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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: July 31, 2011                     Bell Canada (Ext. Consultant)
                                                            Rudiger Geib
                                                        Deutsche Telekom
                                                        Reinhard Schrage
                                                      Schrage Consulting

                                                        January 31, 2011



                  Framework for TCP Throughput Testing
                draft-ietf-ippm-tcp-throughput-tm-11.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 and should be configured as
   recommended.

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 July 31, 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   Test Set-up  . . . . . . . . . . . . . . . . . . . . . . .  5
   2.  Scope and Goals of this methodology. . . . . . . . . . . . . .  5
     2.1   TCP Equilibrium. . . . . . . . . . . . . . . . . . . . . .  6
   3.  TCP Throughput Testing Methodology . . . . . . . . . . . . . .  7
     3.1   Determine Network Path MTU . . . . . . . . . . . . . . . .  9
     3.2.  Baseline Round Trip Time and Bandwidth . . . . . . . . . . 10
         3.2.1  Techniques to Measure Round Trip Time . . . . . . . . 11
         3.2.2  Techniques to Measure end-to-end Bandwidth. . . . . . 12
     3.3.  TCP Throughput Tests . . . . . . . . . . . . . . . . . . . 12
         3.3.1 Calculate minimum required TCP RWND Size. . . . .  . . 12
         3.3.2 Metrics for TCP Throughput Tests . . . . . . . . . . . 15
         3.3.3 Conducting the TCP Throughput Tests. . . . . . . . . . 19
         3.3.4 Single vs. Multiple TCP Connection Testing . . . . . . 19
         3.3.5 Interpretation of the TCP Throughput Results . . . . . 20
         3.3.6 High Performance Network Options . . . . . . . . . . . 20
     3.4. Traffic Management Tests .  . . . . . . . . . . . . . . . . 22
         3.4.1 Traffic Shaping Tests. . . . . . . . . . . . . . . . . 23
          3.4.1.1 Interpretation of Traffic Shaping Test Results. . . 23
         3.4.2 AQM Tests. . . . . . . . . . . . . . . . . . . . . . . 24
          3.4.2.1 Interpretation of AQM Results . . . . . . . . . . . 25
   4.  Security Considerations  . . . . . . . . . . . . . . . . . . . 26
   5.  IANA Considerations  . . . . . . . . . . . . . . . . . . . . . 26
   6.  Acknowledgments  . . . . . . . . . . . . . . . . . . . . . . . 26
   7.  References . . . . . . . . . . . . . . . . . . . . . . . . . . 26
     7.1   Normative References . . . . . . . . . . . . . . . . . . . 26
     7.2   Informative References . . . . . . . . . . . . . . . . . . 27

   Authors' Addresses . . . . . . . . . . . . . . . . . . . . . . . . 27





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1. Introduction

   The SLA (Service Level Agreement) provided to business class
   customers is generally based upon Layer 2/3 criteria such as :
   Guaranteed bandwidth, maximum network latency, maximum packet loss
   percentage and maximum delay variation (i.e. maximum 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 performance, measuring TCP throughput
   provides more 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; i.e. 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 2.1
   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 path bandwidth, 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)
   - Path MSS (Maximum Segment Size)

   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: RFC 2544 was never meant to be used outside a lab environment.

   Sections 2 and 3 of this document provide a general overview of the
   proposed methodology.


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), refers to Layer 4 back and forth delay.


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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 Test Set-up

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

2. Scope and Goals of this Methodology

   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
   initial 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 a results
   interpretation section 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].

   - This methodology is not intended to measure TCP throughput as part
   of an SLA, or to compare the TCP performance between service
   providers or to compare between implementations of this methodology
   in dedicated communications test instruments.

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   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 MSS (Maximum Segment Size) and Socket Buffer sizes) and how
   these affect the outcome of TCP performances over an IP network.
   See section 3.3.3.

   - Provide specific test conditions like link speed, RTT, MSS, 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 2.1 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 3.3.2.

   - In test situations where the recommended procedure does not yield
   the maximum achievable TCP throughput, this methodology provides
   some possible 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 3.3.5.

2.1 TCP Equilibrium

   TCP connections have three (3) fundamental congestion window phases:

   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.




