Network Working Group B. Constantine
Internet-Draft JDSU
Intended status: Informational G. Forget
Expires: May 14, 2011 Bell Canada (Ext. Consultant)
Rudiger Geib
Deutsche Telekom
Reinhard Schrage
Schrage Consulting
November 14, 2010
Framework for TCP Throughput Testing
draft-ietf-ippm-tcp-throughput-tm-08.txt
Abstract
This framework describes a methodology for measuring end-to-end TCP
throughput performance in a managed IP network. The intention is to
provide a practical methodology to validate TCP layer performance.
The goal is to provide a better indication of the user experience.
In this framework, various TCP and IP parameters are identified and
should be tested as part of a managed IP network verification.
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 May 14, 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
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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. . . . . . . . . . . . . . . . . . . . . . 6
3. TCP Throughput Testing Methodology . . . . . . . . . . . . . . 7
3.1 Determine Network Path MTU . . . . . . . . . . . . . . . . 9
3.2. Baseline Round Trip Time and Bandwidth . . . . . . . . . . 10
3.2.1 Techniques to Measure Round Trip Time . . . . . . . . 10
3.2.2 Techniques to Measure end-to-end Bandwidth. . . . . . 11
3.3. TCP Throughput Tests . . . . . . . . . . . . . . . . . . . 12
3.3.1 Calculate Ideal TCP Receive Window Size. . . . . . . . 12
3.3.2 Metrics for TCP Throughput Tests . . . . . . . . . . . 15
3.3.3 Conducting the TCP Throughput Tests. . . . . . . . . . 18
3.3.4 Single vs. Multiple TCP Connection Testing . . . . . . 19
3.3.5 Interpretation of the TCP Throughput Results . . . . . 20
3.4. Traffic Management Tests . . . . . . . . . . . . . . . . . 20
3.4.1 Traffic Shaping Tests. . . . . . . . . . . . . . . . . 21
3.4.1.1 Interpretation of Traffic Shaping Test Results. . . 21
3.4.2 RED Tests. . . . . . . . . . . . . . . . . . . . . . . 22
3.4.2.1 Interpretation of RED Results . . . . . . . . . . . 23
4. Security Considerations . . . . . . . . . . . . . . . . . . . 23
5. IANA Considerations . . . . . . . . . . . . . . . . . . . . . 23
6. Acknowledgments . . . . . . . . . . . . . . . . . . . . . . . 23
7. References . . . . . . . . . . . . . . . . . . . . . . . . . . 24
7.1 Normative References . . . . . . . . . . . . . . . . . . . 24
7.2 Informative References . . . . . . . . . . . . . . . . . . 24
Authors' Addresses . . . . . . . . . . . . . . . . . . . . . . . . 25
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1. Introduction
Network providers are coming to the realization that Layer 2/3
testing is not enough to adequately ensure end-user's satisfaction.
An SLA (Service Level Agreement) is provided to business customers
and is generally based upon Layer 2/3 criteria such as access rate,
latency, packet loss and delay variations. On the other hand,
measuring TCP throughput provides meaningful results with respect to
user experience. Thus, the network provider community desires to
measure IP network throughput performance at the TCP layer.
Additionally, business enterprise customers seek to conduct
repeatable TCP throughput tests between locations. Since these
enterprises rely on the networks of the providers, a common test
methodology with predefined metrics will benefit both parties.
Note that the primary focus of this methodology is managed business
class IP networks; i.e. those Ethernet terminated services for which
businesses are provided an SLA from the network provider. End-users
with "best effort" access between locations can use this methodology,
but this framework and its metrics are intended to be used in a
predictable managed IP service environment.
So the intent behind this document is to define a methodology for
testing sustained TCP layer performance. In this document, the
maximum achievable TCP throughput is that amount of data per unit
time that TCP transports when trying to reach Equilibrium, i.e.
after the initial slow start and congestion avoidance phases. We
refer to this as the maximum achievable TCP Throughput for the TCP
connection(s).
TCP uses a congestion window, (TCP CWND), to determine how many
packets it can send at one time. A larger TCP CWND permits a higher
throughput. TCP "slow start" and "congestion avoidance" algorithms
together determine the TCP CWND size. The Maximum TCP CWND size is
also tributary to the buffer space allocated by the kernel for each
socket. For each socket, there is a default buffer size that can be
changed by the program using a system library called just before
opening the socket. There is also a kernel enforced maximum buffer
size. This buffer size can be adjusted at both ends of the socket
(send and receive). In order to obtain the maximum throughput, it
is critical to use optimal TCP Send and Receive Socket Buffer sizes
as well as the optimal TCP Receive Window size.
