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Requirements for Time-Based Loss Detection

The information below is for an old version of the document that is already published as an RFC.
Document Type
This is an older version of an Internet-Draft that was ultimately published as RFC 8961.
Author Mark Allman
Last updated 2020-11-23 (Latest revision 2020-07-27)
Replaces draft-allman-tcpm-rto-consider
RFC stream Internet Engineering Task Force (IETF)
Intended RFC status Best Current Practice
Additional resources Mailing list discussion
Stream WG state Submitted to IESG for Publication
Document shepherd Yoshifumi Nishida
Shepherd write-up Show Last changed 2020-06-22
IESG IESG state Became RFC 8961 (Best Current Practice)
Action Holders
Consensus boilerplate Yes
Telechat date (None)
Responsible AD Martin Duke
Send notices to Yoshifumi Nishida <>
IANA IANA review state Version Changed - Review Needed
IANA action state No IANA Actions
Internet Engineering Task Force                                M. Allman
INTERNET-DRAFT                                                      ICSI
File: draft-ietf-tcpm-rto-consider-17.txt                  July 27, 2020
Intended Status: Best Current Practice
Expires: January 27, 2021

               Requirements for Time-Based Loss Detection

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
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    This Internet-Draft will expire on January 27, 2021.

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    ( in effect on the date of
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    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, ultimately, protocols 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 unicast communication across the
    Internet.  Within the requirements, implementations have latitude to
    define particulars that best address each situation.


    The key words "MUST", "MUST NOT", "REQUIRED", "SHALL", "SHALL NOT",
    "OPTIONAL" in this document are to be interpreted as described in
    BCP 14 [RFC2119] [RFC8174] when, and only when, they appear in all
    capitals, as shown here.

1   Introduction

    As a network of networks, the Internet consists of a large variety
    of links and systems that support a wide variety of tasks and
    workloads.  The service provided by the network varies from
    best-effort delivery among loosely connected components to highly
    predictable delivery within controlled environments (e.g., between
    physically connected nodes, within a tightly controlled data
    center).  Each path through the network has a set of path
    properties---e.g., available capacity, delay, packet loss.  Given
    the range of networks that make up the Internet, these properties
    range from largely static to highly dynamic.

    This document provides guidelines for developing an understanding of
    one path property: packet loss.  In particular, we offer guidelines
    for developing and implementing time-based loss detectors that have
    been gradually learned over the last several decades.  We focus on
    the general case where the loss properties of a path are (a) unknown
    a priori and (b) dynamically vary over time.  Further, while there
    are numerous root causes of packet loss, we leverage the
    conservative notion that loss is an implicit indication of
    congestion [RFC5681].  While this stance is not always correct, as a
    general assumption it has historically served us well [Jac88].  As
    we discuss further in section 2, the guidelines in this document
    should be viewed as a general default for unicast communication
    across best-effort networks and not as optimal---or even
    applicable---for all situations.

    Given that packet loss is routine in best-effort networks, 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.

            As we mention above, packet loss is often taken as an

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            implicit 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

    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
    Serving both of these goals is difficult as they pull in opposite
    directions [AP99].  By not waiting long enough to accurately
    determine a packet has been lost we may provide a needed
    retransmission in a timely manner, but 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---such as TCP [RFC6298], SCTP
    [RFC4960], SIP [RFC3261]---use their own time-based loss detection
    mechanisms.  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 with respect
    to congestion control [AP99].  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.

2   Context
    This document is different 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 IETF standards process we have derived many time-based

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    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 general, 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.  The requirements in this document
    may not be appropriate in all cases.  In particular, the guidelines
    in section 4 are concerned with the general case, but specific
    situations may allow for more flexibility in terms of loss detection
    because specific facets of the environment are known (e.g., when
    operating over a single physical link or within a tightly controlled
    data center).  Therefore, variants, deviations or wholly different
    time-based loss detectors may be necessary or useful in some cases.
    The correct way to view this document is as the default case and not
    as a one-size-fits-all that is optimal in all cases.

    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.  Therefore, we make the
    following statements about the relationship of this document to past
    and future specifications:

      - 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 future
        time-based loss detection mechanisms.

