Framework for TCP Throughput Testing
draft-ietf-ippm-tcp-throughput-tm-13
The information below is for an old version of the document that is already published as an RFC.
| Document | Type |
This is an older version of an Internet-Draft that was ultimately published as RFC 6349.
|
|
|---|---|---|---|
| Authors | Reinhard Schrage , Gilles Forget , Ruediger Geib , Barry Constantine | ||
| Last updated | 2015-10-14 (Latest revision 2011-05-31) | ||
| Replaces | draft-constantine-ippm-tcp-throughput-tm | ||
| RFC stream | Internet Engineering Task Force (IETF) | ||
| Intended RFC status | Informational | ||
| Formats | |||
| Reviews | |||
| Additional resources | Mailing list discussion | ||
| Stream | WG state | WG Document | |
| Document shepherd | (None) | ||
| IESG | IESG state | Became RFC 6349 (Informational) | |
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||
| Consensus boilerplate | Unknown | ||
| Telechat date | (None) | ||
| Responsible AD | Wesley Eddy | ||
| IESG note | |||
| Send notices to | (None) |
draft-ietf-ippm-tcp-throughput-tm-13
Network Working Group B. Constantine
Internet-Draft JDSU
Intended status: Informational G. Forget
Expires: November 30, 2011 Bell Canada (Ext. Consultant)
Ruediger Geib
Deutsche Telekom
Reinhard Schrage
Schrage Consulting
May 31, 2011
Framework for TCP Throughput Testing
draft-ietf-ippm-tcp-throughput-tm-13.txt
Abstract
This framework describes a practical methodology for measuring end-
to-end TCP Throughput in a managed IP network. The goal is to provide
a better indication in regards to user experience. In this framework,
TCP and IP parameters are specified to optimize TCP throughput.
Requirements Language
The key words "MUST", "MUST NOT", "REQUIRED", "SHALL", "SHALL NOT",
"SHOULD", "SHOULD NOT", "RECOMMENDED", "MAY", and "OPTIONAL" in this
document are to be interpreted as described in RFC 2119 [RFC2119].
Status of this Memo
This Internet-Draft is submitted in full conformance with the
provisions of BCP 78 and BCP 79.
Internet-Drafts are working documents of the Internet Engineering
Task Force (IETF). Note that other groups may also distribute
working documents as Internet-Drafts. The list of current Internet-
Drafts is at http://datatracker.ietf.org/drafts/current/.
Internet-Drafts are draft documents valid for a maximum of six months
and may be updated, replaced, or obsoleted by other documents at any
time. It is inappropriate to use Internet-Drafts as reference
material or to cite them other than as "work in progress."
This Internet-Draft will expire on November 30, 2011.
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Copyright Notice
Copyright (c) 2011 IETF Trust and the persons identified as the
document authors. All rights reserved.
This document is subject to BCP 78 and the IETF Trust's Legal
Provisions Relating to IETF Documents
(http://trustee.ietf.org/license-info) in effect on the date of
publication of this document. Please review these documents
carefully, as they describe your rights and restrictions with respect
to this document. Code Components extracted from this document must
include Simplified BSD License text as described in Section 4.e of
the Trust Legal Provisions and are provided without warranty as
described in the Simplified BSD License.
Table of Contents
1. Introduction . . . . . . . . . . . . . . . . . . . . . . . . . 3
1.1 Terminology. . . . . . . . . . . . . . . . . . . . . . . . 4
1.2 TCP Equilibrium . . . . . . . . . . . . . . . . . . . . . 5
2. Scope and Goals . . . . . . . . . . . . . . . . . . . . . . . 6
3. Methodology. . . . . . . . . . . . . . . . . . . . . . . . . . 7
3.1 Path MTU . . . . . . . . . . . . . . . . . . . . . . . . . 9
3.2 Round Trip Time (RTT) and Bottleneck Bandwidth (BB). . . . 9
3.2.1 Measuring RTT . . . . . . . . . . . . . . . . . . . . 9
3.2.2 Measuring BB . . . . . . . . . . . . . . . . . . . . 10
3.3. Measuring TCP Throughput . . . . . . . . . . . . . . . . . 11
3.3.1 Minimum TCP RWND . . . . . . . . . . . . . . . . . . . 11
4. TCP Metrics . . . . . . . . . . . . . . . . . . . . . . . . . 14
4.1 Transfer Time Ratio. . . . . . . . . . . . . . . . . . . . 14
4.1.1 Maximum Achievable TCP Throughput calculation . . . . 15
4.1.2 Transfer Time and Transfer Time Ratio calculation. . . 16
4.2 TCP Efficiency . . . . . . . . . . . . . . . . . . . . . . 17
4.2.1 TCP Efficiency Percentage calculation . . . . . . . . 17
4.3 Buffer Delay . . . . . . . . . . . . . . . . . . . . . . . 17
4.3.1 Buffer Delay Percentage calculation. . . . . . . . . . 17
5. Conducting TCP Throughput Tests. . . . . . . . . . . . . . . . 18
5.1 Single versus Multiple Connections . . . . . . . . . . . . 18
5.2 Results Interpretation . . . . . . . . . . . . . . . . . . 19
6. Security Considerations . . . . . . . . . . . . . . . . . . . 21
7. IANA Considerations . . . . . . . . . . . . . . . . . . . . . 21
8. Acknowledgments . . . . . . . . . . . . . . . . . . . . . . . 22
9. References . . . . . . . . . . . . . . . . . . . . . . . . . . 22
9.1 Normative References . . . . . . . . . . . . . . . . . . . 22
9.2 Informative References . . . . . . . . . . . . . . . . . . 22
Authors' Addresses . . . . . . . . . . . . . . . . . . . . . . . . 23
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1. Introduction
In the network industry, the SLA (Service Level Agreement) provided
to business class customers is generally based upon Layer 2/3
criteria such as: Bandwidth, latency, packet loss and delay
variations (jitter). Network providers are coming to the realization
that Layer 2/3 testing is not enough to adequately ensure end-user's
satisfaction. In addition to Layer 2/3 testing, this framework
recommends a methodology for measuring TCP Throughput in order to
provide meaningful results with respect to user experience.
Additionally, business class customers seek to conduct repeatable TCP
Throughput tests between locations. Since these organizations rely on
the networks of the providers, a common test methodology with
predefined metrics would benefit both parties.