There are many variables to consider when conducting a TCP throughput
test and this methodology focuses on the most common:
- Path MTU and Maximum Segment Size (MSS)
- RTT and Bottleneck BW
- Ideal TCP Receive Window (including Ideal Receive Socket Buffer)
- Ideal Send Socket Buffer
- TCP Congestion Window (TCP CWND)
- Single Connection and Multiple Connections testing
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This methodology proposes TCP testing that should be performed in
addition to traditional Layer 2/3 type tests. Layer 2/3 tests are
required to verify the integrity of the network before conducting TCP
test. 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 in operational networks may be used.
Note: RFC 2544 was never meant to be used outside a lab environment.
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-to-end operational and managed IP network. A multitude of
network architectures and topologies can be tested. The following
set-up diagram is very general and it only illustrates the
segmentation within end user and network provider domains.
Common terminologies used in the test methodology are:
- 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)
- Customer Provided Equipment (CPE), refers to customer owned
equipment (routers, switches, computers, etc.)
- Customer Edge (CE), refers to provider owned demarcation device.
- End-user: The business enterprise customer. For the purposes of
conducting TCP throughput tests, this may be the IT department.
- Network Under Test (NUT), refers to the tested IP network path.
- Provider Edge (PE), refers to provider's distribution equipment.
- P (Provider), refers to provider core network equipment.
- Round-Trip Time (RTT), refers to Layer 4 back and forth delay.
- Round-Trip Delay (RTD), refers to Layer 1 back and forth delay.
- 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.
+----+ +----+ +----+ +----+ +---+ +---+ +----+ +----+ +----+ +----+
| TCP|-| CPE|-| CE |--| PE |-| P |--| P |-| PE |--| CE |-| CPE|-| TCP|
| TTD| | | | |BB| | | | | | | |BB| | | | | TTD|
+----+ +----+ +----+ +----+ +---+ +---+ +----+ +----+ +----+ +----+
<------------------------ NUT ------------------------>
R >-----------------------------------------------------------|
T |
T <-----------------------------------------------------------|
Note that the NUT may consist of a variety of devices including but
not limited to, load balancers, proxy servers or WAN acceleration
devices. The detailed topology of the NUT should be well understood
when conducting the TCP throughput tests, although this methodology
makes no attempt to characterize specific network architectures.
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2. Scope and Goals of this Methodology
Before defining the goals, it is important to clearly define the
areas that are out-of-scope.
- This methodology is not intended to predict the TCP throughput
during the transient stages of a TCP connection, such as the initial
slow start.
- This 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.
- This methodology is not intended to provide detailed diagnosis
of problems within end-points or within the network itself as
related to non-optimal TCP performance, although a results
interpretation section for each test step may provide insight in
regards with potential issues.
- This methodology does not propose to operate permanently with high
measurement loads. TCP performance and optimization within
operational networks may be captured and evaluated by using data
from the "TCP Extended Statistics MIB" [RFC4898].
- This methodology is not intended to measure TCP throughput as part
of an SLA, or to compare the TCP performance between service
providers or to compare between implementations of this methodology
in dedicated communications test instruments.
In contrast to the above exclusions, a 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 network
to carry end-user TCP applications.
Other specific goals are to :
- Provide a practical test approach that specifies IP hosts
configurable TCP parameters such as TCP Receive Window size, Socket
Buffer size, MSS (Maximum Segment Size), number of connections, and
how these affect the outcome of TCP performance over a network.
See section 3.3.3.
- Provide specific test conditions like link speed, RTT, TCP Receive
Window size, Socket Buffer size and maximum achievable TCP throughput
when trying to reach TCP Equilibrium. For guideline purposes,
provide examples of test conditions and their maximum achievable
TCP throughput. Section 2.1 provides specific details concerning the
definition of TCP Equilibrium within this methodology while section 3
provides specific test conditions with examples.
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- Define three (3) basic metrics to compare the performance of TCP
connections under various network conditions. See section 3.3.2.
- In test situations where the recommended procedure does not yield
the maximum achievable TCP throughput results, 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 on these
issues. See section 3.3.5.
2.1 TCP Equilibrium
TCP connections have three (3) fundamental congestion window phases
as documented in [RFC5681].
These 3 phases are:
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 attainable throughput on an end-to-end network
path. Retransmissions are a natural by-product of the TCP congestion
avoidance algorithm as it seeks to achieve maximum throughput.
3 - The Retransmission Time-out phase, which could include Fast
Retransmit (Tahoe) or Fast Recovery (Reno & New Reno). When multiple
packet lost occurs, Congestion Avoidance phase transitions to Fast
Retransmission or Fast Recovery depending upon TCP implementations.
If a Time-Out occurs, TCP transitions back to the Slow Start phase.
