Internet Engineering Task Force                                M. Allman
INTERNET-DRAFT                                                      ICSI
File: draft-ietf-tcpm-rto-consider-14.txt                   May 13, 2020
Intended Status: Best Current Practice
Expires: November 13, 2020


               Requirements for Time-Based Loss Detection

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Abstract

    Many protocols must detect packet loss for various reasons (e.g., to
    ensure reliability using retransmissions or to understand the level
    of congestion along a network path).  While many mechanisms have
    been designed to detect loss, protocols ultimately can only count on
    the passage of time without delivery confirmation to declare a
    packet "lost".  Each implementation of a time-based loss detection
    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 time-based loss detectors
    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

    Loss detection is a crucial activity for many protocols and
    applications and is generally undertaken for two major reasons:

      (1) Ensuring reliable data delivery.

            This requires a data sender to develop an understanding of
            which transmitted packets have not arrived at the receiver.
            This knowledge allows the sender to retransmit missing data.

      (2) Congestion control.

            Packet loss is often taken as an indication that the sender
            is transmitting too fast and is overwhelming some portion of
            the network path.  Data senders can therefore use loss to
            trigger transmission rate reductions.

    Various mechanisms are used to detect losses in a packet stream.
    Often we use continuous or periodic acknowledgments from the
    recipient to inform the sender's notion of which pieces of data are
    missing.  However, despite our best intentions and most robust
    mechanisms we cannot place ultimate faith in receiving such
    acknowledgments, but can only truly depend on the passage of time.
    Therefore, our ultimate backstop to ensuring that we detect all loss
    is a timeout.  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 concludes the data was lost in transit.

    The specifics of time-based loss detection schemes represent a
    tradeoff between correctness and responsiveness.  In other words we
    wish to simultaneously:

      - wait long enough to ensure the detection of loss is correct, and

      - minimize the amount of delay we impose on applications (before
        repairing loss) and the network (before we reduce the
        congestion).

    Serving both of these goals is difficult as they pull in opposite
    directions [AP99].  By not waiting long enough to accurately

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    determine a packet has been lost we risk sending unnecessary
    ("spurious") retransmissions and needlessly lowering the
    transmission rate.  By waiting long enough that we are unambiguously
    certain a packet has been lost we cannot repair losses in a timely
    manner and we risk prolonging network congestion.

    Many protocols and applications use their own time-based loss
    detection mechanisms (e.g., TCP [RFC6298], SCTP [RFC4960], SIP
    [RFC3261]).  At this point, our experience leads to a recognition
    that often specific tweaks that deviate from standardized time-based
    loss detectors do not materially impact network safety.  Therefore,
    in this document we outline a set of high-level protocol-agnostic
    requirements for time-based loss detection.  The intent is to
    provide a safe foundation on which implementations have the
    flexibility to instantiate mechanisms that best realize their
    specific goals.

   Context

    This document is different from from the way we ideally like to
    engineer systems.  Usually, we strive to understand high-level
    requirements as a starting point.  We then methodically engineer
    specific protocols, algorithms and systems that meet these
    requirements.  Within the standards process we have derived many
    time-based loss detection schemes without benefit from some
    over-arching requirements document---because we had no idea how to
    write such a document!  Therefore, we made the best specific
    decisions we could in response to specific needs.

    At this point, however, the community's experience has matured to
    the point where we can define a set of high-level requirements for
    time-based loss detection schemes.  We now understand how to
    separate the strategies these mechanisms use that are crucial for
    network safety from those small details that do not materially
    impact network safety.  However, adding a requirements umbrella to a
    body of existing specifications is inherently messy and we run the
    risk of creating inconsistencies with both past and future
    mechanisms.  The correct way to view this document is as the default
    case.  Specifically:

      - This document does not update or obsolete any existing RFC.
        These previous specifications---while generally consistent with
        the requirements in this document---reflect community consensus
        and this document does not change that consensus.

