Network Working Group B. Constantine
Internet-Draft JDSU
Intended status: Informational G. Forget
Expires: March 24, 2011 Bell Canada (Ext. Consultant)
Reinhard Schrage
Schrage Consulting
September 24, 2010
Framework for TCP Throughput Testing
draft-ietf-ippm-tcp-throughput-tm-07.txt
Abstract
This document describes a framework for measuring sustained TCP
throughput performance in an end-to-end managed network environment.
This document is intended to provide a practical methodology to help
users validate the TCP layer performance of a managed network, which
should provide a better indication of end-user experience. In the
framework, various TCP and network parameters are identified that
should be tested as part of the network verification at the TCP
layer.
Requirements Language
The key words "MUST", "MUST NOT", "REQUIRED", "SHALL", "SHALL NOT",
"SHOULD", "SHOULD NOT", "RECOMMENDED", "MAY", and "OPTIONAL" in this
document are to be interpreted as described in RFC 2119 [RFC2119].
Status of this Memo
This Internet-Draft is submitted in full conformance with the
provisions of BCP 78 and BCP 79.
Internet-Drafts are working documents of the Internet Engineering
Task Force (IETF). Note that other groups may also distribute
working documents as Internet-Drafts. The list of current Internet-
Drafts is at http://datatracker.ietf.org/drafts/current/.
Internet-Drafts are draft documents valid for a maximum of six months
and may be updated, replaced, or obsoleted by other documents at any
time. It is inappropriate to use Internet-Drafts as reference
material or to cite them other than as "work in progress."
This Internet-Draft will expire on March 24, 2011.
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Copyright Notice
Copyright (c) 2010 IETF Trust and the persons identified as the
document authors. All rights reserved.
This document is subject to BCP 78 and the IETF Trust's Legal
Provisions Relating to IETF Documents
(http://trustee.ietf.org/license-info) in effect on the date of
publication of this document. Please review these documents
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to this document. Code Components extracted from this document must
include Simplified BSD License text as described in Section 4.e of
the Trust Legal Provisions and are provided without warranty as
described in the Simplified BSD License.
Table of Contents
1. Introduction . . . . . . . . . . . . . . . . . . . . . . . . . 3
1.1 Test Set-up and Terminology . . . . . . . . . . . . . . . 4
2. Scope and Goals of this methodology. . . . . . . . . . . . . . 5
2.1 TCP Equilibrium State Throughput . . . . . . . . . . . . . 6
2.2 Metrics for TCP Throughput Tests . . . . . . . . . . . . . 7
3. TCP Throughput Testing Methodology . . . . . . . . . . . . . . 9
3.1 Determine Network Path MTU . . . . . . . . . . . . . . . . 11
3.2. Baseline Round Trip Time and Bandwidth . . . . . . . . . . 13
3.2.1 Techniques to Measure Round Trip Time . . . . . . . . 13
3.2.2 Techniques to Measure End-end Bandwidth . . . . . . . 14
3.3. TCP Throughput Tests . . . . . . . . . . . . . . . . . . . 14
3.3.1 Calculate Optimum TCP Window Size. . . . . . . . . . . 15
3.3.2 Conducting the TCP Throughput Tests. . . . . . . . . . 17
3.3.3 Single vs. Multiple TCP Connection Testing . . . . . . 18
3.3.4 Interpretation of the TCP Throughput Results . . . . . 19
3.4. Traffic Management Tests . . . . . . . . . . . . . . . . . 19
3.4.1 Traffic Shaping Tests. . . . . . . . . . . . . . . . . 20
3.4.1.1 Interpretation of Traffic Shaping Test Results. . . 20
3.4.2 RED Tests. . . . . . . . . . . . . . . . . . . . . . . 21
3.4.2.1 Interpretation of RED Results . . . . . . . . . . . 21
4. Security Considerations . . . . . . . . . . . . . . . . . . . 22
5. IANA Considerations . . . . . . . . . . . . . . . . . . . . . 22
5.1. Registry Specification . . . . . . . . . . . . . . . . . . 22
5.2. Registry Contents . . . . . . . . . . . . . . . . . . . . 22
6. Acknowledgments . . . . . . . . . . . . . . . . . . . . . . . 22
7. References . . . . . . . . . . . . . . . . . . . . . . . . . . 22
7.1 Normative References . . . . . . . . . . . . . . . . . . . 22
7.2 Informative References . . . . . . . . . . . . . . . . . . 23
Authors' Addresses . . . . . . . . . . . . . . . . . . . . . . . . 23
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1. Introduction
Network providers are coming to the realization that Layer 2/3
testing and TCP layer testing are required to more adequately ensure
end-user satisfaction. Testing an operational network prior to
customer activation is referred to as "turn-up" testing and the SLA
(Service Level Agreement) is generally based upon Layer 2/3
information rate, packet delay, loss and delay variation. Therefore,
the network provider community desires to measure network throughput
performance at the TCP layer. Measuring TCP throughput provides a
meaningful measure with respect to the end user experience (and
ultimately reach some level of TCP testing interoperability which
does not exist today).
Additionally, end-users (business enterprises) seek to conduct
repeatable TCP throughput tests between enterprise locations. Since
these enterprises rely on the networks of the providers, a common
test methodology (and metrics) would be equally beneficial to both
parties.
So the intent behind this TCP throughput methodology is to define
a methodology for testing sustained TCP layer performance. In this
document, sustained TCP throughput is that amount of data per unit
time that TCP transports during equilibrium (steady state), i.e.
after the initial slow start phase. We refer to this state as TCP
Equilibrium, and that the equilibrium throughput is the maximum
achievable for the TCP connection(s).