The following diagram depicts these 3 phases.
| Trying to reach TCP Equilibrium >>>>>>>>>>>>>
/\ | High ssthresh TCP CWND 3
/\ | Loss Event * halving Retransmission
/\ | * \ upon loss Time-Out Adjusted
/\ | * \ /\ _______ ssthresh
/\ | * \ / \ /M-Loss | *
TCP | * 2 \/ \ / Events |1 *
Through- | * Congestion\ / |Slow *
put | 1 * Avoidance \/ |Start *
| Slow * Half | *
| Start * TCP CWND *
|___*_______________________Minimum TCP CWND after Time-Out_
Time >>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>
Note : ssthresh = Slow Start threshold.
Through the above 3 phases, TCP is trying to reach Equilibrium, but
since packet loss is currently its only available feedback indicator,
TCP will never reach that goal. Although, a well tuned (and managed)
IP network with well tuned IP hosts and applications should perform
very close to TCP Equilibrium and to the BB (Bottleneck Bandwidth).
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This TCP methodology provides guidelines to measure the maximum
achievable TCP throughput or maximum TCP sustained rate obtained
after TCP CWND has stabilized to an optimal value. All maximum
achievable TCP throughputs specified in section 3 are with respect to
this condition.
It is important to clarify the interaction between the sender's Send
Socket Buffer and the receiver's advertised TCP Receive Window. TCP
test programs such as iperf, ttcp, etc. allow 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 Receive Window size advertised by the
receiver. Implications to the capabilities of the Throughput Test
Device (TTD) are covered at the end of section 3.
3. TCP Throughput Testing Methodology
As stated earlier in section 1, it is considered best practice to
verify the integrity of the network by conducting Layer2/3 tests such
as [RFC2544] or other methods of network stress tests. Although, it
is important to mention here that RFC 2544 was never meant to be used
outside a lab environment.
If the network is not performing properly in terms of packet loss,
jitter, etc. then the TCP layer testing will not be meaningful. A
dysfunctional network will not reach close enough to TCP Equilibrium
to provide optimal TCP throughputs with the available bandwidth.
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 either on the CPE or on the CE device
and not on the PE (Provider Edge) router.
The following represents the sequential order of steps for this
testing methodology:
1. Identify the Path MTU. Packetization Layer Path MTU Discovery
or PLPMTUD, [RFC4821], MUST be conducted to verify the network path
MTU. Conducting PLPMTUD establishes the upper limit for the MSS to
be used in subsequent steps.
2. Baseline Round Trip Time and Bandwidth. This step establishes the
inherent, non-congested Round Trip Time (RTT) and the bottleneck
bandwidth of the end-to-end network path. These measurements are
used to provide estimates of the ideal TCP Receive Window and Send
Socket Buffer sizes that SHOULD be used in subsequent test steps.
These measurements reference [RFC2681] and [RFC4898] to measure RTD
and the associated RTT.
3. TCP Connection Throughput Tests. With baseline measurements
of Round Trip Time and bottleneck bandwidth, single and multiple TCP
connection throughput tests SHOULD be conducted to baseline network
performance expectations.
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4. Traffic Management Tests. Various traffic management and queuing
techniques can be tested in this step, using multiple TCP
connections. Multiple connections testing should verify that the
network is configured properly for traffic shaping versus policing,
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 a dedicated communications test instrument.
In both cases, they must be capable of emulating both client and
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 Reno, TCP options supported, etc. These will
obviously be more important when using dedicated communications test
instruments where the TCP implementation may be customized or tuned
to run in higher performance hardware. When a compliant TCP TTD is
used, the TCP implementation MUST be identified in the test results.
The compliant TCP TTD should be usable for complete end-to-end
testing through network security elements and should also be usable
for testing network sections.
- More important, the TCP test host MUST be capable to generate
and receive stateful TCP test traffic at the full link speed of the
network under test. Stateful TCP test traffic means that the test
host MUST fully implement a TCP stack; this is generally a comment
aimed at dedicated communications test equipments which sometimes
"blast" packets with TCP headers. As a general rule of thumb, testing
TCP throughput at rates greater than 100 Mbit/sec MAY require high
performance server hardware or dedicated hardware based test tools.
- A compliant TCP Throughput Test Device MUST allow adjusting both
Send Socket Buffer and TCP Receive Window sizes. The Receive Socket
Buffer MUST be large enough to accommodate the TCP Receive Window.
- 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 retransmission results in packet captures.
Another option may be to use "TCP Extended Statistics MIB" per
[RFC4898].
- The RFC4821 PLPMTUD test SHOULD be conducted with a dedicated
tester which exposes the ability to run the PLPMTUD algorithm
independent from the OS stack.
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3.1. Determine Network Path MTU
TCP implementations should use Path MTU Discovery techniques (PMTUD).
PMTUD relies on ICMP 'need to frag' messages to learn the path MTU.