Note that the primary focus of this methodology is managed business
class IP networks; e.g. those Ethernet terminated services for which
organizations are provided an SLA from the network provider. Because
of the SLA, the expectation is that the TCP Throughput should achieve
the guaranteed bandwidth. End-users with "best effort" access could
use this methodology, but this framework and its metrics are intended
to be used in a predictable managed IP network. No end-to-end
performance can be guaranteed when only the access portion is being
provisioned to a specific bandwidth capacity.
The intent behind this document is to define a methodology for
testing sustained TCP Layer performance. In this document, the
achievable TCP Throughput is that amount of data per unit time that
TCP transports when in the TCP Equilibrium state. (See Section 1.2
for TCP Equilibrium definition). Throughout this document, maximum
achievable throughput refers to the theoretical achievable throughput
when TCP is in the Equilibrium state.
TCP is connection oriented and at the transmitting side it uses a
congestion window, (TCP CWND). At the receiving end, TCP uses a
receive window, (TCP RWND) to inform the transmitting end on how
many Bytes it is capable to accept at a given time.
Derived from Round Trip Time (RTT) and network Bottleneck Bandwidth
(BB), the Bandwidth Delay Product (BDP) determines the Send and
Received Socket buffers sizes required to achieve the maximum TCP
Throughput. Then, with the help of slow start and congestion
avoidance algorithms, a TCP CWND is calculated based on the IP
network path loss rate. Finally, the minimum value between the
calculated TCP CWND and the TCP RWND advertised by the opposite end
will determine how many Bytes can actually be sent by the
transmitting side at a given time.
Both TCP Window sizes (RWND and CWND) may vary during any given TCP
session, although up to bandwidth limits, larger RWND and larger CWND
will achieve higher throughputs by permitting more in-flight Bytes.
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At both ends of the TCP connection and for each socket, there are
default buffer sizes. There are also kernel enforced maximum buffer
sizes. These buffer sizes can be adjusted at both ends (transmitting
and receiving). Some TCP/IP stack implementations use Receive Window
Auto-Tuning, although in order to obtain the maximum throughput it is
critical to use large enough TCP Send and Receive Socket Buffer
sizes. In fact, they SHOULD be equal to or greater than BDP.
Many variables are involved in TCP Throughput performance, but this
methodology focuses on:
- BB (Bottleneck Bandwidth)
- RTT (Round Trip Time)
- Send and Receive Socket Buffers
- Minimum TCP RWND
- Path MTU (Maximum Transmission Unit)
This methodology proposes TCP testing that SHOULD be performed in
addition to traditional Layer 2/3 type tests. In fact, Layer 2/3
tests are REQUIRED to verify the integrity of the network before
conducting TCP tests. Examples include iperf (UDP mode) and manual
packet layer test techniques where packet throughput, loss, and delay
measurements are conducted. When available, standardized testing
similar to [RFC2544] but adapted for use in operational networks MAY
be used.
Note: [RFC2544] was never meant to be used outside a lab environment.
Sections 2 and 3 of this document provides a general overview of the
proposed methodology. Section 4 defines the metrics while Section 5
explains how to conduct the tests and interpret the results.
1.1 Terminology
The common definitions used in this methodology are:
- TCP Throughput Test Device (TCP TTD), refers to compliant TCP
host that generates traffic and measures metrics as defined in
this methodology. i.e. a dedicated communications test instrument.
- Customer Provided Equipment (CPE), refers to customer owned
equipment (routers, switches, computers, etc.)
- Customer Edge (CE), refers to provider owned demarcation device.
- Provider Edge (PE), refers to provider's distribution equipment.
- Bottleneck Bandwidth (BB), lowest bandwidth along the complete
path. Bottleneck Bandwidth and Bandwidth are used synonymously
in this document. Most of the time the Bottleneck Bandwidth is
in the access portion of the wide area network (CE - PE).
- Provider (P), refers to provider core network equipment.
- Network Under Test (NUT), refers to the tested IP network path.
- Round Trip Time (RTT), is the elapsed time between the clocking in
of the first bit of a TCP segment sent and the receipt of the last
bit of the corresponding TCP Acknowledgment.
- Bandwidth Delay Product (BDP), refers to the product of a data
link's capacity (in bits per second) and its end-to-end delay
(in seconds).
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Figure 1.1 Devices, Links and Paths
+----+ +----+ +----+ +----+ +---+ +---+ +----+ +----+ +----+ +----+
| TCP|-| CPE|-| CE |--| PE |-| P |--| P |-| PE |--| CE |-| CPE|-| TCP|
| TTD| | | | |BB| | | | | | | |BB| | | | | TTD|
+----+ +----+ +----+ +----+ +---+ +---+ +----+ +----+ +----+ +----+
<------------------------ NUT ------------------------->
R >-----------------------------------------------------------|
T |
T <-----------------------------------------------------------|
Note that the NUT may be built with of a variety of devices including
but not limited to, load balancers, proxy servers or WAN acceleration
appliances. The detailed topology of the NUT SHOULD be well known
when conducting the TCP Throughput tests, although this methodology
makes no attempt to characterize specific network architectures.
1.2 TCP Equilibrium
TCP connections have three (3) fundamental congestion window phases,
which are depicted in Figure 1.2.
1 - The Slow Start phase, which occurs at the beginning of a TCP
transmission or after a retransmission time out.
2 - The Congestion Avoidance phase, during which TCP ramps up to
establish the maximum achievable throughput. It is important to note
that retransmissions are a natural by-product of the TCP congestion
avoidance algorithm as it seeks to achieve maximum throughput.
3 - The Loss Recovery phase, which could include Fast Retransmit
(Tahoe) or Fast Recovery (Reno & New Reno). When packet loss occurs,
Congestion Avoidance phase transitions either to Fast Retransmission
or Fast Recovery depending upon the TCP implementation. If a Time-Out
occurs, TCP transitions back to the Slow Start phase.