There are many variables to consider when conducting a TCP throughput
test and this methodology focuses on some of the most common
parameters that MUST be considered such as:
- Path MTU and Maximum Segment Size (MSS)
- RTT and Bottleneck BW
- Ideal TCP Window (Bandwidth Delay Product)
- Single Connection and Multiple Connection testing
This methodology proposes a test which SHOULD be performed in
addition to traditional Layer 2/3 type tests, which are conducted to
verify the integrity of the network before conducting TCP tests.
Examples include iperf (UDP mode) or manual packet layer test
techniques where packet throughput, loss, and delay measurements are
conducted. When available, standardized testing similar to RFC 2544
[RFC2544] but adapted for use on operational networks may be used
(because RFC 2544 methods are not intended for use outside the lab
environment).
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1.1 Test Set-up and Terminology
This section provides a general overview of the test configuration
for this methodology. The test is intended to be conducted on an
end-end operational network, so there are multitudes of network
architectures and topologies that can be tested. This test set-up
diagram is very general and the main intent is to illustrate the
segmentation of the end user and network provider domains.
Common terminologies used in the test methodology are:
- Customer Provided Equipment (CPE), refers to customer owned
- Customer Edge (CE), refers to provider owned demarcation device
- Provider Edge (PE), refers to provider located distribution
equipment
- P (Provider), refers to provider core network equipment
- Bottleneck Bandwidth*, lowest bandwidth along the complete network
path
- Round-Trip Time (RTT), refers to Layer 4 back and forth delay
- Round-Trip Delay (RTD), refers to Layer 1 back and forth delay
- Network Under Test (NUT), refers to the tested IP network path
- TCP Throughput Test Device (TCP TTD), refers to compliant TCP
host that generates traffic and measures metrics as defined in
this methodology
+----+ +----+ +----+ +----+ +---+ +---+ +----+ +----+ +----+ +----+
| | | | | | | | | | | | | | | | | | | |
| TCP|-| CPE|-| CE |--| PE |-| P |--| P |-| PE |--| CE |-| CPE|-| TCP|
| TD | | | | |BB| | | | | | | |BB| | | | | TD |
+----+ +----+ +----+**+----+ +---+ +---+ +----+**+----+ +----+ +----+
<------------------------ NUT ------------------------>
<-------------------------RTT ------------------------>
* Bottleneck Bandwidth and Bandwidth are used synonomously in this
document.
** Most of the time the Bottleneck Bandwidth is in the access portion
of the wide area network (CE - PE)
Note that the NUT may consist of a variety of devices including (and
NOT limited to): load balancers, proxy servers, WAN acceleration
devices. The detailed topology of the NUT MUST be considered when
conducting the TCP throughput tests, but this methodology makes no
attempt to characterize TCP performance related to specific network
architectures.
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2. Scope and Goals of this Methodology
Before defining the goals of this methodology, it is important to
clearly define the areas that are out-of-scope for this
methodology.
- The methodology is not intended to predict TCP throughput
behavior during the transient stages of a TCP connection, such
as initial slow start.
- The methodology is not intended to definitively benchmark TCP
implementations of one OS to another, although some users MAY find
some value in conducting qualitative experiments.
- The methodology is not intended to provide detailed diagnosis
of problems within end-points or the network itself as related to
non-optimal TCP performance, although a results interpretation
section for each test step MAY provide insight into potential
issues within the network.
- The methodology does not propose a method to operate permanently
with high measurement loads. TCP performance and optimization data of
operational networks MAY be captured and evaluated by using data of
the "TCP Extended Statistics MIB" [RFC4898].
- The methodology is not intended to measure TCP throughput as part
of an SLA, or to compare the TCP performance between service
providers or to compare between implementations of this methodology
(test equipment).
In contrast to the above exclusions, the goals of this methodology
are to define a method to conduct a structured, end-to-end
assessment of sustained TCP performance within a managed business
class IP network. A key goal is to establish a set of "best
practices" that an engineer SHOULD apply when validating the
ability of a managed network to carry end-user TCP applications.
The specific goals are to:
- Provide a practical test approach that specifies well understood,
end-user configurable TCP parameters such as TCP Window size, MSS
(Maximum Segment Size), number of connections, and how these affect
the outcome of TCP performance over a network.
- Provide specific test conditions (link speed, RTT, TCP Window size,
etc.) and maximum achievable TCP throughput under TCP Equilibrium
conditions. For guideline purposes, provide examples of these test
conditions and the maximum achievable TCP throughput during the
equilibrium state. Section 2.1 provides specific details concerning
the definition of TCP Equilibrium within the context of this
methodology.
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- Define three (3) basic metrics that can be used to compare the
performance of TCP connections under various network conditions.
- In test situations where the RECOMMENDED procedure does not yield
the maximum achievable TCP throughput result, this methodology
provides some possible areas within the end host or network that
SHOULD be considered for investigation (although again, this
methodology is not intended to provide a detailed diagnosis of these
issues).
2.1 TCP Equilibrium State Throughput
TCP connections have three (3) fundamental congestion window phases
as documented in [RFC5681]. These phases are:
- Slow Start, which occurs during the beginning of a TCP transmission
or after a retransmission time out event.
- Congestion avoidance, which is the phase during which TCP ramps up
to establish the maximum attainable throughput on an end-end network
path. Retransmissions are a natural by-product of the TCP congestion
avoidance algorithm as it seeks to achieve maximum throughput on
the network path.
- Retransmission phase, which include Fast Retransmit (Tahoe) and
Fast Recovery (Reno and New Reno). When a packet is lost, the
Congestion avoidance phase transitions to a Fast Retransmission or
Recovery Phase dependent upon the TCP implementation.
