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Network Working Group                                        V. JacobsonRequest for Comments: 1185                                           LBL                                                               R. Braden                                                                     ISI                                                                L. Zhang                                                                    PARC                                                            October 1990TCP Extension for High-Speed PathsStatus 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 inRFC-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 andRFC-1072 [Jacobson88] propose modest extensions to TCP to extend theJacobson, Braden & Zhang                                        [Page 1]

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 incarnationJacobson, Braden & Zhang                                        [Page 2]

RFC 1185               TCP over High-Speed Paths            October 1990        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 inSection 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:Jacobson, Braden & Zhang                                        [Page 3]

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 inRFC-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, highJacobson, Braden & Zhang                                        [Page 4]

RFC 1185               TCP over High-Speed Paths            October 1990           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 inRFC-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 TimestampsRFC-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 inJacobson, Braden & Zhang                                        [Page 5]

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      sensible receiver).RFC-1072 defined a timestamp echo option for      this purpose.      It is vitally important to use the timestamp echo option 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.   2.3  Avoiding Old Duplicate Segments      Timestamps carried from sender to receiver in TCP Echo options can      also be used to prevent data corruption caused by sequence number      wrap-around, as this section describes.      2.3.1  Basic Algorithm         Assume that every received TCP segment contains a timestamp.         The basic idea is that a segment received with a timestamp that         is earlier than the timestamp of the most recently accepted         segment can be discarded as an old duplicate.  More         specifically, the following processing is to be performed on         normal incoming segments:         R1)  If the timestamp in the arriving segment timestamp is less              than the timestamp of the most recently received in-              sequence segment, treat the arriving segment as not              acceptable:                   If SEG.LEN > 0, send an acknowledgement in reply as                   specified inRFC-793 page 69, and drop the segment;                   otherwise, just silently drop the segment.*_________________________*Sending an ACK segment in reply is not strictly necessary, since  thecase  can  only  arise  when a later in-order segment has already beenreceived.   However,  for  consistency  and  simplicity,  we   suggesttreating  a  timestamp  failure  the  same  way  TCP  treats any otherunacceptable segment.Jacobson, Braden & Zhang                                        [Page 6]

RFC 1185               TCP over High-Speed Paths            October 1990         R2)  If the segment is outside the window, reject it (normal              TCP processing)         R3)  If an arriving segment is in-sequence (i.e, at the left              window edge), accept it normally and record its timestamp.         R4)  Otherwise, treat the segment as a normal in-window, out-              of-sequence TCP segment (e.g., queue it for later delivery              to the user).         Steps R2-R4 are the normal TCP processing steps specified byRFC-793, except that in R3 the latest timestamp is set from         each in-sequence segment that is accepted.  Thus, the latest         timestamp recorded at the receiver corresponds to the left edge         of the window and only advances when the left edge moves         [Jacobson88].         It is important to note that the timestamp is checked only when         a segment first arrives at the receiver, regardless of whether         it is in-sequence or is queued.  Consider the following         example.              Suppose the segment sequence: A.1, B.1, C.1, ..., Z.1 has              been sent, where the letter indicates the sequence number              and the digit represents the timestamp.  Suppose also that              segment B.1 has been lost.  The highest in-sequence              timestamp is 1 (from A.1), so C.1, ..., Z.1 are considered              acceptable and are queued.  When B is retransmitted as              segment B.2 (using the latest timestamp), it fills the              hole and causes all the segments through Z to be              acknowledged and passed to the user.  The timestamps of              the queued segments are *not* inspected again at this              time, since they have already been accepted.  When B.2 is              accepted, the receivers's current timestamp is set to 2.         This rule is vital to allow reasonable performance under loss.         A full window of data is in transit at all times, and after a         loss a full window less one packet will show up out-of-sequence         to be queued at the receiver (e.g., up to ~2**30 bytes of         data); the timestamp option must not result in discarding this         data.         In certain unlikely circumstances, the algorithm of rules R1-R4         could lead to discarding some segments unnecessarily, as shown         in the following example:              Suppose again that segments: A.1, B.1, C.1, ..., Z.1 haveJacobson, Braden & Zhang                                        [Page 7]

