RFC Number : 1323
Title : TCP Extensions for High Performance.
Network Working Group V. Jacobson
Request for Comments: 1323 LBL
Obsoletes: RFC 1072, RFC 1185 R. Braden
ISI
D. Borman
Cray Research
May 1992
TCP Extensions for High Performance
Status of This Memo
This RFC specifies an IAB standards track protocol for the Internet
community, and requests discussion and suggestions for improvements.
Please refer to the current edition of the 'IAB Official Protocol
Standards' for the standardization state and status of this protocol.
Distribution of this memo is unlimited.
Abstract
This memo presents a set of TCP extensions to improve performance
over large bandwidth*delay product paths and to provide reliable
operation over very high-speed paths. It defines new TCP options for
scaled windows and timestamps, which are designed to provide
compatible interworking with TCP's that do not implement the
extensions. The timestamps are used for two distinct mechanisms:
RTTM (Round Trip Time Measurement) and PAWS (Protect Against Wrapped
Sequences). Selective acknowledgments are not included in this memo.
This memo combines and supersedes RFC-1072 and RFC-1185, adding
additional clarification and more detailed specification. Appendix C
summarizes the changes from the earlier RFCs.
The TCP protocol [Postel81] was designed to operate reliably over
almost any transmission medium regardless of transmission rate,
delay, corruption, duplication, or reordering of segments.
Production TCP implementations currently adapt to transfer rates in
the range of 100 bps to 10**7 bps and round-trip delays in the range
1 ms to 100 seconds. Recent work on TCP performance has shown that
TCP can work well over a variety of Internet paths, ranging from 800
Mbit/sec I/O channels to 300 bit/sec dial-up modems [Jacobson88a].
The introduction of fiber optics is resulting in ever-higher
transmission speeds, and the fastest paths are moving out of the
domain for which TCP was originally engineered. This memo defines a
set of modest extensions to TCP to extend the domain of its
application to match this increasing network capability. It is based
upon and obsoletes RFC-1072 [Jacobson88b] and RFC-1185 [Jacobson90b].
There is no one-line answer to the question: 'How fast can TCP go?'.
There are two separate kinds of issues, performance and reliability,
and each depends upon different parameters. We discuss each in turn.
1.1 TCP Performance
TCP performance depends not upon the transfer rate itself, but
rather upon the product of the transfer rate and the round-trip
delay. This 'bandwidth*delay product' measures the amount of data
that would 'fill the pipe'; it is the buffer space required at
sender and receiver to obtain maximum throughput on the TCP
connection over the path, i.e., the amount of unacknowledged data
that TCP must handle in order to keep the pipeline full. TCP
performance problems arise when the bandwidth*delay product is
large. We refer to an Internet path operating in this region as a
'long, fat pipe', and a network containing this path as an 'LFN'
(pronounced 'elephan(t)').
High-capacity packet satellite channels (e.g., DARPA's Wideband
Net) are LFN's. For example, a DS1-speed satellite channel has a
bandwidth*delay product of 10**6 bits or more; this corresponds to
100 outstanding TCP segments of 1200 bytes each. Terrestrial
fiber-optical paths will also fall into the LFN class; for
example, a cross-country delay of 30 ms at a DS3 bandwidth
(45Mbps) also exceeds 10**6 bits.
There are three fundamental performance problems with the current
TCP over LFN paths:
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(1) Window Size Limit
The TCP header uses a 16 bit field to report the receive
window size to the sender. Therefore, the largest window
that can be used is 2**16 = 65K bytes.
To circumvent this problem, Section 2 of this memo defines a
new TCP option, 'Window Scale', to allow windows larger than
2**16. This option defines an implicit scale factor, which
is used to multiply the window size value found in a TCP
header to obtain the true window size.
(2) Recovery from Losses
Packet losses in an LFN can have a catastrophic effect on
throughput. Until recently, properly-operating TCP
implementations would cause the data pipeline to drain with
every packet loss, and require a slow-start action to
recover. Recently, the Fast Retransmit and Fast Recovery
algorithms [Jacobson90c] have been introduced. Their
combined effect is to recover from one packet loss per
window, without draining the pipeline. However, more than
one packet loss per window typically results in a
retransmission timeout and the resulting pipeline drain and
slow start.
Expanding the window size to match the capacity of an LFN
results in a corresponding increase of the probability of
more than one packet per window being dropped. This could
have a devastating effect upon the throughput of TCP over an
LFN. In addition, if a congestion control mechanism based
upon some form of random dropping were introduced into
gateways, randomly spaced packet drops would become common,
possible increasing the probability of dropping more than one
packet per window.
