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Framework for TCP Throughput Testing
draft-ietf-ippm-tcp-throughput-tm-13

The information below is for an old version of the document that is already published as an RFC.
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This is an older version of an Internet-Draft that was ultimately published as RFC 6349.
Authors Reinhard Schrage , Gilles Forget , Ruediger Geib , Barry Constantine
Last updated 2015-10-14 (Latest revision 2011-05-31)
Replaces draft-constantine-ippm-tcp-throughput-tm
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draft-ietf-ippm-tcp-throughput-tm-13
Network Working Group                                     B. Constantine
Internet-Draft                                                      JDSU
Intended status: Informational                                 G. Forget
Expires: November 30, 2011                 Bell Canada (Ext. Consultant)
                                                           Ruediger Geib
                                                        Deutsche Telekom 
                                                        Reinhard Schrage
                                                      Schrage Consulting                                   

                                                            May 31, 2011

                  Framework for TCP Throughput Testing   
                draft-ietf-ippm-tcp-throughput-tm-13.txt

Abstract

   This framework describes a practical methodology for measuring end-
   to-end TCP Throughput in a managed IP network. The goal is to provide 
   a better indication in regards to user experience. In this framework, 
   TCP and IP parameters are specified to optimize TCP throughput. 

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 November 30, 2011.

                
                
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   Copyright Notice

   Copyright (c) 2011 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   Terminology. . . . . . . . . . . . . . . . . . . . . . . .  4
     1.2   TCP Equilibrium  . . . . . . . . . . . . . . . . . . . . .  5
   2.  Scope and Goals  . . . . . . . . . . . . . . . . . . . . . . .  6
   3.  Methodology. . . . . . . . . . . . . . . . . . . . . . . . . .  7
     3.1   Path MTU . . . . . . . . . . . . . . . . . . . . . . . . .  9
     3.2   Round Trip Time (RTT) and Bottleneck Bandwidth (BB). . . .  9
         3.2.1  Measuring RTT . . . . . . . . . . . . . . . . . . . .  9
         3.2.2  Measuring BB  . . . . . . . . . . . . . . . . . . . . 10
     3.3.  Measuring TCP Throughput . . . . . . . . . . . . . . . . . 11
         3.3.1 Minimum TCP RWND . . . . . . . . . . . . . . . . . . . 11
   4.  TCP Metrics  . . . . . . . . . . . . . . . . . . . . . . . . . 14
     4.1   Transfer Time Ratio. . . . . . . . . . . . . . . . . . . . 14
         4.1.1 Maximum Achievable TCP Throughput calculation  . . . . 15
         4.1.2 Transfer Time and Transfer Time Ratio calculation. . . 16
     4.2   TCP Efficiency . . . . . . . . . . . . . . . . . . . . . . 17
         4.2.1 TCP Efficiency Percentage calculation  . . . . . . . . 17
     4.3   Buffer Delay . . . . . . . . . . . . . . . . . . . . . . . 17
         4.3.1 Buffer Delay Percentage calculation. . . . . . . . . . 17
   5.  Conducting TCP Throughput Tests. . . . . . . . . . . . . . . . 18 
     5.1   Single versus Multiple Connections . . . . . . . . . . . . 18
     5.2   Results Interpretation . . . . . . . . . . . . . . . . . . 19
   6.  Security Considerations  . . . . . . . . . . . . . . . . . . . 21
   7.  IANA Considerations  . . . . . . . . . . . . . . . . . . . . . 21
   8.  Acknowledgments  . . . . . . . . . . . . . . . . . . . . . . . 22
   9.  References . . . . . . . . . . . . . . . . . . . . . . . . . . 22
     9.1   Normative References . . . . . . . . . . . . . . . . . . . 22
     9.2   Informative References . . . . . . . . . . . . . . . . . . 22

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

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1. Introduction

   In the network industry, the SLA (Service Level Agreement) provided 
   to business class customers is generally based upon Layer 2/3 
   criteria such as:   Bandwidth, latency, packet loss and delay 
   variations (jitter).  Network providers are coming to the realization 
   that Layer 2/3 testing is not enough to adequately ensure end-user's 
   satisfaction.  In addition to Layer 2/3 testing, this framework 
   recommends a methodology for measuring TCP Throughput in order to 
   provide meaningful results with respect to user experience.

   Additionally, business class customers seek to conduct repeatable TCP
   Throughput tests between locations. Since these organizations rely on 
   the networks of the providers, a common test methodology with 
   predefined metrics would benefit both parties. 

   Note that the primary focus of this methodology is managed business 
   class IP networks; e.g. those Ethernet terminated services for which 
   organizations are provided an SLA from the network provider.  Because 
   of the SLA, the expectation is that the TCP Throughput should achieve
   the guaranteed bandwidth.   End-users with "best effort" access could 
   use this methodology, but this framework and its metrics are intended 
   to be used in a predictable managed IP network.   No end-to-end 
   performance can be guaranteed when only the access portion is being 
   provisioned to a specific bandwidth capacity.

   The intent behind this document is to define a methodology for 
   testing sustained TCP Layer performance.  In this document, the 
   achievable TCP Throughput is that amount of data per unit time that 
   TCP transports when in the TCP Equilibrium state.  (See Section 1.2 
   for TCP Equilibrium definition).  Throughout this document, maximum 
   achievable throughput refers to the theoretical achievable throughput 
   when TCP is in the Equilibrium state.

   TCP is connection oriented and at the transmitting side it uses a 
   congestion window, (TCP CWND).  At the receiving end, TCP uses a 
   receive window, (TCP RWND) to inform the transmitting end on how 
   many Bytes it is capable to accept at a given time.  
   
   Derived from Round Trip Time (RTT) and network Bottleneck Bandwidth
   (BB), the Bandwidth Delay Product (BDP) determines the Send and 
   Received Socket buffers sizes required to achieve the maximum TCP 
   Throughput.  Then, with the help of slow start and congestion 
   avoidance algorithms, a TCP CWND is calculated based on the IP 
   network path loss rate.  Finally, the minimum value between the 
   calculated TCP CWND and the TCP RWND advertised by the opposite end 
   will determine how many Bytes can actually be sent by the
   transmitting side at a given time.