Figure 1.2 TCP CWND Phases
/\ |
/\ |High ssthresh TCP CWND TCP
/\ |Loss Event * halving 3-Loss Recovery Equilibrium
/\ | * \ upon loss
/\ | * \ / \ Time-Out Adjusted
/\ | * \ / \ +--------+ * ssthresh
/\ | * \/ \ / Multiple| *
/\ | * 2-Congestion\ / Loss | *
/\ | * Avoidance \/ Event | *
TCP | * Half | *
Through- | * TCP CWND | * 1-Slow Start
put | * 1-Slow Start Min TCP CWND after T-O
+-----------------------------------------------------------
Time > > > > > > > > > > > > > > > > > > > > > > > > > > >
Note: ssthresh = Slow Start threshold.
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A well tuned and managed IP network with appropriate TCP adjustments
in the IP hosts and applications should perform very close to the
BB when TCP is in the Equilibrium state.
This TCP methodology provides guidelines to measure the maximum
achievable TCP Throughput when TCP is in the Equilibrium state.
All maximum achievable TCP Throughputs specified in Section 3.3 are
with respect to this condition.
It is important to clarify the interaction between the sender's Send
Socket Buffer and the receiver's advertised TCP RWND Size. TCP test
programs such as iperf, ttcp, etc. allows the sender to control the
quantity of TCP Bytes transmitted and unacknowledged (in-flight),
commonly referred to as the Send Socket Buffer. This is done
independently of the TCP RWND Size advertised by the receiver.
2. Scope and Goals
Before defining the goals, it is important to clearly define the
areas that are out-of-scope.
- This methodology is not intended to predict the TCP Throughput
during the transient stages of a TCP connection, such as during the
slow start phase.
- This methodology is not intended to definitively benchmark TCP
implementations of one OS to another, although some users may find
value in conducting qualitative experiments.
- This methodology is not intended to provide detailed diagnosis
of problems within end-points or within the network itself as
related to non-optimal TCP performance, although results
interpretation for each test step may provide insights to potential
issues.
- This methodology does not propose to operate permanently with high
measurement loads. TCP performance and optimization within
operational networks MAY be captured and evaluated by using data
from the "TCP Extended Statistics MIB" [RFC4898].
In contrast to the above exclusions, the primary goal is to define a
method to conduct a practical end-to-end assessment of sustained
TCP performance within a managed business class IP network. Another
key goal is to establish a set of "best practices" that a non-TCP
expert SHOULD apply when validating the ability of a managed IP
network to carry end-user TCP applications.
Specific goals are to:
- Provide a practical test approach that specifies tunable parameters
(such as MTU (Maximum Transmit Unit) and Socket Buffer sizes) and how
these affect the outcome of TCP performances over an IP network.
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- Provide specific test conditions like link speed, RTT, MTU, Socket
Buffer sizes and achievable TCP Throughput when TCP is in the
Equilibrium state. For guideline purposes, provide examples of
test conditions and their maximum achievable TCP Throughput.
Section 1.2 provides specific details concerning the definition of
TCP Equilibrium within this methodology while Section 3 provides
specific test conditions with examples.
- Define three (3) basic metrics to compare the performance of TCP
connections under various network conditions. See Section 4.
- In test situations where the recommended procedure does not yield
the maximum achievable TCP Throughput, this methodology provides
some areas within the end host or the network that SHOULD be
considered for investigation. Although again, this methodology
is not intended to provide detailed diagnosis on these issues.
See Section 5.2.
3. Methodology
This methodology is intended for operational and managed IP networks.
A multitude of network architectures and topologies can be tested.
The diagram in Figure 1.1 is very general and is only there to
illustrate typical segmentation within end-user and network provider
domains.
Also, as stated earlier in Section 1, it is considered best practice
to verify the integrity of the network by conducting Layer 2/3 tests
such as [RFC2544] or other methods of network stress tests.
Although, it is important to mention here that [RFC2544] was never
meant to be used outside a lab environment.
It is not possible to make an accurate TCP Throughput measurement
when the network is dysfunctional. In particular, if the network is
exhibiting high packet loss and/or high jitter, then TCP Layer
Throughput testing will not be meaningful. As a guideline 5% packet
loss and/or 150 ms of jitter may be considered too high for an
accurate measurement.
TCP Throughput testing may require cooperation between the end-user
customer and the network provider. As an example, in an MPLS (Multi-
Protocol Label Switching) network architecture, the testing SHOULD be
conducted either on the CPE or on the CE device and not on the PE
(Provider Edge) router.
The following represents the sequential order of steps for this
testing methodology:
1 - Identify the Path MTU. Packetization Layer Path MTU Discovery
or PLPMTUD, [RFC4821], SHOULD be conducted. It is important to
identify the path MTU so that the TCP TTD is configured properly to
avoid fragmentation.
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2 - Baseline Round Trip Time and Bandwidth. This step establishes the
inherent, non-congested Round Trip Time (RTT) and the Bottleneck
Bandwidth (BB) of the end-to-end network path. These measurements
are used to provide estimates of the TCP RWND and Send Socket Buffer
Sizes that SHOULD be used during subsequent test steps.
3 - TCP Connection Throughput Tests. With baseline measurements
of Round Trip Time and Bottleneck Bandwidth, single and multiple TCP
connection throughput tests SHOULD be conducted to baseline network
performances.
These three (3) steps are detailed in Sections 3.1 - 3.3.
Important to note are some of the key characteristics and
considerations for the TCP test instrument. The test host MAY be a
standard computer or a dedicated communications test instrument.
In both cases, it MUST be capable of emulating both a client and a
server.
The following criteria SHOULD be considered when selecting whether
the TCP test host can be a standard computer or has to be a dedicated
communications test instrument:
- TCP implementation used by the test host, OS version, i.e. LINUX OS
kernel using TCP New Reno, TCP options supported, etc. These will
obviously be more important when using dedicated communications test
instruments where the TCP implementation may be customized or tuned
to run in higher performance hardware. When a compliant TCP TTD is
used, the TCP implementation SHOULD be identified in the test
results. The compliant TCP TTD SHOULD be usable for complete
end-to-end testing through network security elements and SHOULD also
be usable for testing network sections.
- More important, the TCP test host MUST be capable to generate
and receive stateful TCP test traffic at the full BB of the NUT.
Stateful TCP test traffic means that the test host MUST fully
implement a TCP/IP stack; this is generally a comment aimed at
dedicated communications test equipments which sometimes "blast"
packets with TCP headers. As a general rule of thumb, testing TCP
Throughput at rates greater than 100 Mbps may require high
performance server hardware or dedicated hardware based test tools.