To generalize the Fast Retransmit/Fast Recovery mechanism to
handle multiple packets dropped per window, selective
acknowledgments are required. Unlike the normal cumulative
acknowledgments of TCP, selective acknowledgments give the
sender a complete picture of which segments are queued at the
receiver and which have not yet arrived. Some evidence in
favor of selective acknowledgments has been published
[NBS85], and selective acknowledgments have been included in
a number of experimental Internet protocols -- VMTP
[Cheriton88], NETBLT [Clark87], and RDP [Velten84], and
proposed for OSI TP4 [NBS85]. However, in the non-LFN
regime, selective acknowledgments reduce the number of
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packets retransmitted but do not otherwise improve
performance, making their complexity of questionable value.
However, selective acknowledgments are expected to become
much more important in the LFN regime.
RFC-1072 defined a new TCP 'SACK' option to send a selective
acknowledgment. However, there are important technical
issues to be worked out concerning both the format and
semantics of the SACK option. Therefore, SACK has been
omitted from this package of extensions. It is hoped that
SACK can 'catch up' during the standardization process.
(3) Round-Trip Measurement
TCP implements reliable data delivery by retransmitting
segments that are not acknowledged within some retransmission
timeout (RTO) interval. Accurate dynamic determination of an
appropriate RTO is essential to TCP performance. RTO is
determined by estimating the mean and variance of the
measured round-trip time (RTT), i.e., the time interval
between sending a segment and receiving an acknowledgment for
it [Jacobson88a].
Section 4 introduces a new TCP option, 'Timestamps', and then
defines a mechanism using this option that allows nearly
every segment, including retransmissions, to be timed at
negligible computational cost. We use the mnemonic RTTM
(Round Trip Time Measurement) for this mechanism, to
distinguish it from other uses of the Timestamps option.
1.2 TCP Reliability
Now we turn from performance to reliability. High transfer rate
enters TCP performance through the bandwidth*delay product.
However, high transfer rate alone can threaten TCP reliability by
violating the assumptions behind the TCP mechanism for duplicate
detection and sequencing.
An especially serious kind of error may result from an accidental
reuse of TCP sequence numbers in data segments. Suppose that an
'old duplicate segment', e.g., a duplicate data segment that was
delayed in Internet queues, is delivered to the receiver at the
wrong moment, so that its sequence numbers falls somewhere within
the current window. There would be no checksum failure to warn of
the error, and the result could be an undetected corruption of the
data. Reception of an old duplicate ACK segment at the
transmitter could be only slightly less serious: it is likely to
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lock up the connection so that no further progress can be made,
forcing an RST on the connection.
TCP reliability depends upon the existence of a bound on the
lifetime of a segment: the 'Maximum Segment Lifetime' or MSL. An
MSL is generally required by any reliable transport protocol,
since every sequence number field must be finite, and therefore
any sequence number may eventually be reused. In the Internet
protocol suite, the MSL bound is enforced by an IP-layer
mechanism, the 'Time-to-Live' or TTL field.
Duplication of sequence numbers might happen in either of two
ways:
(1) Sequence number wrap-around on the current connection
A TCP sequence number contains 32 bits. At a high enough
transfer rate, the 32-bit sequence space may be 'wrapped'
(cycled) within the time that a segment is delayed in queues.
(2) Earlier incarnation of the connection
Suppose that a connection terminates, either by a proper
close sequence or due to a host crash, and the same
connection (i.e., using the same pair of sockets) is
immediately reopened. A delayed segment from the terminated
connection could fall within the current window for the new
incarnation and be accepted as valid.
Duplicates from earlier incarnations, Case (2), are avoided by
enforcing the current fixed MSL of the TCP spec, as explained in
Section 5.3 and Appendix B. However, case (1), avoiding the
reuse of sequence numbers within the same connection, requires an
MSL bound that depends upon the transfer rate, and at high enough
rates, a new mechanism is required.
More specifically, if the maximum effective bandwidth at which TCP
is able to transmit over a particular path is B bytes per second,
then the following constraint must be satisfied for error-free
operation:
2**31 / B > MSL (secs) [1]
The following table shows the value for Twrap = 2**31/B in
seconds, for some important values of the bandwidth B:
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It is clear that wrap-around of the sequence space is not a
problem for 56kbps packet switching or even 10Mbps Ethernets. On
the other hand, at DS3 and FDDI speeds, Twrap is comparable to the
2 minute MSL assumed by the TCP specification [Postel81]. Moving
towards gigabit speeds, Twrap becomes too small for reliable
enforcement by the Internet TTL mechanism.
