TCP Maintenance (TCPM)                                         D. Borman
Internet-Draft                                       Quantum Corporation
Intended status: Standards Track                               B. Braden
Expires: November 19, 2012                        University of Southern
                                                             V. Jacobson
                                                           Packet Design
                                                   R. Scheffenegger, Ed.
                                                            NetApp, Inc.
                                                            May 18, 2012

                  TCP Extensions for High Performance


   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 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 (Protection Against
   Wrapped Sequences).  Selective acknowledgments are not included in
   this memo.

   This memo updates and obsoletes RFC 1323.

Status of this Memo

   This Internet-Draft is submitted in full conformance with the
   provisions of BCP 78 and BCP 79.

   Internet-Drafts are working documents of the Internet Engineering
   Task Force (IETF).  Note that other groups may also distribute
   working documents as Internet-Drafts.  The list of current Internet-
   Drafts is at

   Internet-Drafts are draft documents valid for a maximum of six months
   and may be updated, replaced, or obsoleted by other documents at any
   time.  It is inappropriate to use Internet-Drafts as reference
   material or to cite them other than as "work in progress."

   This Internet-Draft will expire on November 19, 2012.

Copyright Notice

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   Copyright (c) 2012 IETF Trust and the persons identified as the
   document authors.  All rights reserved.

   This document is subject to BCP 78 and the IETF Trust's Legal
   Provisions Relating to IETF Documents
   ( in effect on the date of
   publication of this document.  Please review these documents
   carefully, as they describe your rights and restrictions with respect
   to this document.  Code Components extracted from this document must
   include Simplified BSD License text as described in Section 4.e of
   the Trust Legal Provisions and are provided without warranty as
   described in the Simplified BSD License.

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Table of Contents

   1.  Introduction . . . . . . . . . . . . . . . . . . . . . . . . .  4
     1.1.  TCP Performance  . . . . . . . . . . . . . . . . . . . . .  4
     1.2.  TCP Reliability  . . . . . . . . . . . . . . . . . . . . .  6
     1.3.  Using TCP options  . . . . . . . . . . . . . . . . . . . .  9
   2.  TCP Window Scale Option  . . . . . . . . . . . . . . . . . . . 10
     2.1.  Introduction . . . . . . . . . . . . . . . . . . . . . . . 10
     2.2.  Window Scale Option  . . . . . . . . . . . . . . . . . . . 10
     2.3.  Using the Window Scale Option  . . . . . . . . . . . . . . 11
     2.4.  Addressing Window Retraction . . . . . . . . . . . . . . . 13
   3.  RTTM -- Round-Trip Time Measurement  . . . . . . . . . . . . . 13
     3.1.  Introduction . . . . . . . . . . . . . . . . . . . . . . . 13
     3.2.  TCP Timestamps Option  . . . . . . . . . . . . . . . . . . 14
     3.3.  The RTTM Mechanism . . . . . . . . . . . . . . . . . . . . 15
     3.4.  Which Timestamp to Echo  . . . . . . . . . . . . . . . . . 17
   4.  PAWS -- Protection Against Wrapped Sequence Numbers  . . . . . 19
     4.1.  Introduction . . . . . . . . . . . . . . . . . . . . . . . 19
     4.2.  The PAWS Mechanism . . . . . . . . . . . . . . . . . . . . 20
       4.2.1.  Basic PAWS Algorithm . . . . . . . . . . . . . . . . . 21
       4.2.2.  Timestamp Clock  . . . . . . . . . . . . . . . . . . . 23
       4.2.3.  Outdated Timestamps  . . . . . . . . . . . . . . . . . 24
       4.2.4.  Header Prediction  . . . . . . . . . . . . . . . . . . 25
       4.2.5.  IP Fragmentation . . . . . . . . . . . . . . . . . . . 26
     4.3.  Duplicates from Earlier Incarnations of Connection . . . . 27
   5.  Conclusions and Acknowledgements . . . . . . . . . . . . . . . 27
   6.  Security Considerations  . . . . . . . . . . . . . . . . . . . 28
   7.  IANA Considerations  . . . . . . . . . . . . . . . . . . . . . 28
   8.  References . . . . . . . . . . . . . . . . . . . . . . . . . . 28
     8.1.  Normative References . . . . . . . . . . . . . . . . . . . 28
     8.2.  Informative References . . . . . . . . . . . . . . . . . . 29
   Appendix A.  Implementation Suggestions  . . . . . . . . . . . . . 31
   Appendix B.  Duplicates from Earlier Connection Incarnations . . . 32
     B.1.  System Crash with Loss of State  . . . . . . . . . . . . . 32
     B.2.  Closing and Reopening a Connection . . . . . . . . . . . . 32
   Appendix C.  Changes from RFC 1072, RFC 1185, and RFC 1323 . . . . 34
   Appendix D.  Summary of Notation . . . . . . . . . . . . . . . . . 36
   Appendix E.  Pseudo-code Summary . . . . . . . . . . . . . . . . . 37
   Appendix F.  Event Processing Summary  . . . . . . . . . . . . . . 39
   Appendix G.  Timestamps Edge Cases . . . . . . . . . . . . . . . . 44
   Authors' Addresses . . . . . . . . . . . . . . . . . . . . . . . . 45

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1.  Introduction

   The TCP protocol [RFC0793] 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 10^10 bps and round-trip delays in the range
   1 ms to 100 seconds.  Work on TCP performance has shown that TCP
   without the extensions described in this memo can work well over a
   variety of Internet paths, ranging from 800 Mbit/sec I/O channels to
   300 bit/sec dial-up modems .

   Over the years, advances in networking technology has resulted in
   ever-higher transmission speeds, and the fastest paths are well
   beyond 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 an
   update to and obsoletes [RFC1323], which in turn is based upon and
   obsoletes [RFC1072] and [RFC1185].

   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 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.  In the past, properly-operating TCP implementations
        would cause the data pipeline to drain with every packet loss,
        and require a slow-start action to recover.  The Fast Retransmit
        and Fast Recovery algorithms [Jacobson90c], [RFC2581] and
        [RFC5681] were introduced, and their combined effect was to
        recover from one packet loss per window, without draining the
        pipeline.  However, more than one packet loss per window
        typically resulted 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, since the publication of RFC 1323, congestion control
        mechanism based upon some form of random dropping have been
        introduced into gateways, and randomly spaced packet drops have
        become common; this increases 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.

        Since the publication of [RFC1323], selective acknowledgments
        (SACK) have become important in the LFN regime.  SACK has been
        published as a [RFC2018], "TCP Selective Acknowledgment
        Options"..  Additional information about SACK can be found in
        [RFC2883], "An Extension to the Selective Acknowledgement (SACK)
        option for TCP" and [RFC3517], "A Conservative Selective

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        Acknowledgment (SACK)-based Loss Recovery Algorithm for TCP".