- A compliant TCP Throughput Test Device MUST allow adjusting both
Send and Receive Socket Buffer sizes. The Socket Buffers MUST be
large enough to fill the BDP.
- Measuring RTT and retransmissions per connection will generally
require a dedicated communications test instrument. In the absence of
dedicated hardware based test tools, these measurements may need to
be conducted with packet capture tools, i.e. conduct TCP Throughput
tests and analyze RTT and retransmissions in packet captures.
Another option MAY be to use "TCP Extended Statistics MIB" per
[RFC4898].
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- The [RFC4821] PLPMTUD test SHOULD be conducted with a dedicated
tester which exposes the ability to run the PLPMTUD algorithm
independently from the OS stack.
3.1. Path MTU
TCP implementations should use Path MTU Discovery techniques (PMTUD).
PMTUD relies on ICMP 'need to frag' messages to learn the path MTU.
When a device has a packet to send which has the Don't Fragment (DF)
bit in the IP header set and the packet is larger than the (MTU) of
the next hop, the packet is dropped and the device sends an ICMP
'need to frag' message back to the host that originated the packet.
The ICMP 'need to frag' message includes the next hop MTU which PMTUD
uses to adjust itself. Unfortunately, because many network managers
completely disable ICMP, this technique does not always prove
reliable.
Packetization Layer Path MTU Discovery or PLPMTUD [RFC4821] MUST then
be conducted to verify the network path MTU. PLPMTUD can be used
with or without ICMP. [RFC4821] specifies search_high and search_low
parameters for the MTU and we recommend to use those. The goal is to
avoid fragmentation during all subsequent tests.
3.2. Round Trip Time (RTT) and Bottleneck Bandwidth (BB)
Before stateful TCP testing can begin, it is important to determine
the baseline RTT (i.e. non-congested inherent delay) and BB of the
end-to-end network to be tested. These measurements are used to
calculate the BDP and to provide estimates of the TCP RWND and
Send Socket Buffer Sizes that SHOULD be used in subsequent test
steps.
3.2.1 Measuring RTT
As previously defined in Section 1.1, RTT is the elapsed time
between the clocking in of the first bit of a TCP segment sent
and the receipt of the last bit of the corresponding TCP
Acknowledgment.
The RTT SHOULD be baselined during off-peak hours in order to obtain
a reliable figure of the inherent network latency. Otherwise,
additional delay caused by network buffering can occur. Also, when
sampling RTT values over a given test interval, the minimum
measured value SHOULD be used as the baseline RTT. This will most
closely estimate the real inherent RTT. This value is also used to
determine the Buffer Delay Percentage metric defined in Section 4.3.
The following list is not meant to be exhaustive, although it
summarizes some of the most common ways to determine Round Trip Time.
The desired measurement precision (i.e. ms versus us) may dictate
whether the RTT measurement can be achieved with ICMP pings or by a
dedicated communications test instrument with precision timers. The
objective in this section is to list several techniques in order of
decreasing accuracy.
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- Use test equipment on each end of the network, "looping" the
far-end tester so that a packet stream can be measured back and forth
from end-to-end. This RTT measurement may be compatible with delay
measurement protocols specified in [RFC5357].
- Conduct packet captures of TCP test sessions using "iperf" or FTP,
or other TCP test applications. By running multiple experiments,
packet captures can then be analyzed to estimate RTT. It is
important to note that results based upon the SYN -> SYN-ACK at the
beginning of TCP sessions SHOULD be avoided since Firewalls might
slow down 3 way handshakes. Also, at the senders side, Ostermann's
LINUX TCPTRACE utility with -l -r arguments can be used to extract
the RTT results directly from the packet captures.
- Obtain RTT statistics available from MIBs defined in [RFC4898].
- ICMP pings may also be adequate to provide Round Trip Time
estimates, provided that the packet size is factored into the
estimates (i.e. pings with different packet sizes might be required).
Some limitations with ICMP Ping may include ms resolution and
whether the network elements are responding to pings or not. Also,
ICMP is often rate-limited or segregated into different buffer
queues. ICMP might not work if QoS (Quality of Service)
reclassification is done at any hop. ICMP is not as reliable and
accurate as in-band measurements.
3.2.2 Measuring BB
Before any TCP Throughput test can be conducted, bandwidth
measurement tests SHOULD be run with stateless IP streams (i.e. not
stateful TCP) in order to determine the BB of the NUT.
These measurements SHOULD be conducted in both directions,
especially in asymmetrical access networks (e.g. ADSL access). These
tests SHOULD be performed at various intervals throughout a business
day or even across a week.
Testing at various time intervals would provide a better
characterization of TCP throughput and better diagnosis insight (for
cases where there are TCP performance issues). The bandwidth tests
SHOULD produce logged outputs of the achieved bandwidths across the
complete test duration.
There are many well established techniques available to provide
estimated measures of bandwidth over a network. It is a common
practice for network providers to conduct Layer 2/3 bandwidth
capacity tests using [RFC2544], although it is understood that
[RFC2544] was never meant to be used outside a lab environment.
These bandwidth measurements SHOULD use network capacity
techniques as defined in [RFC5136].
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3.3. Measuring TCP Throughput
This methodology specifically defines TCP Throughput measurement
techniques to verify maximum achievable TCP performance in a managed
business class IP network.
With baseline measurements of RTT and BB from Section 3.2, a series
of single and / or multiple TCP connection throughput tests SHOULD
be conducted.
The number of trials and single versus multiple TCP connections
choice will be based on the intention of the test. A single TCP
connection test might be enough to measure the achievable throughput
of a Metro Ethernet connectivity. Although, it is important to note
that various traffic management techniques can be used in an IP
network and that some of those can only be tested with multiple
connections. As an example, multiple TCP sessions might be required
to detect traffic shaping versus policing. Multiple sessions might
also be needed to measure Active Queue Management performances.
However, traffic management testing is not within the scope of this
test methodology.
In all circumstances, it is RECOMMENDED to run the tests in each
direction independently first and then to run in both directions
simultaneously. It is also RECOMMENDED to run the tests at
different times of day.
In each case, the TCP Transfer Time Ratio, the TCP Efficiency
Percentage, and the Buffer Delay Percentage MUST be measured in
each direction. These 3 metrics are defined in Section 4.