   Both TCP Window sizes (RWND and CWND) may vary during any given TCP 
   session, although up to bandwidth limits, larger RWND and larger CWND 
   will achieve higher throughputs by permitting more in-flight Bytes.
   

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   At both ends of the TCP connection and for each socket, there are 
   default buffer sizes.  There are also kernel enforced maximum buffer 
   sizes.  These buffer sizes can be adjusted at both ends (transmitting 
   and receiving).  Some TCP/IP stack implementations use Receive Window 
   Auto-Tuning, although in order to obtain the maximum throughput it is 
   critical to use large enough TCP Send and Receive Socket Buffer 
   sizes.  In fact, they SHOULD be equal to or greater than BDP.
      
   Many variables are involved in TCP Throughput performance, but this 
   methodology focuses on: 
   - BB (Bottleneck Bandwidth)
   - RTT (Round Trip Time)   
   - Send and Receive Socket Buffers
   - Minimum TCP RWND
   - Path MTU (Maximum Transmission Unit)
   
   This methodology proposes TCP testing that SHOULD be performed in 
   addition to traditional Layer 2/3 type tests.   In fact, Layer 2/3 
   tests are REQUIRED to verify the integrity of the network before 
   conducting TCP tests.  Examples include iperf (UDP mode) and manual 
   packet layer test techniques where packet throughput, loss, and delay 
   measurements are conducted.  When available, standardized testing 
   similar to [RFC2544] but adapted for use in operational networks MAY 
   be used.
   
   Note: [RFC2544] was never meant to be used outside a lab environment.
   
   Sections 2 and 3 of this document provides a general overview of the 
   proposed methodology.  Section 4 defines the metrics while Section 5
   explains how to conduct the tests and interpret the results.

1.1 Terminology

   The common definitions used in this methodology are:

   - TCP Throughput Test Device (TCP TTD), refers to compliant TCP 
     host that generates traffic and measures metrics as defined in 
     this methodology. i.e. a dedicated communications test instrument.
   - Customer Provided Equipment (CPE), refers to customer owned 
     equipment (routers, switches, computers, etc.)
   - Customer Edge (CE), refers to provider owned demarcation device.
   - Provider Edge (PE), refers to provider's distribution equipment.
   - Bottleneck Bandwidth (BB), lowest bandwidth along the complete 
     path. Bottleneck Bandwidth and Bandwidth are used synonymously 
     in this document. Most of the time the Bottleneck Bandwidth is 
     in the access portion of the wide area network (CE - PE).
   - Provider (P), refers to provider core network equipment.
   - Network Under Test (NUT), refers to the tested IP network path.
   - Round Trip Time (RTT), is the elapsed time between the clocking in 
     of the first bit of a TCP segment sent and the receipt of the last
     bit of the corresponding TCP Acknowledgment.
   - Bandwidth Delay Product (BDP), refers to the product of a data 
     link's capacity (in bits per second) and its end-to-end delay 
     (in seconds). 

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   Figure 1.1 Devices, Links and Paths
 
 +----+ +----+ +----+  +----+ +---+  +---+ +----+  +----+ +----+ +----+
 | TCP|-| CPE|-| CE |--| PE |-| P |--| P |-| PE |--| CE |-| CPE|-| TCP|
 | TTD| |    | |    |BB|    | |   |  |   | |    |BB|    | |    | | TTD|
 +----+ +----+ +----+  +----+ +---+  +---+ +----+  +----+ +----+ +----+
        <------------------------ NUT ------------------------->
    R >-----------------------------------------------------------|
    T                                                             |
    T <-----------------------------------------------------------|

   Note that the NUT may be built with of a variety of devices including 
   but not limited to, load balancers, proxy servers or WAN acceleration 
   appliances.  The detailed topology of the NUT SHOULD be well known 
   when conducting the TCP Throughput tests, although this methodology 
   makes no attempt to characterize specific network architectures.
   
1.2 TCP Equilibrium

   TCP connections have three (3) fundamental congestion window phases,
   which are depicted in Figure 1.2.
   
   1 - The Slow Start phase, which occurs at the beginning of a TCP 
   transmission or after a retransmission time out.

   2 - The Congestion Avoidance phase, during which TCP ramps up to 
   establish the maximum achievable throughput.  It is important to note 
   that retransmissions are a natural by-product of the TCP congestion 
   avoidance algorithm as it seeks to achieve maximum throughput.

   3 - The Loss Recovery phase, which could include Fast Retransmit 
   (Tahoe) or Fast Recovery (Reno & New Reno).  When packet loss occurs, 
   Congestion Avoidance phase transitions either to Fast Retransmission 
   or Fast Recovery depending upon the TCP implementation. If a Time-Out 
   occurs, TCP transitions back to the Slow Start phase.

   Figure 1.2 TCP CWND Phases
   
        /\  |                                                
        /\  |High ssthresh  TCP CWND                         TCP
        /\  |Loss Event *   halving    3-Loss Recovery       Equilibrium
        /\  |          * \  upon loss                           
        /\  |          *  \    /  \        Time-Out            Adjusted 
        /\  |          *   \  /    \      +--------+         * ssthresh
        /\  |          *    \/      \    / Multiple|        * 
        /\  |          * 2-Congestion\  /  Loss    |        *
        /\  |         *    Avoidance  \/   Event   |       *   
   TCP      |        *              Half           |     * 
   Through- |      *                TCP CWND       | * 1-Slow Start 
   put      | * 1-Slow Start                      Min TCP CWND after T-O
            +-----------------------------------------------------------
             Time > > > > > > > > > > > > > > > > > > > > > > > > > > >

   Note: ssthresh = Slow Start threshold.

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   A well tuned and managed IP network with appropriate TCP adjustments
   in the IP hosts and applications should perform very close to the 
   BB when TCP is in the Equilibrium state.

   This TCP methodology provides guidelines to measure the maximum 
   achievable TCP Throughput when TCP is in the Equilibrium state.  
   All maximum achievable TCP Throughputs specified in Section 3.3 are 
   with respect to this condition.

   It is important to clarify the interaction between the sender's Send 
   Socket Buffer and the receiver's advertised TCP RWND Size.  TCP test 
   programs such as iperf, ttcp, etc. allows the sender to control the
   quantity of TCP Bytes transmitted and unacknowledged (in-flight), 
   commonly referred to as the Send Socket Buffer.   This is done 
   independently of the TCP RWND Size advertised by the receiver. 