The 16-bit window field of TCP limits the effective bandwidth B to
2**16/RTT, where RTT is the round-trip time in seconds
[McKenzie89]. If the RTT is large enough, this limits B to a
value that meets the constraint [1] for a large MSL value. For
example, consider a transcontinental backbone with an RTT of 60ms
(set by the laws of physics). With the bandwidth*delay product
limited to 64KB by the TCP window size, B is then limited to
1.1MBps, no matter how high the theoretical transfer rate of the
path. This corresponds to cycling the sequence number space in
Twrap= 2000 secs, which is safe in today's Internet.
It is important to understand that the culprit is not the larger
window but rather the high bandwidth. For example, consider a
(very large) FDDI LAN with a diameter of 10km. Using the speed of
light, we can compute the RTT across the ring as
(2*10**4)/(3*10**8) = 67 microseconds, and the delay*bandwidth
product is then 833 bytes. A TCP connection across this LAN using
a window of only 833 bytes will run at the full 100mbps and can
wrap the sequence space in about 3 minutes, very close to the MSL
of TCP. Thus, high speed alone can cause a reliability problem
with sequence number wrap-around, even without extended windows.
Watson's Delta-T protocol [Watson81] includes network-layer
mechanisms for precise enforcement of an MSL. In contrast, the IP
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mechanism for MSL enforcement is loosely defined and even more
loosely implemented in the Internet. Therefore, it is unwise to
depend upon active enforcement of MSL for TCP connections, and it
is unrealistic to imagine setting MSL's smaller than the current
values (e.g., 120 seconds specified for TCP).
A possible fix for the problem of cycling the sequence space would
be to increase the size of the TCP sequence number field. For
example, the sequence number field (and also the acknowledgment
field) could be expanded to 64 bits. This could be done either by
changing the TCP header or by means of an additional option.
Section 5 presents a different mechanism, which we call PAWS
(Protect Against Wrapped Sequence numbers), to extend TCP
reliability to transfer rates well beyond the foreseeable upper
limit of network bandwidths. PAWS uses the TCP Timestamps option
defined in Section 4 to protect against old duplicates from the
same connection.
1.3 Using TCP options
The extensions defined in this memo all use new TCP options. We
must address two possible issues concerning the use of TCP
options: (1) compatibility and (2) overhead.
We must pay careful attention to compatibility, i.e., to
interoperation with existing implementations. The only TCP option
defined previously, MSS, may appear only on a SYN segment. Every
implementation should (and we expect that most will) ignore
unknown options on SYN segments. However, some buggy TCP
implementation might be crashed by the first appearance of an
option on a non-SYN segment. Therefore, for each of the
extensions defined below, TCP options will be sent on non-SYN
segments only when an exchange of options on the SYN segments has
indicated that both sides understand the extension. Furthermore,
an extension option will be sent in a segment only if
the corresponding option was received in the initial segment.
A question may be raised about the bandwidth and processing
overhead for TCP options. Those options that occur on SYN
segments are not likely to cause a performance concern. Opening a
TCP connection requires execution of significant special-case
code, and the processing of options is unlikely to increase that
cost significantly.
On the other hand, a Timestamps option may appear in any data or
ACK segment, adding 12 bytes to the 20-byte TCP header. We
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believe that the bandwidth saved by reducing unnecessary
retransmissions will more than pay for the extra header bandwidth.
There is also an issue about the processing overhead for parsing
the variable byte-aligned format of options, particularly with a
RISC-architecture CPU. To meet this concern, Appendix A contains
a recommended layout of the options in TCP headers to achieve
reasonable data field alignment. In the spirit of Header
Prediction, a TCP can quickly test for this layout and if it is
verified then use a fast path. Hosts that use this canonical
layout will effectively use the options as a set of fixed-format
fields appended to the TCP header. However, to retain the
philosophical and protocol framework of TCP options, a TCP must be
prepared to parse an arbitrary options field, albeit with less
efficiency.
Finally, we observe that most of the mechanisms defined in this
memo are important for LFN's and/or very high-speed networks. For
low-speed networks, it might be a performance optimization to NOT
use these mechanisms. A TCP vendor concerned about optimal
performance over low-speed paths might consider turning these
extensions off for low-speed paths, or allow a user or
installation manager to disable them.
