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Versions: (draft-allman-tcpm-rto-consider) 00
01 02 03 04 05
Internet Engineering Task Force M. Allman
INTERNET-DRAFT ICSI
File: draft-ietf-tcpm-rto-consider-05.txt March 10, 2017
Intended Status: Best Current Practice
Expires: September 10, 2017
Retransmission Timeout Requirements
Status of this Memo
This document may not be modified, and derivative works of it may
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Abstract
Ensuring reliable communication often manifests in a timeout and
retry mechanism. Each implementation of a retransmission timeout
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 retransmission timeout schemes
appropriate for general use in the Internet. Within the
requirements, implementations have latitude to define particulars
that best address each situation.
Terminology
The key words "MUST", "MUST NOT", "REQUIRED", "SHALL", "SHALL NOT",
"SHOULD", "SHOULD NOT", "RECOMMENDED", "MAY", and "OPTIONAL" in this
document are to be interpreted as described in BCP 14, RFC 2119
[RFC2119].
1 Introduction
Reliable transmission is a key property for many network protocols
and applications. Our protocols use various mechanisms to achieve
reliable data transmission. Often we use continuous or periodic
reports from the recipient to inform the sender's notion of which
pieces of data are missing and need to be retransmitted to ensure
reliability. Alternatively, information coding---e.g., FEC---can be
used to achieve probabilistic reliability without retransmissions.
However, despite our best intentions and most robust mechanisms, the
only thing we can truly depend on is the passage of time and
therefore our ultimate backstop to ensuring reliability is a timeout
and re-try mechanism. 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 assumes the data was lost in transit and therefore
schedules a retransmission. This process of ensuring reliability
via time-based loss detection and resending lost data is commonly
referred to as a "retransmission timeout (RTO)" mechanism.
Various protocols have defined their own RTO mechanisms (e.g., TCP
[RFC6298], SCTP [RFC4960], SIP [RFC3261]). The specifics of
retransmission timeouts often represent a particular tradeoff
between correctness and responsiveness [AP99]. In other words we
want to simultaneously:
- wait long enough to ensure the detection of loss is correct and
therefore a retransmission is in fact needed, and
- bound the delay we impose on applications before repairing
loss.
Serving both of these goals is difficult as they pull in opposite
directions. I.e., towards either (a) withholding needed
retransmissions too long to ensure the original transmission is
truly lost or (b) not waiting long enough---to help application
responsiveness---and hence sending unnecessary (often denoted
"spurious") retransmissions.
We have found that even though the RTO procedure is standardized for
some protocols (e.g., TCP [RFC6298]), implementations often add
their own subtle imprint on the specifics of the process to tilt the
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tradeoff between correctness and responsiveness in some particular
way.
At this point we recognize that often these specific tweaks that
deviate from standardized RTO mechanisms do not materially impact
network safety. Therefore, in this document we outline a set of
high-level protocol-agnostic requirements for RTO mechanisms. The
intent is to provide a safe foundation on which implementations have
the flexibility to instantiate mechanisms that best realize their
specific goals.
2 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 3:
(S.1) The requirements in this document apply only to timer-based
loss detection and retransmission.
While there are a bevy of uses for timers in protocols---from
rate-based pacing to connection failure detection to making
congestion control decisions and beyond---these are outside
the scope of this document.
(S.2) The requirements in this document only apply to cases where
loss detected via a timer is repaired by a retransmission of
the original data.
Other cases are certainly possible---e.g., replacing the lost
data with an updated version---but fall outside the scope of
this document.
(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
hosts B and C, A must use independent RTOs for traffic sent to
B and C.
(S.4) There are cases where state is shared across connections or
flows (e.g., [RFC2140], [RFC3124]). The RTO is one piece
state that is often discussed as sharable. These situations
raise issues that the simple flow-oriented RTO 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 3 are likely applicable,
sharing RTOs across flows is outside the scope of this
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document.
(S.5) The requirements in this document apply to reliable
transmission, but do not assume that all data transmitted
within a connection or flow is reliably sent.
E.g., a protocol like DCCP [RFC4340] could leverage the
requirements in this document for the initial reliable
handshake even though the protocol reverts to unreliable
transmission after the handshake.
E.g., a protocol like SCTP [RFC4960] could leverage the
requirements for data that is sent only "partially reliably".
In this case, the protocol uses two phases for each message.
In the first phase, the protocol attempts to ensure
reliability and can leverage the requirements in this
document. At some point the value of the data is gone and the
protocol transitions to the second phase where the data is
treated as unreliably transmitted and therefore the protocol
will no longer attempt to repair the loss---and hence there
are no more retransmissions and the requirements in this
document are moot.
(S.6) The requirements for RTO mechanisms in this document can be
applied regardless of whether the RTO mechanism is the sole
loss repair strategy or works in concert with other
mechanisms.
E.g., for a simple protocol like UDP-based DNS [] a timeout
and re-try mechanism is likely to act alone to ensure
reliability.