2. Scope and Goals

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

   - This methodology is not intended to predict the TCP Throughput 
   during the transient stages of a TCP connection, such as during the 
   slow start phase.
   
   - This methodology is not intended to definitively benchmark TCP 
   implementations of one OS to another, although some users may find
   value in conducting qualitative experiments.
   
   - This methodology is not intended to provide detailed diagnosis 
   of problems within end-points or within the network itself as 
   related to non-optimal TCP performance, although results 
   interpretation for each test step may provide insights to potential 
   issues.
   
   - This methodology does not propose to operate permanently with high 
   measurement loads.  TCP performance and optimization within 
   operational networks MAY be captured and evaluated by using data 
   from the "TCP Extended Statistics MIB" [RFC4898]. 

   In contrast to the above exclusions, the primary goal is to define a 
   method to conduct a practical end-to-end assessment of sustained 
   TCP performance within a managed business class IP network.  Another 
   key goal is to establish a set of "best practices" that a non-TCP
   expert SHOULD apply when validating the ability of a managed IP 
   network to carry end-user TCP applications.     

   Specific goals are to:

   - Provide a practical test approach that specifies tunable parameters 
   (such as MTU (Maximum Transmit Unit) and Socket Buffer sizes) and how 
   these affect the outcome of TCP performances over an IP network.    

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   - Provide specific test conditions like link speed, RTT, MTU, Socket 
   Buffer sizes and achievable TCP Throughput when TCP is in the  
   Equilibrium state.  For guideline purposes, provide examples of 
   test conditions and their maximum achievable TCP Throughput.  
   Section 1.2 provides specific details concerning the definition of 
   TCP Equilibrium within this methodology while Section 3 provides 
   specific test conditions with examples.

   - Define three (3) basic metrics to compare the performance of TCP 
   connections under various network conditions.  See Section 4.
   
   - In test situations where the recommended procedure does not yield
   the maximum achievable TCP Throughput, this methodology provides 
   some areas within the end host or the network that SHOULD be 
   considered for investigation.   Although again, this methodology 
   is not intended to provide detailed diagnosis on these issues.  
   See Section 5.2.

3. Methodology

   This methodology is intended for operational and managed IP networks.
   A multitude of network architectures and topologies can be tested. 
   The diagram in Figure 1.1 is very general and is only there to 
   illustrate typical segmentation within end-user and network provider 
   domains.   

   Also, as stated earlier in Section 1, it is considered best practice 
   to verify the integrity of the network by conducting Layer 2/3 tests 
   such as [RFC2544] or other methods of network stress tests.  
   Although, it is important to mention here that [RFC2544] was never 
   meant to be used outside a lab environment.  
   
   It is not possible to make an accurate TCP Throughput measurement 
   when the network is dysfunctional. In particular, if the network is 
   exhibiting high packet loss and/or high jitter, then TCP Layer 
   Throughput testing will not be meaningful. As a guideline 5% packet
   loss and/or 150 ms of jitter may be considered too high for an
   accurate measurement.

   TCP Throughput testing may require cooperation between the end-user 
   customer and the network provider.  As an example, in an MPLS (Multi-
   Protocol Label Switching) network architecture, the testing SHOULD be 
   conducted either on the CPE or on the CE device and not on the PE 
   (Provider Edge) router.  

   The following represents the sequential order of steps for this 
   testing methodology:

   1 - Identify the Path MTU.  Packetization Layer Path MTU Discovery 
   or PLPMTUD, [RFC4821], SHOULD be conducted.  It is important to
   identify the path MTU so that the TCP TTD is configured properly to 
   avoid fragmentation.

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   2 - Baseline Round Trip Time and Bandwidth. This step establishes the 
   inherent, non-congested Round Trip Time (RTT) and the Bottleneck 
   Bandwidth (BB) of the end-to-end network path.  These measurements 
   are used to provide estimates of the TCP RWND and Send Socket Buffer 
   Sizes that SHOULD be used during subsequent test steps.

   3 - TCP Connection Throughput Tests.  With baseline measurements
   of Round Trip Time and Bottleneck Bandwidth, single and multiple TCP
   connection throughput tests SHOULD be conducted to baseline network 
   performances.  
   
   These three (3) steps are detailed in Sections 3.1 - 3.3.

   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 a dedicated communications test instrument. 
   In both cases, it MUST be capable of emulating both a client and a
   server.  

   The following criteria SHOULD be considered when selecting whether 
   the TCP test host can be a standard computer or has to be a dedicated 
   communications test instrument: 

   - TCP implementation used by the test host, OS version, i.e. LINUX OS 
   kernel using TCP New Reno, TCP options supported, etc.  These will 
   obviously be more important when using dedicated communications test 
   instruments 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.

   - More important, the TCP test host MUST be capable to generate 
   and receive stateful TCP test traffic at the full BB of the NUT.
   Stateful TCP test traffic means that the test host MUST fully 
   implement a TCP/IP stack; this is generally a comment aimed at 
   dedicated communications test equipments which sometimes "blast" 
   packets with TCP headers. As a general rule of thumb, testing TCP 
   Throughput at rates greater than 100 Mbps may require high 
   performance server hardware or dedicated hardware based test tools.  

   - A compliant TCP Throughput Test Device MUST allow adjusting both 
   Send and Receive Socket Buffer sizes.  The Socket Buffers MUST be 
   large enough to fill the BDP.

   - Measuring RTT and retransmissions per connection will generally 
   require a dedicated communications test instrument. In the absence of 
   dedicated hardware based test tools, these measurements may need to 
   be conducted with packet capture tools, i.e. conduct TCP Throughput 
   tests and analyze RTT and retransmissions in packet captures.  
   Another option MAY be to use "TCP Extended Statistics MIB" per 
   [RFC4898].

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   - The [RFC4821] PLPMTUD test SHOULD be conducted with a dedicated 
   tester which exposes the ability to run the PLPMTUD algorithm 
   independently from the OS stack.
  
3.1. 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 (MTU) of 
   the next hop, 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 adjust itself.   Unfortunately, because many network managers 
   completely disable ICMP, this technique does not always prove 
   reliable.