The following diagram depicts these phases.
| ssthresh
TCP | |
Through- | | Equilibrium
put | |\ /\/\/\/\/\ Retransmit /\/\ ...
| | \ / | Time-out /
| | \ / | _______ _/
| Slow _/ |/ | / | Slow _/
| Start _/ Congestion |/ |Start_/ Congestion
| _/ Avoidance Loss | _/ Avoidance
| _/ Event | _/
| _/ |/
|/__________________________________________________________
Time
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This TCP methodology provides guidelines to measure the equilibrium
throughput which refers to the maximum sustained rate obtained by
congestion avoidance before packet loss conditions occur (which MAY
cause the state change from congestion avoidance to a retransmission
phase). All maximum achievable throughputs specified in Section 3 are
with respect to this equilibrium state.
2.2 Metrics for TCP Throughput Tests
This framework focuses on a TCP throughput methodology and also
provides several basic metrics to compare results of various
throughput tests. It is recognized that the complexity and
unpredictability of TCP makes it impossible to develop a complete
set of metrics that account for the myriad of variables (i.e. RTT
variation, loss conditions, TCP implementation, etc.). However,
these basic metrics will facilitate TCP throughput comparisons
under varying network conditions and between network traffic
management techniques.
The first metric is the TCP Transfer Time, which is simply the
measured time it takes to transfer a block of data across
simultaneous TCP connections. The concept is useful when
benchmarking traffic management techniques, where multiple
connections MAY be REQUIRED.
The TCP Transfer time MAY also be used to provide a normalized ratio
of the actual TCP Transfer Time versus Ideal Transfer Time. This
ratio is called the TCP Transfer Index and is defined as:
Actual TCP Transfer Time
-------------------------
Ideal TCP Transfer Time
The Ideal TCP Transfer time is derived from the network path
bottleneck bandwidth and the various Layer 1/2/3 overheads associated
with the network path. Additionally, the TCP Window size must be
tuned to equal the bandwidth delay product (BDP) as described in
Section 3.3.1.
The following table illustrates a single connection TCP Transfer and
the Ideal TCP Transfer time for a 100 MB file with the ideal TCP
window size based on the BDP.
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Table 2.2: Link Speed, RTT, TCP Throughput, Ideal TCP Transfer time
Link Maximum Achievable Ideal TCP Transfer time
Speed RTT (ms) TCP Throughput(Mbps) Time in seconds
--------------------------------------------------------------------
T1 20 1.17 684.93
T1 50 1.40 570.61
T1 100 1.40 570.61
T3 10 42.05 19.03
T3 15 42.05 19.03
T3 25 41.52 18.82
T3(ATM) 10 36.50 21.92
T3(ATM) 15 36.23 22.14
T3(ATM) 25 36.27 22.05
100M 1 91.98 8.70
100M 2 93.44 8.56
100M 5 93.44 8.56
1Gig 0.1 919.82 0.87
1Gig 0.5 934.47 0.86
1Gig 1 934.47 0.86
10Gig 0.05 9,344.67 0.09
10Gig 0.3 9,344.67 0.09
* Calculation is based on File Size in Bytes X 8 / TCP Throughput.
** TCP Throughput is derived from Table 3.3.
To illustrate the TCP Transfer Time Index, an example would be the
bulk transfer of 100 MB over 5 simultaneous TCP connections (each
connection uploading 100 MB). In this example, the Ethernet service
provides a Committed Access Rate (CAR) of 500 Mbit/s. Each
connection MAY achieve different throughputs during a test and the
overall throughput rate is not always easy to determine (especially
as the number of connections increases).
The ideal TCP Transfer Time would be ~8 seconds, but in this example,
the actual TCP Transfer Time was 12 seconds. The TCP Transfer Index
would be 12/8 = 1.5, which indicates that the transfer across all
connections took 1.5 times longer than the ideal.
The second metric is the TCP Efficiency metric which is the
percentage of bytes that were not retransmitted and is defined as:
Transmitted Bytes - Retransmitted Bytes
--------------------------------------- x 100
Transmitted Bytes
Transmitted bytes are the total number of TCP payload bytes to be
transmitted which includes the original and retransmitted bytes. This
metric provides a comparative measure between various QoS mechanisms
such as traffic management, congestion avoidance, and also various
TCP implementations (i.e. Reno, Vegas, etc.).
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As an example, if 100,000 bytes were sent and 2,000 had to be
retransmitted, the TCP Efficiency SHOULD be calculated as:
102,000 - 2,000
---------------- x 100 = 98.03%
102,000
Note that the retransmitted bytes MAY have occurred more than once,
and these multiple retransmissions are added to the Retransmitted
Bytes count (and the Transmitted Bytes count).
And the third metric is the Buffer Delay Percentage, which represents
the increase in RTT during a TCP throughput test from the inherent
network RTT (baseline RTT). The baseline RTT is the round-trip time
inherent to the network path under non-congested conditions.
The Buffer Delay Percentage is defined as:
Average RTT during Transfer - Baseline RTT
------------------------------------------ x 100
Baseline RTT
As an example, the baseline RTT for the network path is 25 msec.
During the course of a TCP transfer, the average RTT across the
entire transfer increased to 32 msec. In this example, the Buffer
Delay Percentage WOULD be calculated as:
32 - 25
------- x 100 = 28%
25
Note that the TCP Transfer Time, TCP Efficiency, and Buffer Delay
metrics MUST be measured during each throughput test.
Poor TCP Transfer Time Indexes (TCP Transfer Time greater than Ideal
TCP Transfer Times) MAY be diagnosed by correlating with sub-optimal
TCP Efficiency and/or Buffer Delay Percentage metrics.