When a device has a packet to send which has the Don't Fragment (DF)
bit in the IP header set and the packet is larger than the Maximum
Transmission Unit (MTU) of the next hop, the packet is dropped and
the device sends an ICMP 'need to frag' message back to the host that
originated the packet. The ICMP 'need to frag' message includes
the next hop MTU which PMTUD uses to tune the TCP Maximum Segment
Size (MSS). Unfortunately, because many network managers completely
disable ICMP, this technique does not always prove reliable.
Packetization Layer Path MTU Discovery or PLPMTUD [RFC4821] MUST then
be conducted to verify the network path MTU. PLPMTUD can be used
with or without ICMP. The following sections provide a summary of the
PLPMTUD approach and an example using TCP. [RFC4821] specifies a
search_high and a search_low parameter for the MTU. As specified in
[RFC4821], 1024 Bytes is a safe value for search_low in modern
networks.
It is important to determine the links overhead along the IP path,
and then to select a TCP MSS size corresponding to the Layer 3 MTU.
For example, if the MTU is 1024 Bytes and the TCP/IP headers are 40
Bytes, then the MSS would be set to 984 Bytes.
An example scenario is a network where the actual path MTU is 1240
Bytes. The TCP client probe MUST be capable of setting the MSS for
the probe packets and could start at MSS = 984 (which corresponds
to an MTU size of 1024 Bytes).
The TCP client probe would open a TCP connection and advertise the
MSS as 984. Note that the client probe MUST generate these packets
with the DF bit set. The TCP client probe then sends test traffic
per a small default Send Socket Buffer size of ~8KBytes. It should
be kept small to minimize the possibility of congesting the network,
which may induce packet loss. The duration of the test should also
be short (10-30 seconds), again to minimize congestive effects
during the test.
In the example of a 1240 Bytes path MTU, probing with an MSS equal to
984 would yield a successful probe and the test client packets would
be successfully transferred to the test server.
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Also note that the test client MUST verify that the MSS advertised
is indeed negotiated. Network devices with built-in Layer 4
capabilities can intercede during the connection establishment and
reduce the advertised MSS to avoid fragmentation. This is certainly
a desirable feature from a network perspective, but it can yield
erroneous test results if the client test probe does not confirm the
negotiated MSS.
The next test probe would use the search_high value and this would
be set to MSS = 1460 to correspond to a 1500 Bytes MTU. In this
example, the test client will retransmit based upon time-outs, since
no ACKs will be received from the test server. This test probe is
marked as a conclusive failure if none of the test packets are
ACK'ed. If 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 raise it by half also when seeking the upper limit.
3.2. Baseline Round Trip Time and Bandwidth
Before stateful TCP testing can begin, it is important to determine
the baseline Round Trip Time (non-congested inherent delay) and
bottleneck bandwidth of the end-to-end network to be tested. These
measurements are used to provide estimates of the ideal TCP Receive
Window and Send Socket Buffer sizes that SHOULD be used in subsequent
test steps.
3.2.1 Techniques to Measure Round Trip Time
Following the definitions used in section 1.1, Round Trip Time (RTT)
is the elapsed time between the clocking in of the first bit of a
payload sent packet 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. When sampling values of RTT over a test
interval, the minimum value measured SHOULD be used as the baseline
RTT since this will most closely estimate the inherent network
latency. This inherent RTT is also used to determine the Buffer
Delay Percentage metric which is defined in Section 3.3.2
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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 resolution of the measurement (i.e. msec versus usec) may
dictate whether the RTT measurement can be achieved with ICMP pings
or by a dedicated communications test instrument with precision
timers.
The objective in this section is to list several techniques
in order of decreasing accuracy.
- Use test equipment on each end of the network, "looping" the
far-end tester so that a packet stream can be measured back and forth
from end-to-end. This RTT measurement may be compatible with delay
measurement protocols specified in [RFC5357].
- Conduct packet captures of TCP test sessions using "iperf" or FTP,
or other TCP test applications. By running multiple experiments,
packet captures can then be analyzed to estimate RTT based upon the
SYN -> SYN-ACK from the 3 way handshake at the beginning of the TCP
sessions. Although, note that Firewalls might slow down 3 way
handshakes, so it might be useful to compare with measured RTT later
on in the same capture.
- ICMP Pings may also be adequate to provide round trip time
estimations. Some limitations with ICMP Ping may include msec
resolution and whether the network elements are responding to pings
or not. Also, ICMP is often rate-limited and segregated into
different buffer queues, so it is not as reliable and accurate as
in-band measurements.
3.2.2 Techniques to Measure end-to-end Bandwidth
There are many well established techniques available to provide
estimated measures of bandwidth over a network. These measurements
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].
Before any TCP Throughput test can be done, a bandwidth measurement
test MUST be run with stateless IP streams(not stateful TCP) in order
to determine the available bandwidths in each direction. This test
should obviously be performed at various intervals throughout a
business day or even across a week. Ideally, the bandwidth test
should produce logged outputs of the achieved bandwidths across the
test interval.