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   The following diagram depicts these 3 phases.

   Figure 2.1 TCP CWND Phases

        /\  |                                                TCP
        /\  |                                                Equilibrium
        /\  |High ssthresh  TCP CWND
        /\  |Loss Event *   halving    3-Loss Recovery
        /\  |          * \  upon loss                         Adjusted
        /\  |          *  \    /  \        Time-Out           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.

   A well tuned and managed IP network with appropriate TCP adjustments
   in the IP hosts and applications should perform very close to the
   BB (Bottleneck Bandwidth) 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 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.
   Implications to the capabilities of the Throughput Test Device (TTD)
   are covered at the end of section 3.

3. TCP Throughput Testing Methodology

   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 RFC 2544 was never
   meant to be used outside a lab environment.

   If the network is not performing properly in terms of packet loss,
   jitter, etc. then the TCP layer testing will not be meaningful.  A
   dysfunctional network will not achieve optimal TCP throughputs in
   regards with the available bandwidth.

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

   2. Baseline Round Trip Time and Bandwidth. This step establishes the
   inherent, non-congested Round Trip Time (RTT) and the Bottleneck
   Bandwidth of the end-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.   These
   measurements refers to [RFC2681] and [RFC4898] in order to measure
   RTD and associated RTT.

   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.

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

   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 MUST be identified in the test results.
   The compliant TCP TTD should be usable for complete end-to-end
   testing through network security elements and should also be usable
   for testing network sections.


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   - More important, the TCP test host MUST be capable to generate
   and receive stateful TCP test traffic at the full link speed of the
   network under test. 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 Mbit/sec 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].

   - 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. 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, 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.

   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. The following sections provide a summary of the
   PLPMTUD approach and an example using TCP. [RFC4821] specifies a
   search_high and a search_low parameter for the MTU.  As specified in
   [RFC4821], 1024 Bytes is a safe value for search_low in modern
   networks.

   It is important to determine the links overhead along the IP path,
   and then to select a TCP MSS size corresponding to the Layer 3 MTU.
   For example, if the MTU is 1024 Bytes and the TCP/IP headers are 40
   Bytes, (20 for IP + 20 for TCP) then the MSS would be 984 Bytes.


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   An example scenario is a network where the actual path MTU is 1240
   Bytes.  The TCP client probe MUST be capable of setting the MSS for
   the probe packets and could start at MSS = 984 (which corresponds
   to an MTU size of 1024 Bytes).

   The TCP client probe would open a TCP connection and advertise the
   MSS as 984.  Note that the client probe MUST generate these packets
   with the DF bit set. The TCP client probe then sends test traffic
   per a small default Send Socket Buffer size of ~8KBytes.  It should
   be kept small to minimize the possibility of congesting the network,
   which may induce packet 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 Bytes path MTU, probing with an MSS equal to
   984 would yield a successful probe and the test client packets would
   be successfully transferred to the test server.

   Also note that the test client MUST verify that the advertised MSS
   is indeed negotiated.  Network devices with built-in Layer 4
   capabilities can intercede during the connection establishment and
   reduce the advertised MSS to avoid fragmentation.  This is certainly
   a desirable feature from a network perspective, but it 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 it would be
   set to a MSS of 1460 in order to produce a 1500 Bytes MTU.  In this
   example, the test client will 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 none of the test packets are ACK'ed, congestive network
   may be the cause and the test probe is not conclusive.  Re-testing
   at another time is recommended to further isolate.

   The test is repeated until the desired granularity of the MTU is
   discovered.  The method can yield precise results at the expense of
   probing time.  One approach may be to reduce the probe size to
   half between the unsuccessful search_high and successful search_low
   value and raise it by half when seeking the upper limit.

3.2. Baseline Round Trip Time and Bandwidth

   Before stateful TCP testing can begin, it is important to determine
   the baseline Round Trip Time (i.e. non-congested inherent delay) and
   Bottleneck Bandwidth 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.