3. TCP Throughput Testing Methodology
As stated in Section 1, it is considered best practice to verify
the integrity of the network by conducting Layer2/3 stress tests
such as [RFC2544] or other methods of network stress tests. If the
network is not performing properly in terms of packet loss, jitter,
etc. then the TCP layer testing will not be meaningful since the
equilibrium throughput MAY be very difficult to achieve (in a
"dysfunctional" network).
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TCP Throughput testing MAY require cooperation between the end user
customer and the network provider. In a Layer 2/3 VPN architecture,
the testing SHOULD be conducted on the Customer Edge (CE) router and
not the Provider Edge (PE) router.
The following represents the sequential order of steps to conduct the
TCP throughput testing methodology:
1. Identify the Path MTU. Packetization Layer Path MTU Discovery
or PLPMTUD, [RFC4821], MUST be conducted to verify the maximum
network path MTU. Conducting PLPMTUD establishes the upper limit for
the MSS to be used in subsequent steps.
2. Baseline Round Trip Time and Bandwidth. This step establishes the
inherent, non-congested Round Trip Time (RTT) and the bottleneck
bandwidth of the end-end network path. These measurements are used
to provide estimates of the ideal TCP window size, which SHOULD be
used in subsequent test steps. These measurements reference
[RFC2681] and [RFC4898] to measure RTD (and the associated RTT).
Also, [RFC5136] is referenced to measure network capacity.
3. TCP Connection Throughput Tests. With baseline measurements
of Round Trip Time and bottleneck bandwidth, a series of single and
multiple TCP connection throughput tests SHOULD be conducted to
baseline the network performance expectations.
4. Traffic Management Tests. Various traffic management and queuing
techniques SHOULD be tested in this step, using multiple TCP
connections. Multiple connection testing SHOULD verify that the
network is configured properly for traffic shaping versus policing,
various queuing implementations, and RED.
Important to note are some of the key characteristics and
considerations for the TCP test instrument. The test host MAY be a
standard computer or dedicated communications test instrument
and these TCP test hosts be capable of emulating both a client and a
server.
Whether the TCP test host is a standard computer or a compliant TCP
TTD, the following areas SHOULD be considered when selecting
a test host:
- TCP implementation used by the test host OS version, i.e. Linux OS
kernel using TCP Reno, TCP options supported, etc. This will
obviously be more important when using custom test equipment where
the TCP implementation MAY be customized or tuned to run in higher
performance hardware. When a compliant TCP TTD is used, the TCP
implementation SHOULD be identified in the test results. The
compliant TCP TTD SHOULD be usable for complete end-to-end testing
through network security elements and SHOULD also be usable for
testing network sections.
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- Most importantly, the TCP test host must be capable of generating
and receiving stateful TCP test traffic at the full link speed of the
network under test. As a general rule of thumb, testing TCP
throughput at rates greater than 100 Mbit/sec MAY require high
performance server hardware or dedicated hardware based test tools.
Thus, other devices cannot realize higher TCP throughput, and user
expectations SHOULD be set accordingly with user manual or notes on
the results report.
- Measuring RTT and TCP Efficiency per connection will generally
require dedicated hardware based test tools. In the absence of
dedicated hardware based test tools, these measurements MAY need to
be conducted with packet capture tools (conduct TCP throughput tests
and analyze RTT and retransmission results with packet captures).
Another option MAY be to use "TCP Extended Statistics MIB" per
[RFC4898].
- The compliant TCP TTD and its access to the network under test MUST
NOT introduce a performance bottleneck of any kind.
3.1. Determine Network Path MTU
TCP implementations SHOULD use Path MTU Discovery techniques (PMTUD).
PMTUD relies on ICMP 'need to frag' messages to learn the path MTU.
When a device has a packet to send which has the Don't Fragment (DF)
bit in the IP header set and the packet is larger than the Maximum
Transmission Unit (MTU) of the next hop link, the packet is dropped
and the device sends an ICMP 'need to frag' message back to the host
that originated the packet. The ICMP 'need to frag' message includes
the next hop MTU which PMTUD uses to tune the TCP Maximum Segment
Size (MSS). Unfortunately, because many network managers completely
disable ICMP, this technique does not always prove reliable in real
world situations.
Packetization Layer Path MTU Discovery or PLPMTUD [RFC4821] MUST
be conducted to verify the minimum network path MTU. PLPMTUD can
be used with or without ICMP. The following sections provide a
summary of the PLPMTUD approach and an example using the TCP
protocol. [RFC4821] specifies a search_high and a search_low
parameter for the MTU. As specified in [RFC4821], a value of 1024 is
a generally safe value to choose for search_low in modern networks.
It is important to determine the overhead of the links in the path,
and then to select a TCP MSS size corresponding to the Layer 3 MTU.
For example, if the MTU is 1024 bytes and the TCP/IP headers are 40
bytes, then the MSS would be set to 984 bytes.
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An example scenario is a network where the actual path MTU is 1240
bytes. The TCP client probe MUST be capable of setting the MSS for
the probe packets and could start at MSS = 984 (which corresponds
to an MTU size of 1024 bytes).
The TCP client probe would open a TCP connection and advertise the
MSS as 984. Note that the client probe MUST generate these packets
with the DF bit set. The TCP client probe then sends test traffic
per a nominal window size (8KB, etc.). The window size SHOULD be
kept small to minimize the possibility of congesting the network,
which MAY induce congestive loss. The duration of the test should
also be short (10-30 seconds), again to minimize congestive effects
during the test.
