Network Working Group B. Constantine
Internet-Draft JDSU
Intended status: Informational G. Forget
Expires: January 9, 2011 Bell Canada (Ext. Consultant)
L. Jorgenson
nooCore
Reinhard Schrage
Schrage Consulting
July 9, 2010
TCP Throughput Testing Methodology
draft-ietf-ippm-tcp-throughput-tm-04.txt
Abstract
This memo describes a methodology for measuring sustained TCP
throughput performance in an end-to-end managed network environment.
This memo is intended to provide a practical approach to help users
validate the TCP layer performance of a managed network, which should
provide a better indication of end-user application level experience.
In the methodology, various TCP and network parameters are identified
that should be tested as part of the network verification at the TCP
layer.
Status of this Memo
This Internet-Draft is submitted to IETF in full conformance with the
provisions of BCP 78 and BCP 79.
Internet-Drafts are working documents of the Internet Engineering
Task Force (IETF), its areas, and its working groups. Note that
other groups may also distribute working documents as Internet-
Drafts. Creation date July 9, 2010.
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This Internet-Draft will expire on January 9, 2011.
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Table of Contents
1. Introduction . . . . . . . . . . . . . . . . . . . . . . . . . 3
2. Goals of this Methodology. . . . . . . . . . . . . . . . . . . 4
2.1 TCP Equilibrium State Throughput . . . . . . . . . . . . . 5
2.2 Metrics for TCP Throughput Tests . . . . . . . . . . . . . 6
3. TCP Throughput Testing Methodology . . . . . . . . . . . . . . 6
3.1 Determine Network Path MTU . . . . . . . . . . . . . . . . 8
3.2. Baseline Round-trip Delay and Bandwidth. . . . . . . . . . 9
3.2.1 Techniques to Measure Round Trip Time . . . . . . . . 9
3.2.2 Techniques to Measure End-end Bandwidth . . . . . . . 10
3.3. TCP Throughput Tests . . . . . . . . . . . . . . . . . . . 10
3.3.1 Calculate Optimum TCP Window Size. . . . . . . . . . . 11
3.3.2 Conducting the TCP Throughput Tests. . . . . . . . . . 14
3.3.3 Single vs. Multiple TCP Connection Testing . . . . . . 14
3.3.4 Interpretation of the TCP Throughput Results . . . . . 15
3.4. Traffic Management Tests . . . . . . . . . . . . . . . . . 15
3.4.1 Traffic Shaping Tests. . . . . . . . . . . . . . . . . 16
3.4.1.1 Interpretation of Traffic Shaping Test Results. . . 17
3.4.2 RED Tests. . . . . . . . . . . . . . . . . . . . . . . 17
3.4.2.1 Interpretation of RED Results . . . . . . . . . . . 18
4. Acknowledgements . . . . . . . . . . . . . . . . . . . . . . . 18
5. References . . . . . . . . . . . . . . . . . . . . . . . . . . 19
Authors' Addresses . . . . . . . . . . . . . . . . . . . . . . . . 20
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1. Introduction
Even though RFC2544 was meant to benchmark network equipment and
used by network equipment manufacturers (NEMs), network providers
have used it to benchmark operational networks in order to
verify SLAs (Service Level Agreements) before turning on a service
to their business customers. Testing an operational network prior to
customer activation is referred to as "turn-up" testing and the SLA
is generally Layer 2/3 packet throughput, delay, loss and
jitter.
Network providers are coming to the realization that Layer 2/3 testing
and TCP layer testing are required to more adequately ensure end-user
satisfaction. Therefore, the network provider community desires to
measure network throughput performance at the TCP layer. Measuring
TCP throughput provides a meaningful measure with respect to the end
user's application SLA (and ultimately reach some level of TCP
testing interoperability which does not exist today).
Additionally, end-users (business enterprises) seek to conduct
repeatable TCP throughput tests between enterprise locations. Since
these enterprises rely on the networks of the providers, a common test
methodology (and metrics) would be equally beneficial to both parties.