   Packetization Layer Path MTU Discovery or PLPMTUD [RFC4821] MUST then
   be conducted to verify the network path MTU.  PLPMTUD can be used 
   with or without ICMP.  [RFC4821] specifies search_high and search_low 
   parameters for the MTU and we recommend to use those.  The goal is to
   avoid fragmentation during all subsequent tests.   
   
3.2. Round Trip Time (RTT) and Bottleneck Bandwidth (BB)

   Before stateful TCP testing can begin, it is important to determine 
   the baseline RTT (i.e. non-congested inherent delay) and BB of the 
   end-to-end network to be tested.   These measurements are used to 
   calculate the BDP and to provide estimates of the TCP RWND and 
   Send Socket Buffer Sizes that SHOULD be used in subsequent test 
   steps. 

3.2.1 Measuring RTT

   As previously defined in Section 1.1, RTT is the elapsed time 
   between the clocking in of the first bit of a TCP segment sent
   and the receipt of the last bit of the corresponding TCP 
   Acknowledgment.  

   The RTT SHOULD be baselined during off-peak hours in order to obtain 
   a reliable figure of the inherent network latency.  Otherwise,  
   additional delay caused by network buffering can occur.  Also, when 
   sampling RTT values over a given test interval, the minimum 
   measured value SHOULD be used as the baseline RTT.  This will most 
   closely estimate the real inherent RTT.  This value is also used to 
   determine the Buffer Delay Percentage metric defined in Section 4.3.

   The following list is not meant to be exhaustive,  although it 
   summarizes some of the most common ways to determine Round Trip Time. 
   The desired measurement precision (i.e. ms versus us) may dictate 
   whether the RTT measurement can be achieved with ICMP pings or by a 
   dedicated communications test instrument with precision timers.  The 
   objective in this section is to list several techniques in order of 
   decreasing accuracy.

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   - Use test equipment on each end of the network, "looping" the 
   far-end tester so that a packet stream can be measured back and forth
   from end-to-end. This RTT measurement may be compatible with delay 
   measurement protocols specified in [RFC5357].

   - Conduct packet captures of TCP test sessions using "iperf" or FTP, 
   or other TCP test applications.   By running multiple experiments, 
   packet captures can then be analyzed to estimate RTT.  It is 
   important to note that results based upon the SYN -> SYN-ACK at the 
   beginning of TCP sessions SHOULD be avoided since Firewalls might 
   slow down 3 way handshakes.  Also, at the senders side, Ostermann's
   LINUX TCPTRACE utility with -l -r arguments can be used to extract
   the RTT results directly from the packet captures.     

   - Obtain RTT statistics available from MIBs defined in [RFC4898].

   - ICMP pings may also be adequate to provide Round Trip Time 
   estimates, provided that the packet size is factored into the 
   estimates (i.e. pings with different packet sizes might be required).
   Some limitations with ICMP Ping may include ms resolution and 
   whether the network elements are responding to pings or not.  Also, 
   ICMP is often rate-limited or segregated into different buffer 
   queues.   ICMP might not work if QoS (Quality of Service) 
   reclassification is done at any hop.   ICMP is not as reliable and 
   accurate as in-band measurements.
   
3.2.2 Measuring BB

   Before any TCP Throughput test can be conducted, bandwidth 
   measurement tests SHOULD be run with stateless IP streams (i.e. not 
   stateful TCP) in order to determine the BB of the NUT.     
   These measurements SHOULD be conducted in both directions, 
   especially in asymmetrical access networks (e.g. ADSL access). These
   tests SHOULD be performed at various intervals throughout a business
   day or even across a week.  
   
   Testing at various time intervals would provide a better 
   characterization of TCP throughput and better diagnosis insight (for
   cases where there are TCP performance issues).  The bandwidth tests
   SHOULD produce logged outputs of the achieved bandwidths across the
   complete test duration.
   
   There are many well established techniques available to provide 
   estimated measures of bandwidth over a network.  It is a common 
   practice for network providers to conduct Layer 2/3 bandwidth 
   capacity tests using [RFC2544], although it is understood that 
   [RFC2544] was never meant to be used outside a lab environment. 
   These bandwidth measurements SHOULD use network capacity 
   techniques as defined in [RFC5136].

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3.3. Measuring TCP Throughput

   This methodology specifically defines TCP Throughput measurement
   techniques to verify maximum achievable TCP performance in a managed 
   business class IP network.

   With baseline measurements of RTT and BB from Section 3.2, a series 
   of single and / or multiple TCP connection throughput tests SHOULD 
   be conducted.  
   
   The number of trials and single versus multiple TCP connections 
   choice will be based on the intention of the test.  A single TCP 
   connection test might be enough to measure the achievable throughput 
   of a Metro Ethernet connectivity.  Although, it is important to note 
   that various traffic management techniques can be used in an IP 
   network and that some of those can only be tested with multiple 
   connections.  As an example, multiple TCP sessions might be required 
   to detect traffic shaping versus policing.  Multiple sessions might 
   also be needed to measure Active Queue Management performances.  
   However, traffic management testing is not within the scope of this 
   test methodology.

   In all circumstances, it is RECOMMENDED to run the tests in each 
   direction independently first and then to run in both directions 
   simultaneously.  It is also RECOMMENDED to run the tests at 
   different times of day.
   
   In each case, the TCP Transfer Time Ratio, the TCP Efficiency 
   Percentage, and the Buffer Delay Percentage MUST be measured in 
   each direction.  These 3 metrics are defined in Section 4.

3.3.1 Minimum TCP RWND

   The TCP TTD MUST allow the Send Socket Buffer and Receive Window 
   sizes to be set higher than the BDP, otherwise TCP performance will 
   be limited. In the business customer environment, these settings are
   not generally adjustable by the average user.  These settings are 
   either hard coded in the application or configured within the OS as 
   part of a corporate image. In many cases, the user's host Send 
   Socket Buffer and Receive Window size settings are not optimal.

   This section provides derivations of BDPs under various network 
   conditions.  It also provides examples of achievable TCP Throughput 
   with various TCP RWND sizes.  This provides important guidelines 
   showing what can be achieved with settings higher than the BDP, 
   versus what would be achieved in a variety of real world conditions.

   The minimum required TCP RWND Size can be calculated from the 
   Bandwidth Delay Product (BDP), which is:

   BDP (bits) = RTT (sec) x BB (bps)
   Note that the RTT is being used as the "Delay" variable for the BDP. 