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

    (S.1) While there are a bevy of uses for timers in protocols---from
          rate-based pacing to connection failure detection and
          beyond---this document is focused only on loss detection.

    (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

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          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 loss detection 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 other words, ultimately we
          cannot count on acknowledgments to arrive at the data sender
          to indicate which packets never arrived at the receiver.  In
          cases such as these we need a time-based loss detector to
          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 in certain situations or in
          conjunction with endpoint state sharing [DCCM13,CCDJ20,IS20].
          While these mechanisms can aid timely loss recovery, the
          protocol ultimately leans on another more conservative timer
          to ensure reliability when these mechanisms break down.  The
          requirements in this document are only directly applicable to
          last resort loss detection.  However, we expect that many of
          the requirements can serve as useful guidelines for more
          aggressive non-last resort timers, 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

          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 
          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. 

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4   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 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
        framed as assessing the "round-trip time (RTT)" of the network
        path.  The RTT 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.

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        For instance, consider a UDP-based DNS request from a client to
        a recursive resolver [RFC1035].  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

        (a) The RTO SHOULD be set based on multiple observations of the
            FT when available.

            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 recent 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 exponentially-weighted
            moving average (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.

            While multiple FT samples are crucial for capturing the
            delay dynamics of a path, we explicitly do not tightly
            specify the process---including the number of FT samples to
            use and how/when to age samples out of the RTO
            calculation---as the particulars could depend on the
            situation and/or goals of each specific loss detector.

            Finally, FT samples come from packet exchanges between
            peers.  We encourage protocol designers---especially for new
            protocols---to strive to ensure the feedback is not easily
            spoofable by on- or off-path attackers such that they can
            perturb a host's notion of the FT.  Ideally, all messages
            would be cryptographically secure, but given that this is
            not always possible---especially in legacy protocols---using
            a healthy amount of randomness in the packets is encouraged.

        (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.     

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            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

            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.

            Some protocols use non-data exchanges for various
            reasons---e.g., keepalives, heartbeats, control messages.
            To the extent that the latency of these exchanges 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 packet 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 NOT use either version of the FT sample and hence not
            update the RTO (as discussed in [KP87,RFC6298]).  

            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 packets 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) 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 packet [RFC5681]).

        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

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        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]).
    (4) 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.

        As with guideline (3), an exception to this rule exists if an
        IETF standardized mechanism determines that a particular loss is
        not due to congestion.

5   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 TCP implementations use a steady-state minimum
    RTO that is less than the 1 second specified in [RFC6298].  While
    the implication of these deviations from the standard may be more

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    spurious retransmits (per [AP99]), we are aware of no large-scale
    network safety issues caused by this change to the minimum RTO.
    This informs the guidelines in the last section (e.g., there is no
    minimum RTO specified).

    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
    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 time-based
    loss detection mechanisms.  See [RFC6298] for a discussion of these
    within the context of TCP.

7   IANA Considerations

    This document has no IANA considerations.


    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, Stewart Bryant, Martin Duke, Wesley
    Eddy, Gorry Fairhurst, Rahul Arvind Jadhav, Benjamin Kaduk, 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.

    [RFC8174] Leiba, B., "Ambiguity of Uppercase vs Lowercase in RFC
        2119 Key Words", RFC 8174, May 2017.

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.

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    [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. 

    [Jac88] Jacobson, V., "Congestion Avoidance and Control", ACM
        SIGCOMM, August 1988.

    [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.

    [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
        3124, 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.

    [RFC4960] Stweart, R., "Stream Control Transmission Protocol", RFC
        4960, September 2007.
    [RFC5681] Allman, M., V. Paxson, E. Blanton, "TCP Congestion
        Control", RFC 5681, September 2009.
    [RFC5682] Sarolahti, P., M. Kojo, K. Yamamoto, M. Hata, "Forward
        RTO-Recovery (F-RTO): An Algorithm for Detecting Spurious

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        Retransmission Timeouts with TCP", RFC 5682, September 2009.

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

    [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.

    [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


Expires: January 27, 2021                                      [Page 12]