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3.2.1 Techniques to Measure Round Trip Time

   Following the definitions used in section 1.1, Round Trip Time (RTT)
   is the elapsed time between the clocking in of the first bit of a
   payload sent packet and the receipt of the last bit of the
   corresponding Acknowledgment.  Round Trip Delay (RTD) is used
   synonymously to twice the Link Latency.  RTT measurements SHOULD use
   techniques defined in [RFC2681] or statistics available from MIBs
   defined in [RFC4898].

   The RTT SHOULD be baselined during off-peak hours 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 3.3.2

   The following list is not meant to be exhaustive,  although it
   summarizes some of the most common ways to determine Round Trip Time.
   The desired measurement precision (i.e. msec versus usec) may dictate
   whether the RTT measurement can be achieved with ICMP pings or by a
   dedicated communications test instrument with precision timers.

   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 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.

   - 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 msec 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.



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3.2.2 Techniques to Measure end-to-end Bandwidth

   Before any TCP Throughput test can be conducted, bandwidth
   measurement tests MUST be run with stateless IP streams (i.e. not
   stateful TCP) in order to determine the available path bandwidth.
   These measurements SHOULD be conducted in both directions,
   especially in asymmetrical access networks (e.g. ADSL access).
   These tests should obviously be performed at various intervals
   throughout a business day or even across a week.  Ideally, 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.
   Ideally, these bandwidth measurements SHOULD use network capacity
   techniques as defined in [RFC5136].

3.3. TCP Throughput Tests

   This methodology specifically defines TCP throughput techniques to
   verify maximum achievable TCP performance in a managed business
   class IP network, as defined in section 2.1. This document defines
   a method to conduct these maximum achievable TCP throughput tests
   as well as guidelines on 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 SHOULD be conducted in order to measure network
   performance against expectations.  The number of trials and the type
   of testing (i.e. single versus multiple connections) will vary
   according to the intention of the test.   One example would be a
   single connection test in which the throughput achieved by large
   Send and Receive Socket Buffer sizes (i.e. 256KB) is to be measured.
   It would be advisable to test at various times of the business day.

   It is RECOMMENDED to run the tests in each direction independently
   first, then run both directions simultaneously.  In each case, the
   TCP Transfer Time, TCP Efficiency, and Buffer Delay Percentage
   metrics MUST be measured in each direction.  These metrics are
   defined in 3.3.2.

3.3.1 Calculate minimum required TCP RWND Size

   The minimum required TCP RWND Size can be calculated from the
   bandwidth delay product (BDP), which is:

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

   Note that the RTT is being used as the "Delay" variable in the
   BDP calculations.

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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 size
   must be adjusted to the same value at the opposite end of the network
   path.

   Minimum required TCP RWND = BDP / 8

   An example would be a T3 link with 25 msec RTT.  The BDP would equal
   ~1,105,000 bits and the minimum required TCP RWND would be ~138
   KBytes.

   Note that separate calculations are required on asymmetrical paths.
   An asymmetrical path example would be a 90 msec 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 & minimum TCP RWND

      Link                                         Minimum required
      Speed*         RTT              BDP             TCP RWND
      (Mbps)         (ms)            (bits)           (KBytes)
   ---------------------------------------------------------------------
        1.536        20              30,720              3.84
        1.536        50              76,800              9.60
        1.536       100             153,600             19.20
       44.210        10             442,100             55.26
       44.210        15             663,150             82.89
       44.210        25           1,105,250            138.16
      100             1             100,000             12.50
      100             2             200,000             25.00
      100             5             500,000             62.50
    1,000             0.1           100,000             12.50
    1,000             0.5           500,000             62.50
    1,000             1           1,000,000            125.00
   10,000             0.05          500,000             62.50
   10,000             0.3         3,000,000            375.00

   * Note that link speed is the Bottleneck Bandwidth (BB) for the NUT

   The following serial link speeds are used:
   - T1 = 1.536 Mbits/sec (for a B8ZS line encoding facility)
   - T3 = 44.21 Mbits/sec (for a C-Bit Framing facility)

   The above 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

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   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.

