TCP retransmit timer: Computing TCP's Retransmission Timer
The formula TCP senders must use to decide how long to wait before sending unacknowledged data again.
Paxson, V. and M. Allman, "Computing TCP's Retransmission Timer", RFC 2988, DOI 10.17487/RFC2988, November 2000, <https://www.rfc-editor.org/info/rfc2988>.
@misc{rfc2988,
series = {Request for Comments},
number = 2988,
howpublished = {RFC 2988},
publisher = {RFC Editor},
doi = {10.17487/RFC2988},
url = {https://www.rfc-editor.org/info/rfc2988},
author = {V. Paxson and M. Allman},
title = {{Computing TCP's Retransmission Timer}},
pagetotal = 8,
year = 2000,
month = nov,
} RFC text
Contents
Network Working Group V. Paxson
Request for Comments: 2988 ACIRI
Category: Standards Track M. Allman
NASA GRC/BBN
November 2000
Computing TCP's Retransmission Timer
Status of this Memo
This document specifies an Internet standards track protocol for the
Internet community, and requests discussion and suggestions for
improvements. Please refer to the current edition of the "Internet
Official Protocol Standards" (STD 1) for the standardization state
and status of this protocol. Distribution of this memo is unlimited.
Copyright Notice
Copyright (C) The Internet Society (2000). All Rights Reserved.
Abstract
This document defines the standard algorithm that Transmission
Control Protocol (TCP) senders are required to use to compute and
manage their retransmission timer. It expands on the discussion in
section 4.2.3.1 of RFC 1122 and upgrades the requirement of
supporting the algorithm from a SHOULD to a MUST.
1 Introduction
The Transmission Control Protocol (TCP) [Pos81] uses a retransmission
timer to ensure data delivery in the absence of any feedback from the
remote data receiver. The duration of this timer is referred to as
RTO (retransmission timeout). RFC 1122 [Bra89] specifies that the
RTO should be calculated as outlined in [Jac88].
This document codifies the algorithm for setting the RTO. In
addition, this document expands on the discussion in section 4.2.3.1
of RFC 1122 and upgrades the requirement of supporting the algorithm
from a SHOULD to a MUST. RFC 2581 [APS99] outlines the algorithm TCP
uses to begin sending after the RTO expires and a retransmission is
sent. This document does not alter the behavior outlined in RFC 2581
[APS99].
In some situations it may be beneficial for a TCP sender to be more
conservative than the algorithms detailed in this document allow.
However, a TCP MUST NOT be more aggressive than the following
algorithms allow.
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 [Bra97].
2 The Basic Algorithm
To compute the current RTO, a TCP sender maintains two state
variables, SRTT (smoothed round-trip time) and RTTVAR (round-trip
time variation). In addition, we assume a clock granularity of G
seconds.
The rules governing the computation of SRTT, RTTVAR, and RTO are as
follows:
(2.1) Until a round-trip time (RTT) measurement has been made for a
segment sent between the sender and receiver, the sender SHOULD
set RTO <- 3 seconds (per RFC 1122 [Bra89]), though the
"backing off" on repeated retransmission discussed in (5.5)
still applies.
Note that some implementations may use a "heartbeat" timer
that in fact yield a value between 2.5 seconds and 3
seconds. Accordingly, a lower bound of 2.5 seconds is also
acceptable, providing that the timer will never expire
faster than 2.5 seconds. Implementations using a heartbeat
timer with a granularity of G SHOULD not set the timer below
2.5 + G seconds.
(2.2) When the first RTT measurement R is made, the host MUST set
SRTT <- R
RTTVAR <- R/2
RTO <- SRTT + max (G, K*RTTVAR)
where K = 4.
(2.3) When a subsequent RTT measurement R' is made, a host MUST set
RTTVAR <- (1 - beta) * RTTVAR + beta * |SRTT - R'|
SRTT <- (1 - alpha) * SRTT + alpha * R'
The value of SRTT used in the update to RTTVAR is its value
before updating SRTT itself using the second assignment. That
is, updating RTTVAR and SRTT MUST be computed in the above
order.
The above SHOULD be computed using alpha=1/8 and beta=1/4 (as
suggested in [JK88]).
