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RFC 2988

Computing TCP's Retransmission Timer

Pages: 8
Obsoleted by:  6298

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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].
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   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'
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         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).
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   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).
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   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
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   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.
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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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Full Copyright Statement

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Acknowledgement

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