This is a purely informative rendering of an RFC that includes verified errata. This rendering may not be used as a reference.
The following 'Verified' errata have been incorporated in this document:
EID 1308
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.
[JBB92] Jacobson, V., Braden, R., Borman, D., "TCP Extensions for High
Performance", RFC 1323, May 1992.
[JK88] Jacobson, V. and M. Karels, "Congestion Avoidance and
Control", ftp://ftp.ee.lbl.gov/papers/congavoid.ps.Z.
EID 1308 (Verified) is as follows:Section: References
Original Text:
[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.
Corrected Text:
[Jac88] Jacobson, V., "Congestion Avoidance and Control", Computer
Communication Review, vol. 18, no. 4, pp. 314-329, Aug. 1988.
[JBB92] Jacobson, V., Braden, R., Borman, D., "TCP Extensions for High
Performance", RFC 1323, May 1992.
[JK88] Jacobson, V. and M. Karels, "Congestion Avoidance and
Control", ftp://ftp.ee.lbl.gov/papers/congavoid.ps.Z.
Notes:
Reference [JBB92] is mentioned two times in section 3, but it is not included in the reference section.
[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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