In the example of a 1240 byte path MTU, probing with an MSS equal to
984 would yield a successful probe and the test client packets would
be successfully transferred to the test server.
Also note that the test client MUST verify that the MSS advertised
is indeed negotiated. Network devices with built-in Layer 4
capabilities can intercede during the connection establishment
process and reduce the advertised MSS to avoid fragmentation. This
is certainly a desirable feature from a network perspective, but
can yield erroneous test results if the client test probe does not
confirm the negotiated MSS.
The next test probe would use the search_high value and this would
be set to MSS = 1460 to correspond to a 1500 byte MTU. In this
example, the test client MUST retransmit based upon time-outs (since
no ACKs will be received from the test server). This test probe is
marked as a conclusive failure if none of the test packets are
ACK'ed. If any of the test packets are ACK'ed, congestive network
MAY be the cause and the test probe is not conclusive. Re-testing
at other times of the day is RECOMMENDED to further isolate.
The test is repeated until the desired granularity of the MTU is
discovered. The method can yield precise results at the expense of
probing time. One approach MAY be to reduce the probe size to
half between the unsuccessful search_high and successful search_low
value, and increase by increments of 1/2 when seeking the upper
limit.
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3.2. Baseline Round Trip Time and Bandwidth
Before stateful TCP testing can begin, it is important to determine
the baseline Round Trip Time (non-congested inherent delay) and
bottleneck bandwidth of the end-end network to be tested. These
measurements are used to provide estimates of the ideal TCP window
size, which SHOULD be used in subsequent test steps. These latency
and bandwidth tests SHOULD be run during the time of day for which
the TCP throughput tests will occur.
3.2.1 Techniques to Measure Round Trip Time
Following the definitions used in the section 1.1; Round Trip Time
(RTT) is the time elapsed between the clocking in of the first bit
of a payload packet to the receipt of the last bit of the
corresponding Acknowledgment. Round Trip Delay (RTD) is used
synonymously to twice the Link Latency. RTT measurements SHOULD use
techniques defined in [RFC2681] or statistics available from MIBs
defined in [RFC4898].
The RTT SHOULD be baselined during "off-peak" hours to obtain a
reliable figure for inherent network latency versus additional delay
caused by network buffering delays.
During the actual sustained TCP throughput tests, RTT MUST be
measured along with TCP throughput. Buffer delay effects can be
isolated if RTT is concurrently measured.
This is not meant to provide an exhaustive list, but summarizes some
of the more common ways to determine round trip time (RTT) through
the network. The desired resolution of the measurement (i.e. msec
versus usec) may dictate whether the RTT measurement can be achieved
with standard tools such as ICMP ping techniques or whether
specialized test equipment would be required with high precision
timers. The objective in this section is to list several techniques
in order of decreasing accuracy.
- Use test equipment on each end of the network, "looping" the
far-end tester so that a packet stream can be measured end-end. This
test equipment RTT measurement MAY be compatible with delay
measurement protocols specified in [RFC5357].
- Conduct packet captures of TCP test applications using for example
"iperf" or FTP, etc. By running multiple experiments, the packet
captures can be studied to estimate RTT based upon the SYN -> SYN-ACK
handshakes within the TCP connection set-up.
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- ICMP Pings MAY also be adequate to provide round trip time
estimations. Some limitations of ICMP Ping MAY include msec
resolution and whether the network elements respond to pings (or
block them).
3.2.2 Techniques to Measure End-end Bandwidth
There are many well established techniques available to provide
estimated measures of bandwidth over a network. This measurement
SHOULD be conducted in both directions of the network, especially for
access networks which MAY be asymmetrical. Measurements SHOULD use
network capacity techniques defined in [RFC5136].
The bandwidth measurement test MUST be run with stateless IP streams
(not stateful TCP) in order to determine the available bandwidth in
each direction. And this test SHOULD obviously be performed at
various intervals throughout a business day (or even across a week).
Ideally, the bandwidth test SHOULD produce a log output of the
bandwidth achieved across the test interval.
3.3. TCP Throughput Tests
This methodology specifically defines TCP throughput techniques to
verify sustained TCP performance in a managed business network.
Defined in section 2.1, the equilibrium throughput reflects the
maximum rate achieved by a TCP connection within the congestion
avoidance phase on an end-end network path. This section and others
will define the method to conduct these sustained throughput tests
and guidelines of the predicted results.
With baseline measurements of round trip time and bandwidth
from section 3.2, a series of single and multiple TCP connection
throughput tests can be conducted to baseline network performance
against expectations.
It is RECOMMENDED to run the tests in each direction independently
first, then run both directions simultaneously. In each case, the
TCP Transfer Time, TCP Efficiency, and Buffer Delay metrics MUST be
measured in each direction.
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3.3.1 Calculate Ideal TCP Window Size
The ideal TCP window size can be calculated from the bandwidth
delay product (BDP), which is:
BDP (bits) = RTT (sec) x Bandwidth (bps)
By dividing the BDP by 8, the "ideal" TCP window size is calculated.
An example would be a T3 link with 25 msec RTT. The BDP would equal
~1,105,000 bits and the ideal TCP window would equal ~138,000 bytes.
The following table provides some representative network link speeds,
latency, BDP, and associated ideal TCP window size. Sustained
TCP transfers SHOULD reach nearly 100% throughput, minus the overhead
of Layers 1-3 and the divisor of the MSS into the TCP Window.
For this single connection baseline test, the MSS size will effect
the achieved throughput (especially for smaller TCP Window sizes).
Table 3.2 provides the achievable, equilibrium TCP throughput (at
Layer 4) using 1460 byte MSS. Also in this table, the 58 byte L1-L4
overhead including the Ethernet CRC32 is used for simplicity.