2. TCP WINDOW SCALE OPTION
2.1 Introduction
The window scale extension expands the definition of the TCP
window to 32 bits and then uses a scale factor to carry this 32-
bit value in the 16-bit Window field of the TCP header (SEG.WND in
RFC-793). The scale factor is carried in a new TCP option, Window
Scale. This option is sent only in a SYN segment (a segment with
the SYN bit on), hence the window scale is fixed in each direction
when a connection is opened. (Another design choice would be to
specify the window scale in every TCP segment. It would be
incorrect to send a window scale option only when the scale factor
changed, since a TCP option in an acknowledgement segment will not
be delivered reliably (unless the ACK happens to be piggy-backed
on data in the other direction). Fixing the scale when the
connection is opened has the advantage of lower overhead but the
disadvantage that the scale factor cannot be changed during the
connection.)
The maximum receive window, and therefore the scale factor, is
determined by the maximum receive buffer space. In a typical
modern implementation, this maximum buffer space is set by default
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but can be overridden by a user program before a TCP connection is
opened. This determines the scale factor, and therefore no new
user interface is needed for window scaling.
2.2 Window Scale Option
The three-byte Window Scale option may be sent in a SYN segment by
a TCP. It has two purposes: (1) indicate that the TCP is prepared
to do both send and receive window scaling, and (2) communicate a
scale factor to be applied to its receive window. Thus, a TCP
that is prepared to scale windows should send the option, even if
its own scale factor is 1. The scale factor is limited to a power
of two and encoded logarithmically, so it may be implemented by
binary shift operations.
This option is an offer, not a promise; both sides must send
Window Scale options in their SYN segments to enable window
scaling in either direction. If window scaling is enabled,
then the TCP that sent this option will right-shift its true
receive-window values by 'shift.cnt' bits for transmission in
SEG.WND. The value 'shift.cnt' may be zero (offering to scale,
while applying a scale factor of 1 to the receive window).
This option may be sent in an initial segment (i.e., a
segment with the SYN bit on and the ACK bit off). It may also
be sent in a segment, but only if a Window Scale op-
tion was received in the initial segment. A Window Scale
option in a segment without a SYN bit should be ignored.
The Window field in a SYN (i.e., a or ) segment
itself is never scaled.
2.3 Using the Window Scale Option
A model implementation of window scaling is as follows, using the
notation of RFC-793 [Postel81]:
* All windows are treated as 32-bit quantities for storage in
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RFC 1323 TCP Extensions for High Performance May 1992
the connection control block and for local calculations.
This includes the send-window (SND.WND) and the receive-
window (RCV.WND) values, as well as the congestion window.
* The connection state is augmented by two window shift counts,
Snd.Wind.Scale and Rcv.Wind.Scale, to be applied to the
incoming and outgoing window fields, respectively.
* If a TCP receives a segment containing a Window Scale
option, it sends its own Window Scale option in the segment.
* The Window Scale option is sent with shift.cnt = R, where R
is the value that the TCP would like to use for its receive
window.
* Upon receiving a SYN segment with a Window Scale option
containing shift.cnt = S, a TCP sets Snd.Wind.Scale to S and
sets Rcv.Wind.Scale to R; otherwise, it sets both
Snd.Wind.Scale and Rcv.Wind.Scale to zero.
* The window field (SEG.WND) in the header of every incoming
segment, with the exception of SYN segments, is left-shifted
by Snd.Wind.Scale bits before updating SND.WND:
SND.WND = SEG.WND << Snd.Wind.Scale
(assuming the other conditions of RFC793 are met, and using
the 'C' notation '<<' for left-shift).
* The window field (SEG.WND) of every outgoing segment, with
the exception of SYN segments, is right-shifted by
Rcv.Wind.Scale bits:
SEG.WND = RCV.WND >> Rcv.Wind.Scale.
TCP determines if a data segment is 'old' or 'new' by testing
whether its sequence number is within 2**31 bytes of the left edge
of the window, and if it is not, discarding the data as 'old'. To
insure that new data is never mistakenly considered old and vice-
versa, the left edge of the sender's window has to be at most
2**31 away from the right edge of the receiver's window.
Similarly with the sender's right edge and receiver's left edge.
Since the right and left edges of either the sender's or
receiver's window differ by the window size, and since the sender
and receiver windows can be out of phase by at most the window
size, the above constraints imply that 2 * the max window size
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RFC 1323 TCP Extensions for High Performance May 1992
must be less than 2**31, or
max window < 2**30
Since the max window is 2**S (where S is the scaling shift count)
times at most 2**16 - 1 (the maximum unscaled window), the maximum
window is guaranteed to be < 2*30 if S <= 14. Thus, the shift
count must be limited to 14 (which allows windows of 2**30 = 1
Gbyte). If a Window Scale option is received with a shift.cnt
value exceeding 14, the TCP should log the error but use 14
instead of the specified value.