   (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 3.2 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

   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:

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   (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

   Duplicates from earlier incarnations, Case (2), are avoided by
   enforcing the current fixed MSL of the TCP spec, as explained in
   Section 4.3 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     | bits/sec | B bytes/sec |     Twrap secs     |
    |      Dialup      |   56kbps |      7kBps  | 3*10^5 (~3.6 days) |
    |        DS1       |  1.5Mbps |    190kBps  |   10^4 (~3 hours)  |
    |  10MBit Ethernet |   10Mbps |   1.25MBps  |  1700 (~0.5 hours) |
    |        DS3       |   45Mbps |    5.6MBps  |         380        |
    | 100MBit Ethernet |  100Mbps |   12.5MBps  |         170        |
    | Gigabit Ethernet |    1Gbps |    125MBps  |         17         |
    |  10Gig Ethernet  |   10Gbps |   1.25GBps  |         1.7        |

   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 100mbit speeds, Twrap is comparable to the 2 minute
   MSL assumed by the TCP specification [RFC0793].  Moving towards and
   beyond gigabit speeds, Twrap becomes too small for reliable

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   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 [RFC1110].  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 4 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 3.2 to protect against old duplicates from the same

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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.  When RFC 1323 was published, there was
   concern that some buggy TCP implementation might be crashed by the
   first appearance of an option on a non-SYN segment.  However, bugs
   like that can lead to DOS attacks against a TCP, so it is now
   expected that most TCP implementations will properly handle unknown
   options on non-SYN segments.  But it is still prudent to be
   conservative in what you send, and avoiding buggy TCP implementation
   is not the only reason for negotiating TCP options on SYN segments.
   Therefore, for each of the extensions defined below, TCP options will
   be sent on non-SYN segments only after an exchange of options on the
   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

   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.  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

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   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

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 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

   TCP Window Scale Option (WSopt):

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                   Kind: 3

                   Length: 3 bytes

                          | Kind=3  |Length=3 |shift.cnt|

   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 <SYN> segment (i.e., a segment
   with the SYN bit on and the ACK bit off).  It may also be sent in a
   <SYN,ACK> segment, but only if a Window Scale option was received in
   the initial <SYN> segment.  A Window Scale option in a segment
   without a SYN bit should be ignored.

   The Window field in a SYN (i.e., a <SYN> or <SYN,ACK>) 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 [RFC0793]:

   o  All windows are treated as 32-bit quantities for storage in 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.

   o  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.

   o  If a TCP receives a <SYN> segment containing a Window Scale
      option, it sends its own Window Scale option in the <SYN,ACK>

   o  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.

   o  Upon receiving a SYN segment with a Window Scale option containing
      shift.cnt = S, a TCP sets Snd.Wind.Scale to S and sets

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      Rcv.Wind.Scale to R; otherwise, it sets both Snd.Wind.Scale and
      Rcv.Wind.Scale to zero.

   o  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 RFC 793 are met, and using the
      "C" notation "<<" for left-shift).

   o  The window field (SEG.WND) of every outgoing segment, with the
      exception of SYN segments, is right-shifted by Rcv.Wind.Scale

                    SND.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 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

   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.

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2.4.  Addressing Window Retraction

   When a non-zero scale factor is in use, there are instances when a
   retracted window can be offered [Mathis08].  The end of the window
   will be on a boundary based on the granularity of the scale factor
   being used.  If the sequence number is then updated by a number of
   bytes smaller than that granularity, the TCP will have to either
   advertise a new window that is beyond what it previously advertised
   (and perhaps beyond the buffer), or will have to advertise a smaller
   window, which will cause the TCP window to shrink.  Implementations
   should ensure that they handle a shrinking window, as specified in
   section of [RFC1122].

   For the receiver, this implies that:

   1)  The receiver MUST honor, as in-window, any segment that would
       have been in-window for any ACK sent by the receiver.

   2)  When window scaling is in effect, the receiver SHOULD track the
       actual maximum window sequence number (which is likely to be
       greater than the window announced by the most recent ACK, if more
       than one segment has arrived since the application consumed any
       data in the receive buffer).

   On the sender side:

   3)  The initial transmission MUST honor window on most recent ACK.

   4)  On first retransmission, or if it is out-of-window by less than
       (2^Rcv.Wind.Scale) then do normal retransmission(s) without
       regard to receiver window as long as the original segment was in
       window when it was sent.

   5)  On subsequent retransmissions, treat it as zero window probes.

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" [RFC0896] 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
   one packet per window or less.  While this yields an adequate
   approximation to the RTT for small windows, it results in an

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   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
   [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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   TCP Timestamps Option (TSopt):

       Kind: 8

       Length: 10 bytes

        |Kind=8 |  10   |   TS Value (TSval)  |TS Echo Reply (TSecr)|
            1       1              4                     4

   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 valid if the ACK bit is set
   in the TCP header; if it is valid, it echos a timestamp 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.  However, a value of
   zero does not imply TSecr being invalid.  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 <SYN>
   segment (i.e., a segment containing a SYN bit and no ACK bit).  Once
   a TSopt has been sent or received in a non <SYN> segment, it must be
   sent in all segments.  Once a TSopt has been received in a non <SYN>
   segment, then any successive segment that is received without the RST
   bit and without a TSopt may be dropped without further processing,
   and an ACK of the current SND.UNA generated.

   In the case of crossing SYN packets where one SYN contains a TSopt
   and the other doesn't, both sides should put a TSopt in the <SYN,ACK>

3.3.  The RTTM Mechanism

   RTTM places a Timestamps option in every segment, with a TSval that
   is obtained from a (virtual) "timestamp clock".  Values of this clock
   values must be at least approximately proportional to real time, in
   order to measure actual RTT.

   These TSval values are echoed in TSecr values in the reverse
   direction.  The difference between a received TSecr value and the
   current timestamp clock value provides an RTT measurement.

   When timestamps are used, every segment that is received will contain
   a TSecr value; however, these values cannot all be used to update the

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   measured RTT.  The following example illustrates why.  It shows 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 <TSval=x,TSecr=y>.  Each
   TSecr field contains the value most recently received in a TSval

           TCP  A                                          TCP B

                         <A,TSval=1,TSecr=120> ------>

               <---- <ACK(A),TSval=127,TSecr=1>

                         <B,TSval=5,TSecr=127> ------>

               <---- <ACK(B),TSval=131,TSecr=5>

             . . . . . . . . . . . . . . . . . . . . . .

                        <C,TSval=65,TSecr=131> ------>

               <---- <ACK(C),TSval=191,TSecr=65>


   The dotted line marks a pause (60 time units long) in which A had
   nothing to send.  Note that this pause inflates the RTT which B could
   infer from receiving TSecr=131 in data segment C. Thus, in one-way
   data flows, RTTM in the reverse direction measures a value that is
   inflated by gaps in sending data.  However, the following rule
   prevents a resulting inflation of the measured RTT:

      RTTM Rule: A TSecr value received in a segment is used to update
      the averaged RTT measurement only if

      a)  the segment acknowledges some new data, i.e., only if it
          advances the left edge of the send window, and

      b)  the segment does not indicate any loss or reordering, i.e.
          contains SACK options

   Since TCP B is not sending data, the data segment C does not
   acknowledge any new data when it arrives at B. Thus, the inflated
   RTTM measurement is not used to update B's RTTM measurement.