So the intent behind this draft TCP throughput work is to define
a methodology for testing sustained TCP layer performance. In this
document, sustained TCP throughput is that amount of data per unit
time that TCP transports during equilibrium (steady state), i.e.
after the initial slow start phase. We refer to this state as TCP
Equilibrium, and that the equalibrium throughput is the maximum
achievable for the TCP connection(s).
One other important note; the precursor to conducting the TCP tests
test methodlogy is to perform "network stress tests" such as RFC2544
Layer 2/3 tests or other conventional tests. Examples include
OWAMP or manual packet layer test techniques where packet throughput,
loss, and delay measurements are conducted. It is highly recommended
to run traditional Layer 2/3 type test to verify the integrity of the
network before conducting TCP tests.
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2. Goals of this Methodology
Before defining the goals of this methodology, it is important to
clearly define the areas that are not intended to be measured or
analyzed by such a methodology.
- The methodology is not intended to predict TCP throughput
behavior during the transient stages of a TCP connection, such
as initial slow start.
- The methodology is not intended to definitively benchmark TCP
implementations of one OS to another, although some users may find
some value in conducting qualitative experiments
- The methodology is not intended to provide detailed diagnosis
of problems within end-points or the network itself as related to
non-optimal TCP performance, although a results interpretation
section for each test step may provide insight into potential
issues within the network
In contrast to the above exclusions, the goals of this methodology
are to define a method to conduct a structured, end-to-end
assessment of sustained TCP performance within a managed business
class IP network. A key goal is to establish a set of "best
practices" that an engineer should apply when validating the
ability of a managed network to carry end-user TCP applications.
Some specific goals are to:
- Provide a practical test approach that specifies the more well
understood (and end-user configurable) TCP parameters such as Window
size, MSS (Maximum Segment Size), # connections, and how these affect
the outcome of TCP performance over a network.
- Provide specific test conditions (link speed, RTT, window size,
etc.) and maximum achievable TCP throughput under TCP Equilbrium
conditions. For guideline purposes, provide examples of these test
conditions and the maximum achievable TCP throughput during the
equilbrium state. Section 2.1 provides specific details concerning
the definition of TCP Equilibrium within the context of this draft.
- Define two (2) basic metrics that can be used to compare the
performance of TCP connections under various network conditions
- In test situations where the recommended procedure does not yield
the maximum achievable TCP throughput result, this draft provides some
possible areas within the end host or network that should be
considered for investigation (although again, this draft is not
intended to provide a detailed diagnosis of these issues)
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2.1 TCP Equilibrium State Throughput
TCP connections have three (3) fundamental congestion window phases
as documented in RFC2581. These states are:
- Slow Start, which occurs during the beginning of a TCP transmission
or after a retransmission time out event
- Congestion avoidance, which is the phase during which TCP ramps up
to establish the maximum attainable throughput on an end-end network
path. Retransmissions are a natural by-product of the TCP congestion
avoidance algorithm as it seeks to achieve maximum throughput on
the network path.
- Retransmission phase, which include Fast Retransmit (Tahoe) and Fast
Recovery (Reno and New Reno). When a packet is lost, the Congestion
avoidance phase transitions to a Fast Retransmission or Recovery
Phase dependent upon the TCP implementation.
The following diagram depicts these states.
| ssthresh
TCP | |
Through- | | Equilibrium
put | |\ /\/\/\/\/\ Retransmit /\/\ ...
| | \ / | Time-out /
| | \ / | _______ _/
| Slow _/ |/ | / | Slow _/
| Start _/ Congestion |/ |Start_/ Congestion
| _/ Avoidance Loss | _/ Avoidance
| _/ Event | _/
| _/ |/
|/___________________________________________________________
Time
This TCP methodology provides guidelines to measure the equilibrium
throughput which refers to the maximum sustained rate obtained by
congestion avoidance before packet loss conditions occur (which would
cause the state change from congestion avoidance to a retransmission
phase). All maximum achievable throughputs specified in Section 3 are
with respect to this Equilibrium state.
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2.2 Metrics for TCP Throughput Tests
This draft focuses on a TCP throughtput methodology and also
provides two basic metrics to compare results of various throughput
tests. It is recognized that the complexity and unpredictability of
TCP makes it impossible to develop a complete set of metrics that
account for the myriad of variables (i.e. RTT variation, loss
conditions, TCP implementation, etc.). However, these two basic
metrics faciliate TCP throughput comparisons under varying network
conditions and between network traffic management techniques.