RFC 1185               TCP over High-Speed Paths            October 1990              been sent in sequence and that segment B.1 has been lost.              Furthermore, suppose delivery of some of C.1, ... Z.1 is              delayed until AFTER the retransmission B.2 arrives at the              receiver.  These delayed segments will be discarded              unnecessarily when they do arrive, since their timestamps              are now out of date.         This case is very unlikely to occur.  If the retransmission was         triggered by a timeout, some of the segments C.1, ... Z.1 must         have been delayed longer than the RTO time.  This is presumably         an unlikely event, or there would be many spurious timeouts and         retransmissions.  If B's retransmission was triggered by the         "fast retransmit" algorithm, i.e., by duplicate ACK's, then the         queued segments that caused these ACK's must have been received         already.         Even if a segment was delayed past the RTO, the selective         acknowledgment (SACK) facility ofRFC-1072 will cause the         delayed packets to be retransmitted at the same time as B.2,         avoiding an extra RTT and therefore causing a very small         performance penalty.         We know of no case with a significant probability of occurrence         in which timestamps will cause performance degradation by         unnecessarily discarding segments.      2.3.2  Header Prediction         "Header prediction" [Jacobson90] is a high-performance         transport protocol implementation technique that is is most         important for high-speed links.  This technique optimizes the         code for the most common case: receiving a segment correctly         and in order.  Using header prediction, the receiver asks the         question, "Is this segment the next in sequence?"  This         question can be answered in fewer machine instructions than the         question, "Is this segment within the window?"         Adding header prediction to our timestamp procedure leads to         the following sequence for processing an arriving TCP segment:         H1)  Check timestamp (same as step R1 above)         H2)  Do header prediction: if segment is next in sequence and              if there are no special conditions requiring additional              processing, accept the segment, record its timestamp, and              skip H3.         H3)  Process the segment normally, as specified inRFC-793.Jacobson, Braden & Zhang                                        [Page 8]

RFC 1185               TCP over High-Speed Paths            October 1990              This includes dropping segments that are outside the              window and possibly sending acknowledgments, and queueing              in-window, out-of-sequence segments.         However, the timestamp check in step H1 is very unlikely to         fail, and it is a relatively expensive operation since it         requires interval arithmetic on a finite field.  To perform         this check on every single segment seems like poor         implementation engineering, defeating the purpose of header         prediction.  Therefore, we suggest that an implementor         interchange H1 and H2, i.e., perform header prediction FIRST,         performing H1 and H3 only if header prediction fails.  We         believe that this change might gain 5-10% in performance on         high-speed networks.         This reordering does raise a theoretical hazard: a segment from         2**32 bytes in the past may arrive at exactly the wrong time         and be accepted mistakenly by the header-prediction step.  We         make the following argument to show that the probability of         this failure is negligible.              If all segments are equally likely to show up as old              duplicates, then the probability of an old duplicate              exactly matching the left window edge is the maximum              segment size (MSS) divided by the size of the sequence              space.  This ratio must be less than 2**-16, since MSS              must be < 2**16; for example, it will be (2**12)/(2**32) =              2**-20 for an FDDI link.  However, the older a segment is,              the less likely it is to be retained in the Internet, and              under any reasonable model of segment lifetime the              probability of an old duplicate exactly at the left window              edge must be much smaller than 2**16.              The 16 bit TCP checksum also allows a basic unreliability              of one part in 2**16.  A protocol mechanism whose              reliability exceeds the reliability of the TCP checksum              should be considered "good enough", i.e., it won't              contribute significantly to the overall error rate.  We              therefore believe we can ignore the problem of an old              duplicate being accepted by doing header prediction before              checking the timestamp.      2.3.3  Timestamp Frequency         It is important to understand that the receiver algorithm for         timestamps does not involve clock synchronization with the         sender.  The sender's clock is used to stamp the segments, and         the sender uses this fact to measure RTT's.  However, theJacobson, Braden & Zhang                                        [Page 9]

