1. INTRODUCTION
Connected: An Internet Encyclopedia
1. INTRODUCTION
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1. INTRODUCTION
1. INTRODUCTION
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:
- 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.
- 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
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.
- 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
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:
- 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.
- 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:
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 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
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 <SYN,ACK> segment only if
the corresponding option was received in the initial <SYN>
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
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.
Next: 2. TCP WINDOW SCALE OPTION
Connected: An Internet Encyclopedia
1. INTRODUCTION
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