The TCP Efficiency metric is the percentage of bytes that were not
retransmitted and is defined as:
Transmitted Bytes - Retransmitted Bytes
--------------------------------------- x 100
Transmitted Bytes
This metric provides a comparative measure between various QoS
mechanisms such as traffic management, congestion avoidance, and also
various TCP implementations (i.e. Reno, Vegas, etc.).
As an example, if 1000 TCP segments were sent and 20 had to be
retransmitted, the TCP Efficiency would be calculated as:
1000 - 20
--------- x 100 = 98%
1000
The second metric is the TCP Transfer Time, which is simply the time
it takes to transfer a block of data across simultaneous TCP
connections. The concept is useful when benchmarking traffic
management techniques, where multiple connections are generally
required. An example would be the bulk transfer of 10 MB upon 8
separate TCP connections (each connection uploading 10 MB). 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). But by defining the TCP Transfer
Time as the total transfer time of 10MB over all 8 connections, the
single transfer time metric is a useful means to compare various
traffic management techniques (i.e. FiFO, WFQ queuing, WRED, etc.).
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3. TCP Throughput Testing Methodology
This section summarizes the specific test methodology to achieve the
goals listed in Section 2.
As stated in Section 1, it is considered best practice to verify
the integrity of the network by conducting Layer2/3 stress tests
such as RFC2544 (or other methods of network stress tests). If the
network is not performing properly in terms of packet loss, jitter,
etc. then the TCP layer testing will not be meaningful since the
equalibrium throughput would be very difficult to achieve (in a
"dysfunctional" network).
The following represents the sequential order of steps to conduct the
TCP throughput testing methodology:
1. Identify the Path MTU. Packetization Layer Path MTU Discovery
or PLPMTUD (RFC4821) should be conducted to verify the minimum network
path MTU. Conducting PLPMTUD establishes the upper limit for the MSS
to be used in subsequent steps.
2. Baseline Round-trip Delay and Bandwidth. These measurements provide
estimates of the ideal TCP window size, which will be used in
subsequent test steps.
3. TCP Connection Throughput Tests. With baseline measurements
of round trip delay and bandwidth, a series of single and multiple TCP
connection throughput tests can be conducted to baseline the network
performance expectations.
4. Traffic Management Tests. Various traffic management and queuing
techniques are tested in this step, using multiple TCP connections.
Multiple connection testing can verify that the network is configured
properly for traffic shaping versus policing, various queuing
implementations, and RED.
Important to note are some of the key characteristics and
considerations for the TCP test instrument. The test host may be a
standard computer or dedicated communications test instrument
and these TCP test hosts be capable of emulating both a client and a
server.
Whether the TCP test host is a standard computer or dedicated test
instrument, the following areas should be considered when selecting
a test host:
- TCP implementation used by the test host OS, i.e. Linux OS kernel
using TCP Reno, TCP options supported, etc. This will obviously be
more important when using custom test equipment where the TCP
implementation may be customized or tuned to run in higher
performance hardware
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- Most importantly, the TCP test host must be capable of generating
and receiving stateful TCP test traffic at the full link speed of the
network under test. As a general rule of thumb, testing TCP throughput
at rates greater than 100 Mbit/sec generally requires high
performance server hardware or dedicated hardware based test tools.
3.1. Determine Network Path MTU
TCP implementations should use Path MTU Discovery techniques (PMTUD).
PMTUD relies on ICMP 'need to frag' messages to learn the path MTU.
When a device has a packet to send which has the Don't Fragment (DF)
bit in the IP header set and the packet is larger than the Maximum
Transmission Unit (MTU) of the next hop link, the packet is dropped
and the device sends an ICMP 'need to frag' message back to the host
that originated the packet. The ICMP 'need to frag' message includes
the next hop MTU which PMTUD uses to tune the TCP Maximum Segment
Size (MSS). Unfortunately, because many network managers completely
disable ICMP, this technique does not always prove reliable in real
world situations.