   Implementors should note that with Timestamps multiple RTTMs can be

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   taken per RTT.  Many RTO estimators have a weighting factor based on
   an implicit assumption that at most one RTTM will be gotten per RTT.
   When using multiple RTTMs per RTT to update the RTO estimator, the
   weighting factor needs to be decreased to take into account the more
   frequent RTTMs.  For example, an implementation could choose to just
   use one sample per RTT to update the RTO estimator, or or vary the
   gain based on the congestion window, or take an average of all the
   RTTM measurements received over one RTT, and then use that value to
   update the RTO estimator.  This document does not prescribe any
   particular method for modifying the RTO estimator, the important
   point is that the implementation should do something more than just
   feeding additional RTTM samples from one RTT into the RTO estimator.

3.4.  Which Timestamp to Echo

   If more than one Timestamps option is received before a reply segment
   is sent, the TCP must choose only one of the TSvals to echo, ignoring
   the others.  To minimize the state kept in the receiver (i.e., the
   number of unprocessed TSvals), the receiver should be required to
   retain at most one timestamp in the connection control block.

   There are three situations to consider:

   (A)  Delayed ACKs.

        Many TCP's acknowledge only every Kth segment out of a group of
        segments arriving within a short time interval; this policy is
        known generally as "delayed ACKs".  The data-sender TCP must
        measure the effective RTT, including the additional time due to
        delayed ACKs, or else it will retransmit unnecessarily.  Thus,
        when delayed ACKs are in use, the receiver should reply with the
        TSval field from the earliest unacknowledged segment.

   (B)  A hole in the sequence space (segment(s) have been lost).

        The sender will continue sending until the window is filled, and
        the receiver may be generating ACKs as these out-of-order
        segments arrive (e.g., to aid "fast retransmit").

        The lost segment is probably a sign of congestion, and in that
        situation the sender should be conservative about
        retransmission.  Furthermore, it is better to overestimate than
        underestimate the RTT.  An ACK for an out-of-order segment
        should therefore contain the timestamp from the most recent
        segment that advanced the window.

        The same situation occurs if segments are re-ordered by the

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   (C)  A filled hole in the sequence space.

        The segment that fills the hole represents the most recent
        measurement of the network characteristics.  On the other hand,
        an RTT computed from an earlier segment would probably include
        the sender's retransmit time-out, badly biasing the sender's
        average RTT estimate.  Thus, the timestamp from the latest
        segment (which filled the hole) must be echoed.

   An algorithm that covers all three cases is described in the
   following rules for Timestamps option processing on a synchronized

   (1)  The connection state is augmented with two 32-bit slots:

        TS.Recent holds a timestamp to be echoed in TSecr whenever a
        segment is sent, and Last.ACK.sent holds the ACK field from the
        last segment sent.  Last.ACK.sent will equal RCV.NXT except when
        ACKs have been delayed.

   (2)  If:

             SEG.TSval >= TSrecent and SEG.SEQ <= Last.ACK.sent

        then SEG.TSval is copied to TS.Recent; otherwise, it is ignored.

   (3)  When a TSopt is sent, its TSecr field is set to the current
        TS.Recent value.

   The following examples illustrate these rules.  Here A, B, C...
   represent data segments occupying successive blocks of sequence
   numbers, and ACK(A),... represent the corresponding acknowledgment
   segments.  Note that ACK(A) has the same sequence number as B. We
   show only one direction of timestamp echoing, for clarity.

   o  Packets arrive in sequence, and some of the ACKs are delayed.

      By Case (A), the timestamp from the oldest unacknowledged segment
      is echoed.

                  <A, TSval=1> ------------------->
                  <B, TSval=2> ------------------->
                  <C, TSval=3> ------------------->

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                           <---- <ACK(C), TSecr=1>

   o  Packets arrive out of order, and every packet is acknowledged.

      By Case (B), the timestamp from the last segment that advanced the
      left window edge is echoed, until the missing segment arrives; it
      is echoed according to Case (C).  The same sequence would occur if
      segments B and D were lost and retransmitted..

                  <A, TSval=1> ------------------->
                           <---- <ACK(A), TSecr=1>
                  <C, TSval=3> ------------------->
                           <---- <ACK(A), TSecr=1>
                  <B, TSval=2> ------------------->
                           <---- <ACK(C), TSecr=2>
                  <E, TSval=5> ------------------->
                           <---- <ACK(C), TSecr=2>
                  <D, TSval=4> ------------------->
                           <---- <ACK(E), TSecr=4>

4.  PAWS -- Protection Against Wrapped Sequence Numbers

4.1.  Introduction

   Section 4.2describes a simple mechanism to reject old duplicate
   segments that might corrupt an open TCP connection; we call this
   mechanism PAWS (Protection Against Wrapped Sequence numbers).  PAWS
   operates within a single TCP connection, using state that is saved in
   the connection control block.  Section 4.3 and Appendix C discuss the
   implications of the PAWS mechanism for avoiding old duplicates from
   previous incarnations of the same connection.

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4.2.  The PAWS Mechanism

   PAWS uses the same TCP Timestamps option as the RTTM mechanism
   described earlier, and assumes that every received TCP segment
   (including data and ACK segments) contains a timestamp SEG.TSval
   whose values are monotonically non-decreasing in time.  The basic
   idea is that a segment can be discarded as an old duplicate if it is
   received with a timestamp SEG.TSval less than some timestamp recently
   received on this connection.

   In both the PAWS and the RTTM mechanism, the "timestamps" are 32-bit
   unsigned integers in a modular 32-bit space.  Thus, "less than" is
   defined the same way it is for TCP sequence numbers, and the same
   implementation techniques apply.  If s and t are timestamp values,

                       s < t  if 0 < (t - s) < 2^31,

   computed in unsigned 32-bit arithmetic.

   The choice of incoming timestamps to be saved for this comparison
   must guarantee a value that is monotonically increasing.  For
   example, we might save the timestamp from the segment that last
   advanced the left edge of the receive window, i.e., the most recent
   in-sequence segment.  Instead, we choose the value TS.Recent
   introduced in Section 3.4 for the RTTM mechanism, since using a
   common value for both PAWS and RTTM simplifies the implementation of
   both.  As Section 3.4 explained, TS.Recent differs from the timestamp
   from the last in-sequence segment only in the case of delayed ACKs,
   and therefore by less than one window.  Either choice will therefore
   protect against sequence number wrap-around.

   RTTM was specified in a symmetrical manner, so that TSval timestamps
   are carried in both data and ACK segments and are echoed in TSecr
   fields carried in returning ACK or data segments.  PAWS submits all
   incoming segments to the same test, and therefore protects against
   duplicate ACK segments as well as data segments.  (An alternative
   non-symmetric algorithm would protect against old duplicate ACKs: the
   sender of data would reject incoming ACK segments whose TSecr values
   were less than the TSecr saved from the last segment whose ACK field
   advanced the left edge of the send window.  This algorithm was deemed
   to lack economy of mechanism and symmetry.)

   TSval timestamps sent on >SYN< and >SYN,ACK< segments are used to
   initialize PAWS.  PAWS protects against old duplicate non-SYN
   segments, and duplicate SYN segments received while there is a
   synchronized connection.  Duplicate >SYN< and >SYN,ACK< segments
   received when there is no connection will be discarded by the normal
   3-way handshake and sequence number checks of TCP.