Table 3.3: Link Speed, RTT and calculated BDP, TCP Throughput
Link Ideal TCP Maximum Achievable
Speed* RTT (ms) BDP (bits) Window (kBytes) TCP Throughput(Mbps)
---------------------------------------------------------------------
T1 20 30,720 3.84 1.17
T1 50 76,800 9.60 1.40
T1 100 153,600 19.20 1.40
T3 10 442,100 55.26 42.05
T3 15 663,150 82.89 42.05
T3 25 1,105,250 138.16 41.52
T3(ATM) 10 407,040 50.88 36.50
T3(ATM) 15 610,560 76.32 36.23
T3(ATM) 25 1,017,600 127.20 36.27
100M 1 100,000 12.50 91.98
100M 2 200,000 25.00 93.44
100M 5 500,000 62.50 93.44
1Gig 0.1 100,000 12.50 919.82
1Gig 0.5 500,000 62.50 934.47
1Gig 1 1,000,000 125.00 934.47
10Gig 0.05 500,000 62.50 9,344.67
10Gig 0.3 3,000,000 375.00 9,344.67
* Note that link speed is the bottleneck bandwidth for the NUT
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Also, the following link speeds (available payload bandwidth) were
used for the WAN entries:
- T1 = 1.536 Mbits/sec (B8ZS line encoding facility)
- T3 = 44.21 Mbits/sec (C-Bit Framing)
- T3(ATM) = 36.86 Mbits/sec (C-Bit Framing & PLCP, 96000 Cells per
second)
The calculation method used in this document is a 3 step process :
1 - Determine what SHOULD be the optimal TCP Window size value
based on the optimal quantity of "in-flight" octets discovered by
the BDP calculation. We take into consideration that the TCP
Window size has to be an exact multiple value of the MSS.
2 - Calculate the achievable layer 2 throughput by multiplying the
value determined in step 1 with the MSS & (MSS + L2 + L3 + L4
Overheads) divided by the RTT.
3 - Finally, multiply the calculated value of step 2 by the MSS
versus (MSS + L2 + L3 + L4 Overheads) ratio.
This provides the achievable TCP Throughput value. Sometimes, the
maximum achievable throughput is limited by the maximum achievable
quantity of Ethernet Frames per second on the physical media. Then
this value is used in step 2 instead of the calculated one.
The following diagram compares achievable TCP throughputs on a T3 link
with Windows 2000/XP TCP window sizes of 16KB versus 64KB.
45|
| _____42.1M
40| |64K|
TCP | | |
Throughput 35| | | _____34.3M
in Mbps | | | |64K|
30| | | | |
| | | | |
25| | | | |
| | | | |
20| | | | | _____20.5M
| | | | | |64K|
15| 14.5M____| | | | | |
| |16K| | | | | |
10| | | | 9.6M+---+ | | |
| | | | |16K| | 5.8M____+ |
5| | | | | | | |16K| |
|______+___+___+_______+___+___+_______+__ +___+_______
10 15 25
RTT in milliseconds
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The following diagram shows the achievable TCP throughput on a 25ms
T3 when the TCP Window size is increased and with the [RFC1323] TCP
Window scaling option.
45|
| +-----+42.47M
40| | |
TCP | | |
Throughput 35| | |
in Mbps | | |
30| | |
| | |
25| | |
| ______ 21.23M | |
20| | | | |
| | | | |
15| | | | |
| | | | |
10| +----+10.62M | | | |
| _______5.31M | | | | | |
5| | | | | | | | |
|__+_____+______+____+__________+____+________+_____+___
16 32 64 128
TCP Window size in KBytes
3.3.2 Conducting the TCP Throughput Tests
There are several TCP tools that are commonly used in the network
world and one of the most common is the "iperf" tool. With this tool,
hosts are installed at each end of the network segment; one as client
and the other as server. The TCP Window size of both the client and
the server can be manually set and the achieved throughput is
measured, either uni-directionally or bi-directionally. For higher
BDP situations in lossy networks (long fat networks or satellite
links, etc.), TCP options such as Selective Acknowledgment SHOULD be
considered and also become part of the window size / throughput
characterization.
Host hardware performance MUST be well understood before conducting
the TCP throughput tests and other tests in the following sections.
Dedicated test equipment will generally be REQUIRED, especially for
line rates of GigE and 10 GigE. A compliant TCP TTD SHOULD provide a
warning message when the expected test throughput will exceed 10% of
the network bandwidth capacity. If the throughput test is expected
to exceed 10% of the provider bandwidth, then the test SHOULD be
coordinated with the network provider. This does not include the
customer premise bandwidth, the 10% refers directly to the provider's
bandwidth (Provider Edge to Provider router).
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The TCP throughput test SHOULD be run over a long enough duration
to properly exercise network buffers and also characterize
performance during different time periods of the day.
Note that both the TCP Transfer Time, TCP Efficiency, and Buffer
Delay metrics MUST be measured during each throughput test.
Poor TCP Transfer Time Indexes (TCP Transfer Time greater than Ideal
TCP Transfer Times) MAY be diagnosed by correlating with sub-optimal
TCP Efficiency and/or Buffer Delay Percentage metrics.
3.3.3 Single vs. Multiple TCP Connection Testing
The decision whether to conduct single or multiple TCP connection
tests depends upon the size of the BDP in relation to the window
sizes configured in the end-user environment. For example, if the
BDP for a long-fat pipe turns out to be 2MB, then it is probably more
realistic to test this pipe with multiple connections. Assuming
typical host computer window settings of 64 KB, using 32 connections
would realistically test this pipe.
