Requirements for Time-Based Loss Detection
RFC 8961
| Document | Type |
RFC - Best Current Practice
(November 2020; No errata)
Also known as BCP 233
|
|
|---|---|---|---|
| Author | Mark Allman | ||
| Last updated | 2020-11-23 | ||
| Replaces | draft-allman-tcpm-rto-consider | ||
| Stream | Internet Engineering Task Force (IETF) | ||
| Formats | plain text html xml pdf htmlized bibtex | ||
| Reviews | |||
| Stream | WG state | Submitted to IESG for Publication (wg milestone: May 2020 - Submit document on R... ) | |
| Document shepherd | Yoshifumi Nishida | ||
| Shepherd write-up | Show (last changed 2020-06-22) | ||
| IESG | IESG state | RFC 8961 (Best Current Practice) | |
| Action Holders |
(None)
|
||
| Consensus Boilerplate | Yes | ||
| Telechat date | |||
| Responsible AD | Martin Duke | ||
| Send notices to | Yoshifumi Nishida <nsd.ietf@gmail.com> | ||
| IANA | IANA review state | Version Changed - Review Needed | |
| IANA action state | No IANA Actions | ||
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