RFC 1185               TCP over High-Speed Paths            October 1990         receiver treats the timestamp as simply a monotone-increasing         serial number, without any necessary connection to its clock.         From the receiver's viewpoint, the timestamp is acting as a         logical extension of the high-order bits of the sequence         number.         However, the receiver algorithm dpes place some requirements on         the frequency of the timestamp "clock":         (a)  Timestamp clock must not be "too slow".              It must tick at least once for each 2**31 bytes sent.  In              fact, in order to be useful to the sender for round trip              timing, the clock should tick at least once per window's              worth of data, and even with theRFC-1072 window              extension, 2**31 bytes must be at least two windows.              To make this more quantitative, any clock faster than 1              tick/sec will reject old duplicate segments for link              speeds of ~2 Gbps;  a 1ms clock will work up to link              speeds of 2 Tbps (10**12 bps!).         (b)  Timestamp clock must not be "too fast".              Its cycling time must be greater than MSL seconds.  Since              the clock (timestamp) is 32 bits and the worst-case MSL is              255 seconds, the maximum acceptable clock frequency is one              tick every 59 ns.              However, since the sender is using the timestamp for RTT              calculations, the timestamp doesn't need to have much more              resolution than the granularity of the retransmit timer,              e.g., tens or hundreds of milliseconds.         Thus, both limits are easily satisfied with a reasonable clock         rate in the range 1-100ms per tick.         Using the timestamp option relaxes the requirements on MSL for         avoiding sequence number wrap-around.  For example, with a 1 ms         timestamp clock, the 32-bit timestamp will wrap its sign bit in         25 days.  Thus, it will reject old duplicates on the same         connection as long as MSL is 25 days or less.  This appears to         be a very safe figure.  If the timestamp has 10 ms resolution,         the MSL requirement is boosted to 250 days.  An MSL of 25 days         or longer can probably be assumed by the gateway system without         requiring precise MSL enforcement by the TTL value in the IP         layer.Jacobson, Braden & Zhang                                       [Page 10]

RFC 1185               TCP over High-Speed Paths            October 19903.  DUPLICATES FROM EARLIER INCARNATIONS OF CONNECTION   We turn now to the second potential cause of old duplicate packet   errors: packets from an earlier incarnation of the same connection.   The appendix contains a review the mechanisms currently included in   TCP to handle this problem.  These mechanisms depend upon the   enforcement of a maximum segment lifetime (MSL) by the Internet   layer.   The MSL required to prevent failures due to an earlier connection   incarnation does not depend (directly) upon the transfer rate.   However, the timestamp option used as described inSection 2 can   provide additional security against old duplicates from earlier   connections.  Furthermore, we will see that with the universal use of   the timestamp option, enforcement of a maximum segment lifetime would   no longer be required for reliable TCP operation.   There are two cases to be considered (see the appendix for more   explanation):  (1) a system crashing (and losing connection state)   and restarting, and (2) the same connection being closed and reopened   without a loss of host state.  These will be described in the   following two sections.   3.1  System Crash with Loss of State      TCP's quiet time of one MSL upon system startup handles the loss      of connection state in a system crash/restart.  For an      explanation, see for example "When to Keep Quiet" in the TCP      protocol specification [Postel81].  The MSL that is required here      does not depend upon the transfer speed.  The current TCP MSL of 2      minutes seems acceptable as an operational compromise, as many      host systems take this long to boot after a crash.      However, the timestamp option may be used to ease the MSL      requirements (or to provide additional security against data      corruption).  If timestamps are being used and if the timestamp      clock can be guaranteed to be monotonic over a system      crash/restart, i.e., if the first value of the sender's timestamp      clock after a crash/restart can be guaranteed to be greater than      the last value before the restart, then a quiet time will be      unnecessary.      To dispense totally with the quiet time would seem to require that      the host clock be synchronized to a time source that is stable      over the crash/restart period, with an accuracy of one timestamp      clock tick or better.  Fortunately, we can back off from this      strict requirement.  Suppose that the clock is always re-      synchronized to within N timestamp clock ticks and that bootingJacobson, Braden & Zhang                                       [Page 11]