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3.3. TCP Throughput Tests
This methodology specifically defines TCP throughput techniques to
verify sustained TCP performance in a managed business IP network, as
defined in section 2.1. This section and others will define the
method to conduct these sustained TCP throughput tests and guidelines
for the predicted results.
With baseline measurements of round trip time and bandwidth
from section 3.2, a series of single and multiple TCP connection
throughput tests SHOULD be conducted to baseline network performance
against expectations. The number of trials and the type of testing
(single versus multiple connections) will vary according to the
intention of the test. One example would be a single connection test
in which the throughput achieved by large Send Socket Buffer and TCP
Receive Window sizes (i.e. 256KB) is to be measured. It would be
advisable to test performance at various times of the business day.
It is RECOMMENDED to run the tests in each direction independently
first, then run both directions simultaneously. In each case,
TCP Transfer Time, TCP Efficiency, and Buffer Delay Percentage MUST
be measured in each direction. These metrics are defined in 3.3.2.
3.3.1 Calculate Ideal TCP Receive Window Size
The ideal TCP Receive Window size can be calculated from the
bandwidth delay product (BDP), which is:
BDP (bits) = RTT (sec) x Bandwidth (bps)
Note that the RTT is being used as the "Delay" variable in the
BDP calculations.
Then, by dividing the BDP by 8, we obtain the "ideal" TCP Receive
Window size in Bytes. For optimal results, the Send Socket Buffer
size must be adjusted to the same value at the opposite end of the
network path.
Ideal TCP RWIN = BDP / 8
An example would be a T3 link with 25 msec RTT. The BDP would equal
~1,105,000 bits and the ideal TCP Receive Window would be ~138
KBytes.
The following table provides some representative network Link Speeds,
RTT, BDP, and their associated Ideal TCP Receive Window sizes.
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Table 3.3.1: Link Speed, RTT and calculated BDP & TCP Receive Window
Link Ideal TCP
Speed* RTT BDP Receive Window
(Mbps) (ms) (bits) (KBytes)
---------------------------------------------------------------------
1.536 20 30,720 3.84
1.536 50 76,800 9.60
1.536 100 153,600 19.20
44.21 10 442,100 55.26
44.21 15 663,150 82.89
44.21 25 1,105,250 138.16
100 1 100,000 12.50
100 2 200,000 25.00
100 5 500,000 62.50
1,000 0.1 100,000 12.50
1,000 0.5 500,000 62.50
1,000 1 1,000,000 125.00
10,000 0.05 500,000 62.50
10,000 0.3 3,000,000 375.00
* Note that link speed is the bottleneck bandwidth for the NUT
The following serial link speeds are used:
- T1 = 1.536 Mbits/sec (for a B8ZS line encoding facility)
- T3 = 44.21 Mbits/sec (for a C-Bit Framing facility)
The above table illustrates the ideal TCP Receive Window size.
If a smaller TCP Receive Window is used, then the TCP Throughput
is not optimal. To calculate the Ideal TCP Throughput, the following
formula is used: TCP Throughput = TCP RWIN X 8 / RTT
An example could be a 100 Mbps IP path with 5 ms RTT and a TCP
Receive Window size of 16KB, then:
TCP Throughput = 16 KBytes X 8 bits / 5 ms.
TCP Throughput = 128,000 bits / 0.005 sec.
TCP Throughput = 25.6 Mbps.
Another example for a T3 using the same calculation formula is
illustrated on the next page:
TCP Throughput = TCP RWIN X 8 / RTT.
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 Receive Window size exceeds the BDP (i.e. T3 link,
64 KBytes TCP Receive Window on a 10 ms RTT path), the maximum frames
per second limit of 3664 is reached and the calculation formula is:
TCP Throughput = Max FPS X MSS X 8.
TCP Throughput = 3664 FPS X 1460 Bytes X 8 bits.
TCP Throughput = 42.8 Mbps
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The following diagram compares achievable TCP throughputs on a T3
with Send Socket Buffer & TCP Receive Window sizes of 16KB vs. 64KB.
45|
| _______42.8M
40| |64KB |
TCP | | |
Throughput 35| | |
in Mbps | | | +-----+34.1M
30| | | |64KB |
| | | | |
25| | | | |
| | | | |
20| | | | | _______20.5M
| | | | | |64KB |
15| | | | | | |
|12.8M+-----| | | | | |
10| |16KB | | | | | |
| | | |8.5M+-----| | | |
5| | | | |16KB | |5.1M+-----| |
|_____|_____|_____|____|_____|_____|____|16KB |_____|_____
10 15 25
RTT in milliseconds
The following diagram shows the achievable TCP throughput on a 25ms
T3 when Send Socket Buffer & TCP Receive Window sizes are increased.