After the computation, a host MUST update
RTO <- SRTT + max (G, K*RTTVAR)
(2.4) Whenever RTO is computed, if it is less than 1 second then the
RTO SHOULD be rounded up to 1 second.
Traditionally, TCP implementations use coarse grain clocks to
measure the RTT and trigger the RTO, which imposes a large
minimum value on the RTO. Research suggests that a large
minimum RTO is needed to keep TCP conservative and avoid
spurious retransmissions [AP99]. Therefore, this
specification requires a large minimum RTO as a conservative
approach, while at the same time acknowledging that at some
future point, research may show that a smaller minimum RTO is
acceptable or superior.
(2.5) A maximum value MAY be placed on RTO provided it is at least 60
seconds.
3 Taking RTT Samples
TCP MUST use Karn's algorithm [KP87] for taking RTT samples. That
is, RTT samples MUST NOT be made using segments that were
retransmitted (and thus for which it is ambiguous whether the reply
was for the first instance of the packet or a later instance). The
only case when TCP can safely take RTT samples from retransmitted
segments is when the TCP timestamp option [JBB92] is employed, since
the timestamp option removes the ambiguity regarding which instance
of the data segment triggered the acknowledgment.
Traditionally, TCP implementations have taken one RTT measurement at
a time (typically once per RTT). However, when using the timestamp
option, each ACK can be used as an RTT sample. RFC 1323 [JBB92]
suggests that TCP connections utilizing large congestion windows
should take many RTT samples per window of data to avoid aliasing
effects in the estimated RTT. A TCP implementation MUST take at
least one RTT measurement per RTT (unless that is not possible per
Karn's algorithm).
For fairly modest congestion window sizes research suggests that
timing each segment does not lead to a better RTT estimator [AP99].
Additionally, when multiple samples are taken per RTT the alpha and
beta defined in section 2 may keep an inadequate RTT history. A
method for changing these constants is currently an open research
question.
4 Clock Granularity
There is no requirement for the clock granularity G used for
computing RTT measurements and the different state variables.
However, if the K*RTTVAR term in the RTO calculation equals zero,
the variance term MUST be rounded to G seconds (i.e., use the
equation given in step 2.3).
RTO <- SRTT + max (G, K*RTTVAR)
Experience has shown that finer clock granularities (<= 100 msec)
perform somewhat better than more coarse granularities.
Note that [Jac88] outlines several clever tricks that can be used to
obtain better precision from coarse granularity timers. These
changes are widely implemented in current TCP implementations.
5 Managing the RTO Timer
An implementation MUST manage the retransmission timer(s) in such a
way that a segment is never retransmitted too early, i.e. less than
one RTO after the previous transmission of that segment.
The following is the RECOMMENDED algorithm for managing the
retransmission timer:
(5.1) Every time a packet containing data is sent (including a
retransmission), if the timer is not running, start it running
so that it will expire after RTO seconds (for the current value
of RTO).
(5.2) When all outstanding data has been acknowledged, turn off the
retransmission timer.
(5.3) When an ACK is received that acknowledges new data, restart the
retransmission timer so that it will expire after RTO seconds
(for the current value of RTO).
When the retransmission timer expires, do the following:
(5.4) Retransmit the earliest segment that has not been acknowledged
by the TCP receiver.
(5.5) The host MUST set RTO <- RTO * 2 ("back off the timer"). The
maximum value discussed in (2.5) above may be used to provide an
upper bound to this doubling operation.
(5.6) Start the retransmission timer, such that it expires after RTO
seconds (for the value of RTO after the doubling operation
outlined in 5.5).
Note that after retransmitting, once a new RTT measurement is
obtained (which can only happen when new data has been sent and
acknowledged), the computations outlined in section 2 are performed,
including the computation of RTO, which may result in "collapsing"
RTO back down after it has been subject to exponential backoff
(rule 5.5).
Note that a TCP implementation MAY clear SRTT and RTTVAR after
backing off the timer multiple times as it is likely that the
current SRTT and RTTVAR are bogus in this situation. Once SRTT and
RTTVAR are cleared they should be initialized with the next RTT
sample taken per (2.2) rather than using (2.3).