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   Then, by dividing the BDP by 8, we obtain the minimum required TCP 
   RWND Size in Bytes.  For optimal results, the Send Socket Buffer 
   MUST be adjusted to the same value at each end of the network.
   
   Minimum required TCP RWND = BDP / 8 

   As an example on a T3 link with 25 ms RTT, the BDP would equal
   ~1,105,000 bits and the minimum required TCP RWND would be ~138 KB.
   
   Note that separate calculations are REQUIRED on asymmetrical paths.  
   An asymmetrical path example would be a 90 ms RTT ADSL line with 
   5Mbps downstream and 640Kbps upstream. The downstream BDP would equal 
   ~450,000 bits while the upstream one would be only ~57,600 bits.

   The following table provides some representative network Link Speeds,
   RTT, BDP, and their associated minimum required TCP RWND Sizes.     

   Table 3.3.1: Link Speed, RTT, calculated BDP & min. TCP RWND

      Link                                         Minimum required 
      Speed*         RTT              BDP             TCP RWND
      (Mbps)         (ms)            (bits)           (KBytes)
   ---------------------------------------------------------------------
        1.536        20.00           30,720              3.84
        1.536        50.00           76,800              9.60
        1.536       100.00          153,600             19.20
       44.210        10.00          442,100             55.26
       44.210        15.00          663,150             82.89
       44.210        25.00        1,105,250            138.16
      100.000         1.00          100,000             12.50
      100.000         2.00          200,000             25.00
      100.000         5.00          500,000             62.50
    1,000.000         0.10          100,000             12.50
    1,000.000         0.50          500,000             62.50
    1,000.000         1.00        1,000,000            125.00
   10,000.000         0.05          500,000             62.50
   10,000.000         0.30        3,000,000            375.00

   * Note that link speed is the BB for the NUT
   In the above table, the following serial link speeds are used:
   - T1 = 1.536 Mbps (for a B8ZS line encoding facility)
   - T3 = 44.21 Mbps (for a C-Bit Framing facility)

   The previous table illustrates the minimum required TCP RWND. 
   If a smaller TCP RWND Size is used, then the TCP Throughput
   can not be optimal. To calculate the TCP Throughput, the following 
   formula is used: TCP Throughput = TCP RWND X 8 / RTT

   An example could be a 100 Mbps IP path with 5 ms RTT and a TCP RWND 
   of 16KB, then:

   TCP Throughput = 16 KBytes X 8 bits / 5 ms.
   TCP Throughput = 128,000 bits / 0.005 sec.
   TCP Throughput = 25.6 Mbps.

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   Another example for a T3 using the same calculation formula is 
   illustrated in Figure 3.3.1a:
   
   TCP Throughput = 16 KBytes X 8 bits / 10 ms.
   TCP Throughput = 128,000 bits / 0.01 sec.
   TCP Throughput = 12.8 Mbps. *

   
   When the TCP RWND Size exceeds the BDP (T3 link and 64 KBytes TCP 
   RWND on a 10 ms RTT path), the maximum frames per second limit of 
   3664 is reached and then the formula is:

   
   TCP Throughput = Max FPS X (MTU - 40) X 8.
   TCP Throughput = 3664 FPS X 1460 Bytes X 8 bits.
   TCP Throughput = 42.8 Mbps. **   

   
   The following diagram compares achievable TCP Throughputs on a T3 
   with Send Socket Buffer & TCP RWND Sizes of 16KB vs. 64KB.

   
   Figure 3.3.1a TCP Throughputs on a T3 at different RTTs   

   
           45|
             |           _______**42.8
           40|           |64KB |
TCP          |           |     |
Throughput 35|           |     |                     
in Mbps      |           |     |          +-----+34.1  
           30|           |     |          |64KB |
             |           |     |          |     |
           25|           |     |          |     |
             |           |     |          |     |
           20|           |     |          |     |          _______20.5
             |           |     |          |     |          |64KB |
           15|           |     |          |     |          |     |
             |*12.8+-----|     |          |     |          |     |
           10|     |16KB |     |          |     |          |     |
             |     |     |     |8.5 +-----|     |          |     |
            5|     |     |     |    |16KB |     |5.1 +-----|     |       
             |_____|_____|_____|____|_____|_____|____|16KB |_____|_____
                        10               15               25
                                RTT in milliseconds

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   The following diagram shows the achievable TCP Throughput on a 25 ms 
   T3 when Send Socket Buffer & TCP RWND Sizes are increased.
   
   Figure 3.3.1b TCP Throughputs on a T3 with different TCP RWND

           45|
             |
           40|                                             +-----+40.9
TCP          |                                             |     |
Throughput 35|                                             |     |
in Mbps      |                                             |     |
           30|                                             |     |
             |                                             |     |
           25|                                             |     |
             |                                             |     |
           20|                               +-----+20.5   |     |
             |                               |     |       |     |
           15|                               |     |       |     |
             |                               |     |       |     |
           10|                  +-----+10.2  |     |       |     |
             |                  |     |      |     |       |     |
            5|     +-----+5.1   |     |      |     |       |     |
             |_____|_____|______|_____|______|_____|_______|_____|_____
                     16           32           64            128*
                          TCP RWND Size in KBytes
                                                  
   * Note that 128KB requires [RFC1323] TCP Window scaling option.

4. TCP Metrics

   This methodology focuses on a TCP Throughput and provides 3 basic 
   metrics that can be used for better understanding of the results.  
   It is recognized that the complexity and unpredictability of TCP 
   makes it very difficult to develop a complete set of metrics that 
   accounts for the myriad of variables (i.e. RTT variations, loss 
   conditions, TCP implementations, etc.).  However, these 3 metrics 
   facilitate TCP Throughput comparisons under varying network 
   conditions and host buffer size / RWND settings.
   
4.1 Transfer Time Ratio

   The first metric is the TCP Transfer Time Ratio, which is simply the 
   ratio between the Actual versus the Ideal TCP Transfer Times. 
   
   The Actual TCP Transfer Time, is simply the time it takes to transfer
   a block of data across TCP connection(s).  
   
   The Ideal TCP Transfer Time is the predicted time for which a block 
   of data SHOULD transfer across TCP connection(s) considering the BB 
   of the NUT.