Packetization Layer Path MTU Discovery or PLPMTUD (RFC4821) should
be conducted to verify the minimum network path MTU. PLPMTUD can
be used with or without ICMP. The following sections provide a
summary of the PLPMTUD approach and an example using the TCP
protocol. RFC4821 specifies a search_high and search_low parameter
for the MTU. As specified in RFC4821, a value of 1024 is a generally
safe value to choose for search_low in modern networks.
It is important to determine the overhead of the links in the path,
and then to select a TCP MSS size corresponding to the Layer 3 MTU.
For example, if the MTU is 1024 bytes and the TCP/IP headers are 40
bytes, then the MSS would be set to 984 bytes.
An example scenario is a network where the actual path MTU is 1240
bytes. The TCP client probe MUST be capable of setting the MSS for
the probe packets and could start at MSS = 984 (which corresponds
to an MTU size of 1024 bytes).
The TCP client probe would open a TCP connection and advertise the
MSS as 984. Note that the client probe MUST generate these packets
with the DF bit set. The TCP client probe then sends test traffic
per a nominal window size (8KB, etc.). The window size should be
kept small to minimize the possibility of congesting the network,
which could induce congestive loss. The duration of the test should
also be short (10-30 seconds), again to minimize congestive effects
during the test.
In the example of a 1240 byte path MTU, probing with an MSS equal to
984 would yield a successful probe and the test client packets would
be successfully transferred to the test server.
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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
process and reduce the advertised MSS to avoid fragmentation. This
is certainly a desirable feature from a network perspective, but
can yield erroneous test results if the client test probe does not
confirm the negotiated MSS.
The next test probe would use the search_high value and this would
be set to MSS = 1460 to correspond to a 1500 byte MTU. In this
example, the test client would 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 would be to reduce the probe size to
half between the unsuccessful search_high and successful search_low
value, and increase by increments of 1/2 when seeking the upper
limit.
3.2. Baseline Round-trip Delay and Bandwidth
Before stateful TCP testing can begin, it is important to baseline
the round trip delay and bandwidth of the network to be tested.
These measurements provide estimates of the ideal TCP window size,
which will be used in subsequent test steps. These latency and
bandwidth tests should be run over a long enough period of time to
characterize the performance of the network over the course of a
meaningful time period.
One example would be to take samples during various times of the work
day. The goal would be to determine a representative minimum, average,
and maximum RTD and bandwidth for the network under test. Topology
changes are to be avoided during this time of initial convergence
(e.g. in crossing BGP4 boundaries).
In some cases, baselining bandwidth may not be required, since a
network provider's end-to-end topology may be well enough defined.
3.2.1 Techniques to Measure Round Trip Time
Following the definitions used in the references of the appendix;
Round Trip Time (RTT) is the time elapsed between the clocking in of
the first bit of a payload packet to the receipt of the last bit of the
corresponding acknowledgement. Round Trip Delay (RTD) is used
synonymously to twice the Link Latency.
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In any method used to baseline round trip delay between network
end-points, it is important to realize that network latency is the
sum of inherent network delay and congestion. The RTT should be
baselined during "off-peak" hours to obtain a reliable figure for
network latency (versus additional delay caused by congestion).
During the actual sustained TCP throughput tests, it is critical
to measure RTT along with measured TCP throughput. Congestive
effects can be isolated if RTT is concurrently measured.
This is not meant to provide an exhaustive list, but summarizes some
of the more common ways to determine round trip time (RTT) through
the network. The desired resolution of the measurement (i.e. msec
versus usec) may dictate whether the RTT measurement can be achieved
with standard tools such as ICMP ping techniques or whether
specialized test equipment would be required with high precision
timers. The objective in this section is to list several techniques
in order of decreasing accuracy.
- Use test equipment on each end of the network, "looping" the
far-end tester so that a packet stream can be measured end-end. This
test equipment RTT measurement may be compatible with delay
measurement protocols specified in RFC5357.
- Conduct packet captures of TCP test applications using for example
"iperf" or FTP, etc. By running multiple experiments, the packet
captures can be studied to estimate RTT based upon the SYN -> SYN-ACK
handshakes within the TCP connection set-up.