      - The requirements in this document are meant to provide for
        network safety and, as such, SHOULD be used by all time-based
        loss detection mechanisms.

      - The requirements in this document may not be appropriate in all
        cases and, therefore, inconsistent deviations may be necessary
        (hence the "SHOULD" in the last bullet).  However,
        inconsistencies MUST be (a) explained and (b) gather consensus.


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   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:

    (S.1) The requirements in this document apply only to the primary or
          last resort time-based loss detection.

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

    (S.2) The requirements for time-based loss detection mechanisms in
          this document are for the primary or "last resort" loss
          detection mechanism whether the mechanism is the sole loss
          repair strategy or works in concert with other mechanisms.

          While a straightforward time-based loss detector is sufficient
          for simple protocols like DNS [RFC1034,RFC1035], more complex
          protocols often use more advanced loss detectors to aid
          performance.  For instance, TCP and SCTP have methods to
          detect (and repair) loss based on explicit endpoint state
          sharing [RFC2018,RFC4960,RFC6675].  Such mechanisms often
          provide more timely and precise results than time-based loss
          detectors.  However, these mechanisms do not obviate the need
          for a "retransmission timeout" or "RTO" because---as we
          discuss in Section 1---only the passage of time can ultimately
          be relied upon to detect loss.  In cases such as these, the
          time-based loss detector functions as a "last resort".

          Also, note, that some recent proposals have incorporated time
          as a component of advanced loss detection methods---either as
          an aggressive first loss detector or in conjunction with
          endpoint state sharing [DCCM13,CCDJ20,IS20].  Since these
          timers are not used as "last resort" the requirements in this
          document need not be directly used in these cases.  However,
          we expect that many of the requirements are useful for these
          situations, as well.

    (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
          addresses B and C, A needs to use independent time-based loss
          detector instances for traffic sent to B and C.

    (S.4) There are cases where state is shared across connections

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          or flows (e.g., [RFC2140], [RFC3124]).  State pertaining to
          time-based loss detection is often discussed as sharable.
          These situations raise issues that the simple flow-oriented
          time-based loss detection 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 time-based loss
          detection information across flows is outside the scope of
          this document.

   Requirements

    We now list the requirements that apply when designing primary or
    last resort time-based loss detection mechanisms.  For historical
    reasons and ease of exposition, we refer to the time between sending
    a packet and determining the packet has been lost due to lack of
    delivery confirmation as the "retransmission timeout" or "RTO".
    After the RTO passes without delivery confirmation, the sender may
    safely assume the packet is lost.  However, as discussed above, the
    detected loss need not be repaired (i.e., the loss could be detected
    only for congestion control and not reliability purposes).

    (1) As we note above, loss detection happens when a sender does not
        receive delivery confirmation within an some expected period of
        time.  In the absence of any knowledge about the latency of a
        path, the initial RTO MUST be conservatively set to no less than
        1 second.

        Correctness is of the utmost importance when transmitting into a
        network with unknown properties because:

        - Premature loss detection can trigger spurious retransmits that
          could cause issues when a network is already congested.

        - Premature loss detection can needlessly cause congestion
          control to dramatically lower the sender's allowed
          transmission rate---especially since the rate is already
          likely low at this stage of the communication.  Recovering
          from such a rate change can taken a relatively long time.

        - Finally, as discussed below, sometimes using time-based
          loss detection and retransmissions can cause 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) We now specify four requirements that pertain to setting
        an expected time interval for delivery confirmation.

        Often measuring the time required for delivery confirmation is

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        is framed as assessing 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 and the expected latency is better framed as the
        "feedback time" (FT).  In other words, the expectation is not
        always simply a network property, but can include additional
        time before a sender should reasonably expect a response.

        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 network RTT between the client and resolver.

        Therefore, we express the requirements in terms of FT.  Again,
        for ease of exposition we use "RTO" to indicate the interval
        between a packet transmission and the decision the packet has
        been lost---regardless of whether the packet will be
        retransmitted.