The following table is provided to illustrate the relationship of the
BDP, window size, and the number of connections required to utilize
the available capacity. For this example, the network bandwidth is
500 Mbps, RTT is equal to 5 ms, and the BDP equates to 312 KBytes.
#Connections
Window to Fill Link
------------------------
16KB 20
32KB 10
64KB 5
128KB 3
The TCP Transfer Time metric is useful for conducting multiple
connection tests. Each connection SHOULD be configured to transfer
payloads of the same size (i.e. 100 MB), and the TCP Transfer time
SHOULD provide a simple metric to verify the actual versus expected
results.
Note that the TCP transfer time is the time for all connections to
complete the transfer of the configured payload size. From the
example table listed above, the 64KB window is considered. Each of
the 5 connections would be configured to transfer 100MB, and each
TCP should obtain a maximum of 100 Mb/sec per connection. So for
this example, the 100MB payload should be transferred across the
connections in approximately 8 seconds (which would be the ideal TCP
transfer time for these conditions).
Additionally, the TCP Efficiency metric SHOULD be computed for each
connection tested (defined in section 2.2).
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3.3.4 Interpretation of the TCP Throughput Results
At the end of this step, the user will document the theoretical BDP
and a set of Window size experiments with measured TCP throughput for
each TCP window size setting. For cases where the sustained TCP
throughput does not equal the ideal value, some possible causes
are listed:
- Network congestion causing packet loss which MAY be inferred from
a poor TCP Efficiency metric (100% = no loss)
- Network congestion causing an increase in RTT which MAY be inferred
from the Buffer Delay metric (0% = no increase in RTT over baseline)
- Intermediate network devices which actively regenerate the TCP
connection and can alter window size, MSS, etc.
- Rate limiting (policing). More discussion of traffic management
tests follows in section 3.4
3.4. Traffic Management Tests
In most cases, the network connection between two geographic
locations (branch offices, etc.) is lower than the network connection
of the host computers. An example would be LAN connectivity of GigE
and WAN connectivity of 100 Mbps. The WAN connectivity may be
physically 100 Mbps or logically 100 Mbps (over a GigE WAN
connection). In the later case, rate limiting is used to provide the
WAN bandwidth per the SLA.
Traffic management techniques are employed to provide various forms
of QoS, the more common include:
- Traffic Shaping
- Priority queuing
- Random Early Discard (RED, etc.)
Configuring the end-end network with these various traffic management
mechanisms is a complex under-taking. For traffic shaping and RED
techniques, the end goal is to provide better performance for bursty
traffic such as TCP (RED is specifically intended for TCP).
This section of the methodology provides guidelines to test traffic
shaping and RED implementations. As in section 3.3, host hardware
performance MUST be well understood before conducting the traffic
shaping and RED tests. Dedicated test equipment will generally be
REQUIRED for line rates of GigE and 10 GigE. If the throughput test
is expected to exceed 10% of the provider bandwidth, then the test
SHOULD be coordinated with the network provider. This does not
include the customer premise bandwidth, the 10% refers directly to
the provider's bandwidth (Provider Edge to Provider router).
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3.4.1 Traffic Shaping Tests
For services where the available bandwidth is rate limited, there are
two (2) techniques used to implement rate limiting: traffic policing
and traffic shaping.
Simply stated, traffic policing marks and/or drops packets which
exceed the SLA bandwidth (in most cases, excess traffic is dropped).
Traffic shaping employs the use of queues to smooth the bursty
traffic and then send out within the SLA bandwidth limit (without
dropping packets unless the traffic shaping queue is exceeded).
Traffic shaping is generally configured for TCP data services and
can provide improved TCP performance since the retransmissions are
reduced, which in turn optimizes TCP throughput for the given
available bandwidth. Through this section, the available
rate-limited bandwidth shall be referred to as the
"bottleneck bandwidth".
The ability to detect proper traffic shaping is more easily diagnosed
when conducting a multiple TCP connection test. Proper shaping will
provide a fair distribution of the available bottleneck bandwidth,
while traffic policing will not.
The traffic shaping tests are built upon the concepts of multiple
connection testing as defined in section 3.3.3. Calculating the BDP
for the bottleneck bandwidth is first REQUIRED before selecting the
number of connections and TCP Window size per connection.
Similar to the example in section 3.3, a typical test scenario might
be: GigE LAN with a 100Mbps bottleneck bandwidth (rate limited
logical interface), and 5 msec RTT. This would require five (5) TCP
connections of 64 KB window size evenly fill the bottleneck bandwidth
(about 100 Mbps per connection).
The traffic shaping test SHOULD be run over a long enough duration to
properly exercise network buffers (greater than 30 seconds) and also
characterize performance during different time periods of the day.
The throughput of each connection MUST be logged during the entire
test, along with the TCP Transfer Time, TCP Efficiency, and
Buffer Delay metrics.
3.4.1.1 Interpretation of Traffic Shaping Test Results
By plotting the throughput achieved by each TCP connection, the fair
sharing of the bandwidth is generally very obvious when traffic
shaping is properly configured for the bottleneck interface. For the
previous example of 5 connections sharing 500 Mbps, each connection
would consume ~100 Mbps with a smooth variation. If traffic policing
was present on the bottleneck interface, the bandwidth sharing MAY
not be fair and the resulting throughput plot MAY reveal "spikey"
throughput consumption of the competing TCP connections (due to the
retransmissions).
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3.4.2 RED Tests
Random Early Discard techniques are specifically targeted to provide
congestion avoidance for TCP traffic. Before the network element
queue "fills" and enters the tail drop state, RED drops packets at
configurable queue depth thresholds. This action causes TCP
connections to back-off which helps to prevent tail drop, which in
turn helps to prevent global TCP synchronization.