- ICMP Pings may also be adequate to provide round trip time
estimations. Some limitations of ICMP Ping are the msec resolution
and whether the network elements respond to pings (or block them).
3.2.2 Techniques to Measure End-end Bandwidth
There are many well established techniques available to provide
estimated measures of bandwidth over a network. This measurement
should be conducted in both directions of the network, especially for
access networks which are inherently asymmetrical. Some of the
asymmetric implications to TCP performance are documented in RFC-3449
and the results of this work will be further studied to determine
relevance to this draft.
The bandwidth measurement test must be run with stateless IP streams
(not stateful TCP) in order to determine the available bandwidth in
each direction. And this test should obviously be performed at
various intervals throughout a business day (or even across a week).
Ideally, the bandwidth test should produce a log output of the
bandwidth achieved across the test interval AND the round trip delay.
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And during the actual TCP level performance measurements (Sections
3.3 - 3.5), the test tool must be able to track round trip time
of the TCP connection(s) during the test. Measuring round trip time
variation (aka "jitter") provides insight into effects of congestive
delay on the sustained throughput achieved for the TCP layer test.
3.3. TCP Throughput Tests
This draft specifically defines TCP throughput techniques to verify
sustained TCP performance in a managed business network. Defined
in section 2.1, the equalibrium throughput reflects the maximum
rate achieved by a TCP connection within the congestion avoidance
phase on a end-end network path. This section and others will define
the method to conduct these sustained throughput tests and guidelines
of the predicted results.
With baseline measurements of round trip time and bandwidth
from section 3.2, a series of single and multiple TCP connection
throughput tests can be conducted to baseline network performance
against expectations.
3.3.1 Calculate Optimum TCP Window Size
The optimum TCP window size can be calculated from the bandwidth delay
product (BDP), which is:
BDP (bits) = RTT (sec) x Bandwidth (bps)
By dividing the BDP by 8, the "ideal" TCP window size is calculated.
An example would be a T3 link with 25 msec RTT. The BDP would equal
~1,105,000 bits and the ideal TCP window would equal ~138,000 bytes.
The following table provides some representative network link speeds,
latency, BDP, and associated "optimum" TCP window size. Sustained
TCP transfers should reach nearly 100% throughput, minus the overhead
of Layers 1-3 and the divisor of the MSS into the window.
For this single connection baseline test, the MSS size will effect
the achieved throughput (especially for smaller TCP window sizes).
Table 3.2 provides the achievable, equalibrium TCP throughput (at
Layer 4) using 1460 byte MSS. Also in this table, the case of 58 byte
L1-L4 overhead including the Ethernet CRC32 is used for simplicity.
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Table 3.2: Link Speed, RTT and calculated BDP, TCP Throughput
Link Ideal TCP Maximum Achievable
Speed* RTT (ms) BDP (bits) Window (kbytes) TCP Throughput(Mbps)
----------------------------------------------------------------------
T1 20 30,720 3.84 1.17
T1 50 76,800 9.60 1.40
T1 100 153,600 19.20 1.40
T3 10 442,100 55.26 42.05
T3 15 663,150 82.89 42.05
T3 25 1,105,250 138.16 41.52
T3(ATM) 10 407,040 50.88 36.50
T3(ATM) 15 610,560 76.32 36.23
T3(ATM) 25 1,017,600 127.20 36.27
100M 1 100,000 12.50 91.98
100M 2 200,000 25.00 93.44
100M 5 500,000 62.50 93.44
1Gig 0.1 100,000 12.50 919.82
1Gig 0.5 500,000 62.50 934.47
1Gig 1 1,000,000 125.00 934.47
10Gig 0.05 500,000 62.50 9,344.67
10Gig 0.3 3,000,000 375.00 9,344.67
* Note that link speed is the minimum link speed throughput a network;
i.e. WAN with T1 link, etc.