   Another example for a T3 using the same calculation formula is
   illustrated on the next page:

   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 MSS X 8.
   TCP Throughput = 3664 FPS X 1460 Bytes X 8 bits.
   TCP Throughput = 42.8 Mbps

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


   Figure 3.3.1a TCP Throughputs on a T3 at different RTTs


           45|
             |           _______42.8M
           40|           |64KB |
TCP          |           |     |
Throughput 35|           |     |
in Mbps      |           |     |          +-----+34.1M
           30|           |     |          |64KB |
             |           |     |          |     |
           25|           |     |          |     |
             |           |     |          |     |
           20|           |     |          |     |          _______20.5M
             |           |     |          |     |          |64KB |
           15|           |     |          |     |          |     |
             |12.8M+-----|     |          |     |          |     |
           10|     |16KB |     |          |     |          |     |
             |     |     |     |8.5M+-----|     |          |     |
            5|     |     |     |    |16KB |     |5.1M+-----|     |
             |_____|_____|_____|____|_____|_____|____|16KB |_____|_____
                        10               15               25
                                RTT in milliseconds




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   The following diagram shows the achievable TCP throughput on a 25ms
   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|                                             +-----+40.9M
TCP          |                                             |     |
Throughput 35|                                             |     |
in Mbps      |                                             |     |
           30|                                             |     |
             |                                             |     |
           25|                                             |     |
             |                                             |     |
           20|                               +-----+20.5M  |     |
             |                               |     |       |     |
           15|                               |     |       |     |
             |                               |     |       |     |
           10|                  +-----+10.2M |     |       |     |
             |                  |     |      |     |       |     |
            5|     +-----+5.1M  |     |      |     |       |     |
             |_____|_____|______|_____|______|_____|_______|_____|_____
                     16           32           64            128*
                          TCP RWND Size in KBytes

   * Note that 128KB requires [RFC1323] TCP Window scaling option.

3.3.2 Metrics for TCP Throughput Tests

   This framework focuses on a TCP throughput methodology and also
   provides several basic metrics to compare results between various
   throughput tests.  It is recognized that the complexity and
   unpredictability of TCP makes it impossible to develop a complete
   set of metrics that accounts for the myriad of variables (i.e. RTT
   variation, loss conditions, TCP implementation, etc.).  However,
   these basic metrics will facilitate TCP throughput comparisons
   under varying network conditions and between network traffic
   management techniques.

   The first metric is the TCP Transfer Time, which is simply the
   measured time required to transfer a block of data across
   simultaneous TCP connections.  This concept is useful when
   benchmarking traffic management techniques and when multiple
   TCP connections are required.

   TCP Transfer time may also be used to provide a normalized ratio of
   the actual TCP Transfer Time versus the Ideal Transfer Time.  This
   ratio is called the TCP Transfer Index and is defined as:

                     Actual TCP Transfer Time
                    -------------------------
                     Ideal TCP Transfer Time

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   The Ideal TCP Transfer time is derived from the network path
   Bottleneck Bandwidth and Layer 1/2/3/4 overheads associated with the
   network path.  Additionally, both the TCP RWND and the Send Socket
   Buffer Sizes must be tuned to equal or exceed the bandwidth delay
   product (BDP) as described in section 3.3.1.

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

   Table 3.3.2: Link Speed, RTT, BDP, TCP Throughput, and
                Ideal TCP Transfer time for a 100 MB File

       Link                             Maximum            Ideal TCP
       Speed                   BDP      Achievable TCP     Transfer time
       (Mbps)     RTT (ms)   (KBytes)   Throughput(Mbps)   (seconds)
   --------------------------------------------------------------------
         1.536    50            9.6            1.4             571
        44.21     25          138.2           42.8              18
       100         2           25.0           94.9               9
     1,000         1          125.0          949.2               1
    10,000         0.05        62.5        9,492                 0.1

    Transfer times are rounded for simplicity.

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

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

   The maximum achievable layer 2 throughput on T1 and T3 Interfaces
   is based on the maximum frames per second (FPS) permitted by the
   actual layer 1 speed with an MTU of 1500 Bytes.

