Internet Engineering Task Force                                M. Allman
INTERNET-DRAFT                                                      ICSI
File: draft-ietf-tcpm-rto-consider-05.txt                 March 10, 2017
Intended Status: Best Current Practice
Expires: September 10, 2017


                  Retransmission Timeout Requirements

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Abstract

    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

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    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.

Terminology

    The key words "MUST", "MUST NOT", "REQUIRED", "SHALL", "SHALL NOT",
    "SHOULD", "SHOULD NOT", "RECOMMENDED", "MAY", and "OPTIONAL" in this
    document are to be interpreted as described in BCP 14, RFC 2119
    [RFC2119].

   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
    reports 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]).  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
        loss.

    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.

    We have found that even though the RTO procedure is standardized for
    some protocols (e.g., TCP [RFC6298]), implementations often add
    their own subtle imprint on the specifics of the process to tilt the

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    tradeoff between correctness and responsiveness in some particular
    way.

    At this point we recognize that often these 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 mechanisms.  The
    intent is to provide a safe foundation on which implementations have
    the flexibility to instantiate mechanisms that best realize their
    specific goals.

   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 3:

    (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
          document.

          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 3 are likely applicable,
          sharing RTOs across flows is outside the scope of this

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          document.

    (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
          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
          mechanisms.

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

          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.

   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

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        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.

        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 be based on a 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
            communication.

            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

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            once-per-RTT definition of "regularly" is explicitly a
            "SHOULD" and not a "MUST".

            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.

            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]).

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        This ensures network safety.

        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]).

   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.

   Security Considerations

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

Acknowledgments

    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,
    Yuchung Cheng, David Black, Gorry Fairhurst, Mirja Kuhlewind,
    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,

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        "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
        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
   http://www.icir.org/mallman











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