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   RFC 1323 recommended that RST segments NOT carry timestamps, and that
   they be acceptable regardless of their timestamp.  At that time, the
   thinking was that old duplicate RST segments should be exceedingly
   unlikely, and their cleanup function should take precedence over
   timestamps.  More recently, discussion about various blind attacks on
   TCP connections have raised the suggestion that if the Timestamps
   option is present, SEG.TSecr could be used to provide stricter
   acceptance tests for RST packets.  While still under discussion, to
   enable research into this area it is now recommended that when
   generating a RST, that if the packet causing the RST to be generated
   contained a Timestamps option that the RST also contain a Timestamps
   option.  In the RST segment, SEG.TSecr should be set to SEG.TSval
   from the incoming packet and SEG.TSval should be set to zero.  If a
   RST is being generated because of a user abort, and Snd.TS.OK is set,
   then a Timestamps option should be included in the RST.  When a RST
   packet is received, it must not be subjected to PAWS checks, and
   information from the Timestamps option must not be use to update
   connection state information.  SEG.TSecr may be used to provide
   stricter RST acceptance checks.

4.2.1.  Basic PAWS Algorithm

   The PAWS algorithm requires the following processing to be performed
   on all incoming segments for a synchronized connection:

   R1)  If there is a Timestamps option in the arriving segment,
        SEG.TSval < TS.Recent, TS.Recent is valid (see later discussion)
        and the RST bit is not set, then treat the arriving segment as
        not acceptable:

           Send an acknowledgement in reply as specified in RFC 793 page
           69 and drop the segment.

           Note: it is necessary to send an ACK segment in order to
           retain TCP's mechanisms for detecting and recovering from
           half-open connections.  For example, see Figure 10 of RFC

   R2)  If the segment is outside the window, reject it (normal TCP

   R3)  If an arriving segment satisfies: SEG.SEQ <= Last.ACK.sent (see
        Section 3.4), then record its timestamp in TS.Recent.

   R4)  If an arriving segment is in-sequence (i.e., at the left window
        edge), then accept it normally.

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   R5)  Otherwise, treat the segment as a normal in-window, out- of-
        sequence TCP segment (e.g., queue it for later delivery to the

   Steps R2, R4, and R5 are the normal TCP processing steps specified by
   RFC 793.

   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 it must be queued for later delivery.

   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 timestamp in TS.TStamp 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, TS.Stamp is set to

   This rule allows 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-R5 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 have 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 ACKs, then the queued

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   segments that caused these ACKs must have been received already.

   Even if a segment were delayed past the RTO, the Fast Retransmit
   mechanism [Jacobson90c] 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.

4.2.2.  Timestamp Clock

   It is important to understand that the PAWS algorithm does not
   require clock synchronization between sender and receiver.  The
   sender's timestamp clock is used to stamp the segments, and the
   sender uses the echoed timestamp to measure RTT's.  However, the
   receiver treats the timestamp as simply a monotonically 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.

   The receiver algorithm does place some requirements on the frequency
   of the timestamp clock.

   (a)  The 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 the window extension defined in Section 2.2, 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 ~8 Gbps.
        A 1ms timestamp clock will work at link speeds up to 8 Tbps
        (8*10^12) bps!

   (b)  The timestamp clock must not be "too fast".

        Its recycling 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, it is desirable to establish a much longer recycle
        period, in order to handle outdated timestamps on idle
        connections (see Section 4.2.3), and to relax the MSL

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        requirement for preventing sequence number wrap-around.  With a
        1 ms timestamp clock, the 32-bit timestamp will wrap its sign
        bit in 24.8 days.  Thus, it will reject old duplicates on the
        same connection if MSL is 24.8 days or less.  This appears to be
        a very safe figure; an MSL of 24.8 days or longer can probably
        be assumed by the gateway system without requiring precise MSL
        enforcement by the TTL value in the IP layer.

   Based upon these considerations, we choose a timestamp clock
   frequency in the range 1 ms to 1 sec per tick.  This range also
   matches the requirements of the RTTM mechanism, which does not need
   much more resolution than the granularity of the retransmit timer,
   e.g., tens or hundreds of milliseconds.

   The PAWS mechanism also puts a strong monotonicity requirement on the
   sender's timestamp clock.  The method of implementation of the
   timestamp clock to meet this requirement depends upon the system
   hardware and software.

   o  Some hosts have a hardware clock that is guaranteed to be
      monotonic between hardware resets.

   o  A clock interrupt may be used to simply increment a binary integer
      by 1 periodically.

   o  The timestamp clock may be derived from a system clock that is
      subject to being abruptly changed, by adding a variable offset
      value.  This offset is initialized to zero.  When a new timestamp
      clock value is needed, the offset can be adjusted as necessary to
      make the new value equal to or larger than the previous value
      (which was saved for this purpose).

4.2.3.  Outdated Timestamps

   If a connection remains idle long enough for the timestamp clock of
   the other TCP to wrap its sign bit, then the value saved in TS.Recent
   will become too old; as a result, the PAWS mechanism will cause all
   subsequent segments to be rejected, freezing the connection (until
   the timestamp clock wraps its sign bit again).

   With the chosen range of timestamp clock frequencies (1 sec to 1 ms),
   the time to wrap the sign bit will be between 24.8 days and 24800
   days.  A TCP connection that is idle for more than 24 days and then
   comes to life is exceedingly unusual.  However, it is undesirable in
   principle to place any limitation on TCP connection lifetimes.

   We therefore require that an implementation of PAWS include a
   mechanism to "invalidate" the TS.Recent value when a connection is

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   idle for more than 24 days.  (An alternative solution to the problem
   of outdated timestamps would be to send keep-alive segments at a very
   low rate, but still more often than the wrap-around time for
   timestamps, e.g., once a day.  This would impose negligible overhead.
   However, the TCP specification has never included keep-alives, so the
   solution based upon invalidation was chosen.)

   Note that a TCP does not know the frequency, and therefore, the
   wraparound time, of the other TCP, so it must assume the worst.  The
   validity of TS.Recent needs to be checked only if the basic PAWS
   timestamp check fails, i.e., only if SEG.TSval < TS.Recent.  If
   TS.Recent is found to be invalid, then the segment is accepted,
   regardless of the failure of the timestamp check, and rule R3 updates
   TS.Recent with the TSval from the new segment.

   To detect how long the connection has been idle, the TCP may update a
   clock or timestamp value associated with the connection whenever
   TS.Recent is updated, for example.  The details will be

4.2.4.  Header Prediction

   "Header prediction" [Jacobson90a] is a high-performance transport
   protocol implementation technique that 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 recommended sequence for processing an arriving TCP

   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 in RFC 793.  This
        includes dropping segments that are outside the window and
        possibly sending acknowledgments, and queueing in-window, out-
        of-sequence segments.

   Another possibility would be to interchange steps H1 and H2, i.e., to
   perform the header prediction step H2 FIRST, and perform H1 and H3
   only when header prediction fails.  This could be a performance

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   improvement, since the timestamp check in step H1 is very unlikely to
   fail, and it requires unsigned modulo arithmetic, a relatively
   expensive operation.  To perform this check on every single segment
   is contrary to the philosophy of header prediction.  We believe that
   this change might produce a measurable reduction in CPU time for TCP
   protocol processing on high-speed networks.