                              Actual TCP Transfer Time
   TCP Transfer Time Ratio =  -------------------------
                              Ideal TCP Transfer Time

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   The Ideal TCP Transfer Time is derived from the Maximum Achievable
   TCP Throughput, which is related to the BB and Layer 1/2/3/4 
   overheads associated with the network path.  The following sections 
   provide derivations for the Maximum Achievable TCP Throughput and 
   example calculations for the TCP Transfer Time Ratio.

4.1.1 Maximum Achievable TCP Throughput calculation

   This section provides formulas to calculate the Maximum Achievable 
   TCP Throughput with examples for T3 (44.21 Mbps) and Ethernet. 
   
   All calculations are based on IP version 4 with TCP/IP headers of 
   20 Bytes each (20 for TCP + 20 for IP) within an MTU of 1500 Bytes.  

   First, the maximum achievable Layer 2 throughput of a T3 Interface 
   is limited by the maximum quantity of Frames Per Second (FPS) 
   permitted by the actual physical layer (Layer 1) speed.

   The calculation formula is:
   FPS = T3 Physical Speed / ((MTU + PPP + Flags + CRC16) X 8)
   FPS = (44.21Mbps /((1500 Bytes + 4 Bytes + 2 Bytes + 2 Bytes) X 8 )))
   FPS = (44.21Mbps /(1508 Bytes X 8))  
   FPS = 44.21Mbps / 12064 bits
   FPS = 3664
  
   Then, to obtain the Maximum Achievable TCP Throughput (Layer 4), we
   simply use: (MTU - 40) in Bytes X 8 bits X max FPS. 

   For a T3, the maximum TCP Throughput = 1460 Bytes X 8 bits X 3664 FPS
   Maximum TCP Throughput = 11680 bits X 3664 FPS
   Maximum TCP Throughput = 42.8 Mbps.

   On Ethernet, the maximum achievable Layer 2 throughput is limited by 
   the maximum Frames Per Second permitted by the IEEE802.3 standard.

   The maximum FPS for 100 Mbps Ethernet is 8127 and the calculation is:
   FPS = (100Mbps /(1538 Bytes X 8 bits))
   
   The maximum FPS for GigE is 81274 and the calculation formula is:
   FPS = (1Gbps /(1538 Bytes X 8 bits))
   
   The maximum FPS for 10GigE is 812743 and the calculation formula is: 
   FPS = (10Gbps /(1538 Bytes X 8 bits))
   
   The 1538 Bytes equates to:

   MTU + Ethernet + CRC32 + IFG + Preamble + SFD
        (IFG = Inter-Frame Gap and SFD = Start of Frame Delimiter)
   Where MTU is 1500 Bytes, Ethernet is 14 Bytes, CRC32 is 4 Bytes, 
   IFG is 12 Bytes, Preamble is 7 Bytes and SFD is 1 Byte.

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   Then, to obtain the Maximum Achievable TCP Throughput (Layer 4), we
   simply use: (MTU - 40) in Bytes X 8 bits X max FPS. 
   For a 100Mbps, the max TCP Throughput = 1460Bytes X 8 bits X 8127 FPS
   Maximum TCP Throughput = 11680 bits X 8127 FPS
   Maximum TCP Throughput = 94.9 Mbps.

   It is important to note that better results could be obtained with 
   jumbo frames on Gigabit and 10 Gigabit Ethernet interfaces.  

4.1.2 TCP Transfer Time and Transfer Time Ratio calculation

   The following table illustrates the Ideal TCP Transfer time of a 
   single TCP connection when its TCP RWND and Send Socket Buffer Sizes 
   equals or exceeds the BDP. 

   Table 4.1.1: Link Speed, RTT, BDP, TCP Throughput, and 
                Ideal TCP Transfer time for a 100 MB File

       Link                             Maximum            Ideal TCP
       Speed                   BDP      Achievable TCP     Transfer time
       (Mbps)     RTT (ms)   (KBytes)   Throughput(Mbps)   (seconds)*
   --------------------------------------------------------------------
         1.536    50.00         9.6            1.4             571.0
        44.210    25.00       138.2           42.8              18.0 
       100.000     2.00        25.0           94.9               9.0
     1,000.000     1.00       125.0          949.2               1.0
    10,000.000     0.05        62.5        9,492.0               0.1

    * Transfer times are rounded for simplicity.

   For a 100MB file (100 x 8 = 800 Mbits), the Ideal TCP Transfer Time 
   is derived as follows:

                                           800 Mbits
       Ideal TCP Transfer Time = -----------------------------------
                                  Maximum Achievable TCP Throughput

   To illustrate the TCP Transfer Time Ratio, an example would be the 
   bulk transfer of 100 MB over 5 simultaneous TCP connections  (each 
   connection transferring 100 MB).  In this example, the Ethernet 
   service provides a Committed Access Rate (CAR) of 500 Mbps.  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 Time 
   Ratio would then be 12/8 = 1.5, which indicates that the transfer 
   across all connections took 1.5 times longer than the ideal.

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4.2 TCP Efficiency

   The second metric represents the percentage of Bytes that were not
   retransmitted.

                       Transmitted Bytes - Retransmitted Bytes
   TCP Efficiency % =  ---------------------------------------  X 100
                                Transmitted Bytes

   Transmitted Bytes are the total number of TCP Bytes to be transmitted 
   including the original and the retransmitted Bytes.      

4.2.1 TCP Efficiency Percentage calculation

   As an example, if 100,000 Bytes were sent and 2,000 had to be 
   retransmitted, the TCP Efficiency Percentage would be calculated as:

                        102,000 - 2,000
   TCP Efficiency % =  -----------------  x 100 = 98.03%
                          102,000

   Note that the Retransmitted Bytes may have occurred more than once,
   if so, then these multiple retransmissions are added to the 
   Retransmitted Bytes and to the Transmitted Bytes counts.

4.3 Buffer Delay
  
   The third metric is the Buffer Delay Percentage, which represents 
   the increase in RTT during a TCP Throughput test versus the inherent 
   or baseline RTT.  The baseline RTT is the Round Trip Time inherent to 
   the network path under non-congested conditions as defined in Section
   3.2.1.  The average RTT is derived from the total of all measured 
   RTTs during the actual test at every second divided by the test 
   duration in seconds.
   
