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+Internet Engineering Task Force (IETF) M. Allman
+Request for Comments: 8961 ICSI
+BCP: 233 November 2020
+Category: Best Current Practice
+ISSN: 2070-1721
+
+
+ Requirements for Time-Based Loss Detection
+
+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, 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; therefore,
+ no implementation suits all situations. This document 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.
+
+Status of This Memo
+
+ This memo documents an Internet Best Current Practice.
+
+ This document is a product of the Internet Engineering Task Force
+ (IETF). It represents the consensus of the IETF community. It has
+ received public review and has been approved for publication by the
+ Internet Engineering Steering Group (IESG). Further information on
+ BCPs is available in Section 2 of RFC 7841.
+
+ Information about the current status of this document, any errata,
+ and how to provide feedback on it may be obtained at
+ https://www.rfc-editor.org/info/rfc8961.
+
+Copyright Notice
+
+ Copyright (c) 2020 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
+ (https://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.
+
+Table of Contents
+
+ 1. Introduction
+ 1.1. Terminology
+ 2. Context
+ 3. Scope
+ 4. Requirements
+ 5. Discussion
+ 6. Security Considerations
+ 7. IANA Considerations
+ 8. References
+ 8.1. Normative References
+ 8.2. Informative References
+ Acknowledgments
+ Author's Address
+
+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, and 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 varying 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 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 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
+ determine a packet has been lost, we may provide a needed
+ retransmission in a timely manner but risk both 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], and 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.
+
+1.1. Terminology
+
+ The key words "MUST", "MUST NOT", "REQUIRED", "SHALL", "SHALL NOT",
+ "SHOULD", "SHOULD NOT", "RECOMMENDED", "NOT RECOMMENDED", "MAY", and
+ "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.
+
+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 loss
+ detection schemes without the benefit of 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
+ one-size-fits-all guidance 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; 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
+ 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, we ultimately 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 function
+ 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
+ document.
+
+ Protocols such as SCTP [RFC4960] and Multipath TCP (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. That is, 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] and [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.
+
+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 take 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 round-trip times (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 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 misleading, and the expected latency
+ is better framed as the "feedback time" (FT). In other words,
+ the expectation is not always simply a network property; it 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 [RFC1035]. When the request can be served
+ from the resolver's cache, the feedback time (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 that the packet
+ has been lost, regardless of whether the packet will be
+ retransmitted.
+
+ (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;
+ 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.
+
+ 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,
+ 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, and 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. 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;
+ 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] and [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. For example, 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
+ 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 that
+ 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], Forward
+ RTO-Recovery (F-RTO) [RFC5682], and Duplicate Selective
+ Acknowledgement (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
+ 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 safely 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 actions.
+
+8. References
+
+8.1. Normative References
+
+ [RFC2119] Bradner, S., "Key words for use in RFCs to Indicate
+ Requirement Levels", BCP 14, RFC 2119,
+ DOI 10.17487/RFC2119, March 1997,
+ <https://www.rfc-editor.org/info/rfc2119>.
+
+ [RFC8174] Leiba, B., "Ambiguity of Uppercase vs Lowercase in RFC
+ 2119 Key Words", BCP 14, RFC 8174, DOI 10.17487/RFC8174,
+ May 2017, <https://www.rfc-editor.org/info/rfc8174>.
+
+8.2. Informative References
+
+ [AP99] Allman, M. and V. Paxson, "On Estimating End-to-End
+ Network Path Properties", Proceedings of the ACM SIGCOMM
+ Technical Symposium, September 1999.
+
+ [CCDJ20] Cheng, Y., Cardwell, N., Dukkipati, N., and P. Jha, "The
+ RACK-TLP loss detection algorithm for TCP", Work in
+ Progress, Internet-Draft, draft-ietf-tcpm-rack-13, 2
+ November 2020,
+ <https://tools.ietf.org/html/draft-ietf-tcpm-rack-13>.
+
+ [DCCM13] Dukkipati, N., Cardwell, N., Cheng, Y., and M. Mathis,
+ "Tail Loss Probe (TLP): An Algorithm for Fast Recovery of
+ Tail Losses", Work in Progress, Internet-Draft, draft-
+ dukkipati-tcpm-tcp-loss-probe-01, 25 February 2013,
+ <https://tools.ietf.org/html/draft-dukkipati-tcpm-tcp-
+ loss-probe-01>.
+
+ [IS20] Iyengar, J., Ed. and I. Swett, Ed., "QUIC Loss Detection
+ and Congestion Control", Work in Progress, Internet-Draft,
+ draft-ietf-quic-recovery-32, 20 October 2020,
+ <https://tools.ietf.org/html/draft-ietf-quic-recovery-32>.
+
+ [Jac88] Jacobson, V., "Congestion avoidance and control", ACM
+ SIGCOMM, DOI 10.1145/52325.52356, August 1988,
+ <https://doi.org/10.1145/52325.52356>.
+
+ [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",
+ STD 13, RFC 1034, DOI 10.17487/RFC1034, November 1987,
+ <https://www.rfc-editor.org/info/rfc1034>.
+
+ [RFC1035] Mockapetris, P., "Domain names - implementation and
+ specification", STD 13, RFC 1035, DOI 10.17487/RFC1035,
+ November 1987, <https://www.rfc-editor.org/info/rfc1035>.
