run both directions simultaneously.  In each case, the
   TCP Transfer Time, TCP Efficiency, and Buffer Delay metrics MUST be
   measured in each direction.










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3.3.1 Calculate Ideal TCP Window Size

   The ideal TCP window size can be calculated from the bandwidth
   delay product (BDP), which is:

   BDP (bits) = RTT (sec) x Bandwidth (bps)

   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 ideal 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 TCP 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, 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

   Link                               Ideal TCP      Maximum Achievable
   Speed*    RTT (ms)  BDP (bits)  Window (kBytes)  TCP Throughput(Mbps)
   ---------------------------------------------------------------------
    T1         20        30,720          3.84              1.17
    T1         50        76,800          9.60              1.40
    T1        100       153,600         19.20              1.40
    T3         10       442,100         55.26             42.05
    T3         15       663,150         82.89             42.05
    T3         25     1,105,250        138.16             41.52
    T3(ATM)    10       407,040         50.88             36.50
    T3(ATM)    15       610,560         76.32             36.23
    T3(ATM)    25     1,017,600        127.20             36.27
    100M        1       100,000         12.50             91.98
    100M        2       200,000         25.00             93.44
    100M        5       500,000         62.50             93.44
    1Gig      0.1       100,000         12.50            919.82
    1Gig      0.5       500,000         62.50            934.47
    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



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   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
     second)

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

   1 - 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 - 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
   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.

  The following diagram compares achievable TCP throughputs on a T3 link
  with Windows 2000/XP TCP window sizes of 16KB versus 64KB.


           45|
             |          _____42.1M
           40|          |64K|
TCP          |          |   |
Throughput 35|          |   |           _____34.3M
in Mbps      |          |   |           |64K|
           30|          |   |           |   |
             |          |   |           |   |
           25|          |   |           |   |
             |          |   |           |   |
           20|          |   |           |   |           _____20.5M
             |          |   |           |   |           |64K|
           15| 14.5M____|   |           |   |           |   |
             |      |16K|   |           |   |           |   |
           10|      |   |   |   9.6M+---+   |           |   |
             |      |   |   |       |16K|   |   5.8M____+   |
            5|      |   |   |       |   |   |       |16K|   |
             |______+___+___+_______+___+___+_______+__ +___+_______
                        10              15              25
                                RTT in milliseconds


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   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.

           45|
             |                                             +-----+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 KBytes

3.3.2 Conducting the TCP Throughput Tests

   There are several TCP tools that are commonly used in the network
   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
   the TCP throughput tests and other tests in the following sections.
   Dedicated test equipment 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).



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   The TCP throughput test SHOULD be run over a long enough duration
   to properly exercise network buffers and also characterize
   performance during different time periods of the day.

   Note that both the TCP Transfer Time, TCP Efficiency, and Buffer
   Delay metrics 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 Single vs. Multiple TCP Connection Testing

   The decision whether to conduct single or multiple TCP connection
   tests depends upon the size of the BDP in relation to the window
   sizes configured in the end-user environment.  For example, if the
   BDP for a long-fat pipe turns out to be 2MB, then it is probably more
   realistic to test this pipe with multiple connections. Assuming
   typical host computer window settings of 64 KB, using 32 connections
   would realistically test this pipe.

   The following table is provided to illustrate the relationship of the
   BDP, window size, and the number of connections required to utilize
   the available capacity.  For this example, the network bandwidth is
   500 Mbps, RTT is equal to 5 ms, and the BDP equates to 312 KBytes.

              #Connections
    Window    to Fill Link
   ------------------------
    16KB          20
    32KB          10
    64KB           5
    128KB          3

   The TCP Transfer Time metric is useful for conducting multiple
   connection tests.  Each connection SHOULD be configured to transfer
   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
   results.

   Note that the TCP transfer time is the time for all connections to
   complete the transfer of the configured payload size.  From the
   example table listed above, the 64KB window is considered.  Each of
   the 5 connections would be configured to transfer 100MB, and each
   TCP should obtain a maximum of 100 Mb/sec per connection.  So for
   this example, the 100MB payload should be transferred across the
   connections in approximately 8 seconds (which would be the ideal TCP
   transfer time for these conditions).

   Additionally, the TCP Efficiency metric SHOULD be computed for each
   connection tested (defined in section 2.2).

