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IETF RFC 6298
Computing TCP's Retransmission Timer
Last modified on Thursday, June 23rd, 2011
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Internet Engineering Task Force (IETF) V. Paxson
Request for Comments: 6298 ICSI/UC Berkeley
Obsoletes: 2988 M. Allman
Updates: 1122 ICSI
Category: Standards Track J. Chu
ISSN: 2070-1721 Google
M. Sargent
CWRU
June 2011
Computing TCP's Retransmission Timer
 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. This document
obsoletes RFC 2988.
Status of This Memo
This is an Internet Standards Track document.
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
Internet Standards is available in Section 2 of RFC 5741.
Information about the current status of this document, any errata,
and how to provide feedback on it may be obtained at
http://www.rfc-editor.org/info/RFC 6298.
Paxson, et al. Standards Track PAGE 1 
RFC 6298 Computing TCP's Retransmission Timer June 2011
Copyright Notice
Copyright (c) 2011 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
(http://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.
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 5681 [APB09] 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 5681
[APB09].
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. This document obsoletes RFC 2988 [PA00].
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.
Paxson, et al. Standards Track PAGE 2
RFC 6298 Computing TCP's Retransmission Timer June 2011
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 <- 1 second, though the "backing off" on repeated
retransmission discussed in (5.5) still applies.
Note that the previous version of this document used an initial
RTO of 3 seconds [PA00]. A TCP implementation MAY still use
this value (or any other value > 1 second). This change in the
lower bound on the initial RTO is discussed in further detail
in Appendix A.
(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
Paxson, et al. Standards Track PAGE 3
RFC 6298 Computing TCP's Retransmission Timer June 2011
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 coarser granularities.
Paxson, et al. Standards Track PAGE 4
RFC 6298 Computing TCP's Retransmission Timer June 2011
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).
(5.7) If the timer expires awaiting the ACK of a SYN segment and the
TCP implementation is using an RTO less than 3 seconds, the RTO
MUST be re-initialized to 3 seconds when data transmission
begins (i.e., after the three-way handshake completes).
This represents a change from the previous version of this
document [PA00] and is discussed in Appendix A.
Paxson, et al. Standards Track PAGE 5
RFC 6298 Computing TCP's Retransmission Timer June 2011
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 back off (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 5681) 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.
The security considerations in RFC 5681 [APB09] are also applicable
to this document.
Paxson, et al. Standards Track PAGE 6
RFC 6298 Computing TCP's Retransmission Timer June 2011
7. Changes from RFC 2988
This document reduces the initial RTO from the previous 3 seconds
[PA00] to 1 second, unless the SYN or the ACK of the SYN is lost, in
which case the default RTO is reverted to 3 seconds before data
transmission begins.
8. Acknowledgments
The RTO algorithm described in this memo was originated by Van
Jacobson in [Jac88].
Much of the data that motivated changing the initial RTO from 3
seconds to 1 second came from Robert Love, Andre Broido, and Mike
Belshe.
9. References
9.1. Normative References
[APB09] Allman, M., Paxson, V., and E. Blanton, "TCP Congestion
Control", RFC 5681, September 2009.
[Bra89] Braden, R., Ed., "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.
[JBB92] Jacobson, V., Braden, R., and D. Borman, "TCP Extensions for
High Performance", RFC 1323, May 1992.
[Pos81] Postel, J., "Transmission Control Protocol", STD 7, RFC 793,
September 1981.
9.2. Informative References
[AP99] Allman, M. and V. Paxson, "On Estimating End-to-End Network
Path Properties", SIGCOMM 99.
[Chu09] Chu, J., "Tuning TCP Parameters for the 21st Century",
http://www.ietf.org/proceedings/75/slides/tcpm-1.pdf, July
2009.
[SLS09] Schulman, A., Levin, D., and Spring, N., "CRAWDAD data set
umd/sigcomm2008 (v. 2009-03-02)",
http://crawdad.cs.dartmouth.edu/umd/sigcomm2008, March, 2009.
Paxson, et al. Standards Track PAGE 7
RFC 6298 Computing TCP's Retransmission Timer June 2011
[HKA04] Henderson, T., Kotz, D., and Abyzov, I., "CRAWDAD trace
dartmouth/campus/tcpdump/fall03 (v. 2004-11-09)",
http://crawdad.cs.dartmouth.edu/dartmouth/campus/
tcpdump/fall03, November 2004.
[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.
[PA00] Paxson, V. and M. Allman, "Computing TCP's Retransmission
Timer", RFC 2988, November 2000.
Paxson, et al. Standards Track PAGE 8
RFC 6298 Computing TCP's Retransmission Timer June 2011
Appendix A. Rationale for Lowering the Initial RTO
Choosing a reasonable initial RTO requires balancing two competing
considerations:
1. The initial RTO should be sufficiently large to cover most of the
end-to-end paths to avoid spurious retransmissions and their
associated negative performance impact.
2. The initial RTO should be small enough to ensure a timely recovery
from packet loss occurring before an RTT sample is taken.
Traditionally, TCP has used 3 seconds as the initial RTO [Bra89]
[PA00]. This document calls for lowering this value to 1 second
using the following rationale:
- Modern networks are simply faster than the state-of-the-art was at
the time the initial RTO of 3 seconds was defined.