45|
|
40| +-----+40.9M
TCP | | |
Throughput 35| | |
in Mbps | | |
30| | |
| | |
25| | |
| | |
20| +-----+20.5M | |
| | | | |
15| | | | |
| | | | |
10| +-----+10.2M | | | |
| | | | | | |
5| +-----+5.1M | | | | | |
|_____|_____|______|_____|______|_____|_______|_____|_____
16 32 64 128*
TCP Receive Window size in KBytes
* Note that 128KB requires [RFC1323] TCP Window scaling option.
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3.3.2 Metrics for TCP Throughput Tests
This framework focuses on a TCP throughput methodology and also
provides several basic metrics to compare results 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 accounts for the myriad of variables (i.e. RTT
variation, loss conditions, TCP implementation, etc.). However,
these basic metrics will facilitate TCP throughput comparisons
under varying network conditions and between network traffic
management techniques.
The first metric is the TCP Transfer Time, which is simply the
measured time it takes to transfer a block of data across
simultaneous TCP connections. This concept is useful when
benchmarking traffic management techniques and where multiple
TCP connections are required.
TCP Transfer time may also be used to provide a normalized ratio of
the actual TCP Transfer Time versus the Ideal Transfer Time. This
ratio is called the TCP Transfer Index and is defined as:
Actual TCP Transfer Time
-------------------------
Ideal TCP Transfer Time
The Ideal TCP Transfer time is derived from the network path
bottleneck bandwidth and various Layer 1/2/3/4 overheads associated
with the network path. Additionally, both the TCP Receive Window and
the Send Socket Buffer sizes must be tuned to equal the bandwidth
delay product (BDP) as described in section 3.3.1.
The following table illustrates the Ideal TCP Transfer time of a
single TCP connection when its TCP Receive Window and Send Socket
Buffer sizes are equal to the BDP.
Table 3.3.2: Link Speed, RTT, BDP, TCP Throughput, and
Ideal TCP Transfer time for a 100 MB File
Link Maximum Ideal TCP
Speed BDP Achievable TCP Transfer time
(Mbps) RTT (ms) (KBytes) Throughput(Mbps) (seconds)
--------------------------------------------------------------------
1.536 50 9.6 1.4 571
44.21 25 138.2 42.8 18
100 2 25.0 94.9 9
1,000 1 125.0 949.2 1
10,000 0.05 62.5 9,492 0.1
Transfer times are rounded for simplicity.
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For a 100MB file(100 x 8 = 800 Mbits), the Ideal TCP Transfer Time
is derived as follows:
800 Mbits
Ideal TCP Transfer Time = -----------------------------------
Maximum Achievable TCP Throughput
The maximum achievable layer 2 throughput on T1 and T3 Interfaces
is based on the maximum frames per second (FPS) permitted by the
actual layer 1 speed when the MTU is 1500 Bytes.
The maximum FPS for a T1 is 127 and the calculation formula is:
FPS = T1 Link Speed / ((MTU + PPP + Flags + CRC16) X 8)
FPS = (1.536M /((1500 Bytes + 4 Bytes + 2 Bytes + 2 Bytes) X 8 )))
FPS = (1.536M / (1508 Bytes X 8))
FPS = 1.536 Mbps / 12064 bits
FPS = 127
The maximum FPS for a T3 is 3664 and the calculation formula is:
FPS = T3 Link Speed / ((MTU + PPP + Flags + CRC16) X 8)
FPS = (44.21M /((1500 Bytes + 4 Bytes + 2 Bytes + 2 Bytes) X 8 )))
FPS = (44.21M / (1508 Bytes X 8))
FPS = 44.21 Mbps / 12064 bits
FPS = 3664
The 1508 equates to:
MTU + PPP + Flags + CRC16
Where MTU is 1500 Bytes, PPP is 4 Bytes, Flags are 2 Bytes and CRC16
is 2 Bytes.
Then, to obtain the Maximum Achievable TCP Throughput (layer 4), we
simply use: MSS in Bytes X 8 bits X max FPS.
For a T3, the maximum TCP Throughput = 1460 Bytes X 8 bits X 3664 FPS
Maximum TCP Throughput = 11680 bits X 3664 FPS
Maximum TCP Throughput = 42.8 Mbps.
The maximum achievable layer 2 throughput on Ethernet Interfaces is
based on the maximum frames per second permitted by the IEEE802.3
standard when the MTU is 1500 Bytes.