   However, putting H2 first would create a hazard: a segment from 2^32
   bytes in the past might arrive at exactly the wrong time and be
   accepted mistakenly by the header-prediction step.  The following
   reasoning has been introduced in [RFC1185] 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 a 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.

   However, this probabilistic argument is not universally accepted, and
   the consensus at present is that the performance gain does not
   justify the hazard in the general case.  It is therefore recommended
   that H2 follow H1.

4.2.5.  IP Fragmentation

   At high data rates, the protection against old packets provided by
   PAWS can be circumvented by errors in IP fragment reassembly (see
   [RFC4963]).  The only way to protect against incorrect IP fragment
   reassembly is to not allow the packets to be fragmented.  This is
   done by setting the Don't Fragment (DF) bit in the IP header.
   Setting the DF bit implies the use of Path MTU Discovery as described
   in [RFC1191], thus any TCP implementation that implements PAWS must
   also implement Path MTU Discovery.

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4.3.  Duplicates from Earlier Incarnations of Connection

   The PAWS mechanism protects against errors due to sequence number
   wrap-around on high-speed connection.  Segments from an earlier
   incarnation of the same connection are also a potential cause of old
   duplicate errors.  In both cases, the TCP mechanisms to prevent such
   errors depend upon the enforcement of a maximum segment lifetime
   (MSL) by the Internet (IP) layer (see Appendix of RFC 1185 for a
   detailed discussion).  Unlike the case of sequence space wrap-around,
   the MSL required to prevent old duplicate errors from earlier
   incarnations does not depend upon the transfer rate.  If the IP layer
   enforces the recommended 2 minute MSL of TCP, and if the TCP rules
   are followed, TCP connections will be safe from earlier incarnations,
   no matter how high the network speed.  Thus, the PAWS mechanism is
   not required for this case.

   We may still ask whether the PAWS mechanism can provide additional
   security against old duplicates from earlier connections, allowing us
   to relax the enforcement of MSL by the IP layer.  Appendix B explores
   this question, showing that further assumptions and/or mechanisms are
   required, beyond those of PAWS.  This is not part of the current

5.  Conclusions and Acknowledgements

   This memo presented a set of extensions to TCP to provide efficient
   operation over large-bandwidth*delay-product paths and reliable
   operation over very high-speed paths.  These extensions are designed
   to provide compatible interworking with TCP's that do not implement
   the extensions.

   These mechanisms are implemented using new TCP options for scaled
   windows and timestamps.  The timestamps are used for two distinct
   mechanisms: RTTM (Round Trip Time Measurement) and PAWS (Protect
   Against Wrapped Sequences).

   The Window Scale option was originally suggested by Mike St. Johns of
   USAF/DCA.  The present form of the option was suggested by Mike
   Karels of UC Berkeley in response to a more cumbersome scheme defined
   by Van Jacobson.  Lixia Zhang helped formulate the PAWS mechanism
   description in RFC 1185.

   Finally, much of this work originated as the result of discussions
   within the End-to-End Task Force on the theoretical limitations of
   transport protocols in general and TCP in particular.  Task force
   members and other on the end2end-interest list have made valuable
   contributions by pointing out flaws in the algorithms and the

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   documentation.  Continued discussion and development since the
   publication of RFC 1323 originally occurred in the IETF TCP Large
   Windows Working Group, later on in the End-to-End Task Force, and
   most recently in the IETF TCP Maintenance Working Group.  The authors
   are grateful for all these contributions.

6.  Security Considerations

   The TCP sequence space is a fixed size, and as the window becomes
   larger it becomes easier for an attacker to generate forged packets
   that can fall within the TCP window, and be accepted as valid
   packets.  While use of Timestamps and PAWS can help to mitigate this,
   when using PAWS, if an attacker is able to forge a packet that is
   acceptable to the TCP connection, a timestamp that is in the future
   would cause valid packets to be dropped due to PAWS checks.  Hence,
   implementors should take care to not open the TCP window drastically
   beyond the requirements of the connection.

   Middle boxes and options If a middle box removes TCP options from the
   SYN, such as TSopt, a high speed connection that needs PAWS would not
   have that protection.  In this situation, an implementor could
   provide a mechanism for the application to determine whether or not
   PAWS is in use on the connection, and chose to terminate the
   connection if that protection doesn't exist.

   Mechanisms to protect the TCP header from modification should also
   protect the TCP options.

   Expanding the TCP window beyond 64K for IPv6 allows Jumbograms
   [RFC2675] to be used when the local network supports packets larger
   than 64K. When larger TCP packets are used, the TCP checksum becomes

7.  IANA Considerations

   This document has no actions for IANA.

8.  References

8.1.  Normative References

   [RFC0793]  Postel, J., "Transmission Control Protocol", STD 7,
              RFC 793, September 1981.

   [RFC1191]  Mogul, J. and S. Deering, "Path MTU discovery", RFC 1191,

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              November 1990.

8.2.  Informative References

              Garlick, L., Rom, R., and J. Postel, "Issues in Reliable
              Host-to-Host Protocols", Proc. Second Berkeley Workshop on
              Distributed Data Management and Computer Networks ,
              May 1977, <>.

              Hamming, R., "Digital Filters", Prentice Hall, Englewood
              Cliffs, N.J. ISBN 0-13-212571-4, 1977.

              Jacobson, V., "Congestion Avoidance and Control", SIGCOMM
              '88, Stanford, CA. , August 1988,

              Jacobson, V., "4BSD Header Prediction", ACM Computer
              Communication Review , April 1990.

              Jacobson, V., "Modified TCP congestion avoidance
              algorithm", Message to end2end-interest mailing list ,
              April 1990,

   [Jain86]   Jain, R., "Divergence of Timeout Algorithms for Packet
              Retransmissions", Proc. Fifth Phoenix Conf. on Comp. and
              Comm., Scottsdale, Arizona , March 1986,

   [Karn87]   Karn, P. and C. Partridge, "Estimating Round-Trip Times in
              Reliable Transport Protocols", Proc. SIGCOMM '87 ,
              August 1987.

              Martin, D., "[Tsvwg] RFC 1323.bis", Message to the tsvwg
              mailing list , September 2003, <

              Mathis, M., "[tcpm] Example of 1323 window retraction
              problemPer my comments at the microphone at TCPM...",
              Message to the tcpm mailing list , March 2008, <http://

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   [RFC0896]  Nagle, J., "Congestion control in IP/TCP internetworks",
              RFC 896, January 1984.

   [RFC1072]  Jacobson, V. and R. Braden, "TCP extensions for long-delay
              paths", RFC 1072, October 1988.

   [RFC1110]  McKenzie, A., "Problem with the TCP big window option",
              RFC 1110, August 1989.

   [RFC1122]  Braden, R., "Requirements for Internet Hosts -
              Communication Layers", STD 3, RFC 1122, October 1989.

   [RFC1185]  Jacobson, V., Braden, B., and L. Zhang, "TCP Extension for
              High-Speed Paths", RFC 1185, October 1990.

   [RFC1323]  Jacobson, V., Braden, B., and D. Borman, "TCP Extensions
              for High Performance", RFC 1323, May 1992.

   [RFC2018]  Mathis, M., Mahdavi, J., Floyd, S., and A. Romanow, "TCP
              Selective Acknowledgment Options", RFC 2018, October 1996.

