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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: January 9, 2011                     Bell Canada (Ext. Consultant)
                                                              L. Jorgenson
                                                                   nooCore
                                                          Reinhard Schrage
                                                        Schrage Consulting
                                                              July 9, 2010



                    TCP Throughput Testing Methodology
                draft-ietf-ippm-tcp-throughput-tm-04.txt

Abstract

   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
   layer.



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 July 9, 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
   http://www.ietf.org/ietf/1id-abstracts.txt.

   The list of Internet-Draft Shadow Directories can be accessed at
   http://www.ietf.org/shadow.html.


           This Internet-Draft will expire on January 9, 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 BSD License.


Table of Contents

   1.  Introduction . . . . . . . . . . . . . . . . . . . . . . . . .  3
   2.  Goals of this Methodology. . . . . . . . . . . . . . . . . . .  4
     2.1   TCP Equilibrium State Throughput . . . . . . . . . . . . .  5
     2.2   Metrics for TCP Throughput Tests . . . . . . . . . . . . .  6
   3.  TCP Throughput Testing Methodology . . . . . . . . . . . . . .  6
     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 . . . . . . . .  9
         3.2.2  Techniques to Measure End-end Bandwidth . . . . . . . 10
     3.3.  TCP Throughput Tests . . . . . . . . . . . . . . . . . . . 10
         3.3.1 Calculate Optimum TCP Window Size. . . . . . . . . . . 11
         3.3.2 Conducting the TCP Throughput Tests. . . . . . . . . . 14
         3.3.3 Single vs. Multiple TCP Connection Testing . . . . . . 14
         3.3.4 Interpretation of the TCP Throughput Results . . . . . 15
     3.4. Traffic Management Tests .  . . . . . . . . . . . . . . . . 15
         3.4.1 Traffic Shaping Tests. . . . . . . . . . . . . . . . . 16
          3.4.1.1 Interpretation of Traffic Shaping Test Results. . . 17
         3.4.2 RED Tests. . . . . . . . . . . . . . . . . . . . . . . 17
          3.4.2.1 Interpretation of RED Results . . . . . . . . . . . 18
   4.  Acknowledgements . . . . . . . . . . . . . . . . . . . . . . . 18
   5.  References . . . . . . . . . . . . . . . . . . . . . . . . . . 19
   Authors' Addresses . . . . . . . . . . . . . . . . . . . . . . . . 20











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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
   jitter.

   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. 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).

   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 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.  Examples include
   OWAMP or manual packet layer test techniques where packet throughput,
   loss, and delay measurements are conducted.  It is  highly recommended
   to run traditional Layer 2/3 type test to verify the integrity of the
   network before conducting TCP tests.











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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 (Maximum Segment Size), # 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.

   - Define two (2) 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 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     | _/
            | _/                                   |/
            |/___________________________________________________________
                                                            Time


   This TCP methodology provides guidelines to measure the equilibrium
   throughput which refers to the maximum sustained rate obtained by
   congestion avoidance before packet loss conditions occur (which 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 complete set of metrics that
   account for the myriad of variables (i.e. RTT variation, loss
   conditions, TCP implementation, etc.).  However, these two basic
   metrics faciliate TCP throughput comparisons under varying network
   conditions and between network traffic management techniques.

   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 provides a comparative measure between various QoS
   mechanisms such as traffic management, congestion avoidance, and also
   various TCP implementations (i.e. Reno, Vegas, etc.).

   As an example, if 1000 TCP segments were sent and 20 had to be
   retransmitted, the TCP Efficiency would be calculated as:


                   1000 - 20
                   ---------  x 100 = 98%
                     1000


   The second metric is the TCP Transfer Time, which is simply the 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 are generally
   required.  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 TCP Transfer
   Time as the total transfer time of 10MB over all 8 connections, the
   single transfer time metric is a useful means to compare various
   traffic management techniques (i.e. FiFO, WFQ queuing, WRED, etc.).





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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).

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

   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

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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 generally requires high
   performance server hardware or dedicated hardware based test tools.


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.

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   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 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
   limit.

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

   Following the definitions used in the references of the appendix;
   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.


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   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).

   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.


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   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.


3.3. TCP 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 and multiple TCP connection
   throughput tests can be conducted to baseline network performance
   against expectations.


3.3.1 Calculate Optimum TCP Window Size

   The optimum 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 "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 1460 byte MSS.  Also in this table, the case of 58 byte
   L1-L4 overhead including the Ethernet CRC32 is used for simplicity.






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   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              1.17
    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 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
     second)

   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.


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  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

   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


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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 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.

   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.

   The TCP throughput test should be run over a a long enough duration
   to properly exercise network buffers and also characterize performance
   during different time periods of the day.  The results must be logged
   at the desired interval and 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.

   Additionally, the TCP Efficiency and TCP Transfer time metrics should
   be logged in order to further characterize the window size tests.

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
   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.





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              #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
   a certain payload (i.e. 100 MB), and the TCP Transfer time provides
   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).

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 predicted value, some possible causes
   are listed:

   - Network congestion causing packet loss; the TCP Efficiency metric
   is a useful gauge to compare network performance
   - Network congestion not causing packet loss but increasing RTT
   - Intermediate network devices which actively regenerate the TCP
   connection and can alter window size, MSS, etc.
   - Over utilization of available link or 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.


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   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, especially for line rates of GigE and 10 GigE.

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 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 and then selecting
   the number of connections / 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).

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   The traffic shaping should be run over a long enough duration to
   properly exercise network buffers 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
   Efficiency and TCP Transfer time metric. Additionally, it is
   recommended to log RTT and retransmissions per connection over the test
   interval.

3.4.1.1 Interpretation of Traffic Shaping Test Restults

   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 would not
   be fair and the resulting throughput plot would reveal "spikey"
   connection throughput consumption of the competing TCP connections
   (due to the retransmissions).

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 equalibrium 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 secion 3.3.3.  Calculating the BDP for the
   bottleneck bandwidth is first required and then selecting the number of
   connections / window size per connection.






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   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 would produce TCP bursts of 64KB for each connection.
   If the timing of the TCP tester permits, these TCP bursts could stress
   queue sizes in the 512KB range.  Again experimentation will be required
   and the proper number of TCP connections / 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.

   By plotting the aggregate throughput achieved on the bottleneck
   interface, proper RED operation can 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 would be greatly reduced
   (generally 20-40%) and the TCP Transfer time would be proportionally
   longer then the ideal transfer time.

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

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.



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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
              2001.
   [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

   [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
              Addison-Wesley


















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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

   Loki Jorgenson
   nooCore

   Phone: (604) 908-5833
   ljorgenson@nooCore.com


   Reinhard Schrage
   Schrage Consulting

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





















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