   The maximum FPS for a T1 is 127 and the calculation formula is:
   FPS = T1 Link Speed / ((MTU + PPP + Flags + CRC16) X 8)
   FPS = (1.536M /((1500 Bytes + 4 Bytes + 2 Bytes + 2 Bytes) X 8 )))
   FPS = (1.536M / (1508 Bytes X 8))
   FPS = 1.536 Mbps / 12064 bits
   FPS = 127

   The maximum FPS for a T3 is 3664 and the calculation formula is:
   FPS = T3 Link Speed / ((MTU + PPP + Flags + CRC16) X 8)
   FPS = (44.21M /((1500 Bytes + 4 Bytes + 2 Bytes + 2 Bytes) X 8 )))
   FPS = (44.21M / (1508 Bytes X 8))
   FPS = 44.21 Mbps / 12064 bits
   FPS = 3664



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   The 1508 equates to:

     MTU + PPP + Flags + CRC16

   Where the MTU is 1500 Bytes, PPP is 4 Bytes, the 2 Flags are 1 Byte
   each and the CRC16 is 2 Bytes.

   Then, to obtain the Maximum Achievable TCP Throughput (layer 4), we
   simply use: MSS in Bytes X 8 bits X max FPS.
   For a T3, the maximum TCP Throughput = 1460 Bytes X 8 bits X 3664 FPS
   Maximum TCP Throughput = 11680 bits X 3664 FPS
   Maximum TCP Throughput = 42.8 Mbps.

   The maximum achievable layer 2 throughput on Ethernet Interfaces is
   based on the maximum frames per second permitted by the IEEE802.3
   standard when the MTU is 1500 Bytes.

   The maximum FPS for 100M Ethernet is 8127 and the calculation is:
   FPS = (100Mbps /(1538 Bytes X 8 bits))

   The maximum FPS for GigE is 81274 and the calculation formula is:
   FPS = (1Gbps /(1538 Bytes X 8 bits))

   The maximum FPS for 10GigE is 812743 and the calculation formula is:
   FPS = (10Gbps /(1538 Bytes X 8 bits))

   The 1538 equates to:

     MTU + Eth + CRC32 + IFG + Preamble + SFD
        (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.

   Note that better results could be obtained with jumbo frames on
   GigE and 10 GigE.

   Then, to obtain the Maximum Achievable TCP Throughput (layer 4), we
   simply use: MSS in Bytes X 8 bits X max FPS.
   For a 100M, the maximum TCP Throughput = 1460 B X 8 bits X 8127 FPS
   Maximum TCP Throughput = 11680 bits X 8127 FPS
   Maximum TCP Throughput = 94.9 Mbps.

   To illustrate the TCP Transfer Time Index, an example would be the
   bulk transfer of 100 MB over 5 simultaneous TCP connections  (each
   connection transferring 100 MB).  In this example, the Ethernet
   service provides a Committed Access Rate (CAR) of 500 Mbit/s.  Each
   connection may achieve different throughputs during a test and the
   overall throughput rate is not always easy to determine (especially
   as the number of connections increases).

   The ideal TCP Transfer Time would be ~8 seconds, but in this example,
   the actual TCP Transfer Time was 12 seconds.  The TCP Transfer Index
   would then be 12/8 = 1.5, which indicates that the transfer across
   all connections took 1.5 times longer than the ideal.

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   The second metric is TCP Efficiency, which is the percentage of Bytes
   that were not retransmitted and is defined as:

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

   Transmitted Bytes are the total number of TCP Bytes to be transmitted
   including the original and the retransmitted Bytes.   This metric
   provides comparative results between various traffic management and
   congestion avoidance mechanisms.   Performance between different TCP
   implementations could also be compared. (e.g. Reno, Vegas, etc).

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

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

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

   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.
   (See 3.2.1 for details concerning the baseline RTT measurements).

   The Buffer Delay Percentage is defined as:

              Average RTT during Transfer - Baseline RTT
              ------------------------------------------ x 100
                             Baseline RTT

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

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

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



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3.3.3 Conducting the TCP Throughput Tests

   Several TCP tools are currently used in the network world and one of
   the most common is "iperf".  With this tool, hosts are installed at
   each end of the network path; one acts as client and the other as
   a server.  The Send Socket Buffer and the TCP 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 or
   satellite links, etc.), TCP options such as Selective Acknowledgment
   SHOULD be considered and become part of the window size / throughput
   characterization.

