Internet DRAFT - draft-ietf-tcpm-rto-consider
draft-ietf-tcpm-rto-consider
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
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Abstract
Many protocols must detect packet loss for various reasons (e.g., to
ensure reliability using retransmissions or to understand the level
of congestion along a network path). While many mechanisms have
been designed to detect loss, 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.
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.
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
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 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
document.
Protocols such as SCTP [RFC4960] and MP-TCP [RFC6182] that
communicate in a unicast fashion with multiple specific
endpoints can leverage the requirements in this document
provided they track state and follow the requirements for each
endpoint independently. I.e., if host A communicates with
addresses B and C, A needs to use independent time-based loss
detector instances for traffic sent to B and C.
(S.4) There are cases where state is shared across connections
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
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 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
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, 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.
Acknowledgments
This document benefits from years of discussions with Ethan Blanton,
Sally Floyd, Jana Iyengar, Shawn Ostermann, Vern Paxson, and the
members of the TCPM and TCP-IMPL working groups. Ran Atkinson,
Yuchung Cheng, David Black, 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
EMail: mallman@icir.org
http://www.icir.org/mallman
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