RFC 2988 PROPOSED STANDARD TRANSPORT · TIMERS · RELIABILITY

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.

On rfc-editor.org ↗
PUBLISHED
November 2000
AUTHORS
V. Paxson, M. Allman
STREAM
IETF
OBSOLETED BY
RFC 6298 — TCP RTO
INITIAL RTO
3 seconds
MINIMUM RTO (SHOULD)
1 second
MAXIMUM RTO (IF ANY)
at least 60 seconds
K
4
ALPHA AND BETA
1/8 and 1/4
BACKOFF ON EXPIRY
RTO doubled
DOI
10.17487/RFC2988
ERRATA
Reported errata

RFC text

Contents
Original text · RFC Editor View on rfc-editor.org ↗
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

Full Copyright Statement

   Copyright (C) The Internet Society (2000).  All Rights Reserved.

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Acknowledgement

   Funding for the RFC Editor function is currently provided by the
   Internet Society.

AI ELI5

On this page
PROBLEMTCP needs a retransmission timer to ensure delivery when no feedback arrives from the receiver, and the length of that timer (the RTO) has to be set sensibly.
MECHANISMThe sender tracks a smoothed round-trip time (SRTT) and its variation (RTTVAR), sets RTO to SRTT plus four times RTTVAR, never below 1 second if it follows the SHOULD, and doubles the RTO each time the timer expires.
STATUS TODAYRFC 2988 turned the algorithm from a SHOULD in RFC 1122 into a MUST. It has since been obsoleted by RFC 6298.

§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 set SRTT = R, RTTVAR = R/2 and RTO = SRTT + max(G, K*RTTVAR), where K = 4.
  • On each later measurement R', MUST update RTTVAR = (1 - beta) * RTTVAR + beta * |SRTT - R'| first, then SRTT = (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.

THE STATE KEPT BY THE SENDER (RFC 2988)
  • 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.

§6References