6 Security Considerations
This document requires a TCP to wait for a given interval before
retransmitting an unacknowledged segment. An attacker could cause a
TCP sender to compute a large value of RTO by adding delay to a
timed packet's latency, or that of its acknowledgment. However,
the ability to add delay to a packet's latency often coincides with
the ability to cause the packet to be lost, so it is difficult to
see what an attacker might gain from such an attack that could cause
more damage than simply discarding some of the TCP connection's
packets.
The Internet to a considerable degree relies on the correct
implementation of the RTO algorithm (as well as those described in
RFC 2581) in order to preserve network stability and avoid
congestion collapse. An attacker could cause TCP endpoints to
respond more aggressively in the face of congestion by forging
acknowledgments for segments before the receiver has actually
received the data, thus lowering RTO to an unsafe value. But to do
so requires spoofing the acknowledgments correctly, which is
difficult unless the attacker can monitor traffic along the path
between the sender and the receiver. In addition, even if the
attacker can cause the sender's RTO to reach too small a value, it
appears the attacker cannot leverage this into much of an attack
(compared to the other damage they can do if they can spoof packets
belonging to the connection), since the sending TCP will still back
off its timer in the face of an incorrectly transmitted packet's
loss due to actual congestion.
Acknowledgments
The RTO algorithm described in this memo was originated by Van
Jacobson in [Jac88].
References
[AP99] Allman, M. and V. Paxson, "On Estimating End-to-End Network
Path Properties", SIGCOMM 99.
[APS99] Allman, M., Paxson V. and W. Stevens, "TCP Congestion
Control", RFC 2581, April 1999.
[Bra89] Braden, R., "Requirements for Internet Hosts --
Communication Layers", STD 3, RFC 1122, October 1989.
[Bra97] Bradner, S., "Key words for use in RFCs to Indicate
Requirement Levels", BCP 14, RFC 2119, March 1997.
[Jac88] Jacobson, V., "Congestion Avoidance and Control", Computer
Communication Review, vol. 18, no. 4, pp. 314-329, Aug. 1988.
[JK88] Jacobson, V. and M. Karels, "Congestion Avoidance and
Control", ftp://ftp.ee.lbl.gov/papers/congavoid.ps.Z.
[KP87] Karn, P. and C. Partridge, "Improving Round-Trip Time
Estimates in Reliable Transport Protocols", SIGCOMM 87.
[Pos81] Postel, J., "Transmission Control Protocol", STD 7, RFC 793,
September 1981.
Author's Addresses
Vern Paxson
ACIRI / ICSI
1947 Center Street
Suite 600
Berkeley, CA 94704-1198
Phone: 510-666-2882
Fax: 510-643-7684
EMail: vern@aciri.org
http://www.aciri.org/vern/
Mark Allman
NASA Glenn Research Center/BBN Technologies
Lewis Field
21000 Brookpark Rd. MS 54-2
Cleveland, OH 44135
Phone: 216-433-6586
Fax: 216-433-8705
EMail: mallman@grc.nasa.gov
http://roland.grc.nasa.gov/~mallman
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AI ELI5
On this page
§1Why TCP Needs a Retransmission Timer
TCP uses a retransmission timer to make sure data is delivered even when the remote receiver sends no feedback. The duration of this timer is the RTO (retransmission timeout). RFC 1122 already said the RTO should be calculated as outlined in Jacobson's 1988 paper.
This document codifies that algorithm and upgrades the requirement to support it from a SHOULD to a MUST. It does not change how TCP begins sending after the timer expires; that is outlined in RFC 2581.
A sender may be more conservative than the algorithm allows, but a TCP MUST NOT be more aggressive.
TCP sends little packages and waits for a thank-you note. If no note comes, it sends the package again, so it needs a rule for how long to wait.
§2The Basic Algorithm
The sender keeps two state variables: SRTT (smoothed round-trip time) and RTTVAR (round-trip time variation). It also assumes a clock granularity of G seconds. The rules are:
- Before any round-trip time (RTT) measurement exists, the sender SHOULD set RTO to 3 seconds.
