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Versions: (RFC 2988) 00 01 02 RFC 6298
Internet Engineering Task Force V. Paxson
INTERNET DRAFT ICSI/UC Berkeley
File: draft-paxson-tcpm-rfc2988bis-02.txt M. Allman
Intended status: Proposed Standard ICSI
J. Chu
Google
M. Sargent
CWRU
March 14, 2011
Computing TCP's Retransmission Timer
Status of this Memo
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Abstract
This document defines the standard algorithm that Transmission
Control Protocol (TCP) senders are required to use to compute and
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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 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.
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 <- 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
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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
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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
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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.
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 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
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loss due to actual congestion.
7 IANA Considerations
None
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.
Normative References
[APB09] Allman, M., Paxson V. and E. Blanton, "TCP Congestion
Control", RFC 5681, September 2009.
[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.
[JBB92] Jacobson, V., R. Braden, D. Borman, "TCP Extensions for High
Performance", RFC 1323, May 1992.
[Pos81] Postel, J., "Transmission Control Protocol", STD 7, RFC 793,
September 1981.
Non-Normative 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.
[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.
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[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.
Author's Addresses
Vern Paxson
ICSI
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
Appendix A
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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 criteria 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
[RFC5681] 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 three 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.
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.
--------------------------------------------------------
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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).
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.
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