E.g., within a complex protocol like TCP or SCTP we have
designed methods to detect and repair loss based on explicit
endpoint state sharing [RFC2018,RFC4960,RFC6675]. These
mechanisms are preferred over the RTO as they are often more
timely and precise than the coarse-grained RTO. In these
cases, the RTO becomes a last resort when the more advanced
mechanisms fail.
3 Requirements
We now list the requirements that apply when designing
retransmission timeout (RTO) mechanisms.
(1) In the absence of any knowledge about the latency of a path, the
RTO MUST be conservatively set to no less than 1 second.
This requirement ensures two important aspects of the RTO.
First, when transmitting into an unknown network,
retransmissions will not be sent before an ACK would reasonably
be expected to arrive and hence possibly waste scarce network
resources. Second, as noted below, sometimes retransmissions
can lead to ambiguities in assessing the latency of a network
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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) As we note above, loss detection happens when a sender does not
receive delivery confirmation within an some expected period of
time. We now specify four requirements that pertain to setting
the length of this expectation.
Often measuring the time required for delivery confirmation is
is framed as involving the "round-trip time (RTT)" of the
network path as this is the minimum amount of time required to
receive delivery confirmation and also often follows protocol
behavior whereby acknowledgments are generated quickly after
data arrives. For instance, this is the case for the RTO used
by TCP [RFC6298] and SCTP [RFC4960]. However, this is somewhat
mis-leading as the expected latency is better framed as the
"feedback time" (FT). In other words, the expectation is not
always simply a network property, but includes additional time
before a sender should reasonably expect a response to a query.
For instance, consider a UDP-based DNS request from a client to
a recursive resolver. When the request can be served from the
resolver's cache the FT likely well approximates the network RTT
between the client and resolver. However, on a cache miss the
resolver will request the needed information from one or more
authoritative DNS servers, which will non-trivially increase the
FT compared to the RTT between the client and resolver.
Therefore, we express the following requirements in terms of FT:
(a) In steady state the RTO SHOULD be set based on recent
observations of both the FT and the variance of the FT.
In other words, the RTO should be based on a reasonable
amount of time that the sender should wait for delivery
confirmation before retransmitting the given data.
(b) FT observations SHOULD be taken regularly.
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.
The notion of "regularly" SHOULD be defined as at least once
per RTT or as frequently as data is exchanged in cases where
that happens less frequently than once per RTT. However, we
also recognize that it may not always be practical to take
an FT sample this often in all cases. Hence, this
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once-per-RTT definition of "regularly" is explicitly a
"SHOULD" and not a "MUST".
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 keepalives, heartbeats or other messages
to exchange control information. To the extent that the
latency of these transactions mirrors data exchange, they
can be leveraged to take FT samples within the RTO
mechanism. Such samples can help protocols keep their RTO
accurate during lulls in data transmission. However, given
that these messages may not be subject to the same delays as
data transmission, we do not take a general view on whether
this is useful or not.
(d) An RTO mechanism MUST NOT use ambiguous FT samples.
Assume two copies of some segment X are transmitted at times
t0 and t1 and then at time t2 the sender receives
confirmation that X in fact arrived. In some cases, it is
not clear which copy of X triggered the confirmation and
hence the actual FT is either t2-t1 or t2-t0, but which is a
mystery. Therefore, in this situation an implementation
MUST use Karn's algorithm [KP87,RFC6298] and use neither
version of the FT sample and hence not update the RTO.
There are cases where two copies of some data are
transmitted in a way whereby the sender can tell which is
being acknowledged by an incoming ACK. E.g., TCP's
timestamp option [RFC7323] allows for segments to be
uniquely identified and hence avoid the ambiguity. In such
cases there is no ambiguity and the resulting samples can
update the RTO.
(3) Each time the RTO is used to detect a loss and a retransmission
is scheduled, 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 the successful repair of the
lost data and subsequent transmission of non-retransmitted data.
A maximum value MAY be placed on the RTO. The maximum RTO MUST
NOT be less than 60 seconds (a la [RFC6298]).
This ensures network safety.
(4) Retransmissions triggered by the RTO mechanism MUST be taken as
indications of network congestion and the sending rate adapted
using a standard mechanism (e.g., TCP collapses the congestion
window to one segment [RFC5681]).
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This ensures network safety.
Exception could be made to this rule if an IETF standardized
mechanism is used to determine 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,
RTO-triggered congestion control actions may be reversed when a
standard mechanism determines that the cause of the loss was not
congestion after all (e.g., [RFC5682]).
4 Discussion
We note that research has shown the tension between the
responsiveness and correctness of retransmission timeouts 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 EWMA gains,
lowering the minimum RTO, etc.) can reduce the time spent waiting on
needed retransmissions. However, at the same time, such
aggressiveness leads to more needless retransmissions. 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 an RTO expiration carries a requirement to invoke
a congestion response and hence slow transmission down.