3.3.1 Minimum TCP RWND
The TCP TTD MUST allow the Send Socket Buffer and Receive Window
sizes to be set higher than the BDP, otherwise TCP performance will
be limited. In the business customer environment, these settings are
not generally adjustable by the average user. These settings are
either hard coded in the application or configured within the OS as
part of a corporate image. In many cases, the user's host Send
Socket Buffer and Receive Window size settings are not optimal.
This section provides derivations of BDPs under various network
conditions. It also provides examples of achievable TCP Throughput
with various TCP RWND sizes. This provides important guidelines
showing what can be achieved with settings higher than the BDP,
versus what would be achieved in a variety of real world conditions.
The minimum required TCP RWND Size can be calculated from the
Bandwidth Delay Product (BDP), which is:
BDP (bits) = RTT (sec) x BB (bps)
Note that the RTT is being used as the "Delay" variable for the BDP.
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Then, by dividing the BDP by 8, we obtain the minimum required TCP
RWND Size in Bytes. For optimal results, the Send Socket Buffer
MUST be adjusted to the same value at each end of the network.
Minimum required TCP RWND = BDP / 8
As an example on a T3 link with 25 ms RTT, the BDP would equal
~1,105,000 bits and the minimum required TCP RWND would be ~138 KB.
Note that separate calculations are REQUIRED on asymmetrical paths.
An asymmetrical path example would be a 90 ms RTT ADSL line with
5Mbps downstream and 640Kbps upstream. The downstream BDP would equal
~450,000 bits while the upstream one would be only ~57,600 bits.
The following table provides some representative network Link Speeds,
RTT, BDP, and their associated minimum required TCP RWND Sizes.
Table 3.3.1: Link Speed, RTT, calculated BDP & min. TCP RWND
Link Minimum required
Speed* RTT BDP TCP RWND
(Mbps) (ms) (bits) (KBytes)
---------------------------------------------------------------------
1.536 20.00 30,720 3.84
1.536 50.00 76,800 9.60
1.536 100.00 153,600 19.20
44.210 10.00 442,100 55.26
44.210 15.00 663,150 82.89
44.210 25.00 1,105,250 138.16
100.000 1.00 100,000 12.50
100.000 2.00 200,000 25.00
100.000 5.00 500,000 62.50
1,000.000 0.10 100,000 12.50
1,000.000 0.50 500,000 62.50
1,000.000 1.00 1,000,000 125.00
10,000.000 0.05 500,000 62.50
10,000.000 0.30 3,000,000 375.00
* Note that link speed is the BB for the NUT
In the above table, the following serial link speeds are used:
- T1 = 1.536 Mbps (for a B8ZS line encoding facility)
- T3 = 44.21 Mbps (for a C-Bit Framing facility)
The previous table illustrates the minimum required TCP RWND.
If a smaller TCP RWND Size is used, then the TCP Throughput
can not be optimal. To calculate the TCP Throughput, the following
formula is used: TCP Throughput = TCP RWND X 8 / RTT
An example could be a 100 Mbps IP path with 5 ms RTT and a TCP RWND
of 16KB, then:
TCP Throughput = 16 KBytes X 8 bits / 5 ms.
TCP Throughput = 128,000 bits / 0.005 sec.
TCP Throughput = 25.6 Mbps.
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Another example for a T3 using the same calculation formula is
illustrated in Figure 3.3.1a:
TCP Throughput = 16 KBytes X 8 bits / 10 ms.
TCP Throughput = 128,000 bits / 0.01 sec.
TCP Throughput = 12.8 Mbps. *
When the TCP RWND Size exceeds the BDP (T3 link and 64 KBytes TCP
RWND on a 10 ms RTT path), the maximum frames per second limit of
3664 is reached and then the formula is:
TCP Throughput = Max FPS X (MTU - 40) X 8.
TCP Throughput = 3664 FPS X 1460 Bytes X 8 bits.
TCP Throughput = 42.8 Mbps. **
The following diagram compares achievable TCP Throughputs on a T3
with Send Socket Buffer & TCP RWND Sizes of 16KB vs. 64KB.
Figure 3.3.1a TCP Throughputs on a T3 at different RTTs
45|
| _______**42.8
40| |64KB |
TCP | | |
Throughput 35| | |
in Mbps | | | +-----+34.1
30| | | |64KB |
| | | | |
25| | | | |
| | | | |
20| | | | | _______20.5
| | | | | |64KB |
15| | | | | | |
|*12.8+-----| | | | | |
10| |16KB | | | | | |
| | | |8.5 +-----| | | |
5| | | | |16KB | |5.1 +-----| |
|_____|_____|_____|____|_____|_____|____|16KB |_____|_____
10 15 25
RTT in milliseconds
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The following diagram shows the achievable TCP Throughput on a 25 ms
T3 when Send Socket Buffer & TCP RWND Sizes are increased.
Figure 3.3.1b TCP Throughputs on a T3 with different TCP RWND
45|
|
40| +-----+40.9
TCP | | |
Throughput 35| | |
in Mbps | | |
30| | |
| | |
25| | |
| | |
20| +-----+20.5 | |
| | | | |
15| | | | |
| | | | |
10| +-----+10.2 | | | |
| | | | | | |
5| +-----+5.1 | | | | | |
|_____|_____|______|_____|______|_____|_______|_____|_____
16 32 64 128*
TCP RWND Size in KBytes
* Note that 128KB requires [RFC1323] TCP Window scaling option.
4. TCP Metrics
This methodology focuses on a TCP Throughput and provides 3 basic
metrics that can be used for better understanding of the results.
It is recognized that the complexity and unpredictability of TCP
makes it very difficult to develop a complete set of metrics that
accounts for the myriad of variables (i.e. RTT variations, loss
conditions, TCP implementations, etc.). However, these 3 metrics
facilitate TCP Throughput comparisons under varying network
conditions and host buffer size / RWND settings.
4.1 Transfer Time Ratio
The first metric is the TCP Transfer Time Ratio, which is simply the
ratio between the Actual versus the Ideal TCP Transfer Times.
The Actual TCP Transfer Time, is simply the time it takes to transfer
a block of data across TCP connection(s).
The Ideal TCP Transfer Time is the predicted time for which a block
of data SHOULD transfer across TCP connection(s) considering the BB
of the NUT.
