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Versions: (draft-allman-tcpm-rto-consider) 00 01   Best Current Practice
          02 03 04 05 06 07 08 09 10 11 12 13 14                        
          15 16 17 rfc8961                                              
Internet Engineering Task Force                                M. Allman
INTERNET-DRAFT                                                      ICSI
File: draft-ietf-tcpm-rto-consider-08.txt              February 22, 2019
Intended Status: Best Current Practice
Expires: August 22, 2019

                  Retransmission Timeout Requirements

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), its areas,
    and its working groups. Note that other groups may also distribute
    working documents as Internet-Drafts.

    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."

    The list of current Internet-Drafts can be accessed at

    The list of Internet-Draft Shadow Directories can be accessed at

    This Internet-Draft will expire on August 22, 2019.

Copyright Notice

    Copyright (c) 2019 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
    (http://trustee.ietf.org/license-info) 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.


    Ensuring reliable communication often manifests in a timeout and
    retry mechanism.  Each implementation of a retransmission timeout
    mechanism represents a balance between correctness and timeliness
    and therefore no implementation suits all situations.  This document
    provides high-level requirements for retransmission timeout schemes
    appropriate for general use in the Internet.  Within the
    requirements, implementations have latitude to define particulars
    that best address each situation.

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    The key words "MUST", "MUST NOT", "REQUIRED", "SHALL", "SHALL NOT",
    document are to be interpreted as described in BCP 14, RFC 2119

1   Introduction

    Reliable transmission is a key property for many network protocols
    and applications.  Our protocols use various mechanisms to achieve
    reliable data transmission.  Often we use continuous or periodic
    acknowledgments from the recipient to inform the sender's notion of
    which pieces of data are missing and need to be retransmitted to
    ensure reliability.  Alternatively, information coding---e.g.,
    FEC---can be used to achieve probabilistic reliability without
    retransmissions.  However, despite our best intentions and most
    robust mechanisms, the only thing we can truly depend on is the
    passage of time and therefore our ultimate backstop to ensuring
    reliability is a timeout and re-try mechanism.  That is, the sender
    sets some expectation for how long to wait for confirmation of
    delivery for a given piece of data.  When this time period passes
    without delivery confirmation the sender assumes the data was lost
    in transit and therefore schedules a retransmission.  This process
    of ensuring reliability via time-based loss detection and resending
    lost data is commonly referred to as a "retransmission timeout
    (RTO)" mechanism.

    Various protocols have defined their own RTO mechanisms (e.g., TCP
    [RFC6298], SCTP [RFC4960], SIP [RFC3261]).  In this document, our
    use of "RTO" does not refer to any one specific scheme, but rather
    is a generic term that includes all timer-based retransmission
    mechanisms.  The specifics of retransmission timeouts often
    represent a particular tradeoff between correctness and
    responsiveness [AP99].  In other words we want to simultaneously:

      - wait long enough to ensure the detection of loss is correct and
        therefore a retransmission is in fact needed, and

      - bound the delay we impose on applications before repairing

    Serving both of these goals is difficult as they pull in opposite
    directions.  I.e., towards either (a) withholding needed
    retransmissions too long to ensure the original transmission is
    truly lost or (b) not waiting long enough---to help application
    responsiveness---and hence sending unnecessary (often denoted
    "spurious") retransmissions.

    At this point, our experience has lead to a recognition that often
    specific tweaks that deviate from standardized RTO mechanisms do not
    materially impact network safety.  Therefore, in this document we
    outline a set of high-level protocol-agnostic requirements for RTO

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    mechanisms.  The intent is to provide a safe foundation on which
    implementations have the flexibility to instantiate mechanisms that
    best realize their specific goals.

2   Context

    This document is a bit "weird" in that it is backwards from the way
    we generally like to engineer systems.  Usually, we strive to
    understand high-level requirements as a starting point.  We then
    methodically proceed to engineer specific protocols, algorithms and
    systems that meet these requirements.  Within the standards process
    we have derived many retransmission timeouts without benefit from
    some over-arching requirements document---because we had no idea how
    to write such a requirements document!  Therefore, we made the best
    specific decisions we could in response to specific needs.