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3.3.4 Interpretation of the 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 ideal value, some possible causes
   are listed:

   - Network congestion causing packet loss which MAY be inferred from
   a poor TCP Efficiency metric (100% = no loss)
   - Network congestion causing an increase in RTT which MAY be inferred
   from the Buffer Delay metric (0% = no increase in RTT over baseline)
   - Intermediate network devices which actively regenerate the TCP
   connection and can alter window size, MSS, etc.
   - Rate limiting (policing).  More discussion of traffic management
   tests follows in section 3.4

3.4. Traffic Management Tests

   In most cases, the network connection between two geographic
   locations (branch offices, etc.) is lower than the network connection
   of the host computers.  An example would be LAN connectivity of GigE
   and WAN connectivity of 100 Mbps.  The WAN connectivity may be
   physically 100 Mbps or logically 100 Mbps (over a GigE WAN
   connection).  In the later case, rate limiting is used to provide the
   WAN bandwidth per the SLA.

   Traffic management techniques are employed to provide various forms
   of QoS, the more common include:

   - Traffic Shaping
   - Priority queuing
   - Random Early Discard (RED, etc.)

   Configuring the end-end network with these various traffic management
   mechanisms is a complex under-taking.  For traffic shaping and RED
   techniques, the end goal is to provide better performance for bursty
   traffic such as TCP (RED is specifically intended for TCP).

   This section of the methodology provides guidelines to test traffic
   shaping and RED implementations.  As in section 3.3, host hardware
   performance MUST be well understood before conducting the traffic
   shaping and RED tests. Dedicated test equipment will generally be
   REQUIRED for line rates of GigE and 10 GigE.  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).



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3.4.1 Traffic Shaping Tests

   For services where the available bandwidth is rate limited, there are
   two (2) techniques used to implement rate limiting: traffic policing
   and traffic shaping.

   Simply stated, traffic policing marks and/or drops packets which
   exceed the SLA bandwidth (in most cases, excess traffic is dropped).
   Traffic shaping employs the use of queues to smooth the bursty
   traffic and then send out within the SLA bandwidth limit (without
   dropping packets unless the traffic shaping queue is exceeded).

   Traffic shaping is generally configured for TCP data services and
   can provide improved TCP performance since the retransmissions are
   reduced, which in turn optimizes TCP throughput for the given
   available bandwidth.  Through this section, the available
   rate-limited bandwidth shall be referred to as the
   "bottleneck bandwidth".

   The ability to detect proper traffic shaping is more easily diagnosed
   when conducting a multiple TCP connection test.  Proper shaping will
   provide a fair distribution of the available bottleneck bandwidth,
   while traffic policing will not.

   The traffic shaping tests are built upon the concepts of multiple
   connection testing as defined in section 3.3.3.  Calculating the BDP
   for the bottleneck bandwidth is first REQUIRED before selecting the
   number of connections and TCP Window size per connection.

   Similar to the example in section 3.3, a typical test scenario might
   be:  GigE LAN with a 100Mbps bottleneck bandwidth (rate limited
   logical interface), and 5 msec RTT.  This would require five (5) TCP
   connections of 64 KB window size evenly fill the bottleneck bandwidth
   (about 100 Mbps per connection).

   The traffic shaping test SHOULD be run over a long enough duration to
   properly exercise network buffers (greater than 30 seconds) and also
   characterize performance during different time periods of the day.
   The throughput of each connection MUST be logged during the entire
   test, along with the TCP Transfer Time, TCP Efficiency, and
   Buffer Delay metrics.

3.4.1.1 Interpretation of Traffic Shaping Test Results

   By plotting the throughput achieved by each TCP connection, the fair
   sharing of the bandwidth is generally very obvious when traffic
   shaping is properly configured for the bottleneck interface.  For the
   previous example of 5 connections sharing 500 Mbps, each connection
   would consume ~100 Mbps with a smooth variation.  If traffic policing
   was present on the bottleneck interface, the bandwidth sharing MAY
   not be fair and the resulting throughput plot MAY reveal "spikey"
   throughput consumption of the competing TCP connections (due to the
   retransmissions).

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3.4.2 RED Tests

   Random Early Discard techniques are specifically targeted to provide
   congestion avoidance for TCP traffic.  Before the network element
   queue "fills" and enters the tail drop state, RED drops packets at
   configurable queue depth thresholds.  This action causes TCP
   connections to back-off which helps to prevent tail drop, which in
   turn helps to prevent global TCP synchronization.

   Again, rate limited interfaces can benefit greatly from RED based
   techniques.  Without RED, TCP is generally not able to achieve the
   full bandwidth of the bottleneck interface.  With RED enabled, TCP
   congestion avoidance throttles the connections on the higher speed
   interface (i.e. LAN) and can reach equilibrium with the bottleneck
   bandwidth (achieving closer to full throughput).