Also, the following link speeds (available payload bandwidth) were
used for the WAN entries:
- T1 = 1.536 Mbits/sec (B8ZS line encoding facility)
- T3 = 44.21 Mbits/sec (C-Bit Framing)
- T3(ATM) = 36.86 Mbits/sec (C-Bit Framing & PLCP, 96000 Cells per
second)
The calculation method used in this document is a 3 step process :
1 - We determine what should be the optimal TCP Window size value
based on the optimal quantity of "in-flight" octets discovered by
the BDP calculation. We take into consideration that the TCP
Window size has to be an exact multiple value of the MSS.
2 - Then we calculate the achievable layer 2 throughput by multiplying
the value determined in step 1 with the MSS & (MSS + L2 + L3 + L4
Overheads) divided by the RTT.
3 - Finally, we multiply the calculated value of step 2 by the MSS
versus (MSS + L2 + L3 + L4 Overheads) ratio.
This gives us the achievable TCP Throughput value. Sometimes, the
maximum achievable throughput is limited by the maximum achievable
quantity of Ethernet Frames per second on the physical media. Then
this value is used in step 2 instead of the calculated one.
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The following diagram compares achievable TCP throughputs on a T3 link
with Windows 2000/XP TCP window sizes of 16KB versus 64KB.
45|
| _____42.1M
40| |64K|
TCP | | |
Throughput 35| | | _____34.3M
in Mbps | | | |64K|
30| | | | |
| | | | |
25| | | | |
| | | | |
20| | | | | _____20.5M
| | | | | |64K|
15| 14.5M____| | | | | |
| |16K| | | | | |
10| | | | 9.6M+---+ | | |
| | | | |16K| | 5.8M____+ |
5| | | | | | | |16K| |
|______+___+___+_______+___+___+_______+__ +___+_______
10 15 25
RTT in milliseconds
The following diagram shows the achievable TCP throughput on a 25ms T3
when the TCP Window size is increased and with the RFC1323 TCP Window
scaling option.
45|
| +-----+42.47M
40| | |
TCP | | |
Throughput 35| | |
in Mbps | | |
30| | |
| | |
25| | |
| ______ 21.23M | |
20| | | | |
| | | | |
15| | | | |
| | | | |
10| +----+10.62M | | | |
| _______5.31M | | | | | |
5| | | | | | | | |
|__+_____+______+____+___________+____+________+_____+___
16 32 64 128
TCP Window size in KBytes
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3.3.2 Conducting the TCP Throughput Tests
There are several TCP tools that are commonly used in the network
world and one of the most common is the "iperf" tool. With this tool,
hosts are installed at each end of the network segment; one as client
and the other as server. The TCP Window size of both the client and
the server can be maunally set and the achieved throughput is measured,
either uni-directionally or bi-directionally. For higher BDP
situations in lossy networks (long fat networks or satellite links,
etc.), TCP options such as Selective Acknowledgment should be
considered and also become part of the window size / throughput
characterization.
Host hardware performance must be well understood before conducting
the TCP throughput tests and other tests in the following sections.
Dedicated test equipment will generally be required, especially for
line rates of GigE and 10 GigE.
The TCP throughput test should be run over a a long enough duration
to properly exercise network buffers and also characterize performance
during different time periods of the day. The results must be logged
at the desired interval and the test must record RTT and TCP
retransmissions at each interval.
This correlation of retransmissions and RTT over the course of the
test will clearly identify which portions of the transfer reached
TCP Equilbrium state and to what effect increased RTT (congestive
effects) may have been the cause of reduced equilibrium performance.
Additionally, the TCP Efficiency and TCP Transfer time metrics should
be logged in order to further characterize the window size tests.
3.3.3 Single vs. Multiple TCP Connection Testing
The decision whether to conduct single or multiple TCP connection
tests depends upon the size of the BDP in relation to the window sizes
configured in the end-user environment. For example, if the BDP for a
long-fat pipe turns out to be 2MB, then it is probably more realistic
to test this pipe with multiple connections. Assuming typical host
computer window settings of 64 KB, using 32 connections would
realistically test this pipe.
The following table is provided to illustrate the relationship of the
BDP, window size, and the number of connections required to utilize the
the available capacity. For this example, the network bandwidth is
500 Mbps, RTT is equal to 5 ms, and the BDP equates to 312 KBytes.
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#Connections
Window to Fill Link
------------------------
16KB 20
32KB 10
64KB 5
128KB 3
The TCP Transfer Time metric is useful for conducting multiple
connection tests. Each connection should be configured to transfer
a certain payload (i.e. 100 MB), and the TCP Transfer time provides
a simple metric to verify the actual versus expected results.