   Host hardware performance must be well understood before conducting
   the tests described in the following sections.  A dedicated
   communications test instrument will generally be required, especially
   for line rates of GigE and 10 GigE.  A compliant TCP TTD SHOULD
   provide a warning message when the expected test throughput will
   exceed 10% of the network bandwidth capacity.  If the throughput test
   is expected to exceed 10% of the provider bandwidth, then the test
   should be coordinated with the network provider.  This does not
   include the customer premise bandwidth, the 10% refers directly to
   the provider's bandwidth (Provider Edge to Provider router).

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

3.3.4 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 TCP RWND
   configured in the end-user environment. For example, if the BDP for
   a long fat network turns out to be 2MB, then it is probably more
   realistic to test this network path with multiple connections.
   Assuming typical host computer TCP RWND Sizes of 64 KB (i.e. Windows
   XP), using 32 TCP connections would emulate a typical 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.

   Table 3.3.4 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

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   The TCP Transfer Time metric is useful for conducting multiple
   connection tests.  Each connection should be configured to transfer
   payloads of the same size (i.e. 100 MB), and the TCP Transfer time
   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
   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 Mb/sec.  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 metric MUST be computed for each
   connection as defined in section 3.3.2.


3.3.5 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.  For cases where the sustained TCP throughput
   does not equal the ideal value, some possible causes are:

   - Network congestion causing packet loss which MAY be inferred from
     a poor TCP Efficiency % (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)
   - Intermediate network devices which actively regenerate the TCP
     connection and can alter TCP RWND Size, MSS, etc.
   - Rate limiting (policing).  More details on traffic management
     tests follows in section 3.4


3.3.6 High Performance Network Options

   For cases where the network outperforms the client/server IP hosts
   some possible causes are:

   - 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.


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   - 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 Long
   Fat Networks. (i.e., networks with large BDP, see Figure 3.3.1b).


   - 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.






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   - 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 must 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.

3.4. Traffic Management Tests

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

   Traffic management techniques might be employed and the most common
   are:

   - Traffic Policing and/or Shaping
   - Priority queuing
   - Active Queue Management (AQM)

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   Configuring the end-to-end network with these various traffic
   management mechanisms is a complex under-taking. For traffic shaping
   and AQM techniques, the end goal is to provide better performance to
   bursty traffic.

   This section of the methodology provides guidelines to test traffic
   shaping and AQM implementations.  As in section 3.3, host hardware
   performance must be well understood before conducting the traffic
   shaping and AQM tests. Dedicated communications test instrument will
   generally be REQUIRED for line rates of GigE and 10 GigE.  If the
   throughput test is expected to exceed 10% of the provider bandwidth,
   then the test should be coordinated with the network provider.  This
   does not include the customer premises bandwidth, the 10% refers to
   the provider's bandwidth (Provider Edge to Provider router). Note
   that GigE and 10 GigE interfaces might benefit from hold-queue
   adjustments in order to prevent the saw-tooth TCP traffic pattern.


3.4.1 Traffic Shaping Tests

   For services where the available bandwidth is rate limited, two (2)
   techniques can be used: traffic policing or 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 exhausted).

   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 available
   bandwidth.  Throughout this section, the 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 connections test.  Proper shaping will
   provide a fair distribution of the available Bottleneck Bandwidth,
   while traffic policing will not.

   The traffic shaping tests are built upon the concepts of multiple
   connections testing as defined in section 3.3.3.  Calculating the BDP
   for the Bottleneck Bandwidth is first required before selecting the
   number of connections, the Send Socket Buffer and TCP RWND Sizes 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 Send Socket Buffer and TCP RWND Sizes to evenly
   fill the Bottleneck Bandwidth (~100 Mbps per connection).


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   The traffic shaping 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.  The
   throughput of each connection MUST be logged during the entire test,
   along with the TCP Transfer Time, TCP Efficiency, and Buffer Delay
   Percentage.

3.4.1.1 Interpretation of Traffic Shaping Test Results

   By plotting the throughput achieved by each TCP connection, we should
   see fair sharing of the bandwidth when traffic shaping is properly
   configured.  For the previous example of 5 connections sharing 500
   Mbps, each connection would consume ~100 Mbps with smooth variations.