    At this point, however, we believe the community's experience has
    matured to the point where we can define a set of high-level
    requirements for retransmission timers.  That is, we now understand
    how to separate the aspects of retransmission timers that are
    crucial for network safety from those small details that do not
    materially impact network safety.  There are two basic benefits of
    writing this high-level document post-facto:

      - Existing retransmission timer mechanisms may be revisited with
        an eye towards changing the small and less crucial details to
        facilitate some benefit (e.g., performance), while at the same
        time not sacrificing network safety.

      - Future retransmission timers will have a solid basis of
        experience to lean on rather than cobbling together a new
        retransmission timer from scratch and/or pieces parts of other

    However, adding a requirements umbrella to a body of existing
    specific retransmission timer specifications is inherently messy and
    we run the risk of creating "inconsistencies".  The correct way to
    view this document is as the default case and these other
    specifications as agreed upon deviations from the default.  For
    instance, [RFC3261] uses a smaller initial timeout than this
    document specifies (requirement (1) in section 4).  This situation
    does not render useless the general guidance in this document, but
    rather develops an initial retransmission timeout that is
    appropriate in a specific context.  Likewise, TCP's retransmission
    timer has a minimum value of 1 second [RFC6298], whereas this
    document does not specify that a minimum retransmission timeout is
    necessary at all.  Again, this situation should be viewed as
    [RFC6298] providing a refinement for a specific case.

3   Scope

    The principles we outline in this document are protocol-agnostic and
    widely applicable.  We make the following scope statements about
    the application of the requirements discussed in Section 4:

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    (S.1) The requirements in this document apply only to timer-based
          loss detection and retransmission.

          While there are a bevy of uses for timers in protocols---from
          rate-based pacing to connection failure detection to making
          congestion control decisions and beyond---these are outside
          the scope of this document.

    (S.2) The requirements in this document only apply to cases where
          loss detected via a timer is repaired by a retransmission of
          the original data.

          Other cases are certainly possible---e.g., replacing the lost
          data with an updated version---but fall outside the scope of
          this document.

    (S.3) The requirements in this document apply only to endpoint-to-
          endpoint unicast communication.  Reliable multicast (e.g.,
          [RFC5740]) protocols are explicitly outside the scope of this

          Protocols such as SCTP [RFC4960] and MP-TCP [RFC6182] that
          communicate in a unicast fashion with multiple specific
          endpoints can leverage the requirements in this document
          provided they track state and follow the requirements for each
          endpoint independently.  I.e., if host A communicates with
          hosts B and C, A must use independent RTOs for traffic sent to
          B and C.

    (S.4) There are cases where state is shared across connections or
          flows (e.g., [RFC2140], [RFC3124]).  The RTO is one piece
          state that is often discussed as sharable.  These situations
          raise issues that the simple flow-oriented RTO mechanism
          discussed in this document does not consider (e.g., how long
          to preserve state between connections).  Therefore, while the
          general principles given in Section 4 are likely applicable,
          sharing RTOs across flows is outside the scope of this

    (S.5) The requirements in this document apply to reliable
          transmission, but do not assume that all data transmitted
          within a connection or flow is reliably sent.

          E.g., a protocol like DCCP [RFC4340] could leverage the
          requirements in this document for the initial reliable
          handshake even though the protocol reverts to unreliable
          transmission after the handshake.

          E.g., a protocol like SCTP [RFC4960] could leverage the
          requirements for data that is sent only "partially reliably".
          In this case, the protocol uses two phases for each message.
          In the first phase, the protocol attempts to ensure
          reliability and can leverage the requirements in this

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          document.  At some point the value of the data is gone and the
          protocol transitions to the second phase where the data is
          treated as unreliably transmitted and therefore the protocol
          will no longer attempt to repair the loss---and hence there
          are no more retransmissions and the requirements in this
          document are moot.