- Studies have found that the round-trip times of more than 97.5% of
the connections observed in a large scale analysis were less than 1
second [Chu09], suggesting that 1 second meets criterion 1 above.
- In addition, the studies observed retransmission rates within the
three-way handshake of roughly 2%. This shows that reducing the
initial RTO has benefit to a non-negligible set of connections.
- However, roughly 2.5% of the connections studied in [Chu09] have an
RTT longer than 1 second. For those connections, a 1 second
initial RTO guarantees a retransmission during connection
establishment (needed or not).
When this happens, this document calls for reverting to an initial
RTO of 3 seconds for the data transmission phase. Therefore, the
implications of the spurious retransmission are modest: (1) an
extra SYN is transmitted into the network, and (2) according to RFC
5681 [APB09] the initial congestion window will be limited to 1
segment. While (2) clearly puts such connections at a
disadvantage, this document at least resets the RTO such that the
connection will not continually run into problems with a short
timeout. (Of course, if the RTT is more than 3 seconds, the
connection will still encounter difficulties. But that is not a
new issue for TCP.)
In addition, we note that when using timestamps, TCP will be able
to take an RTT sample even in the presence of a spurious
retransmission, facilitating convergence to a correct RTT estimate
when the RTT exceeds 1 second.
Paxson, et al. Standards Track PAGE 9
RFC 6298 Computing TCP's Retransmission Timer June 2011
As an additional check on the results presented in [Chu09], we
analyzed packet traces of client behavior collected at four different
vantage points at different times, as follows:
Name Dates Pkts. Cnns. Clnts. Servs.
--------------------------------------------------------
LBL-1 Oct/05--Mar/06 292M 242K 228 74K
LBL-2 Nov/09--Feb/10 1.1B 1.2M 1047 38K
ICSI-1 Sep/11--18/07 137M 2.1M 193 486K
ICSI-2 Sep/11--18/08 163M 1.9M 177 277K
ICSI-3 Sep/14--21/09 334M 3.1M 170 253K
ICSI-4 Sep/11--18/10 298M 5M 183 189K
Dartmouth Jan/4--21/04 1B 4M 3782 132K
SIGCOMM Aug/17--21/08 11.6M 133K 152 29K
The "LBL" data was taken at the Lawrence Berkeley National
Laboratory, the "ICSI" data from the International Computer Science
Institute, the "SIGCOMM" data from the wireless network that served
the attendees of SIGCOMM 2008, and the "Dartmouth" data was collected
from Dartmouth College's wireless network. The latter two datasets
are available from the CRAWDAD data repository [HKA04] [SLS09]. The
table lists the dates of the data collections, the number of packets
collected, the number of TCP connections observed, the number of
local clients monitored, and the number of remote servers contacted.
We consider only connections initiated near the tracing vantage
point.
Analysis of these datasets finds the prevalence of retransmitted SYNs
to be between 0.03% (ICSI-4) to roughly 2% (LBL-1 and Dartmouth).
We then analyzed the data to determine the number of additional and
spurious retransmissions that would have been incurred if the initial
RTO was assumed to be 1 second. In most of the datasets, the
proportion of connections with spurious retransmits was less than
0.1%. However, in the Dartmouth dataset, approximately 1.1% of the
connections would have sent a spurious retransmit with a lower
initial RTO. We attribute this to the fact that the monitored
network is wireless and therefore susceptible to additional delays
from RF effects.
Finally, there are obviously performance benefits from retransmitting
lost SYNs with a reduced initial RTO. Across our datasets, the
percentage of connections that retransmitted a SYN and would realize
at least a 10% performance improvement by using the smaller initial
RTO specified in this document ranges from 43% (LBL-1) to 87%
(ICSI-4). The percentage of connections that would realize at least
a 50% performance improvement ranges from 17% (ICSI-1 and SIGCOMM) to
73% (ICSI-4).
Paxson, et al. Standards Track PAGE 10
RFC 6298 Computing TCP's Retransmission Timer June 2011
From the data to which we have access, we conclude that the lower
initial RTO is likely to be beneficial to many connections, and
harmful to relatively few.
Authors' Addresses
Vern Paxson
ICSI/UC Berkeley
1947 Center Street
Suite 600
Berkeley, CA 94704-1198
Phone: 510-666-2882
EMail: vern@icir.org
http://www.icir.org/vern/
Mark Allman
ICSI
1947 Center Street
Suite 600
Berkeley, CA 94704-1198
Phone: 440-235-1792
EMail: mallman@icir.org
http://www.icir.org/mallman/
H.K. Jerry Chu
Google, Inc.
1600 Amphitheatre Parkway
Mountain View, CA 94043
Phone: 650-253-3010
EMail: hkchu@google.com
Matt Sargent
Case Western Reserve University
Olin Building
10900 Euclid Avenue
Room 505
Cleveland, OH 44106
Phone: 440-223-5932
EMail: mts71@case.edu
Paxson, et al. Standards Track PAGE 11
Computing TCP's Retransmission Timer
RFC TOTAL SIZE: 22454 bytes
PUBLICATION DATE: Thursday, June 23rd, 2011
LEGAL RIGHTS: The IETF Trust (see BCP 78)
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