   [RFC2581]  Allman, M., Paxson, V., and W. Stevens, "TCP Congestion
              Control", RFC 2581, April 1999.

   [RFC2675]  Borman, D., Deering, S., and R. Hinden, "IPv6 Jumbograms",
              RFC 2675, August 1999.

   [RFC2883]  Floyd, S., Mahdavi, J., Mathis, M., and M. Podolsky, "An
              Extension to the Selective Acknowledgement (SACK) Option
              for TCP", RFC 2883, July 2000.

   [RFC3517]  Blanton, E., Allman, M., Fall, K., and L. Wang, "A
              Conservative Selective Acknowledgment (SACK)-based Loss
              Recovery Algorithm for TCP", RFC 3517, April 2003.

   [RFC4963]  Heffner, J., Mathis, M., and B. Chandler, "IPv4 Reassembly
              Errors at High Data Rates", RFC 4963, July 2007.

   [RFC5681]  Allman, M., Paxson, V., and E. Blanton, "TCP Congestion
              Control", RFC 5681, September 2009.

              Watson, R., "Timer-based Mechanisms in Reliable Transport
              Protocol Connection Management", Computer Networks, Vol.
              5 , 1981.

   [Zhang86]  Zhang, L., "Why TCP Timers Don't Work Well", Proc. SIGCOMM
              '86, Stowe, VT , August 1986.

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Appendix A.  Implementation Suggestions

   TCP Option Layout

      The following layouts are recommended for sending options on non-
      SYN segments, to achieve maximum feasible alignment of 32-bit and
      64-bit machines.

                   |   NOP  |  NOP   |  TSopt |   10   |
                   |           TSval  timestamp        |
                   |           TSecr  timestamp        |

   Interaction with the TCP Urgent Pointer

      The TCP Urgent pointer, like the TCP window, is a 16 bit value.
      Some of the original discussion for the TCP Window Scale option
      included proposals to increase the Urgent pointer to 32 bits.  As
      it turns out, this is unnecessary.  There are two observations
      that should be made:

      (1)  With IP Version 4, the largest amount of TCP data that can be
           sent in a single packet is 65495 bytes (64K - 1 - size of
           fixed IP and TCP headers).

      (2)  Updates to the urgent pointer while the user is in "urgent
           mode" are invisible to the user.

      This means that if the Urgent Pointer points beyond the end of the
      TCP data in the current packet, then the user will remain in
      urgent mode until the next TCP packet arrives.  That packet will
      update the urgent pointer to a new offset, and the user will never
      have left urgent mode.

      Thus, to properly implement the Urgent Pointer, the sending TCP
      only has to check for overflow of the 16 bit Urgent Pointer field
      before filling it in.  If it does overflow, than a value of 65535
      should be inserted into the Urgent Pointer.

      The same technique applies to IP Version 6, except in the case of
      IPv6 Jumbograms.  When IPv6 Jumbograms are supported, [RFC2675]
      requires additional steps for dealing with the Urgent Pointer,
      these are described in section 5.2 of [RFC2675].

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Appendix B.  Duplicates from Earlier Connection Incarnations

   There are two cases to be considered: (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.

B.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
   [RFC0793].  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 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.  We can back off from this strict requirement to take
   advantage of approximate clock synchronization.  Suppose that the
   clock is always re-synchronized to within N timestamp clock ticks and
   that booting (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.

B.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 [RFC0793].
   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.  The TIME- WAIT delay serves two
   different purposes:

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   (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.  (It could be
        argued that the side that is sending a FIN knows what degree of
        reliability it needs, and therefore it should be able to
        determine the length of the TIME-WAIT delay for the FIN's
        recipient.  This could be accomplished with an appropriate TCP
        option in FIN segments.)

        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 be no worse at
        high transfer speeds.

   (b)  Allow old duplicate segments to expire.

        To replace this function of TIME-WAIT state, a mechanism would
        have to operate across connections.  PAWS is defined strictly
        within a single connection; the last timestamp (TS.Recent) is
        kept in the connection control block, and discarded when a
        connection is closed.

        An additional mechanism could be added to the TCP, a per-host
        cache of the last timestamp received from any connection.  This
        value could then be used in the PAWS mechanism to reject old
        duplicate segments from earlier incarnations of the connection,
        if the timestamp clock can be guaranteed to have ticked at least
        once since the old connection was open.  This would require that
        the TIME-WAIT delay plus the RTT together must be at least one
        tick of the sender's timestamp clock.  Such an extension is not
        part of the proposal of this RFC.

        Note that this is a variant on the mechanism proposed by
        Garlick, Rom, and Postel [Garlick77], which required each host
        to maintain connection records containing the highest sequence

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        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.

Appendix C.  Changes from RFC 1072, RFC 1185, and RFC 1323

   The protocol extensions defined in RFC 1323 document differ in
   several important ways from those defined in RFC 1072 and RFC 1185.

   (a)  SACK has been split off into a separate document, [RFC2018].

   (b)  The detailed rules for sending timestamp replies (see
        Section 3.4) differ in important ways.  The earlier rules could
        result in an under-estimate of the RTT in certain cases (packets
        dropped or out of order).

   (c)  The same value TS.Recent is now shared by the two distinct
        mechanisms RTTM and PAWS.  This simplification became possible
        because of change (b).

   (d)  An ambiguity in RFC 1185 was resolved in favor of putting
        timestamps on ACK as well as data segments.  This supports the
        symmetry of the underlying TCP protocol.

   (e)  The echo and echo reply options of RFC 1072 were combined into a
        single Timestamps option, to reflect the symmetry and to
        simplify processing.

   (f)  The problem of outdated timestamps on long-idle connections,
        discussed in Section 4.2.2, was realized and resolved.

   (g)  RFC 1185 recommended that header prediction take precedence over
        the timestamp check.  Based upon some skepticism about the
        probabilistic arguments given in Section 4.2.4, it was decided
        to recommend that the timestamp check be performed first.

   (h)  The spec was modified so that the extended options will be sent
        on <SYN,ACK> segments only when they are received in the
        corresponding <SYN> segments.  This provides the most
        conservative possible conditions for interoperation with
        implementations without the extensions.

   In addition to these substantive changes, the present RFC attempts to
   specify the algorithms unambiguously by presenting modifications to
   the Event Processing rules of RFC 793; see Appendix F.

   There are additional changes in this document from RFC 1323.  These

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   changes are:

   (a)  The description of which TSecr values can be used to update the
        measured RTT has been clarified.  Specifically, with Timestamps,
        the Karn algorithm [Karn87] is disabled.  The Karn algorithm
        disables all RTT measurements during retransmission, since it is
        ambiguous whether the ACK is is for the original packet, or the
        retransmitted packet.  With Timestamps, that ambiguity is
        removed since the TSecr in the ACK will contain the TSval from
        whichever data packet made it to the destination.

   (b)  In RFC1323, section 3.4, step (2) of the algorithm to control
        which timestamp is echoed was incorrect in two regards:

        (1)  It failed to update TSrecent for a retransmitted segment
             that resulted from a lost ACK.

        (2)  It failed if SEG.LEN = 0.

        In the new algorithm, the case of SEG.TSval = TSrecent is
        included for consistency with the PAWS test.