While the tradeoff between responsiveness and correctness seems
fundamental, the tradeoff can be made less relevant if the sender
can detect and recover from spurious RTOs. 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 needless retransmits.
Also, note, that in addition to the experiments discussed in [AP99],
the Linux TCP implementation has been using various non-standard RTO
mechanisms for many years seemingly without large scale problems
(e.g., using different EWMA gains than specified in [RFC6298]).
Further, a number of implementations use minimum RTOs that are 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.
Finally, we note that while allowing implementations to be more
aggressive may in fact increase the number of needless
retransmissions the above requirements fail safe in that they insist
on exponential backoff of the RTO 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.
5 Security Considerations
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This document does not alter the security properties of
retransmission timeout mechanisms. See [RFC6298] for a discussion
of these within the context of TCP.
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, Gorry Fairhurst, Mirja Kuhlewind,
Jonathan Looney and Michael Scharf provided useful comments on a
previous version of this draft.
Normative References
[RFC2119] Bradner, S., "Key words for use in RFCs to Indicate
Requirement Levels", BCP 14, RFC 2119, March 1997.
Informative References
[AP99] Allman, M., V. Paxson, "On Estimating End-to-End Network Path
Properties", Proceedings of the ACM SIGCOMM Technical Symposium,
September 1999.
[KP87] Karn, P. and C. Partridge, "Improving Round-Trip Time
Estimates in Reliable Transport Protocols", SIGCOMM 87.
[RFC2018] Mathis, M., Mahdavi, J., Floyd, S., and A. Romanow, "TCP
Selective Acknowledgment Options", RFC 2018, October 1996.
[RFC2140] Touch, J., "TCP Control Block Interdependence", RFC 2140,
April 1997.
[RFC2883] Floyd, S., Mahdavi, J., Mathis, M., and M. Podolsky, "An
Extension to the Selective Acknowledgement (SACK) Option for
TCP", RFC 2883, July 2000.
[RFC3124] Balakrishnan, H., S. Seshan, "The Congestion Manager", RFC
2134, June 2001.
[RFC3261] Rosenberg, J., Schulzrinne, H., Camarillo, G., Johnston,
A., Peterson, J., Sparks, R., Handley, M., and E. Schooler,
"SIP: Session Initiation Protocol", RFC 3261, June 2002.
[RFC3522] Ludwig, R., M. Meyer, "The Eifel Detection Algorithm for
TCP", RFC 3522, april 2003.
[RFC3708] Blanton, E., M. Allman, "Using TCP Duplicate Selective
Acknowledgement (DSACKs) and Stream Control Transmission
Protocol (SCTP) Duplicate Transmission Sequence Numbers (TSNs)
to Detect Spurious Retransmissions", RFC 3708, February 2004.
[RFC3940] Adamson, B., C. Bormann, M. Handley, J. Macker,
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"Negative-acknowledgment (NACK)-Oriented Reliable Multicast
(NORM) Protocol", November 2004, RFC 3940.
[RFC4340] Kohler, E., M. Handley, S. Floyd, "Datagram Congestion
Control Protocol (DCCP)", March 2006, RFC 4340.
[RFC4960] Stweart, R., "Stream Control Transmission Protocol", RFC
4960, September 2007.
[RFC5682] Sarolahti, P., M. Kojo, K. Yamamoto, M. Hata, "Forward
RTO-Recovery (F-RTO): An Algorithm for Detecting Spurious
Retransmission Timeouts with TCP", RFC 5682, September 2009.
[RFC5740] Adamson, B., C. Bormann, M. Handley, J. Macker,
"NACK-Oriented Reliable Multicast (NORM) Transport Protocol",
November 2009, RFC 5740.
[RFC6182] Ford, A., C. Raiciu, M. Handley, S. Barre, J. Iyengar,
"Architectural Guidelines for Multipath TCP Development", March
2011, RFC 6182.
[RFC6298] Paxson, V., M. Allman, H.K. Chu, M. Sargent, "Computing
TCP's Retransmission Timer", June 2011, RFC 6298.
[RFC6582] Henderson, T., S. Floyd, A. Gurtov, Y. Nishida, "The
NewReno Modification to TCP's Fast Recovery Algorithm", April
2012, RFC 6582.
[RFC6675] Blanton, E., M. Allman, L. Wang, I. Jarvinen, M. Kojo,
Y. Nishida, "A Conservative Loss Recovery Algorithm Based on
Selective Acknowledgment (SACK) for TCP", August 2012, RFC 6675.
[RFC7323] Borman D., B. Braden, V. Jacobson, R. Scheffenegger, "TCP
Extensions for High Performance", September 2014, RFC 7323.
Authors' Addresses
Mark Allman
International Computer Science Institute
1947 Center St. Suite 600
Berkeley, CA 94704
EMail: mallman@icir.org
http://www.icir.org/mallman
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