    (S.6) The requirements for RTO mechanisms in this document can be
          applied regardless of whether the RTO mechanism is the sole
          loss repair strategy or works in concert with other

          E.g., for a simple protocol like UDP-based DNS [] a timeout
          and re-try mechanism is likely to act alone to ensure

          E.g., within a complex protocol like TCP or SCTP we have
          designed methods to detect and repair loss based on explicit
          endpoint state sharing [RFC2018,RFC4960,RFC6675].  These
          mechanisms are preferred over the RTO as they are often more
          timely and precise than the coarse-grained RTO.  In these
          cases, the RTO becomes a last resort when the more advanced
          mechanisms fail.

          E.g., some protocols may leverage more than one retransmission
          timer simultaneously.  In these cases, the general guidance in
          this document can be applied to all such timers.

4   Requirements

    We now list the requirements that apply when designing
    retransmission timeout (RTO) mechanisms.

    (1) In the absence of any knowledge about the latency of a path, the
        RTO MUST be conservatively set to no less than 1 second.

        This requirement ensures two important aspects of the RTO.
        First, when transmitting into an unknown network,
        retransmissions will not be sent before an ACK would reasonably
        be expected to arrive and hence possibly waste scarce network
        resources.  Second, as noted below, sometimes retransmissions
        can lead to ambiguities in assessing the latency of a network
        path.  Therefore, it is especially important for the first
        latency sample to be free of ambiguities such that there is a
        baseline for the remainder of the communication.

        The specific constant (1 second) comes from the analysis of
        Internet RTTs found in Appendix A of [RFC6298].

    (2) As we note above, loss detection happens when a sender does not
        receive delivery confirmation within an some expected period of
        time.  We now specify four requirements that pertain to setting
        the length of this expectation.

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        Often measuring the time required for delivery confirmation is
        is framed as involving the "round-trip time (RTT)" of the
        network path as this is the minimum amount of time required to
        receive delivery confirmation and also often follows protocol
        behavior whereby acknowledgments are generated quickly after
        data arrives.  For instance, this is the case for the RTO used
        by TCP [RFC6298] and SCTP [RFC4960].  However, this is somewhat
        mis-leading as the expected latency is better framed as the
        "feedback time" (FT).  In other words, the expectation is not
        always simply a network property, but includes additional time
        before a sender should reasonably expect a response to a query.

        For instance, consider a UDP-based DNS request from a client to
        a recursive resolver.  When the request can be served from the
        resolver's cache the FT likely well approximates the network RTT
        between the client and resolver.  However, on a cache miss the
        resolver will request the needed information from one or more
        authoritative DNS servers, which will non-trivially increase the
        FT compared to the RTT between the client and resolver.

        Therefore, we express the following requirements in terms of FT:

        (a) In steady state the RTO SHOULD be set based on recent
            observations of both the FT and the variance of the FT.

            In other words, the RTO should represent an
            empirically-derived reasonable amount of time that the
            sender should wait for delivery confirmation before
            retransmitting the given data.

        (b) FT observations SHOULD be taken regularly.

            Internet measurements show that taking only a single FT
            sample per TCP connection results in a relatively poorly
            performing RTO mechanism [AP99], hence this requirement that
            the FT be sampled continuously throughout the lifetime of

            The notion of "regularly" SHOULD be defined as at least once
            per RTT or as frequently as data is exchanged in cases where
            that happens less frequently than once per RTT.  However, we
            also recognize that it may not always be practical to take
            an FT sample this often in all cases.  Hence, this
            once-per-RTT definition of "regularly" is explicitly a
            "SHOULD" and not a "MUST".

            As an example, TCP takes an FT sample roughly once per RTT,
            or if using the timestamp option [RFC7323] on each
            acknowledgment arrival.  [AP99] shows that both these
            approaches result in roughly equivalent performance for the
            RTO estimator.

        (c) FT observations MAY be taken from non-data exchanges.

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            Some protocols use keepalives, heartbeats or other messages
            to exchange control information.  To the extent that the
            latency of these transactions mirrors data exchange, they
            can be leveraged to take FT samples within the RTO
            mechanism.  Such samples can help protocols keep their RTO
            accurate during lulls in data transmission.  However, given
            that these messages may not be subject to the same delays as
            data transmission, we do not take a general view on whether
            this is useful or not.