   (c)  One correction was made to the Event Processing Summary in
        Appendix F.  In SEND CALL/ESTABLISHED STATE, RCV.WND is used to
        fill in the SEG.WND value, not SND.WND.

   (d)  New pseudo-code summary has been added in Appendix E.

   (e)  Appendix A has been expanded with information about the TCP MSS
        option and the TCP Urgent Pointer.

   (f)  It is now recommended that Timestamps options be included in RST
        packets if the incoming packet contained a Timestamps option.

   (g)  RST packets are explicitly excluded from PAWS processing.

   (h)  Snd.TSoffset and Snd.TSclock variables have been added.
        Snd.TSclock is the sum of my.TSclock and Snd.TSoffset.  This
        allows the starting points for timestamps to be randomized on a
        per-connection basis.  Setting Snd.TSoffset to zero yields the
        same results as [RFC1323].

   (i)  RTTM update processing explicitly excludes packets containing
        SACK options.  This addresses inflation of the RTT during
        episodes of packet loss in both directions.

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   (j)  In Section 3.2 the if-clause allowing sending of timestamps only
        when received in a <SYN> or <SYN,ACK> was removed, to allow for
        late timestamp negotiation.

   (k)  Section 2.4 was added describing the unavoidable window
        retraction issue, and explicitly describing the mitigation steps

Appendix D.  Summary of Notation

   The following notation has been used in this document.


      WSopt:            TCP Window Scale Option
      TSopt:            TCP Timestamps Option

   Option Fields

      shift.cnt:        Window scale byte in WSopt
      TSval:            32-bit Timestamp Value field in TSopt
      TSecr:            32-bit Timestamp Reply field in TSopt

   Option Fields in Current Segment

      SEG.TSval:        TSval field from TSopt in current segment
      SEG.TSecr:        TSecr field from TSopt in current segment
      SEG.WSopt:        8-bit value in WSopt

   Clock Values

      my.TSclock:       System wide source of 32-bit timestamp values
      my.TSclock.rate:  Period of my.TSclock (1 ms to 1 sec)
      Snd.TSoffset:     A offset for randomizing Snd.TSclock
      Snd.TSclock:      my.TSclock + Snd.TSoffset

   Per-Connection State Variables

      TS.Recent:        Latest received Timestamp
      Last.ACK.sent:    Last ACK field sent
      Snd.TS.OK:        1-bit flag
      Snd.WS.OK:        1-bit flag
      Rcv.Wind.Scale:   Receive window scale power

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      Snd.Wind.Scale:   Send window scale power
      Start.Time:       Snd.TSclock value when segment being timed was
                        sent (used by pre-1323 code).


      Update_SRTT(m)    Procedure to update the smoothed RTT and RTT
                        variance estimates, using the rules of
                        [Jacobson88a], given m, a new RTT measurement

Appendix E.  Pseudo-code Summary

   Create new TCB => {
       Rcv.wind.scale =
             MIN( 14, MAX(0, floor(log2(receive buffer space)) - 15) );
       Snd.wind.scale = 0;
       Last.ACK.sent = 0;
       Snd.TS.OK = Snd.WS.OK = FALSE;
       Snd.TSoffset = random 32 bit value

   Send initial <SYN> segment => {
       SEG.WND = MIN( RCV.WND, 65535 );
       Include in segment: TSopt(TSval=Snd.TSclock, TCecr=0);
       Include in segment: WSopt = Rcv.wind.scale;

   Send <SYN,ACK> segment => {
       SEG.ACK = Last.ACK.sent = RCV.NXT;
       SEG.WND = MIN( RCV.WND, 65535 );
       if (Snd.TS.OK) then
             Include in segment:
                   TSopt(TSval=Snd.TSclock, TSecr=TS.Recent);
       if (Snd.WS.OK) then
             Include in segment: WSopt = Rcv.wind.scale;

   Receive <SYN> or <SYN,ACK> segment => {
       if (Segment contains TSopt) then {
             TS.Recent = SEG.TSval;
             Snd.TS.OK = TRUE;
             if (is <SYN,ACK> segment) then
                          (Snd.TSclock - SEG.TSecr)/my.TSclock.rate);
       if (Segment contains WSopt) then {

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             Snd.wind.scale = SEG.WSopt;
             Snd.WS.OK = TRUE;
             if (the ACK bit is not set, and Rcv.wind.scale has not been
               initialized by the user) then
                   Rcv.wind.scale = Snd.wind.scale;
             Rcv.wind.scale = Snd.wind.scale = 0;

   Send non-SYN segment => {
       SEG.ACK = Last.ACK.sent = RCV.NXT;
       SEG.WND = MIN( RCV.WND >> Rcv.wind.scale, 65535 );
       if (Snd.TS.OK) then
             Include in segment:
                   TSopt(TSval=Snd.TSclock, TSecr=TS.Recent);

   Receive non-SYN segment in (state >= ESTABLISHED) => {
       Window = (SEG.WND << Snd.wind.scale);
             /* Use 32-bit 'Window' instead of 16-bit 'SEG.WND'
              * in rest of processing.
       if (Segment contains TSopt) then {
             if (SEG.TSval < TS.Recent && Idle less than 24 days) then {
                   if (Send.TS.OK AND (NOT RST) ) then {
                               /* Timestamp too old =>
                                *    segment is unacceptable.
                         Send ACK segment;
                         Discard segment and return;
             else {
                   if (SEG.SEQ =< Last.ACK.sent) then
                               TS.Recent = SEG.TSval;
       if (SEG.ACK > SND.UNA) then {
                    /* (At least part of) first segment in
                     * retransmission queue has been ACKd
             if (Segment contains TSopt) then
                          (Snd.TSclock - SEG.TSecr)/my.TSclock.rate);
                   Update_SRTT( /* for compatibility */
                          (Snd.TSclock - Start.Time)/my.TSclock.rate);

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Appendix F.  Event Processing Summary

   OPEN Call


      An initial send sequence number (ISS) is selected.  Send a SYN
      segment of the form:



   SEND Call

      CLOSED STATE (i.e., TCB does not exist)



         If the foreign socket is specified, then change the connection
         from passive to active, select an ISS.  Send a SYN segment
         containing the options: <TSval=Snd.TSclock> and
         <WSopt=Rcv.Wind.Scale>.  Set SND.UNA to ISS, SND.NXT to ISS+1.
         Enter SYN-SENT state. ...




         Segmentize the buffer and send it with a piggybacked
         acknowledgment (acknowledgment value = RCV.NXT). ...

         If the urgent flag is set ...

         If the Snd.TS.OK flag is set, then include the TCP Timestamps
         option <TSval=Snd.TSclock,TSecr=TS.Recent> in each data

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         Scale the receive window for transmission in the segment

                   SEG.WND = (RCV.WND >> Rcv.Wind.Scale).



      If the state is LISTEN then

         first check for an RST


         second check for an ACK


         third check for a SYN

            if the SYN bit is set, check the security.  If the ...


            if the SEG.PRC is less than the TCB.PRC then continue.

            Check for a Window Scale option (WSopt); if one is found,
            save SEG.WSopt in Snd.Wind.Scale and set Snd.WS.OK flag on.
            Otherwise, set both Snd.Wind.Scale and Rcv.Wind.Scale to
            zero and clear Snd.WS.OK flag.