RFC 1185               TCP over High-Speed Paths            October 1990      (extended with a quiet time, if necessary) takes more than N      ticks.  This will guarantee monotonicity of the timestamps, which      can then be used to reject old duplicates even without an enforced      MSL.   3.2  Closing and Reopening a Connection      When a TCP connection is closed, a delay of 2*MSL in TIME-WAIT      state ties up the socket pair for 4 minutes (see Section 3.5 of      [Postel81].  Applications built upon TCP that close one connection      and open a new one (e.g., an FTP data transfer connection using      Stream mode) must choose a new socket pair each time.  This delay      serves two different purposes:      (a)  Implement the full-duplex reliable close handshake of TCP.           The proper time to delay the final close step is not really           related to the MSL; it depends instead upon the RTO for the           FIN segments and therefore upon the RTT of the path.*           Although there is no formal upper-bound on RTT, common           network engineering practice makes an RTT greater than 1           minute very unlikely.  Thus, the 4 minute delay in TIME-WAIT           state works satisfactorily to provide a reliable full-duplex           TCP close.  Note again that this is independent of MSL           enforcement and network speed.           The TIME-WAIT state could cause an indirect performance           problem if an application needed to repeatedly close one           connection and open another at a very high frequency, since           the number of available TCP ports on a host is less than           2**16.  However, high network speeds are not the major           contributor to this problem; the RTT is the limiting factor           in how quickly connections can be opened and closed.           Therefore, this problem will no worse at high transfer           speeds.      (b)  Allow old duplicate segements to expire.           Suppose that a host keeps a cache of the last timestamp           received from each remote host.  This can be used to reject           old duplicate segments from earlier incarnations of the_________________________*Note: It could be argued that the side that is sending  a  FIN  knowswhat  degree  of reliability it needs, and therefore it should be ableto  determine  the  length  of  the  TIME-WAIT  delay  for  the  FIN'srecipient.   This could be accomplished with an appropriate TCP optionin FIN segments.Jacobson, Braden & Zhang                                       [Page 12]

RFC 1185               TCP over High-Speed Paths            October 1990           connection, if the timestamp clock can be guaranteed to have           ticked at least once since the old conennection was open.           This requires that the TIME-WAIT delay plus the RTT together           must be at least one tick of the sender's timestamp clock.           Note that this is a variant on the mechanism proposed by           Garlick, Rom, and Postel (see the appendix), which required           each host to maintain connection records containing the           highest sequence numbers on every connection.  Using           timestamps instead, it is only necessary to keep one quantity           per remote host, regardless of the number of simultaneous           connections to that host.      We conclude that if all hosts used the TCP timestamp algorithm      described inSection 2, enforcement of a maximum segment lifetime      would be unnecessary and the quiet time at system startup could be      shortened or removed.  In any case, the timestamp mechanism can      provide additional security against old duplicates from earlier      connection incarnations.   However, a 4 minute TIME-WAIT delay      (unrelated to MSL enforcement or network speed) must be retained      to provide the reliable close handshake of TCP.4. CONCLUSIONS   We have presented a mechanism, based upon the TCP timestamp echo   option ofRFC-1072, that will allow very high TCP transfer rates   without reliability problems due to old duplicate segments on the   same connection.  This mechanism also provides additional security   against intrusion of old duplicates from earlier incarnations of the   same connection.  If the timestamp mechanism were used by all hosts,   the quiet time at system startup could be eliminated and enforcement   of a maximum segment lifetime (MSL) would no longer be necessary.REFERENCES   [Cerf76]  Cerf, V., "TCP Resynchronization", Tech Note #79, Digital   Systems Lab, Stanford, January 1976.   [Dalal74]  Dalal, Y., "More on Selecting Sequence Numbers", INWG   Protocol Note #4, October 1974.   [Garlick77]  Garlick, L., R. Rom, and J. Postel, "Issues in Reliable   Host-to-Host Protocols", Proc. Second Berkeley Workshop on   Distributed Data Management and Computer Networks, May 1977.   [Hamming77]  Hamming, R., "Digital Filters", ISBN 0-13-212571-4,   Prentice Hall, Englewood Cliffs, N.J., 1977.Jacobson, Braden & Zhang                                       [Page 13]