The maximum FPS for 100M Ethernet is 8127 and the calculation is:
FPS = (100Mbps /(1538 Bytes X 8 bits))
The maximum FPS for GigE is 81274 and the calculation formula is:
FPS = (1Gbps /(1538 Bytes X 8 bits))
The maximum FPS for 10GigE is 812743 and the calculation formula is:
FPS = (10Gbps /(1538 Bytes X 8 bits))
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The 1538 equates to:
MTU + Eth + CRC32 + IFG + Preamble + SFD
Where MTU is 1500 Bytes, Ethernet is 14 Bytes, CRC32 is 4 Bytes,
IFG is 12 Bytes, Preamble is 7 Bytes and SFD is 1 Byte.
Note that better results could be obtained with jumbo frames on
GigE and 10 GigE.
Then, to obtain the Maximum Achievable TCP Throughput (layer 4), we
simply use: MSS in Bytes X 8 bits X max FPS.
For a 100M, the maximum TCP Throughput = 1460 B X 8 bits X 8127 FPS
Maximum TCP Throughput = 11680 bits X 8127 FPS
Maximum TCP Throughput = 94.9 Mbps.
To illustrate the TCP Transfer Time Index, an example would be the
bulk transfer of 100 MB over 5 simultaneous TCP connections (each
connection 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 then 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 TCP Efficiency, which is the percentage of Bytes
that were not retransmitted and is defined as:
Transmitted Bytes - Retransmitted Bytes
--------------------------------------- x 100
Transmitted Bytes
Transmitted Bytes are the total number of TCP payload Bytes to be
transmitted which includes the original and retransmitted Bytes. This
metric provides a comparative measure between various QoS mechanisms
like traffic management or congestion avoidance. Various TCP
implementations like Reno, Vegas, etc. could also be compared.
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).
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The third metric is the Buffer Delay Percentage, which represents the
increase in RTT during a TCP throughput test with respect to
inherent or baseline network RTT. The baseline RTT is the round-trip
time inherent to the network path under non-congested conditions.
(See 3.2.1 for details concerning the baseline RTT measurements).
The Buffer Delay Percentage is defined as:
Average RTT during Transfer - Baseline RTT
------------------------------------------ x 100
Baseline RTT
As an example, 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
Percentage MUST be measured during each throughput test. Poor TCP
Transfer Time Indexes (TCP Transfer Time greater than Ideal TCP
Transfer Times) may be diagnosed by correlating with sub-optimal TCP
Efficiency and/or Buffer Delay Percentage metrics.
3.3.3 Conducting the TCP Throughput Tests
Several TCP tools are currently used in the network world and one of
the most common is "iperf". With this tool, hosts are installed at
each end of the network path; one acts as client and the other as
a server. The Send Socket Buffer and the TCP Receive Window sizes
of both client and server can be manually set. The achieved
throughput can then be measured, either uni-directionally or
bi-directionally. For higher BDP situations in lossy networks
(long fat networks or satellite links, etc.), TCP options such as
Selective Acknowledgment SHOULD be considered and become part of
the window size / throughput characterization.
Host hardware performance must be well understood before conducting
the tests described in the following sections. A dedicated
communications test instrument will generally be required, especially
for line rates of GigE and 10 GigE. A compliant TCP TTD SHOULD
provide a warning message when the expected test throughput will
exceed 10% of the network bandwidth capacity. If the throughput test
is expected to exceed 10% of the provider bandwidth, then the test
should be coordinated with the network provider. This does not
include the customer premise bandwidth, the 10% refers directly to
the provider's bandwidth (Provider Edge to Provider router).
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The TCP throughput test should be run over a long enough duration
to properly exercise network buffers (greater than 30 seconds) and
also characterize performance at different time periods of the day.
3.3.4 Single vs. Multiple TCP Connection Testing
The decision whether to conduct single or multiple TCP connection
tests depends upon the size of the BDP in relation to the configured
TCP Receive Window sizes configured in the end-user environment.
For example, if the BDP for a long fat network turns out to be 2MB,
then it is probably more realistic to test this network path with
multiple connections. Assuming typical host computer TCP Receive
Window Sizes of 64 KB, using 32 TCP connections would realistically
test this path.
The following table is provided to illustrate the relationship
between the TCP Receive Window size and the number of TCP connections
required to utilize 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.
TCP Number of TCP Connections
Window to fill available bandwidth
-------------------------------------
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
previous table, the 64KB window is considered. Each of the 5
TCP connections would be configured to transfer 100MB, and each one
should obtain a maximum of 100 Mb/sec. So for this example, the
100MB payload should be transferred across the connections in
approximately 8 seconds (which would be the ideal TCP transfer time
under these conditions).
Additionally, the TCP Efficiency metric MUST be computed for each
connection tested as defined in section 3.3.2.