   The ability to detect proper RED configuration is more easily
   diagnosed when conducting a multiple TCP connection test.  Multiple
   TCP connections provide the multiple bursty sources that emulate the
   real-world conditions for which RED was intended.

   The RED tests also build upon the concepts of multiple connection
   testing as defined in section 3.3.3.  Calculating the BDP for the
   bottleneck bandwidth is first REQUIRED before selecting the number
   of connections and TCP Window size per connection.

   For RED testing, the desired effect is to cause the TCP connections
   to burst beyond the bottleneck bandwidth so that queue drops will
   occur.  Using the same example from section 3.4.1 (traffic shaping),
   the 500 Mbps bottleneck bandwidth requires 5 TCP connections (with
   window size of 64Kb) to fill the capacity.  Some experimentation is
   REQUIRED, but it is RECOMMENDED to start with double the number of
   connections to stress the network element buffers / queues.  In this
   example, 10 connections SHOULD produce TCP bursts of 64KB for each
   connection.  If the timing of the TCP tester permits, these TCP
   bursts SHOULD stress queue sizes in the 512KB range.  Again
   experimentation will be REQUIRED and the proper number of TCP
   connections and TCP window size will be dictated by the size the
   network element queue.


3.4.2.1 Interpretation of RED Results

   The default queuing technique for most network devices is FIFO based.
   Without RED, the FIFO based queue will cause excessive loss to all of
   the TCP connections and in the worst case global TCP synchronization.




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   By plotting the aggregate throughput achieved on the bottleneck
   interface, proper RED operation MAY be determined if the bottleneck
   bandwidth is fully utilized.  For the previous example of 10
   connections (window = 64 KB) sharing 500 Mbps, each connection SHOULD
   consume ~50 Mbps.  If RED was not properly enabled on the interface,
   then the TCP connections will retransmit at a higher rate and the
   net effect is that the bottleneck bandwidth is not fully utilized.

   Another means to study non-RED versus RED implementation is to use
   the TCP Transfer Time metric for all of the connections.  In this
   example, a 100 MB payload transfer SHOULD take ideally 16 seconds
   across all 10 connections (with RED enabled).  With RED not enabled,
   the throughput across the bottleneck bandwidth MAY be greatly
   reduced (generally 20-40%) and the actual TCP Transfer time MAY be
   proportionally longer then the Ideal TCP Transfer time.

   Additionally, the TCP Transfer Efficiency metric is useful, since
   non-RED implementations MAY exhibit a lower TCP Transfer Efficiency.

4.  Security Considerations

   The security considerations that apply to any active measurement of
   live networks are relevant here as well.  See [RFC4656] and
   [RFC5357].

5.  IANA Considerations

   This document does not REQUIRE an IANA registration for ports
   dedicated to the TCP testing described in this document.

6.  Acknowledgments

   Thanks to Matt Mathis, Matt Zekauskas, Al Morton, Rudi Geib, and
   Yaakov Stein, Loki Jorgenson for many good comments and for pointing
   us to great sources of information pertaining to past works in the
   TCP capacity area.

7.  References

7.1 Normative References

   [RFC2119]  Bradner, S., "Key words for use in RFCs to Indicate
              Requirement Levels", BCP 14, RFC 2119, March 1997.

   [RFC4656]  Shalunov, S., Teitelbaum, B., Karp, A., Boote, J., and M.
              Zekauskas, "A One-way Active Measurement Protocol
              (OWAMP)", RFC 4656, September 2006.

   [RFC5681]  Allman, M., Paxson, V., Stevens W., "TCP Congestion
              Control", RFC 5681, September 2009.

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

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

   [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

   [RFC2681]  Almes G., Kalidindi S., Zekauskas, M., "A Round-trip Delay
              Metric for IPPM", RFC 2681, September, 1999

   [RFC4898]  Mathis, M., Heffner, J., Raghunarayan, R., "TCP Extended
              Statistics MIB", May 2007

   [RFC5136]  Chimento P., Ishac, J., "Defining Network Capacity",
              February 2008

   [RFC1323]  Jacobson, V., Braden, R., Borman D., "TCP Extensions for
              High Performance", May 1992

7.2.  Informative References


Authors' Addresses

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

   Phone: +1 240 404 2227
   barry.constantine@jdsu.com

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

   Phone: (514) 895-8212
   gilles.forget@sympatico.ca

   Reinhard Schrage
   Schrage Consulting

   Phone: +49 (0) 5137 909540
   reinhard@schrageconsult.com

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