Again, rate limited interfaces can benefit greatly from RED based
techniques. Without RED, TCP is generally not able to achieve the
full bandwidth of the bottleneck interface. With RED enabled, TCP
congestion avoidance throttles the connections on the higher speed
interface (i.e. LAN) and can reach equilibrium with the bottleneck
bandwidth (achieving closer to full throughput).
The ability to detect proper RED configuration is more easily
diagnosed when conducting a multiple TCP connection test. Multiple
TCP connections provide the multiple bursty sources that emulate the
real-world conditions for which RED was intended.
The RED tests also build upon the concepts of multiple connection
testing as defined in section 3.3.3. Calculating the BDP for the
bottleneck bandwidth is first REQUIRED before selecting the number
of connections and TCP Window size per connection.
For RED testing, the desired effect is to cause the TCP connections
to burst beyond the bottleneck bandwidth so that queue drops will
occur. Using the same example from section 3.4.1 (traffic shaping),
the 500 Mbps bottleneck bandwidth requires 5 TCP connections (with
window size of 64Kb) to fill the capacity. Some experimentation is
REQUIRED, but it is RECOMMENDED to start with double the number of
connections to stress the network element buffers / queues. In this
example, 10 connections SHOULD produce TCP bursts of 64KB for each
connection. If the timing of the TCP tester permits, these TCP
bursts SHOULD stress queue sizes in the 512KB range. Again
experimentation will be REQUIRED and the proper number of TCP
connections and TCP window size will be dictated by the size the
network element queue.
3.4.2.1 Interpretation of RED Results
The default queuing technique for most network devices is FIFO based.
Without RED, the FIFO based queue will cause excessive loss to all of
the TCP connections and in the worst case global TCP synchronization.
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By plotting the aggregate throughput achieved on the bottleneck
interface, proper RED operation MAY be determined if the bottleneck
bandwidth is fully utilized. For the previous example of 10
connections (window = 64 KB) sharing 500 Mbps, each connection SHOULD
consume ~50 Mbps. If RED was not properly enabled on the interface,
then the TCP connections will retransmit at a higher rate and the
net effect is that the bottleneck bandwidth is not fully utilized.
Another means to study non-RED versus RED implementation is to use
the TCP Transfer Time metric for all of the connections. In this
example, a 100 MB payload transfer SHOULD take ideally 16 seconds
across all 10 connections (with RED enabled). With RED not enabled,
the throughput across the bottleneck bandwidth MAY be greatly
reduced (generally 20-40%) and the actual TCP Transfer time MAY be
proportionally longer then the Ideal TCP Transfer time.
Additionally, the TCP Transfer Efficiency metric is useful, since
non-RED implementations MAY exhibit a lower TCP Transfer Efficiency.
4. Security Considerations
The security considerations that apply to any active measurement of
live networks are relevant here as well. See [RFC4656] and
[RFC5357].
5. IANA Considerations
This document does not REQUIRE an IANA registration for ports
dedicated to the TCP testing described in this document.
6. Acknowledgments
Thanks to Matt Mathis, Matt Zekauskas, Al Morton, Rudi Geib, and
Yaakov Stein, Loki Jorgenson for many good comments and for pointing
us to great sources of information pertaining to past works in the
TCP capacity area.
7. References
7.1 Normative References
[RFC2119] Bradner, S., "Key words for use in RFCs to Indicate
Requirement Levels", BCP 14, RFC 2119, March 1997.
[RFC4656] Shalunov, S., Teitelbaum, B., Karp, A., Boote, J., and M.
Zekauskas, "A One-way Active Measurement Protocol
(OWAMP)", RFC 4656, September 2006.
[RFC5681] Allman, M., Paxson, V., Stevens W., "TCP Congestion
Control", RFC 5681, September 2009.
[RFC2544] Bradner, S., McQuaid, J., "Benchmarking Methodology for
Network Interconnect Devices", RFC 2544, June 1999
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[RFC5357] Hedayat, K., Krzanowski, R., Morton, A., Yum, K., Babiarz,
J., "A Two-Way Active Measurement Protocol (TWAMP)",
RFC 5357, October 2008
[RFC4821] Mathis, M., Heffner, J., "Packetization Layer Path MTU
Discovery", RFC 4821, June 2007
draft-ietf-ippm-btc-cap-00.txt Allman, M., "A Bulk
Transfer Capacity Methodology for Cooperating Hosts",
August 2001
[RFC2681] Almes G., Kalidindi S., Zekauskas, M., "A Round-trip Delay
Metric for IPPM", RFC 2681, September, 1999
[RFC4898] Mathis, M., Heffner, J., Raghunarayan, R., "TCP Extended
Statistics MIB", May 2007
[RFC5136] Chimento P., Ishac, J., "Defining Network Capacity",
February 2008
[RFC1323] Jacobson, V., Braden, R., Borman D., "TCP Extensions for
High Performance", May 1992
7.2. Informative References
Authors' Addresses
Barry Constantine
JDSU, Test and Measurement Division
One Milesone Center Court
Germantown, MD 20876-7100
USA
Phone: +1 240 404 2227
barry.constantine@jdsu.com
Gilles Forget
Independent Consultant to Bell Canada.
308, rue de Monaco, St-Eustache
Qc. CANADA, Postal Code : J7P-4T5
Phone: (514) 895-8212
gilles.forget@sympatico.ca
Reinhard Schrage
Schrage Consulting
Phone: +49 (0) 5137 909540
reinhard@schrageconsult.com
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