Note that the TCP transfer time is the time for all connections to
complete the transfer of the configured payload size. From the
example table listed above, the 64KB window is considered. Each of
the 5 connections would be configured to transfer 100MB, and each
TCP should obtain a maximum of 100 Mb/sec per connection. So for this
example, the 100MB payload should be transferred across the connections
in approximately 8 seconds (which would be the ideal TCP transfer time
for these conditions).
Additionally, the TCP Efficiency metric should be computed for each
connection tested (defined in section 2.2).
3.3.4 Interpretation of the TCP Throughput Results
At the end of this step, the user will document the theoretical BDP
and a set of Window size experiments with measured TCP throughput for
each TCP window size setting. For cases where the sustained TCP
throughput does not equal the predicted value, some possible causes
are listed:
- Network congestion causing packet loss; the TCP Efficiency metric
is a useful gauge to compare network performance
- Network congestion not causing packet loss but increasing RTT
- Intermediate network devices which actively regenerate the TCP
connection and can alter window size, MSS, etc.
- Over utilization of available link or rate limiting (policing). More
discussion of traffic management tests follows in section 3.4
3.4. Traffic Management Tests
In most cases, the network connection between two geographic locations
(branch offices, etc.) is lower than the network connection of the
host computers. An example would be LAN connectivity of GigE and
WAN connectivity of 100 Mbps. The WAN connectivity may be physically
100 Mbps or logically 100 Mbps (over a GigE WAN connection). In the
later case, rate limiting is used to provide the WAN bandwidth per the
SLA.
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Traffic management techniques are employed to provide various forms of
QoS, the more common include:
- Traffic Shaping
- Priority Queuing
- Random Early Discard (RED, etc.)
Configuring the end-end network with these various traffic management
mechanisms is a complex under-taking. For traffic shaping and RED
techniques, the end goal is to provide better performance for bursty
traffic such as TCP (RED is specifically intended for TCP).
This section of the methodology provides guidelines to test traffic
shaping and RED implementations. As in section 3.3, host hardware
performance must be well understood before conducting the traffic
shaping and RED tests. Dedicated test equipment will generally be
required, especially for line rates of GigE and 10 GigE.
3.4.1 Traffic Shaping Tests
For services where the available bandwidth is rate limited, there are
two (2) techniques used to implement rate limiting: traffic policing
and traffic shaping.
Simply stated, traffic policing marks and/or drops packets which
exceed the SLA bandwidth (in most cases, excess traffic is dropped).
Traffic shaping employs the use of queues to smooth the bursty
traffic and then send out within the SLA bandwidth limit (without
dropping packets unless the traffic shaping queue is exceeded).
Traffic shaping is generally configured for TCP data services and
can provide improved TCP performance since the retransmissions are
reduced, which in turn optimizes TCP throughput for the given
available bandwidth. Through this section, the available rate-limited
bandwidth shall be referred to as the "bottleneck bandwidth".
The ability to detect proper traffic shaping is more easily diagnosed
when conducting a multiple TCP connection test. Proper shaping will
provide a fair distribution of the available bottleneck bandwidth,
while traffic policing will not.
The traffic shaping tests build upon the concepts of multiple
connection testing as defined in section 3.3.3. Calculating the BDP
for the bottleneck bandwidth is first required and then selecting
the number of connections / window size per connection.
Similar to the example in section 3.3, a typical test scenario might
be: GigE LAN with a 100Mbps bottleneck bandwidth (rate limited logical
interface), and 5 msec RTT. This would require five (5) TCP
connections of 64 KB window size evenly fill the bottleneck bandwidth
(about 100 Mbps per connection).
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The traffic shaping should be run over a long enough duration to
properly exercise network buffers 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
Efficiency and TCP Transfer time metric. Additionally, it is
recommended to log RTT and retransmissions per connection over the test
interval.