   If traffic shaping is not configured properly or if traffic policing
   is present on the bottleneck interface,  the bandwidth sharing may
   not be fair.  The resulting throughput plot may reveal "spikey"
   throughput consumption of the competing TCP connections (due to the
   high rate of TCP retransmissions).

3.4.2 AQM Tests

   Active Queue Management techniques are specifically targeted to
   provide congestion avoidance to TCP traffic.  As an example, before
   the network element queue "fills" and enters the tail drop state, an
   AQM implementation like RED (Random Early Discard) drops packets at
   pre-configurable queue depth thresholds.  This action causes TCP
   connections to back-off which helps prevent tail drops and in
   turn helps avoid global TCP synchronization.

   RED is just an example and other AQM implementations like WRED
   (Weighted Random Early Discard) or REM (Random Exponential Marking)
   or AREM (Adaptive Random Exponential Marking), just to name a few,
   could be used.

   Again, rate limited interfaces may benefit greatly from AQM based
   techniques.  With a default FIFO queue, bloated buffering is
   increasingly a common encounter and has dire effects on TCP
   connections.  However, the main effect is the delayed congestion
   feedback (poor TCP control loop response) and enormous queuing
   delays on all other traffic flows.

   In a FIFO based queue, the TCP traffic may not be able to achieve
   the full throughput available on the Bottleneck Bandwidth link.
   While with an AQM implementation, TCP congestion avoidance would
   throttle the connections on the higher speed interface (i.e. LAN)
   and could help achieve the full throughput (up to the Bottleneck
   Bandwidth).  The bursty nature of TCP traffic is a key factor in the
   overall effectiveness of AQM techniques; steady state bulk transfer
   flows will generally not benefit from AQM because with bulk transfer
   flows, network device queues gracefully throttle the effective
   throughput rates due to increased delays.

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

   AQM testing also builds upon the concepts of multiple connections
   testing as defined in section 3.3.3.  Calculating the BDP for the
   Bottleneck Bandwidth is first required before selecting the number
   of connections, the Send Socket Buffer size and the TCP RWND Size
   per connection.

   For AQM 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 in order to stress the network element buffers / queues
   (10 connections for this example).

   The TCP TTD must be configured to generate these connections as
   shorter (bursty) flows versus bulk transfer type flows.  These TCP
   bursts should stress queue sizes in the 512KB range.  Again
   experimentation will be required; the proper number of TCP
   connections, the Send Socket Buffer and TCP RWND Sizes will be
   dictated by the size of the network element queue.

3.4.2.1 Interpretation of AQM Results

   The default queuing technique for most network devices is FIFO based.
   Under heavy traffic conditions, FIFO based queue management may cause
   enormous queuing delays plus delayed congestion feedback to all TCP
   applications. This can cause excessive loss on all of the TCP
   connections and in the worst cases, global TCP synchronization.

   AQM implementation can be detected by plotting individual and
   aggregate throughput results achieved by multiple TCP connections on
   the bottleneck interface. Proper AQM operation may be determined if
   the TCP throughput is fully utilized (up to the Bottleneck Bandwidth)
   and fairly shared between TCP connections.  For the previous example
   of 10 connections (window = 64 KB) sharing 500 Mbps, each connection
   should consume ~50 Mbps.  If AQM was not properly enabled on the
   interface, then the TCP connections would retransmit at higher rates
   and the net effect is that the Bottleneck Bandwidth is not fully
   utilized.

   Another means to study non-AQM versus AQM implementations is to use
   the Buffer Delay Percent metric for all of the connections.  The
   Buffer Delay Percentage should be significantly lower in AQM
   implementations versus default FIFO queuing.

   Additionally, non-AQM implementations may exhibit a lower TCP
   Transfer Efficiency.

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4. Security Considerations

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

5. IANA Considerations

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

6. Acknowledgments

   Thanks to 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.

7. References

7.1 Normative References

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

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

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

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

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

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

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

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

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

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

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7.2. Informative References

Authors' Addresses

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

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

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

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

   Rudiger 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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