RFC 1185               TCP over High-Speed Paths            October 1990   [Jacobson88]  Jacobson, V., and R. Braden, "TCP Extensions for   Long-Delay Paths",RFC 1072, LBL and USC/Information Sciences   Institute, October 1988.   [Jacobson90]  Jacobson, V., "4BSD Header Prediction", ACM Computer   Communication Review, April 1990.   [McKenzie89]  McKenzie, A., "A Problem with the TCP Big Window   Option",RFC 1110, BBN STC, August 1989.   [Postel81]  Postel, J., "Transmission Control Protocol",RFC 793,   DARPA, September 1981.   [Tomlinson74]  Tomlinson, R., "Selecting Sequence Numbers", INWG   Protocol Note #2, September 1974.   [Watson81]  Watson, R., "Timer-based Mechanisms in Reliable   Transport Protocol Connection Management", Computer Networks,   Vol. 5, 1981.Jacobson, Braden & Zhang                                       [Page 14]

RFC 1185               TCP over High-Speed Paths            October 1990APPENDIX -- Protection against Old Duplicates in TCP   During the development of TCP, a great deal of effort was devoted to   the problem of protecting a TCP connection from segments left from   earlier incarnations of the same connection.  Several different   mechanisms were proposed for this purpose [Tomlinson74] [Dalal74]   [Cerf76] [Garlick77].   The connection parameters that are required in this discussion are:           Tc = Connection duration in seconds.           Nc = Total number of bytes sent on connection.           B = Effective bandwidth of connection = Nc/Tc.   Tomlinson proposed a scheme with two parts: a clock-driven selection   of ISN (Initial Sequence Number) for a connection, and a   resynchronization procedure [Tomlinson74]. The clock-driven scheme   chooses:      ISN = (integer(R*t)) mod 2**32                 [2]   where t is the current time relative to an arbitrary origin, and R is   a constant.  R was intended to be chosen so that ISN will advance   faster than sequence numbers will be used up on the connection.   However, at high speeds this will not be true; the consequences of   this will be discussed below.   The clock-driven choice of ISN in formula [2] guarantees freedom from   old duplicates matching a reopened connection if the original   connection was "short-lived" and "slow".  By "short-lived", we mean a   connection that stayed open for a time Tc less than the time to cycle   the ISN, i.e., Tc < 2**32/R seconds.  By "slow", we mean that the   effective transfer rate B is less than R.   This is illustrated in Figure 1, where sequence numbers are plotted   against time.  The asterisks show the ISN lines from formula [2],   while the circles represent the trajectories of several short-lived   incarnations of the same connection, each terminating at the "x".        Note: allowing rapid reuse of connections was believed to be an        important goal during the early TCP development.  This        requirement was driven by the hope that TCP would serve as a        basis for user-level transaction protocols as well as        connection-oriented protocols.  The paradigm discussed was the        "Christmas Tree" or "Kamikazee" segment that contained SYN and        FIN bits as well as data.  Enthusiasm for this was somewhatJacobson, Braden & Zhang                                       [Page 15]