        (a) If/when available, the RTO SHOULD be set based on multiple
            observations 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 deciding the given
            data is lost.  Network paths are inherently dynamic and
            therefore it is crucial to incorporate multiple FT samples
            in the RTO to take into account the delay variation across
            time.

            For example, TCP's RTO [RFC6298] would satisfy this
            requirement due to its use of an EWMA to combine multiple FT
            samples into a "smoothed RTT".  In the name of
            conservativeness, TCP goes further to also include an
            explicit variance term when computing the RTO.

        (b) FT observations SHOULD be taken and incorporated into the
            RTO at least once per RTT or as frequently as data is
            exchanged in cases where that happens less frequently than
            once per RTT.

            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.

            As an example, TCP takes an FT sample roughly once per RTT,

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            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, 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
        either (a) the subsequent successful transmission of
        non-retransmitted data, or (b) an RTO passes without detecting
        additional losses.  The former will generally be quicker.  The
        latter covers cases where loss is detected, but not repaired.

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

        This ensures network safety.

    (4) Loss detected by the RTO mechanism MUST be taken as an
        indication 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.

        An exception to this rule is if an IETF standardized mechanism
        determines 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, congestion
        control actions taken based on time-based loss detection could
        be reversed when a standard mechanism post-facto determines that
        the cause of the loss was not congestion (e.g., [RFC5682]).

   Discussion

    We note that research has shown the tension between the
    responsiveness and correctness of time-based loss detection 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
    exponentially weighted moving average (EWMA) gains, lowering the
    minimum RTO, etc.) can reduce the time required to detect actual
    loss.  However, at the same time, such aggressiveness leads to more
    cases of mistakenly declaring packets lost that ultimately arrived
    at the receiver.  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 detecting loss---even if
    falsely---carries a requirement to invoke a congestion response
    which will ultimately reduce the transmission rate.

    While the tradeoff between responsiveness and correctness seems
    fundamental, the tradeoff can be made less relevant if the sender
    can detect and recover from mistaken loss detection.  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
    mistakenly declaring packets to be lost.

    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 a steady-state minimum RTO
    that are less than the 1 second specified in [RFC6298] (which is
    different from the initial RTO we specify in Section 4, Requirement
    1).  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 could in fact increase the number of needless
    retransmissions the above requirements fail safe in that they insist
    on exponential backoff 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

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    there is a downside to being aggressive, the incentives for proper
    behavior are retained in the mechanism.

   Security Considerations

    This document does not alter the security properties of time-based
    loss detection mechanisms.  See [RFC6298] for a discussion of these
    within the context of TCP.

   IANA Considerations

    This document has no IANA considerations.

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, Martin Duke, Gorry Fairhurst, Rahul
    Arvind Jadhav, Mirja Kuhlewind, Nicolas Kuhn, Jonathan Looney and
    Michael Scharf provided useful comments on previous versions of this
    document.

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.

    [CCDJ20] Cheng, Y., N. Cardwell, N. Dukkipati, P. Jha, "RACK: a
        time-based fast loss detection algorithm for TCP",
        Internet-Draft draft-ietf-tcpm-rack-08.txt (work in progress),
        March 2020.

    [DCCM13] Dukkipati, N., N. Cardwell, Y. Cheng, M. Mathis, "Tail Loss
        Probe (TLP): An Algorithm for Fast Recovery of Tail Losses",
        Internet-Draft draft-dukkipati-tcpm-tcp-loss-probe-01.txt (work
        in progress), February 2013.

    [IS20] Iyengar, I., I. Swett, "QUIC Loss Detection and Congestion
        Control", Internet-Draft
        draft-ietf-quic-recovery-27.txt (work in progress), March 2020.

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

    [RFC1034] Mockapetris, P.  "Domain Names - Concepts and Facilities",
        RFC 1034, November 1987.


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    [RFC1035] Mockapetris, P.  "Domain Names - Implementation and
        Specification", RFC 1035, November 1987.

    [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
        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

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        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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