3.4.1.1 Interpretation of Traffic Shaping Test Restults
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 would not
be fair and the resulting throughput plot would reveal "spikey"
connection throughput consumption of the competing TCP connections
(due to the retransmissions).
3.4.2 RED Tests
Random Early Discard techniques are specifically targeted to provide
congestion avoidance for TCP traffic. Before the network element queue
"fills" and enters the tail drop state, RED drops packets at
configurable queue depth thresholds. This action causes TCP
connections to back-off which helps to prevent tail drop, which in
turn helps to prevent global TCP synchronization.
Again, rate limited interfaces can benefit greatly from RED based
techniques. Without RED, TCP is generally not able to achieve the full
bandwidth of the bottleneck interface. With RED enabled, TCP
congestion avoidance throttles the connections on the higher speed
interface (i.e. LAN) and can reach equalibrium with the bottleneck
bandwidth (achieving closer to full throughput).
The ability to detect proper RED configuration is more easily diagnosed
when conducting a multiple TCP connection test. Multiple TCP
connections provide the multiple bursty sources that emulate the
real-world conditions for which RED was intended.
The RED tests also build upon the concepts of multiple connection
testing as defined in secion 3.3.3. Calculating the BDP for the
bottleneck bandwidth is first required and then selecting the number of
connections / window size per connection.
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For RED testing, the desired effect is to cause the TCP connections to
burst beyond the bottleneck bandwidth so that queue drops will occur.
Using the same example from section 3.4.1 (traffic shaping), the
500 Mbps bottleneck bandwidth requires 5 TCP connections (with window
size of 64Kb) to fill the capacity. Some experimentation is required,
but it is recommended to start with double the number of connections
to stress the network element buffers / queues. In this example, 10
connections would produce TCP bursts of 64KB for each connection.
If the timing of the TCP tester permits, these TCP bursts could stress
queue sizes in the 512KB range. Again experimentation will be required
and the proper number of TCP connections / window size will be dictated
by the size the network element queue.
3.4.2.1 Interpretation of RED Results
The default queuing technique for most network devices is FIFO based.
Without RED, the FIFO based queue will cause excessive loss to all of
the TCP connections and in the worst case global TCP synchronization.
By plotting the aggregate throughput achieved on the bottleneck
interface, proper RED operation can 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 would be greatly reduced
(generally 20-40%) and the TCP Transfer time would be proportionally
longer then the ideal transfer time.
Additionally, the TCP Transfer Efficiency metric is useful, since
non-RED implementations will exhibit a lower TCP Tranfer Efficiency
than RED implementations.
4. Acknowledgements
The author would like to thank Gilles Forget, Loki Jorgenson,
and Reinhard Schrage for technical review and contributions to this
draft-03 memo.
Also thanks to Matt Mathis and Matt Zekauskas for many good comments
through email exchange and for pointing us to great sources of
information pertaining to past works in the TCP capacity area.
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5. References
[RFC2581] Allman, M., Paxson, V., Stevens W., "TCP Congestion
Control", RFC 2581, June 1999.
[RFC3148] Mathis M., Allman, M., "A Framework for Defining
Empirical Bulk Transfer Capacity Metrics", RFC 3148, July
2001.
[RFC2544] Bradner, S., McQuaid, J., "Benchmarking Methodology for
Network Interconnect Devices", RFC 2544, June 1999
[RFC3449] Balakrishnan, H., Padmanabhan, V. N., Fairhurst, G.,
Sooriyabandara, M., "TCP Performance Implications of
Network Path Asymmetry", RFC 3449, December 2002
[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
[MSMO] The Macroscopic Behavior of the TCP Congestion Avoidance
Algorithm Mathis, M.,Semke, J, Mahdavi, J, Ott, T
July 1997 SIGCOMM Computer Communication Review,
Volume 27 Issue 3
[Stevens Vol1] TCP/IP Illustrated, Vol1, The Protocols
Addison-Wesley
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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
Loki Jorgenson
nooCore
Phone: (604) 908-5833
ljorgenson@nooCore.com
Reinhard Schrage
Schrage Consulting
Phone: +49 (0) 5137 909540
reinhard@schrageconsult.com
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