RFC 1185               TCP over High-Speed Paths            October 1990        dampened when it was observed that the 3-way SYN handshake and        the FIN handshake mean that 5 packets are required for a minimum        exchange. Furthermore, the TIME-WAIT state delay implies that        the same connection really cannot be reopened immediately.  No        further work has been done in this area, although existing        applications (especially SMTP) often generate very short TCP        sessions.  The reuse problem is generally avoided by using a        different port pair for each connection.        |- 2**32       ISN             ISN        |              *               *        |             *               *        |            *               *        |           *x              *        |          o               *    ^   |         *               *    |   |        *  x            *        |       * o             *    S   |      *o              *    e   |     o               *    q   |    *               *        |   *               *    #   |  * x             *        | *o              *        |o_______________*____________                         ^         Time -->                       4.55hrs     Figure 1.  Clock-Driven ISN  avoiding duplication on                short-Lived, slow connections.   However, clock-driven ISN selection does not protect against old   duplicate packets for a long-lived or fast connection:  the   connection may close (or crash) just as the ISN has cycled around and   reached the same value again.  If the connection is then reopened, a   datagram still in transit from the old connection may fall into the   current window.  This is illustrated by Figure 2 for a slow, long-   lived connection, and by Figures 3 and 4 for fast connections.  In   each case, the point "x" marks the place at which the original   connection closes or crashes.  The arrow in Figure 2 illustrates an   old duplicate segment.  Figure 3 shows a connection whose total byte   count Nc < 2**32, while Figure 4 concerns Nc >= 2**32.   To prevent the duplication illustrated in Figure 2, Tomlinson   proposed to "resynchronize" the connection sequence numbers if theyJacobson, Braden & Zhang                                       [Page 16]

RFC 1185               TCP over High-Speed Paths            October 1990   came within an MSL of the ISN.  Resynchronization might take the form   of a delay (point "y") or the choice of a new sequence number (point   "z").        |- 2**32       ISN               ISN        |              *                 *        |             *                 *        |            *                 *        |           *                 *        |          *                 *    ^   |         *                 *    |   |        *                 *        |       *                 *    S   |      *                 *    e   |     *                x* y    q   |    *           o     *        |   *      o          *z    #   |  *o                *        | *                 *        |*_________________*____________                           ^         Time -->                          4.55hrs        Figure 2.  Resynchronization to Avoid Duplication                   on Slow, Long-Lived Connection        |- 2**32       ISN               ISN        |              *                 *        |       x   o *                 *        |            *                 *        |      o-->o*                 *        |          *                 *    ^   |     o   o                 *    |   |        *                 *        |    o  *                 *    S   |      *                 *    e   |   o *                 *    q   |    *                 *        |  o*                 *    #   |  *                 *        | o                 *        |*_________________*____________                           ^         Time -->                          4.55hrs     Figure 3.  Duplication on Fast Connection: Nc < 2**32 bytesJacobson, Braden & Zhang                                       [Page 17]

RFC 1185               TCP over High-Speed Paths            October 1990        |- 2**32       ISN               ISN        |      o       *                 *        |           x *                 *        |            *                 *        |     o     *                 *        |          o                 *    ^   |         *                 *    |   |    o   *                 *        |       * o               *    S   |      *                *    e   |   o *                 *    q   |    *   o             *        |   *                 *    #   |  o                 *        | *     o           *        |*_________________*____________                           ^         Time -->                          4.55hrs     Figure 4.  Duplication on Fast Connection: Nc > 2**32 bytes   In summary, Figures 1-4 illustrated four possible failure modes for   old duplicate packets from an earlier incarnation.  We will call   these four modes F1 , F2, F3, and F4:   F1:  B < R, Tc < 4.55 hrs. (Figure 1)   F2:  B < R, Tc >= 4.55 hrs. (Figure 2)   F3:  B >= R, Nc < 2**32 (Figure 3)   F4:  B >= R, Nc >= 2**32 (Figure 4)   Another limitation of clock-driven ISN selection should be mentioned.   Tomlinson assumed that the current time t in formula [2] is obtained   from a clock that is persistent over a system crash.  For his scheme   to work correctly, the clock must be restarted with an accuracy of   1/R seconds (e.g, 4 microseconds in the case of TCP).  While this may   be possible for some hosts and some crashes, in most cases there will   be an uncertainty in the clock after a crash that ranges from a   second to several minutes.   As a result of this random clock offset after system   reinitialization, there is a possibility that old segments sent   before the crash may fall into the window of a new connection   incarnation.  The solution to this problem that was adopted in theJacobson, Braden & Zhang                                       [Page 18]