Actual TCP Transfer Time
TCP Transfer Time Ratio = -------------------------
Ideal TCP Transfer Time
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The Ideal TCP Transfer Time is derived from the Maximum Achievable
TCP Throughput, which is related to the BB and Layer 1/2/3/4
overheads associated with the network path. The following sections
provide derivations for the Maximum Achievable TCP Throughput and
example calculations for the TCP Transfer Time Ratio.
4.1.1 Maximum Achievable TCP Throughput calculation
This section provides formulas to calculate the Maximum Achievable
TCP Throughput with examples for T3 (44.21 Mbps) and Ethernet.
All calculations are based on IP version 4 with TCP/IP headers of
20 Bytes each (20 for TCP + 20 for IP) within an MTU of 1500 Bytes.
First, the maximum achievable Layer 2 throughput of a T3 Interface
is limited by the maximum quantity of Frames Per Second (FPS)
permitted by the actual physical layer (Layer 1) speed.
The calculation formula is:
FPS = T3 Physical Speed / ((MTU + PPP + Flags + CRC16) X 8)
FPS = (44.21Mbps /((1500 Bytes + 4 Bytes + 2 Bytes + 2 Bytes) X 8 )))
FPS = (44.21Mbps /(1508 Bytes X 8))
FPS = 44.21Mbps / 12064 bits
FPS = 3664
Then, to obtain the Maximum Achievable TCP Throughput (Layer 4), we
simply use: (MTU - 40) in Bytes X 8 bits X max FPS.
For a T3, the maximum TCP Throughput = 1460 Bytes X 8 bits X 3664 FPS
Maximum TCP Throughput = 11680 bits X 3664 FPS
Maximum TCP Throughput = 42.8 Mbps.
On Ethernet, the maximum achievable Layer 2 throughput is limited by
the maximum Frames Per Second permitted by the IEEE802.3 standard.
The maximum FPS for 100 Mbps Ethernet is 8127 and the calculation is:
FPS = (100Mbps /(1538 Bytes X 8 bits))
The maximum FPS for GigE is 81274 and the calculation formula is:
FPS = (1Gbps /(1538 Bytes X 8 bits))
The maximum FPS for 10GigE is 812743 and the calculation formula is:
FPS = (10Gbps /(1538 Bytes X 8 bits))
The 1538 Bytes equates to:
MTU + Ethernet + CRC32 + IFG + Preamble + SFD
(IFG = Inter-Frame Gap and SFD = Start of Frame Delimiter)
Where MTU is 1500 Bytes, Ethernet is 14 Bytes, CRC32 is 4 Bytes,
IFG is 12 Bytes, Preamble is 7 Bytes and SFD is 1 Byte.
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Then, to obtain the Maximum Achievable TCP Throughput (Layer 4), we
simply use: (MTU - 40) in Bytes X 8 bits X max FPS.
For a 100Mbps, the max TCP Throughput = 1460Bytes X 8 bits X 8127 FPS
Maximum TCP Throughput = 11680 bits X 8127 FPS
Maximum TCP Throughput = 94.9 Mbps.
It is important to note that better results could be obtained with
jumbo frames on Gigabit and 10 Gigabit Ethernet interfaces.
4.1.2 TCP Transfer Time and Transfer Time Ratio calculation
The following table illustrates the Ideal TCP Transfer time of a
single TCP connection when its TCP RWND and Send Socket Buffer Sizes
equals or exceeds the BDP.
Table 4.1.1: Link Speed, RTT, BDP, TCP Throughput, and
Ideal TCP Transfer time for a 100 MB File
Link Maximum Ideal TCP
Speed BDP Achievable TCP Transfer time
(Mbps) RTT (ms) (KBytes) Throughput(Mbps) (seconds)*
--------------------------------------------------------------------
1.536 50.00 9.6 1.4 571.0
44.210 25.00 138.2 42.8 18.0
100.000 2.00 25.0 94.9 9.0
1,000.000 1.00 125.0 949.2 1.0
10,000.000 0.05 62.5 9,492.0 0.1
* Transfer times are rounded for simplicity.
For a 100MB file (100 x 8 = 800 Mbits), the Ideal TCP Transfer Time
is derived as follows:
800 Mbits
Ideal TCP Transfer Time = -----------------------------------
Maximum Achievable TCP Throughput
To illustrate the TCP Transfer Time Ratio, an example would be the
bulk transfer of 100 MB over 5 simultaneous TCP connections (each
connection transferring 100 MB). In this example, the Ethernet
service provides a Committed Access Rate (CAR) of 500 Mbps. Each
connection may achieve different throughputs during a test and the
overall throughput rate is not always easy to determine (especially
as the number of connections increases).
The ideal TCP Transfer Time would be ~8 seconds, but in this example,
the actual TCP Transfer Time was 12 seconds. The TCP Transfer Time
Ratio would then be 12/8 = 1.5, which indicates that the transfer
across all connections took 1.5 times longer than the ideal.
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4.2 TCP Efficiency
The second metric represents the percentage of Bytes that were not
retransmitted.
Transmitted Bytes - Retransmitted Bytes
TCP Efficiency % = --------------------------------------- X 100
Transmitted Bytes
Transmitted Bytes are the total number of TCP Bytes to be transmitted
including the original and the retransmitted Bytes.
4.2.1 TCP Efficiency Percentage calculation
As an example, if 100,000 Bytes were sent and 2,000 had to be
retransmitted, the TCP Efficiency Percentage would be calculated as:
102,000 - 2,000
TCP Efficiency % = ----------------- x 100 = 98.03%
102,000
Note that the Retransmitted Bytes may have occurred more than once,
if so, then these multiple retransmissions are added to the
Retransmitted Bytes and to the Transmitted Bytes counts.
4.3 Buffer Delay
The third metric is the Buffer Delay Percentage, which represents
the increase in RTT during a TCP Throughput test versus the inherent
or baseline RTT. The baseline RTT is the Round Trip Time inherent to
the network path under non-congested conditions as defined in Section
3.2.1. The average RTT is derived from the total of all measured
RTTs during the actual test at every second divided by the test
duration in seconds.
Total RTTs during transfer
Average RTT during transfer = -----------------------------
Transfer duration in seconds
Average RTT during Transfer - Baseline RTT
Buffer Delay % = ------------------------------------------ X 100
Baseline RTT
4.3.1 Buffer Delay calculation
As an example, consider a network path with a baseline RTT of 25 ms.