+
+ [RFC2018] Mathis, M., Mahdavi, J., Floyd, S., and A. Romanow, "TCP
+ Selective Acknowledgment Options", RFC 2018,
+ DOI 10.17487/RFC2018, October 1996,
+ <https://www.rfc-editor.org/info/rfc2018>.
+
+ [RFC2140] Touch, J., "TCP Control Block Interdependence", RFC 2140,
+ DOI 10.17487/RFC2140, April 1997,
+ <https://www.rfc-editor.org/info/rfc2140>.
+
+ [RFC2883] Floyd, S., Mahdavi, J., Mathis, M., and M. Podolsky, "An
+ Extension to the Selective Acknowledgement (SACK) Option
+ for TCP", RFC 2883, DOI 10.17487/RFC2883, July 2000,
+ <https://www.rfc-editor.org/info/rfc2883>.
+
+ [RFC3124] Balakrishnan, H. and S. Seshan, "The Congestion Manager",
+ RFC 3124, DOI 10.17487/RFC3124, June 2001,
+ <https://www.rfc-editor.org/info/rfc3124>.
+
+ [RFC3261] Rosenberg, J., Schulzrinne, H., Camarillo, G., Johnston,
+ A., Peterson, J., Sparks, R., Handley, M., and E.
+ Schooler, "SIP: Session Initiation Protocol", RFC 3261,
+ DOI 10.17487/RFC3261, June 2002,
+ <https://www.rfc-editor.org/info/rfc3261>.
+
+ [RFC3522] Ludwig, R. and M. Meyer, "The Eifel Detection Algorithm
+ for TCP", RFC 3522, DOI 10.17487/RFC3522, April 2003,
+ <https://www.rfc-editor.org/info/rfc3522>.
+
+ [RFC3708] Blanton, E. and 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,
+ DOI 10.17487/RFC3708, February 2004,
+ <https://www.rfc-editor.org/info/rfc3708>.
+
+ [RFC4960] Stewart, R., Ed., "Stream Control Transmission Protocol",
+ RFC 4960, DOI 10.17487/RFC4960, September 2007,
+ <https://www.rfc-editor.org/info/rfc4960>.
+
+ [RFC5681] Allman, M., Paxson, V., and E. Blanton, "TCP Congestion
+ Control", RFC 5681, DOI 10.17487/RFC5681, September 2009,
+ <https://www.rfc-editor.org/info/rfc5681>.
+
+ [RFC5682] Sarolahti, P., Kojo, M., Yamamoto, K., and M. Hata,
+ "Forward RTO-Recovery (F-RTO): An Algorithm for Detecting
+ Spurious Retransmission Timeouts with TCP", RFC 5682,
+ DOI 10.17487/RFC5682, September 2009,
+ <https://www.rfc-editor.org/info/rfc5682>.
+
+ [RFC5740] Adamson, B., Bormann, C., Handley, M., and J. Macker,
+ "NACK-Oriented Reliable Multicast (NORM) Transport
+ Protocol", RFC 5740, DOI 10.17487/RFC5740, November 2009,
+ <https://www.rfc-editor.org/info/rfc5740>.
+
+ [RFC6182] Ford, A., Raiciu, C., Handley, M., Barre, S., and J.
+ Iyengar, "Architectural Guidelines for Multipath TCP
+ Development", RFC 6182, DOI 10.17487/RFC6182, March 2011,
+ <https://www.rfc-editor.org/info/rfc6182>.
+
+ [RFC6298] Paxson, V., Allman, M., Chu, J., and M. Sargent,
+ "Computing TCP's Retransmission Timer", RFC 6298,
+ DOI 10.17487/RFC6298, June 2011,
+ <https://www.rfc-editor.org/info/rfc6298>.
+
+ [RFC6675] Blanton, E., Allman, M., Wang, L., Jarvinen, I., Kojo, M.,
+ and Y. Nishida, "A Conservative Loss Recovery Algorithm
+ Based on Selective Acknowledgment (SACK) for TCP",
+ RFC 6675, DOI 10.17487/RFC6675, August 2012,
+ <https://www.rfc-editor.org/info/rfc6675>.
+
+ [RFC7323] Borman, D., Braden, B., Jacobson, V., and R.
+ Scheffenegger, Ed., "TCP Extensions for High Performance",
+ RFC 7323, DOI 10.17487/RFC7323, September 2014,
+ <https://www.rfc-editor.org/info/rfc7323>.
+
+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 TCPIMPL Working Groups. Ran Atkinson,
+ Yuchung Cheng, David Black, Stewart Bryant, Martin Duke, Wesley Eddy,
+ Gorry Fairhurst, Rahul Arvind Jadhav, Benjamin Kaduk, Mirja
+ Kühlewind, Nicolas Kuhn, Jonathan Looney, and Michael Scharf provided
+ useful comments on previous draft versions of this document.
+
+Author's Address
+
+ Mark Allman
+ International Computer Science Institute
+ 2150 Shattuck Ave., Suite 1100
+ Berkeley, CA 94704
+ United States of America
+
+ Email: mallman@icir.org
+ URI: https://www.icir.org/mallman