The scale factor applies only to the Window field as transmitted
in the TCP header; each TCP using extended windows will maintain
the window values locally as 32-bit numbers. For example, the
'congestion window' computed by Slow Start and Congestion
Avoidance is not affected by the scale factor, so window scaling
will not introduce quantization into the congestion window.
3. RTTM: ROUND-TRIP TIME MEASUREMENT
3.1 Introduction
Accurate and current RTT estimates are necessary to adapt to
changing traffic conditions and to avoid an instability known as
'congestion collapse' [Nagle84] in a busy network. However,
accurate measurement of RTT may be difficult both in theory and in
implementation.
Many TCP implementations base their RTT measurements upon a sample
of only one packet per window. While this yields an adequate
approximation to the RTT for small windows, it results in an
unacceptably poor RTT estimate for an LFN. If we look at RTT
estimation as a signal processing problem (which it is), a data
signal at some frequency, the packet rate, is being sampled at a
lower frequency, the window rate. This lower sampling frequency
violates Nyquist's criteria and may therefore introduce 'aliasing'
artifacts into the estimated RTT [Hamming77].
A good RTT estimator with a conservative retransmission timeout
calculation can tolerate aliasing when the sampling frequency is
'close' to the data frequency. For example, with a window of 8
packets, the sample rate is 1/8 the data frequency -- less than an
order of magnitude different. However, when the window is tens or
hundreds of packets, the RTT estimator may be seriously in error,
resulting in spurious retransmissions.
If there are dropped packets, the problem becomes worse. Zhang
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RFC 1323 TCP Extensions for High Performance May 1992
[Zhang86], Jain [Jain86] and Karn [Karn87] have shown that it is
not possible to accumulate reliable RTT estimates if retransmitted
segments are included in the estimate. Since a full window of
data will have been transmitted prior to a retransmission, all of
the segments in that window will have to be ACKed before the next
RTT sample can be taken. This means at least an additional
window's worth of time between RTT measurements and, as the error
rate approaches one per window of data (e.g., 10**-6 errors per
bit for the Wideband satellite network), it becomes effectively
impossible to obtain a valid RTT measurement.
A solution to these problems, which actually simplifies the sender
substantially, is as follows: using TCP options, the sender places
a timestamp in each data segment, and the receiver reflects these
timestamps back in ACK segments. Then a single subtract gives the
sender an accurate RTT measurement for every ACK segment (which
will correspond to every other data segment, with a sensible
receiver). We call this the RTTM (Round-Trip Time Measurement)
mechanism.
It is vitally important to use the RTTM mechanism with big
windows; otherwise, the door is opened to some dangerous
instabilities due to aliasing. Furthermore, the option is
probably useful for all TCP's, since it simplifies the sender.
3.2 TCP Timestamps Option
TCP is a symmetric protocol, allowing data to be sent at any time
in either direction, and therefore timestamp echoing may occur in
either direction. For simplicity and symmetry, we specify that
timestamps always be sent and echoed in both directions. For
efficiency, we combine the timestamp and timestamp reply fields
into a single TCP Timestamps Option.
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RFC 1323 TCP Extensions for High Performance May 1992
The Timestamps option carries two four-byte timestamp fields.
The Timestamp Value field (TSval) contains the current value of
the timestamp clock of the TCP sending the option.
The Timestamp Echo Reply field (TSecr) is only valid if the ACK
bit is set in the TCP header; if it is valid, it echos a times-
tamp value that was sent by the remote TCP in the TSval field
of a Timestamps option. When TSecr is not valid, its value
must be zero. The TSecr value will generally be from the most
recent Timestamp option that was received; however, there are
exceptions that are explained below.
A TCP may send the Timestamps option (TSopt) in an initial
segment (i.e., segment containing a SYN bit and no ACK
bit), and may send a TSopt in other segments only if it re-
ceived a TSopt in the initial segment for the connection.
3.3 The RTTM Mechanism
The timestamp value to be sent in TSval is to be obtained from a
(virtual) clock that we call the 'timestamp clock'. Its values
must be at least approximately proportional to real time, in order
to measure actual RTT.
The following example illustrates a one-way data flow with
segments arriving in sequence without loss. Here A, B, C...
represent data blocks occupying successive blocks of sequence
numbers, and ACK(A),... represent the corresponding cumulative
acknowledgments. The two timestamp fields of the Timestamps
option are shown symbolically as . Each TSecr
field contains the value most recently received in a TSval field.
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