- On the first measurement
R, MUST setSRTT = R,RTTVAR = R/2andRTO = SRTT + max(G, K*RTTVAR), whereK = 4. - On each later measurement
R', MUST updateRTTVAR = (1 - beta) * RTTVAR + beta * |SRTT - R'|first, thenSRTT = (1 - alpha) * SRTT + alpha * R', and then recompute RTO. The constants SHOULD be alpha = 1/8 and beta = 1/4. - If the computed RTO is under 1 second, it SHOULD be rounded up to 1 second.
- A maximum MAY be placed on RTO, provided it is at least 60 seconds.
The order matters: the old SRTT is used when updating RTTVAR. The RFC explains that a large minimum RTO keeps TCP conservative and avoids spurious retransmissions, while allowing that future research might show a smaller minimum is acceptable.
Implementations with a heartbeat timer may yield a value between 2.5 and 3 seconds for the initial RTO. A lower bound of 2.5 seconds is acceptable if the timer never expires sooner, and a heartbeat timer of granularity G SHOULD NOT go below 2.5 + G seconds.
TCP keeps a running average of how long replies usually take, plus a note of how much that time wobbles. It waits the average plus a few wobbles, and never less than one second.
- SRTTSmoothed round-trip time, updated with alpha = 1/8.
- RTTVARRound-trip time variation, updated with beta = 1/4.
- RTOSRTT + max(G, K*RTTVAR) with K = 4; at least 1 second (SHOULD).
- Initial RTO3 seconds until the first RTT measurement.
§3Taking RTT Samples and Clock Granularity
TCP MUST use Karn's algorithm for taking RTT samples: samples MUST NOT come from segments that were retransmitted, because it is ambiguous whether the reply answered the first copy or a later one. The only safe exception is when the TCP timestamp option is in use, since it removes that ambiguity.
A TCP implementation MUST take at least one RTT measurement per RTT, unless Karn's algorithm prevents it. For modest congestion windows, research suggests timing every segment does not give a better estimator. How to adjust alpha and beta when many samples are taken per RTT is described as an open research question.
There is no requirement on the clock granularity G. If the K*RTTVAR term equals zero, though, the variance term MUST be rounded to G seconds. Experience shows that finer clocks (100 msec or less) perform somewhat better than coarser ones.
TCP only times a reply when it is sure which message the reply belongs to, like not guessing which of two identical letters a friend is answering. A finer clock helps it time things more exactly.
§4Managing the RTO Timer
An implementation MUST manage its timers so a segment is never retransmitted less than one RTO after its previous transmission. The RECOMMENDED procedure is:
- When a packet containing data is sent (including a retransmission) and the timer is not running, start it to expire after RTO seconds.
- When all outstanding data has been acknowledged, turn the timer off.
- When an ACK acknowledges new data, restart the timer for RTO seconds.
When the timer expires, the sender retransmits the earliest unacknowledged segment, then MUST set RTO = RTO * 2 (exponential backoff), then restarts the timer with the doubled value. The maximum from the basic algorithm may cap the doubling.
Once a new RTT measurement is obtained after a retransmission, the normal computation runs again and can collapse the RTO back down. A TCP MAY also clear SRTT and RTTVAR after backing off several times, since they are likely bogus by then; they are then initialized from the next sample as for a first measurement.
Start a stopwatch when you send something. If the thank-you arrives, reset it. If time runs out, send again and wait twice as long next time.
§5Security Considerations
An attacker who can add delay to a packet or its acknowledgment could make a sender compute a large RTO. The RFC notes that the ability to add delay usually comes with the ability to drop the packet, so it is hard to see what extra damage this would do.
The Internet relies on correct RTO behavior, together with the algorithms in RFC 2581, to preserve stability and avoid congestion collapse. An attacker could try to push RTO to an unsafe low value by forging acknowledgments, but this requires correct spoofing, which is difficult unless the attacker can monitor traffic on the path. Even then, the sender still backs off when a packet is lost to real congestion, so the RFC sees little to gain from it.
A trickster could try to make TCP wait too long or too little. But anyone who can mess with the mail in that way can already do worse things, so this adds little danger.