                                      Total RTTs during transfer 
      Average RTT during transfer = -----------------------------
                                     Transfer duration in seconds
   
     
                     Average RTT during Transfer - Baseline RTT
    Buffer Delay % = ------------------------------------------ X 100
                                 Baseline RTT    

4.3.1 Buffer Delay calculation  

   As an example, consider a network path with a baseline RTT of 25 ms. 
   During the course of a TCP transfer, the average RTT across 
   the entire transfer increases to 32 ms.  Then, the Buffer Delay 
   Percentage would be calculated as:

                     32 - 25
    Buffer Delay % = ------- x 100 = 28%
                       25

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   Note that the TCP Transfer Time Ratio, TCP Efficiency Percentage, and 
   the Buffer Delay Percentage MUST all be measured during each 
   throughput test.  Poor TCP Transfer Time Ratio (i.e. TCP Transfer 
   Time greater than the Ideal TCP Transfer Time) may be diagnosed by 
   correlating with sub-optimal TCP Efficiency Percentage and/or Buffer 
   Delay Percentage metrics.  

5. Conducting TCP Throughput Tests

   Several TCP tools are currently used in the network world and one of 
   the most common is "iperf".  With this tool, hosts are installed at 
   each end of the network path; one acts as client and the other as 
   a server.  The Send Socket Buffer and the TCP RWND Sizes of both 
   client and server can be manually set.  The achieved throughput can 
   then be measured, either uni-directionally or bi-directionally.  For 
   higher BDP situations in lossy networks (Long Fat Networks (LFNs) or 
   satellite links, etc.), TCP options such as Selective Acknowledgment 
   SHOULD become part of the window size / throughput characterization.  
   
   Host hardware performance must be well understood before conducting
   the tests described in the following sections.  A dedicated 
   communications test instrument will generally be REQUIRED, especially 
   for line rates of GigE and 10 GigE.  A compliant TCP TTD SHOULD 
   provide a warning message when the expected test throughput will 
   exceed the subscribed customer SLA.  If the throughput test is 
   expected to exceed the subscribed customer SLA, then the test 
   SHOULD be coordinated with the network provider.

   The TCP Throughput test SHOULD be run over a long enough duration 
   to properly exercise network buffers (i.e. greater than 30 seconds) 
   and SHOULD also characterize performance at different times of day.  

5.1 Single versus Multiple TCP Connections

   The decision whether to conduct single or multiple TCP connection 
   tests depends upon the size of the BDP in relation to the TCP RWND 
   configured in the end-user environment. For example, if the BDP for 
   a Long Fat Network (LFN) turns out to be 2MB, then it is probably 
   more realistic to test this network path with multiple connections.  
   Assuming typical host TCP RWND Sizes of 64 KB (i.e. Windows XP), 
   using 32 TCP connections would emulate a small office scenario.

   The following table is provided to illustrate the relationship
   between the TCP RWND and the number of TCP connections required to 
   fill the available capacity of a given BDP. For this example, the 
   network bandwidth is 500 Mbps and the RTT is 5 ms, then the BDP 
   equates to 312.5 KBytes.

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   Table 5.1 Number of TCP connections versus TCP RWND
   
                 Number of TCP Connections
      TCP RWND   to fill available bandwidth 
     -------------------------------------
       16KB             20
       32KB             10
       64KB              5
      128KB              3

   The TCP Transfer Time Ratio metric is useful when conducting multiple
   connection tests.  Each connection SHOULD be configured to transfer
   payloads of the same size (i.e. 100 MB), then the TCP Transfer Time 
   Ratio provides a simple metric to verify the actual versus expected 
   results.

   Note that the TCP Transfer Time is the time required for each 
   connection to complete the transfer of the predetermined payload 
   size.  From the previous table, the 64KB window is considered.  Each 
   of the 5 TCP connections would be configured to transfer 100MB, and 
   each one should obtain a maximum of 100 Mbps.  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 
   under these conditions).

   Additionally, the TCP Efficiency Percentage metric MUST be computed 
   for each connection as defined in Section 4.2. 

5.2 Results Interpretation

   At the end, a TCP Throughput Test Device (TCP TTD) SHOULD generate a
   report with the calculated BDP and a set of Window Size experiments.
   Window Size refers to the minimum of the Send Socket Buffer and TCP 
   RWND.  The report SHOULD include TCP Throughput results for each TCP 
   Window Size tested.  The goal is to provide clear achievable versus 
   actual TCP Throughputs results with respect to the TCP Window Size 
   when no fragmentation occurs.  The report SHOULD also include the 
   results for the 3 metrics defined in Section 4. The goal is to 
   provide a clear relationship between these 3 metrics and user 
   experience.  As an example, for the same results in regards with 
   Transfer Time Ratio, a better TCP Efficiency could be obtained at the
   cost of higher Buffer Delays.   
      
   For cases where the test results are not equal to the ideal values, 
   some possible causes are:

   - Network congestion causing packet loss which may be inferred from 
   a poor TCP Efficiency % (i.e., higher TCP Efficiency % = less packet 
   loss)
   
   - Network congestion causing an increase in RTT which may be inferred
   from the Buffer Delay Percentage (i.e., 0% = no increase in RTT over
   baseline)

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   - Intermediate network devices which actively regenerate the TCP
   connection and can alter TCP RWND Size, MTU, etc.
   
   - Rate limiting by policing instead of shaping. 

   - Maximum TCP Buffer space.  All operating systems have a global 
   mechanism to limit the quantity of system memory to be used by TCP 
   connections. On some systems, each connection is subject to a memory 
   limit that is applied to the total memory used for input data, output 
   data and controls. On other systems, there are separate limits for 
   input and output buffer spaces per connection.  Client/server IP 
   hosts might be configured with Maximum Buffer Space limits that are 
   far too small for high performance networks.

   - Socket Buffer Sizes.  Most operating systems support separate per 
   connection send and receive buffer limits that can be adjusted as 
   long as they stay within the maximum memory limits.  These socket 
   buffers MUST be large enough to hold a full BDP of TCP Bytes plus
   some overhead.  There are several methods that can be used to adjust
   socket buffer sizes, but TCP Auto-Tuning automatically adjusts these
   as needed to optimally balance TCP performance and memory usage.
   