        (d) An RTO mechanism MUST NOT use ambiguous FT samples.

            Assume two copies of some segment X are transmitted at times
            t0 and t1 and then at time t2 the sender receives
            confirmation that X in fact arrived.  In some cases, it is
            not clear which copy of X triggered the confirmation and
            hence the actual FT is either t2-t1 or t2-t0, but which is a
            mystery.  Therefore, in this situation an implementation
            MUST use Karn's algorithm [KP87,RFC6298] and use neither
            version of the FT sample and hence not update the RTO.

            There are cases where two copies of some data are
            transmitted in a way whereby the sender can tell which is
            being acknowledged by an incoming ACK.  E.g., TCP's
            timestamp option [RFC7323] allows for segments to be
            uniquely identified and hence avoid the ambiguity.  In such
            cases there is no ambiguity and the resulting samples can
            update the RTO.

    (3) Each time the RTO is used to detect a loss and a retransmission
        is scheduled, the value of the RTO MUST be exponentially backed
        off such that the next firing requires a longer interval.  The
        backoff SHOULD be removed after the successful repair of the
        lost data and subsequent transmission of non-retransmitted data.

        A maximum value MAY be placed on the RTO.  The maximum RTO MUST
        NOT be less than 60 seconds (a la [RFC6298]).

        This ensures network safety.

    (4) Retransmissions triggered by the RTO mechanism MUST be taken as
        indications of network congestion and the sending rate adapted
        using a standard mechanism (e.g., TCP collapses the congestion
        window to one segment [RFC5681]).

        This ensures network safety.

        An exception could be made to this rule if an IETF standardized
        mechanism is used to determine that a particular loss is due to
        a non-congestion event (e.g., packet corruption).  In such a
        case a congestion control action is not required.  Additionally,
        RTO-triggered congestion control actions may be reversed when a
        standard mechanism determines that the cause of the loss was not
        congestion after all (e.g., [RFC5682]).

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5   Discussion

    We note that research has shown the tension between the
    responsiveness and correctness of retransmission timeouts seems to
    be a fundamental tradeoff in the context of TCP [AP99].  That is,
    making the RTO more aggressive (e.g., via changing TCP's EWMA gains,
    lowering the minimum RTO, etc.) can reduce the time spent waiting on
    needed retransmissions.  However, at the same time, such
    aggressiveness leads to more needless retransmissions.  Therefore,
    being as aggressive as the requirements given in the previous
    section allow in any particular situation may not be the best course
    of action because an RTO expiration carries a requirement to invoke
    a congestion response and hence slow transmission down.

    While the tradeoff between responsiveness and correctness seems
    fundamental, the tradeoff can be made less relevant if the sender
    can detect and recover from spurious RTOs.  Several mechanisms have
    been proposed for this purpose, such as Eifel [RFC3522], F-RTO
    [RFC5682] and DSACK [RFC2883,RFC3708].  Using such mechanisms may
    allow a data originator to tip towards being more responsive without
    incurring (as much of) the attendant costs of needless retransmits.

    Also, note, that in addition to the experiments discussed in [AP99],
    the Linux TCP implementation has been using various non-standard RTO
    mechanisms for many years seemingly without large scale problems
    (e.g., using different EWMA gains than specified in [RFC6298]).
    Further, a number of implementations use minimum RTOs that are less
    than the 1 second specified in [RFC6298].  While the implication of
    these deviations from the standard may be more spurious retransmits
    (per [AP99]), we are aware of no large scale network safety issues
    caused by this change to the minimum RTO.

    Finally, we note that while allowing implementations to be more
    aggressive may in fact increase the number of needless
    retransmissions the above requirements fail safe in that they insist
    on exponential backoff of the RTO and a transmission rate reduction.
    Therefore, providing implementers more latitude than they have
    traditionally been given in IETF specifications of RTO mechanisms
    does not somehow open the flood gates to aggressive behavior.  Since
    there is a downside to being aggressive the incentives for proper
    behavior are retained in the mechanism.

6   Security Considerations

    This document does not alter the security properties of
    retransmission timeout mechanisms.  See [RFC6298] for a discussion
    of these within the context of TCP.