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3.3.5 Interpretation of the TCP Throughput Results
At the end of this step, the user will document the theoretical BDP
and a set of Window size experiments with measured TCP throughput for
each TCP window size. For cases where the sustained TCP throughput
does not equal the ideal value, some possible causes are:
- Network congestion causing packet loss which MAY be inferred from
a poor TCP Efficiency % (higher TCP Efficiency % = less packet
loss)
- Network congestion causing an increase in RTT which MAY be inferred
from the Buffer Delay Percentage (i.e., 0% = no increase in RTT
over baseline)
- Intermediate network devices which actively regenerate the TCP
connection and can alter TCP Receive Window size, MSS, etc.
- Rate limiting (policing). More details on 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
to host computers. An example would be LAN connectivity of GigE
and WAN connectivity of 100 Mbps. The WAN connectivity may be
physically 100 Mbps or logically 100 Mbps (over a GigE WAN
connection). In the later case, rate limiting is used to provide the
WAN bandwidth per the SLA.
Traffic management techniques are employed to provide various forms
of QoS, the more common include:
- Traffic Shaping
- Priority queuing
- Random Early Discard (RED)
Configuring the end-to-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 to
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 communications test instrument will
generally be REQUIRED for line rates of GigE and 10 GigE. If the
throughput test is expected to exceed 10% of the provider bandwidth,
then the test should be coordinated with the network provider. This
does not include the customer premises bandwidth, the 10% refers to
the provider's bandwidth (Provider Edge to Provider router). Note
that GigE and 10 GigE interfaces might benefit from hold-queue
adjustments in order to prevent the saw-tooth TCP traffic pattern.
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3.4.1 Traffic Shaping Tests
For services where the available bandwidth is rate limited, two (2)
techniques can be used: traffic policing or traffic shaping.
Simply stated, traffic policing marks and/or drops packets which
exceed the SLA bandwidth (in most cases, excess traffic is dropped).
Traffic shaping employs the use of queues to smooth the bursty
traffic and then send out within the SLA bandwidth limit (without
dropping packets unless the traffic shaping queue is exhausted).
Traffic shaping is generally configured for TCP data services and
can provide improved TCP performance since the retransmissions are
reduced, which in turn optimizes TCP throughput for the available
bandwidth. Through this section, the rate-limited bandwidth shall
be referred to as the "bottleneck bandwidth".
The ability to detect proper traffic shaping is more easily diagnosed
when conducting a multiple TCP connections test. Proper shaping will
provide a fair distribution of the available bottleneck bandwidth,
while traffic policing will not.
The traffic shaping tests are built upon the concepts of multiple
connections testing as defined in section 3.3.3. Calculating the BDP
for the bottleneck bandwidth is first required before selecting the
number of connections and Send Buffer and TCP Receive Window sizes
per connection.
Similar to the example in section 3.3, a typical test scenario might
be: GigE LAN with a 100Mbps bottleneck bandwidth (rate limited
logical interface), and 5 msec RTT. This would require five (5) TCP
connections of 64 KB Send Socket Buffer and TCP Receive Window sizes
to evenly fill the bottleneck bandwidth (~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 Percentage.
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 TCP 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 may benefit greatly from RED based
techniques. Without RED, TCP may not be able to achieve the full
bottleneck bandwidth. With RED enabled, TCP congestion avoidance
throttles the connections on the higher speed interface (i.e. LAN)
and can help achieve the full bottleneck bandwidth. The burstiness
of TCP traffic is a key factor in the overall effectiveness of RED
techniques; steady state bulk transfer flows will generally not
benefit from RED. With bulk transfer flows, network device queues
gracefully throttle the effective throughput rates due to increased
delays.
The ability to detect proper RED configuration is more easily
diagnosed when conducting a multiple TCP connections test. Multiple
TCP connections provide the bursty sources that emulate the
real-world conditions for which RED was intended.
The RED tests also builds upon the concepts of multiple connections
testing as defined in section 3.3.3. Calculating the BDP for the
bottleneck bandwidth is first required before selecting the number
of connections, the Send Socket Buffer size and the TCP Receive
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 (10
connections for this example).
The TCP TTD must be configured to generate these connections as
shorter (bursty) flows versus bulk transfer type flows. These TCP
bursts should stress queue sizes in the 512KB range. Again
experimentation will be required; the proper number of TCP
connections, the Send Socket Buffer and TCP Receive Window sizes will
be dictated by the size of the network element queue.
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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 may cause excessive loss to all of
the TCP connections and in the worst case global TCP synchronization.
By plotting the aggregate throughput achieved on the bottleneck
interface, proper RED operation 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 10-20%) and the actual TCP Transfer time may be
proportionally longer then the Ideal TCP Transfer time.
Additionally, 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 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.
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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
[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
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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
Rudiger Geib
Heinrich-Hertz-Strasse (Number: 3-7)
Darmstadt, Germany, 64295
Phone: +49 6151 6282747
Ruediger.Geib@telekom.de
Reinhard Schrage
Schrage Consulting
Phone: +49 (0) 5137 909540
reinhard@schrageconsult.com
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