RFC 1185               TCP over High-Speed Paths            October 1990   final TCP spec is a "quiet time" of MSL seconds when the system is   initialized [Postel81, p. 28].  No TCP connection can be opened until   the expiration of this quiet time.   A different approach was suggested by Garlick, Rom, and Postel   [Garlick77].  Rather than using clock-driven ISN selection, they   proposed to maintain connection records containing the last ISN used   on every connection.  To immediately open a new incarnation of a   connection, the ISN is taken to be greater than the last sequence   number of the previous incarnation, so that the new incarnation will   have unique sequence numbers.  To handle a system crash, they   proposed a quiet time, i.e., a delay at system startup time to allow   old duplicates to expire.  Note that the connection records need be   kept only for MSL seconds; after that, no collision is possible, and   a new connection can start with sequence number zero.   The scheme finally adopted for TCP combines features of both these   proposals.  TCP uses three mechanisms:   (A)  ISN selection is clock-driven to handle short-lived connections.        The parameter R =  250KBps, so that the ISN value cycles in        2**32/R = 4.55 hours.   (B)  (One end of) a closed connection is left in a "busy" state,        known as "TIME-WAIT" state, for a time of 2*MSL.  TIME-WAIT        state handles the proper close of a long-lived connection        without resynchronization.  It also allows reliable completion        of the full-duplex close handshake.   (C)  There is a quiet time of one MSL at system startup.  This        handles a crash of a long-lived connection and avoids time        resynchronization problems in (A).   Notice that (B) and (C) together are logically sufficient to prevent   accidental reuse of sequence numbers from a different incarnation,   for any of the failure modes F1-F4.  (A) is not logically necessary   since the close delay (B) makes it impossible to reopen the same TCP   connection immediately.  However, the use of (A) does give additional   assurance in a common case, perhaps compensating for a host that has   set its TIME-WAIT state delay too short.   Some TCP implementations have permitted a connection in the TIME-WAIT   state to be reopened immediately by the other side, thus short-   circuiting mechanism (B).  Specifically, a new SYN for the same   socket pair is accepted when the earlier incarnation is still in   TIME-WAIT state.  Old duplicates in one direction can be avoided by   choosing the ISN to be the next unused sequence number from the   preceding connection (i.e., FIN+1); this is essentially anJacobson, Braden & Zhang                                       [Page 19]

RFC 1185               TCP over High-Speed Paths            October 1990   application of the scheme of Garlick, Rom, and Postel, using the   connection block in TIME-WAIT state as the connection record.   However, the connection is still vulnerable to old duplicates in the   other direction.  Mechanism (A) prevents trouble in mode F1, but   failures can arise in F2, F3, or F4; of these, F2, on short, fast   connections, is the most dangerous.   Finally, we note TCP will operate reliably without any MSL-based   mechanisms in the following restricted domain:   *    Total data sent is less then 2**32 octets, and   *    Effective sustained rate less than 250KBps, and   *    Connection duration less than 4.55 hours.   At the present time, the great majority of current TCP usage falls   into this restricted domain.  The third component, connection   duration, is the most commonly violated.Security Considerations   Security issues are not discussed in this memo.Authors' Addresses   Van Jacobson   University of California   Lawrence Berkeley Laboratory   Mail Stop 46A   Berkeley, CA 94720   Phone: (415) 486-6411   EMail: van@CSAM.LBL.GOV   Bob Braden   University of Southern California   Information Sciences Institute   4676 Admiralty Way   Marina del Rey, CA 90292   Phone: (213) 822-1511   EMail: Braden@ISI.EDUJacobson, Braden & Zhang                                       [Page 20]

RFC 1185               TCP over High-Speed Paths            October 1990   Lixia Zhang   XEROX Palo Alto Research Center   3333 Coyote Hill Road   Palo Alto, CA 94304   Phone: (415) 494-4415   EMail: lixia@PARC.XEROX.COMJacobson, Braden & Zhang                                       [Page 21]