During the course of a TCP transfer, the average RTT across
the entire transfer increases to 32 ms. Then, the Buffer Delay
Percentage would be calculated as:
32 - 25
Buffer Delay % = ------- x 100 = 28%
25
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Note that the TCP Transfer Time Ratio, TCP Efficiency Percentage, and
the Buffer Delay Percentage MUST all be measured during each
throughput test. Poor TCP Transfer Time Ratio (i.e. TCP Transfer
Time greater than the Ideal TCP Transfer Time) may be diagnosed by
correlating with sub-optimal TCP Efficiency Percentage and/or Buffer
Delay Percentage metrics.
5. Conducting TCP Throughput Tests
Several TCP tools are currently used in the network world and one of
the most common is "iperf". With this tool, hosts are installed at
each end of the network path; one acts as client and the other as
a server. The Send Socket Buffer and the TCP RWND Sizes of both
client and server can be manually set. The achieved throughput can
then be measured, either uni-directionally or bi-directionally. For
higher BDP situations in lossy networks (Long Fat Networks (LFNs) or
satellite links, etc.), TCP options such as Selective Acknowledgment
SHOULD become part of the window size / throughput characterization.
Host hardware performance must be well understood before conducting
the tests described in the following sections. A dedicated
communications test instrument will generally be REQUIRED, especially
for line rates of GigE and 10 GigE. A compliant TCP TTD SHOULD
provide a warning message when the expected test throughput will
exceed the subscribed customer SLA. If the throughput test is
expected to exceed the subscribed customer SLA, then the test
SHOULD be coordinated with the network provider.
The TCP Throughput test SHOULD be run over a long enough duration
to properly exercise network buffers (i.e. greater than 30 seconds)
and SHOULD also characterize performance at different times of day.
5.1 Single versus Multiple TCP Connections
The decision whether to conduct single or multiple TCP connection
tests depends upon the size of the BDP in relation to the TCP RWND
configured in the end-user environment. For example, if the BDP for
a Long Fat Network (LFN) turns out to be 2MB, then it is probably
more realistic to test this network path with multiple connections.
Assuming typical host TCP RWND Sizes of 64 KB (i.e. Windows XP),
using 32 TCP connections would emulate a small office scenario.
The following table is provided to illustrate the relationship
between the TCP RWND and the number of TCP connections required to
fill the available capacity of a given BDP. For this example, the
network bandwidth is 500 Mbps and the RTT is 5 ms, then the BDP
equates to 312.5 KBytes.
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Table 5.1 Number of TCP connections versus TCP RWND
Number of TCP Connections
TCP RWND to fill available bandwidth
-------------------------------------
16KB 20
32KB 10
64KB 5
128KB 3
The TCP Transfer Time Ratio metric is useful when conducting multiple
connection tests. Each connection SHOULD be configured to transfer
payloads of the same size (i.e. 100 MB), then the TCP Transfer Time
Ratio provides a simple metric to verify the actual versus expected
results.
Note that the TCP Transfer Time is the time required for each
connection to complete the transfer of the predetermined payload
size. From the previous table, the 64KB window is considered. Each
of the 5 TCP connections would be configured to transfer 100MB, and
each one should obtain a maximum of 100 Mbps. So for this example,
the 100MB payload should be transferred across the connections in
approximately 8 seconds (which would be the Ideal TCP Transfer Time
under these conditions).
Additionally, the TCP Efficiency Percentage metric MUST be computed
for each connection as defined in Section 4.2.
5.2 Results Interpretation
At the end, a TCP Throughput Test Device (TCP TTD) SHOULD generate a
report with the calculated BDP and a set of Window Size experiments.
Window Size refers to the minimum of the Send Socket Buffer and TCP
RWND. The report SHOULD include TCP Throughput results for each TCP
Window Size tested. The goal is to provide clear achievable versus
actual TCP Throughputs results with respect to the TCP Window Size
when no fragmentation occurs. The report SHOULD also include the
results for the 3 metrics defined in Section 4. The goal is to
provide a clear relationship between these 3 metrics and user
experience. As an example, for the same results in regards with
Transfer Time Ratio, a better TCP Efficiency could be obtained at the
cost of higher Buffer Delays.
For cases where the test results are not equal to the ideal values,
some possible causes are:
- Network congestion causing packet loss which may be inferred from
a poor TCP Efficiency % (i.e., higher TCP Efficiency % = less packet
loss)
- Network congestion causing an increase in RTT which may be inferred
from the Buffer Delay Percentage (i.e., 0% = no increase in RTT over
baseline)
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- Intermediate network devices which actively regenerate the TCP
connection and can alter TCP RWND Size, MTU, etc.
- Rate limiting by policing instead of shaping.
- Maximum TCP Buffer space. All operating systems have a global
mechanism to limit the quantity of system memory to be used by TCP
connections. On some systems, each connection is subject to a memory
limit that is applied to the total memory used for input data, output
data and controls. On other systems, there are separate limits for
input and output buffer spaces per connection. Client/server IP
hosts might be configured with Maximum Buffer Space limits that are
far too small for high performance networks.
- Socket Buffer Sizes. Most operating systems support separate per
connection send and receive buffer limits that can be adjusted as
long as they stay within the maximum memory limits. These socket
buffers MUST be large enough to hold a full BDP of TCP Bytes plus
some overhead. There are several methods that can be used to adjust
socket buffer sizes, but TCP Auto-Tuning automatically adjusts these
as needed to optimally balance TCP performance and memory usage.
It is important to note that Auto-Tuning is enabled by default in
LINUX since the kernel release 2.6.6 and in UNIX since FreeBSD 7.0.
It is also enabled by default in Windows since Vista and in MAC since
OS X version 10.5 (leopard). Over buffering can cause some
applications to behave poorly, typically causing sluggish interactive
response and risk running the system out of memory. Large default
socket buffers have to be considered carefully on multi-user systems.
- TCP Window Scale Option, [RFC1323]. This option enables TCP to
support large BDP paths. It provides a scale factor which is
required for TCP to support window sizes larger than 64KB. Most
systems automatically request WSCALE under some conditions, such as
when the receive socket buffer is larger than 64KB or when the other
end of the TCP connection requests it first. WSCALE can only be
negotiated during the 3 way handshake. If either end fails to
request WSCALE or requests an insufficient value, it cannot be
renegotiated. Different systems use different algorithms to select
WSCALE, but it is very important to have large enough buffer
sizes. Note that under these constraints, a client application
wishing to send data at high rates may need to set its own receive
buffer to something larger than 64K Bytes before it opens the
connection to ensure that the server properly negotiates WSCALE.