   It is important to note that Auto-Tuning is enabled by default in
   LINUX since the kernel release 2.6.6 and in UNIX since FreeBSD 7.0.
   It is also enabled by default in Windows since Vista and in MAC since
   OS X version 10.5 (leopard).  Over buffering can cause some
   applications to behave poorly, typically causing sluggish interactive
   response and risk running the system out of memory.   Large default
   socket buffers have to be considered carefully on multi-user systems.

   - TCP Window Scale Option, [RFC1323].  This option enables TCP to 
   support large BDP paths.  It provides a scale factor which is 
   required for TCP to support window sizes larger than 64KB. Most 
   systems automatically request WSCALE under some conditions, such as 
   when the receive socket buffer is larger than 64KB or when the other 
   end of the TCP connection requests it first.  WSCALE can only be 
   negotiated during the 3 way handshake.  If either end fails to 
   request WSCALE or requests an insufficient value, it cannot be 
   renegotiated. Different systems use different algorithms to select
   WSCALE, but it is very important to have large enough buffer
   sizes.  Note that under these constraints, a client application 
   wishing to send data at high rates may need to set its own receive 
   buffer to something larger than 64K Bytes before it opens the 
   connection to ensure that the server properly negotiates WSCALE.  
   A system administrator might have to explicitly enable [RFC1323] 
   extensions.  Otherwise, the client/server IP host would not support 
   TCP window sizes (BDP) larger than 64KB.  Most of the time, 
   performance gains will be obtained by enabling this option in LFNs.

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   - TCP Timestamps Option, [RFC1323].  This feature provides better 
   measurements of the Round Trip Time and protects TCP from data 
   corruption that might occur if packets are delivered so late that the
   sequence numbers wrap before they are delivered.  Wrapped sequence 
   numbers do not pose a serious risk below 100 Mbps, but the risk 
   increases at higher data rates. Most of the time, performance gains
   will be obtained by enabling this option in Gigabit bandwidth
   networks.

   - TCP Selective Acknowledgments Option (SACK), [RFC2018]. This allows
   a TCP receiver to inform the sender about exactly which data segment 
   is missing and needs to be retransmitted.  Without SACK, TCP has to 
   estimate which data segment is missing, which works just fine if all 
   losses are isolated (i.e. only one loss in any given round trip).  
   Without SACK, TCP takes a very long time to recover after multiple 
   and consecutive losses.  SACK is now supported by most operating 
   systems, but it may have to be explicitly enabled by the system 
   administrator. In networks with unknown load and error patterns, TCP 
   SACK will improve throughput performances.  On the other hand, 
   security appliances vendors might have implemented TCP randomization 
   without considering TCP SACK and under such circumstances, SACK might 
   need to be disabled in the client/server IP hosts until the vendor 
   corrects the issue.  Also, poorly implemented SACK algorithms might
   cause extreme CPU loads and might need to be disabled.

   - Path MTU.  The client/server IP host system SHOULD use the largest
   possible MTU for the path.  This may require enabling Path MTU
   Discovery [RFC1191] & [RFC4821].  Since [RFC1191] is flawed, it is
   sometimes not enabled by default and may need to be explicitly
   enabled by the system administrator. [RFC4821] describes a new, more
   robust algorithm for MTU discovery and ICMP black hole recovery.

   - TOE (TCP Offload Engine). Some recent Network Interface Cards (NIC)
   are equipped with drivers that can do part or all of the TCP/IP 
   protocol processing.  TOE implementations require additional work 
   (i.e. hardware-specific socket manipulation) to set up and tear down 
   connections.  Because TOE NICs configuration parameters are vendor 
   specific and not necessarily RFC-compliant,  they are poorly 
   integrated with UNIX & LINUX.  Occasionally, TOE might need to be 
   disabled in a server because its NIC does not have enough memory 
   resources to buffer thousands of connections.
      
   Note that both ends of a TCP connection MUST be properly tuned.

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6. Security Considerations

   Measuring TCP network performance raises security concerns.  Metrics 
   produced within this framework may create security issues.

6.1 Denial of Service Attacks

   TCP network performance metrics, as defined in this document attempts 
   to fill the NUT with a stateful connection.  However, since the test 
   MAY use stateless IP streams as specified in Section 3.2.2, it might 
   appear to network operators as a Denial Of Service attack. Thus, as 
   mentioned at the beginning of section 3, TCP Throughput testing may 
   require cooperation between the end-user customer and the network 
   provider. 

6.2 User data confidentiality

   Metrics within this framework generate packets from a sample, rather
   than taking samples based on user data.  Thus, our framework does not
   threaten user data confidentiality.

6.3 Interference with metrics

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

7. IANA Considerations

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

8. Acknowledgments

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

9. References

9.1 Normative References

   [RFC1191]  Mogul, A., Deering, S., "Path MTU Discovery", 1990

   [RFC1323]  Jacobson, V., Braden, R., Borman D., "TCP Extensions for 
              High Performance", May 1992
                          
   [RFC2018]  Mathis, M., Mahdavi, J., Floyd, S., Romanow, A., "TCP 
              Selective Acknowledgment Options", 1996

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

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   [RFC2544]  Bradner, S., McQuaid, J., "Benchmarking Methodology for 
              Network Interconnect Devices", RFC 2544, June 1999  
   
   [RFC4656]  Shalunov, S., Teitelbaum, B., Karp, A., Boote, J., and M.
              Zekauskas, "A One-way Active Measurement Protocol
              (OWAMP)", RFC 4656, September 2006.

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

   [RFC4898]  Mathis, M., Heffner, J., Raghunarayan, R., "TCP Extended 
              Statistics MIB", May 2007
                          
   [RFC5136]  Chimento P., Ishac, J., "Defining Network Capacity", 
              February 2008 
                          
   [RFC5357]  Hedayat, K., Krzanowski, R., Morton, A., Yum, K., Babiarz,
              J., "A Two-Way Active Measurement Protocol (TWAMP)", 
              RFC 5357, October 2008

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

9.2. Informative References

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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

   Ruediger Geib
   Heinrich-Hertz-Strasse (Number: 3-7)
   Darmstadt, Germany, 64295

   Phone: +49 6151 6282747
   Ruediger.Geib@telekom.de

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

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