rfc1185.txt
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Network Working Group V. Jacobson
Request for Comments: 1185 LBL
R. Braden
ISI
L. Zhang
PARC
October 1990
TCP Extension for High-Speed Paths
Status of This Memo
This memo describes an Experimental Protocol extension to TCP 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.
Summary
This memo describes a small extension to TCP to support reliable
operation over very high-speed paths, using sender timestamps
transmitted using the TCP Echo option proposed in RFC-1072.
1. INTRODUCTION
TCP uses positive acknowledgments and retransmissions to provide
reliable end-to-end delivery over a full-duplex virtual circuit
called a connection [Postel81]. A connection is defined by its two
end points; each end point is a "socket", i.e., a (host,port) pair.
To protect against data corruption, TCP uses an end-to-end checksum.
Duplication and reordering are handled using a fine-grained sequence
number space, with each octet receiving a distinct sequence number.
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. In
practice, proper TCP implementations have demonstrated remarkable
robustness in adapting to a wide range of network characteristics.
For example, 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.
However, 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 and RFC-
1072 [Jacobson88] propose modest extensions to TCP to extend the
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RFC 1185 TCP over High-Speed Paths October 1990
domain of its application to higher speeds.
There is no one-line answer to the question: "How fast can TCP go?".
The issues are reliability and performance, and these depend upon the
round-trip delay and the maximum time that segments may be queued in
the Internet, as well as upon the transmission speed. We must think
through these relationships very carefully if we are to successfully
extend TCP's domain.
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. RFC-1072 proposed a set of TCP extensions to improve TCP
efficiency for "LFNs" (long fat networks), i.e., networks with large
bandwidth*delay products.
On the other hand, high transfer rate can threaten TCP reliability by
violating the assumptions behind the TCP mechanism for duplicate
detection and sequencing. The present memo specifies a solution for
this problem, extending TCP reliability to transfer rates well beyond
the foreseeable upper limit of bandwidth.
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, was delivered to the receiver at the wrong moment
so that its sequence numbers fell 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 lock up the connection so that
no further progress can be made and a RST is required to
resynchronize the two ends.
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 may be delayed in
queues. Section 2 discusses this case and proposes a mechanism
to reject old duplicates on the current connection.
(2) Segment from an earlier connection incarnation
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Suppose 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. This case is discussed in Section 3.
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.
Watson's Delta-T protocol [Watson81] includes network-layer
mechanisms for precise enforcement of an MSL. In contrast, the IP
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). The timestamp algorithm
described in the following section gives a way out of this dilemma
for high-speed networks.
2. SEQUENCE NUMBER WRAP-AROUND
2.1 Background
Avoiding reuse of sequence numbers within the same connection is
simple in principle: enforce a segment lifetime shorter than the
time it takes to cycle the sequence space, whose size is
effectively 2**31.
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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RFC 1185 TCP over High-Speed Paths October 1990
Network B*8 B Twrap
bits/sec bytes/sec secs
_______ _______ ______ ______
ARPANET 56kbps 7KBps 3*10**5 (~3.6 days)
DS1 1.5Mbps 190KBps 10**4 (~3 hours)
Ethernet 10Mbps 1.25MBps 1700 (~30 mins)
DS3 45Mbps 5.6MBps 380
FDDI 100Mbps 12.5MBps 170
Gigabit 1Gbps 125MBps 17
It is clear why wrap-around of the sequence space was 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.
Based on this reasoning, an earlier RFC [McKenzie89] has cautioned
that expanding the TCP window space as proposed in RFC-1072 will
lead to sequence wrap-around and hence to possible data
corruption. We believe that this is mis-identifying the culprit,
which 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
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speed alone can cause a reliability problem with sequence
number wrap-around, even without extended windows.
An "obvious" fix for the problem of cycling the sequence space is
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. However, the proposals for
making such a change while maintaining compatibility with current
TCP have tended towards complexity and ugliness.
This memo proposes a simple solution to the problem, using the TCP
echo options defined in RFC-1072. Section 2.2 which follows
describes the original use of these options to carry timestamps in
order to measure RTT accurately. Section 2.3 proposes a method of
using these same timestamps to reject old duplicate segments that
could corrupt an open TCP connection. Section 3 discusses the
application of this mechanism to avoiding old duplicates from
previous incarnations.
2.2 TCP Timestamps
RFC-1072 defined two TCP options, Echo and Echo Reply. Echo
carries a 32-bit number, and the receiver of the option must
return this same value to the source host in an Echo Reply option.
RFC-1072 furthermore describes the use of these options to contain
32-bit timestamps, for measuring the RTT. A TCP sending data
would include Echo options containing the current clock value.
The receiver would echo these timestamps in returning segments
(generally, ACK segments). The difference between a timestamp
from an Echo Reply option and the current time would then measure
the RTT at the sender.
This mechanism was designed to solve the following problem: almost
all TCP implementations base their RTT measurements on a sample of
only one packet per window. 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). Unfortunately, this lower sampling frequency
violates Nyquist's criteria and may introduce "aliasing" artifacts
into the estimated RTT [Hamming77].
A good RTT estimator with a conservative retransmission timeout
calculation can tolerate the 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
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RFC 1185 TCP over High-Speed Paths October 1990
error, resulting in spurious retransmissions.
A solution to the aliasing problem that actually simplifies the
sender substantially (since the RTT code is typically the single
biggest protocol cost for TCP) is as follows: the will sender
place a timestamp in each segment and the receiver will reflect
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
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