A system administrator might have to explicitly enable [RFC1323]
extensions. Otherwise, the client/server IP host would not support
TCP window sizes (BDP) larger than 64KB. Most of the time,
performance gains will be obtained by enabling this option in LFNs.
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- TCP Timestamps Option, [RFC1323]. This feature provides better
measurements of the Round Trip Time and protects TCP from data
corruption that might occur if packets are delivered so late that the
sequence numbers wrap before they are delivered. Wrapped sequence
numbers do not pose a serious risk below 100 Mbps, but the risk
increases at higher data rates. Most of the time, performance gains
will be obtained by enabling this option in Gigabit bandwidth
networks.
- TCP Selective Acknowledgments Option (SACK), [RFC2018]. This allows
a TCP receiver to inform the sender about exactly which data segment
is missing and needs to be retransmitted. Without SACK, TCP has to
estimate which data segment is missing, which works just fine if all
losses are isolated (i.e. only one loss in any given round trip).
Without SACK, TCP takes a very long time to recover after multiple
and consecutive losses. SACK is now supported by most operating
systems, but it may have to be explicitly enabled by the system
administrator. In networks with unknown load and error patterns, TCP
SACK will improve throughput performances. On the other hand,
security appliances vendors might have implemented TCP randomization
without considering TCP SACK and under such circumstances, SACK might
need to be disabled in the client/server IP hosts until the vendor
corrects the issue. Also, poorly implemented SACK algorithms might
cause extreme CPU loads and might need to be disabled.
- Path MTU. The client/server IP host system SHOULD use the largest
possible MTU for the path. This may require enabling Path MTU
Discovery [RFC1191] & [RFC4821]. Since [RFC1191] is flawed, it is
sometimes not enabled by default and may need to be explicitly
enabled by the system administrator. [RFC4821] describes a new, more
robust algorithm for MTU discovery and ICMP black hole recovery.
- TOE (TCP Offload Engine). Some recent Network Interface Cards (NIC)
are equipped with drivers that can do part or all of the TCP/IP
protocol processing. TOE implementations require additional work
(i.e. hardware-specific socket manipulation) to set up and tear down
connections. Because TOE NICs configuration parameters are vendor
specific and not necessarily RFC-compliant, they are poorly
integrated with UNIX & LINUX. Occasionally, TOE might need to be
disabled in a server because its NIC does not have enough memory
resources to buffer thousands of connections.
Note that both ends of a TCP connection MUST be properly tuned.
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6. Security Considerations
Measuring TCP network performance raises security concerns. Metrics
produced within this framework may create security issues.
6.1 Denial of Service Attacks
TCP network performance metrics, as defined in this document attempts
to fill the NUT with a stateful connection. However, since the test
MAY use stateless IP streams as specified in Section 3.2.2, it might
appear to network operators as a Denial Of Service attack. Thus, as
mentioned at the beginning of section 3, TCP Throughput testing may
require cooperation between the end-user customer and the network
provider.
6.2 User data confidentiality
Metrics within this framework generate packets from a sample, rather
than taking samples based on user data. Thus, our framework does not
threaten user data confidentiality.
6.3 Interference with metrics
The security considerations that apply to any active measurement of
live networks are relevant here as well. See [RFC4656] and
[RFC5357].
7. IANA Considerations
This document does not REQUIRE an IANA registration for ports
dedicated to the TCP testing described in this document.
8. Acknowledgments
Thanks to Lars Eggert, Al Morton, Matt Mathis, Matt Zekauskas,
Yaakov Stein, and Loki Jorgenson for many good comments and for
pointing us to great sources of information pertaining to past works
in the TCP capacity area.
9. References
9.1 Normative References
[RFC1191] Mogul, A., Deering, S., "Path MTU Discovery", 1990
[RFC1323] Jacobson, V., Braden, R., Borman D., "TCP Extensions for
High Performance", May 1992
[RFC2018] Mathis, M., Mahdavi, J., Floyd, S., Romanow, A., "TCP
Selective Acknowledgment Options", 1996
[RFC2119] Bradner, S., "Key words for use in RFCs to Indicate
Requirement Levels", BCP 14, RFC 2119, March 1997.
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[RFC2544] Bradner, S., McQuaid, J., "Benchmarking Methodology for
Network Interconnect Devices", RFC 2544, June 1999
[RFC4656] Shalunov, S., Teitelbaum, B., Karp, A., Boote, J., and M.
Zekauskas, "A One-way Active Measurement Protocol
(OWAMP)", RFC 4656, September 2006.
[RFC4821] Mathis, M., Heffner, J., "Packetization Layer Path MTU
Discovery", RFC 4821, June 2007
[RFC4898] Mathis, M., Heffner, J., Raghunarayan, R., "TCP Extended
Statistics MIB", May 2007
[RFC5136] Chimento P., Ishac, J., "Defining Network Capacity",
February 2008
[RFC5357] Hedayat, K., Krzanowski, R., Morton, A., Yum, K., Babiarz,
J., "A Two-Way Active Measurement Protocol (TWAMP)",
RFC 5357, October 2008
draft-ietf-ippm-btc-cap-00.txt Allman, M., "A Bulk
Transfer Capacity Methodology for Cooperating Hosts",
August 2001
9.2. Informative References
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Authors' Addresses
Barry Constantine
JDSU, Test and Measurement Division
One Milesone Center Court
Germantown, MD 20876-7100
USA
Phone: +1 240 404 2227
barry.constantine@jdsu.com
Gilles Forget
Independent Consultant to Bell Canada.
308, rue de Monaco, St-Eustache
Qc. CANADA, Postal Code: J7P-4T5
Phone: (514) 895-8212
gilles.forget@sympatico.ca
Ruediger Geib
Heinrich-Hertz-Strasse (Number: 3-7)
Darmstadt, Germany, 64295
Phone: +49 6151 6282747
Ruediger.Geib@telekom.de
Reinhard Schrage
Schrage Consulting
Phone: +49 (0) 5137 909540
reinhard@schrageconsult.com
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