            Check for a TSopt option; if one is found, save SEG.TSval in
            the variable TS.Recent and turn on the Snd.TS.OK bit.

            Set RCV.NXT to SEG.SEQ+1, IRS is set to SEG.SEQ and any
            other control or text should be queued for processing later.
            ISS should be selected and a SYN segment sent of the form:


            If the Snd.WS.OK bit is on, include a WSopt option
            <WSopt=Rcv.Wind.Scale> in this segment.  If the Snd.TS.OK
            bit is on, include a TSopt
            <TSval=Snd.TSclock,TSecr=TS.Recent> in this segment.
            Last.ACK.sent is set to RCV.NXT.

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            SND.NXT is set to ISS+1 and SND.UNA to ISS.  The connection
            state should be changed to SYN-RECEIVED.  Note that any
            other incoming control or data (combined with SYN) will be
            processed in the SYN-RECEIVED state, but processing of SYN
            and ACK should not be repeated.  If the listen was not fully
            specified (i.e., the foreign socket was not fully
            specified), then the unspecified fields should be filled in

         fourth other text or control


      If the state is SYN-SENT then

         first check the ACK bit



         fourth check the SYN bit


            If the SYN bit is on and the security/compartment and
            precedence are acceptable then, RCV.NXT is set to SEG.SEQ+1,
            IRS is set to SEG.SEQ, and any acknowledgements on the
            retransmission queue which are thereby acknowledged should
            be removed.

            Check for a Window Scale option (WSopt); if it is found,
            save SEG.WSopt in Snd.Wind.Scale; otherwise, set both
            Snd.Wind.Scale and Rcv.Wind.Scale to zero.

            Check for a TSopt option; if one is found, save SEG.TSval in
            variable TS.Recent and turn on the Snd.TS.OK bit in the
            connection control block.  If the ACK bit is set, use
            Snd.TSclock - SEG.TSecr as the initial RTT estimate.

            If SND.UNA > ISS (our SYN has been ACKed), change the
            connection state to ESTABLISHED, form an ACK segment:


            and send it.  If the Snd.Echo.OK bit is on, include a TSopt
            option <TSval=Snd.TSclock,TSecr=TS.Recent> in this ACK
            segment.  Last.ACK.sent is set to RCV.NXT.

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            Data or controls which were queued for transmission may be
            included.  If there are other controls or text in the
            segment then continue processing at the sixth step below
            where the URG bit is checked, otherwise return.

            Otherwise enter SYN-RECEIVED, form a SYN,ACK segment:


            and send it.  If the Snd.Echo.OK bit is on, include a TSopt
            option <TSval=Snd.TSclock,TSecr=TS.Recent> in this segment.
            If the Snd.WS.OK bit is on, include a WSopt option
            <WSopt=Rcv.Wind.Scale> in this segment.  Last.ACK.sent is
            set to RCV.NXT.

            If there are other controls or text in the segment, queue
            them for processing after the ESTABLISHED state has been
            reached, return.

         fifth, if neither of the SYN or RST bits is set then drop the
         segment and return.


      First, check sequence number

         FIN-WAIT-1 STATE
         FIN-WAIT-2 STATE

            Segments are processed in sequence.  Initial tests on
            arrival are used to discard old duplicates, but further
            processing is done in SEG.SEQ order.  If a segment's
            contents straddle the boundary between old and new, only the
            new parts should be processed.

            Rescale the received window field:

                  TrueWindow = SEG.WND << Snd.Wind.Scale,

            and use "TrueWindow" in place of SEG.WND in the following

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            Check whether the segment contains a Timestamps option and
            bit Snd.TS.OK is on.  If so:

               If SEG.TSval < TS.Recent and the RST bit is off, then
               test whether connection has been idle less than 24 days;
               if all are true, then the segment is not acceptable;
               follow steps below for an unacceptable segment.

               If SEG.SEQ is equal to Last.ACK.sent, then save SEG.TSval
               in variable TS.Recent.

            There are four cases for the acceptability test for an
            incoming segment:


            If an incoming segment is not acceptable, an acknowledgment
            should be sent in reply (unless the RST bit is set, if so
            drop the segment and return):


            Last.ACK.sent is set to SEG.ACK of the acknowledgment.  If
            the Snd.Echo.OK bit is on, include the Timestamps option
            <TSval=Snd.TSclock,TSecr=TS.Recent> in this ACK segment.
            Set Last.ACK.sent to SEG.ACK and send the ACK segment.
            After sending the acknowledgment, drop the unacceptable
            segment and return.


      fifth check the ACK field.

         if the ACK bit is off drop the segment and return.

         if the ACK bit is on



               If SND.UNA < SEG.ACK <= SND.NXT then, set SND.UNA <-
               SEG.ACK.  Also compute a new estimate of round-trip time.
               If Snd.TS.OK bit is on, use Snd.TSclock - SEG.TSecr;
               otherwise use the elapsed time since the first segment in
               the retransmission queue was sent.  Any segments on the
               retransmission queue which are thereby entirely

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      Seventh, process the segment text.

         FIN-WAIT-1 STATE
         FIN-WAIT-2 STATE


            Send an acknowledgment of the form:


            If the Snd.TS.OK bit is on, include Timestamps option
            <TSval=Snd.TSclock,TSecr=TS.Recent> in this ACK segment.
            Set Last.ACK.sent to SEG.ACK of the acknowledgment, and send
            it.  This acknowledgment should be piggy-backed on a segment
            being transmitted if possible without incurring undue delay.


Appendix G.  Timestamps Edge Cases

   While the rules laid out for when to calculate RTTM produce the
   correct results most of the time, there are some edge cases where an
   incorrect RTTM can be calculated.  All of these situations involve
   the loss of packets.  It is felt that these scenarios are rare, and
   that if they should happen, they will cause a single RTTM measurement
   to be inflated, which mitigates its effects on RTO calculations.

   [Martin03] cites two similar cases when the returning ACK is lost,
   and before the retransmission timer fires, another returning packet
   arrives, which ACKs the data.  In this case, the RTTM calculated will
   be inflated:

             tc=1   <A, TSval=1> ------------------->

             tc=2   (lost) <---- <ACK(A), TSecr=1, win=n>
                 (RTTM would have been 1)

                    (receive window opens, window update is sent)
             tc=5        <---- <ACK(A), TSecr=1, win=m>
                    (RTTM is calculated at 4)

   One thing to note about this situation is that it is somewhat bounded

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   by RTO + RTT, limiting how far off the RTTM calculation will be.
   While more complex scenarios can be constructed that produce larger
   inflations (e.g., retransmissions are lost), those scenarios involve
   multiple packet losses, and the connection will have other more
   serious operational problems than using an inflated RTTM in the RTO

Authors' Addresses

   David Borman
   Quantum Corporation
   Mendota Heights  MN 55120


   Bob Braden
   University of Southern California
   4676 Admiralty Way
   Marina del Rey  CA 90292


   Van Jacobson
   Packet Design
   2465 Latham Street
   Mountain View  CA 94040


   Richard Scheffenegger (editor)
   NetApp, Inc.
   Am Euro Platz 2
   Vienna,   1120


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