    This document benefits from years of discussions with Ethan Blanton,
    Sally Floyd, Jana Iyengar, Shawn Ostermann, Vern Paxson, and the
    members of the TCPM and TCP-IMPL working groups.  Ran Atkinson,

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    Yuchung Cheng, David Black, Gorry Fairhurst, Mirja Kuhlewind,
    Nicolas Kuhn, Jonathan Looney and Michael Scharf provided useful
    comments on a previous version of this draft.

Normative References

    [RFC2119] Bradner, S., "Key words for use in RFCs to Indicate
        Requirement Levels", BCP 14, RFC 2119, March 1997.

Informative References

    [AP99] Allman, M., V. Paxson, "On Estimating End-to-End Network Path
        Properties", Proceedings of the ACM SIGCOMM Technical Symposium,
        September 1999.

    [KP87] Karn, P. and C. Partridge, "Improving Round-Trip Time
        Estimates in Reliable Transport Protocols", SIGCOMM 87.

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

    [RFC2140] Touch, J., "TCP Control Block Interdependence", RFC 2140,
        April 1997.

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

    [RFC3124] Balakrishnan, H., S. Seshan, "The Congestion Manager", RFC
        2134, June 2001.

    [RFC3261] Rosenberg, J., Schulzrinne, H., Camarillo, G., Johnston,
        A., Peterson, J., Sparks, R., Handley, M., and E. Schooler,
        "SIP: Session Initiation Protocol", RFC 3261, June 2002.

    [RFC3522] Ludwig, R., M. Meyer, "The Eifel Detection Algorithm for
        TCP", RFC 3522, april 2003.

    [RFC3708] Blanton, E., M. Allman, "Using TCP Duplicate Selective
        Acknowledgement (DSACKs) and Stream Control Transmission
        Protocol (SCTP) Duplicate Transmission Sequence Numbers (TSNs)
        to Detect Spurious Retransmissions", RFC 3708, February 2004.

    [RFC3940] Adamson, B., C. Bormann, M. Handley, J. Macker,
        "Negative-acknowledgment (NACK)-Oriented Reliable Multicast
        (NORM) Protocol", November 2004, RFC 3940.

    [RFC4340] Kohler, E., M. Handley, S. Floyd, "Datagram Congestion
        Control Protocol (DCCP)", March 2006, RFC 4340.

    [RFC4960] Stweart, R., "Stream Control Transmission Protocol", RFC
        4960, September 2007.

    [RFC5682] Sarolahti, P., M. Kojo, K. Yamamoto, M. Hata, "Forward

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        RTO-Recovery (F-RTO): An Algorithm for Detecting Spurious
        Retransmission Timeouts with TCP", RFC 5682, September 2009.

    [RFC5740] Adamson, B., C. Bormann, M. Handley, J. Macker,
        "NACK-Oriented Reliable Multicast (NORM) Transport Protocol",
        November 2009, RFC 5740.

    [RFC6182] Ford, A., C. Raiciu, M. Handley, S. Barre, J. Iyengar,
        "Architectural Guidelines for Multipath TCP Development", March
        2011, RFC 6182.

    [RFC6298] Paxson, V., M. Allman, H.K. Chu, M. Sargent, "Computing
        TCP's Retransmission Timer", June 2011, RFC 6298.

    [RFC6582] Henderson, T., S. Floyd, A. Gurtov, Y. Nishida, "The
        NewReno Modification to TCP's Fast Recovery Algorithm", April
        2012, RFC 6582.

    [RFC6675] Blanton, E., M. Allman, L. Wang, I. Jarvinen, M.  Kojo,
        Y. Nishida, "A Conservative Loss Recovery Algorithm Based on
        Selective Acknowledgment (SACK) for TCP", August 2012, RFC 6675.

    [RFC7323] Borman D., B. Braden, V. Jacobson, R. Scheffenegger, "TCP
        Extensions for High Performance", September 2014, RFC 7323.

Authors' Addresses

   Mark Allman
   International Computer Science Institute
   1947 Center St.  Suite 600
   Berkeley, CA  94704

   EMail: mallman@icir.org

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