Skip to main content

Computing TCP's Retransmission Timer

The information below is for an old version of the document that is already published as an RFC.
Document Type
This is an older version of an Internet-Draft that was ultimately published as RFC 6298.
Authors Matt Sargent , Jerry Chu , Dr. Vern Paxson , Mark Allman
Last updated 2015-10-14 (Latest revision 2011-03-14)
RFC stream Internet Engineering Task Force (IETF)
Intended RFC status Proposed Standard
Additional resources Mailing list discussion
Stream WG state WG Document
Document shepherd (None)
IESG IESG state Became RFC 6298 (Proposed Standard)
Action Holders
Consensus boilerplate Unknown
Telechat date (None)
Responsible AD Wesley Eddy
IESG note
Send notices to (None)
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
                                                              M. Sargent
                                                          March 14, 2011

                  Computing TCP's Retransmission Timer

Status of this Memo

    This Internet-Draft is submitted to IETF in full conformance with
    the provisions of BCP 78 and BCP 79.

    Internet-Drafts are working documents of the Internet Engineering
    Task Force (IETF), its areas, and its working groups.  Note that
    other groups may also distribute working documents as Internet-

    Internet-Drafts are draft documents valid for a maximum of six
    months and may be updated, replaced, or obsoleted by other documents
    at any time.  It is inappropriate to use Internet-Drafts as
    reference material or to cite them other than as "work in progress."

    The list of current Internet-Drafts can be accessed at

    The list of Internet-Draft Shadow Directories can be accessed at

    This Internet-Draft will expire on September 14, 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
    ( 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.


   This document defines the standard algorithm that Transmission
   Control Protocol (TCP) senders are required to use to compute and

Expires: September 14, 2011                                     [Page 1]
draft-paxson-tcpm-rfc2988bis-02.txt                           March 2011

   manage their retransmission timer.  It expands on the discussion in
   section 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
   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
   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",
   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

   The rules governing the computation of SRTT, RTTVAR, and RTO are as

   (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

Expires: September 14, 2011                                     [Page 2]
draft-paxson-tcpm-rfc2988bis-02.txt                           March 2011

            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

         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

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

Expires: September 14, 2011                                     [Page 3]
draft-paxson-tcpm-rfc2988bis-02.txt                           March 2011

   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

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

Expires: September 14, 2011                                     [Page 4]
draft-paxson-tcpm-rfc2988bis-02.txt                           March 2011

         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

   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

Expires: September 14, 2011                                     [Page 5]
draft-paxson-tcpm-rfc2988bis-02.txt                           March 2011

   loss due to actual congestion.

7  IANA Considerations    


   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

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", 
 , July

   [SLS09] Schulman, A., Levin, D., and Spring, N., "CRAWDAD data set
           umd/sigcomm2008 (v. 2009-03-02)", 
 , March,

   [HKA04] Henderson, T., Kotz, D., and Abyzov, I., "CRAWDAD trace
           dartmouth/campus/tcpdump/fall03 (v. 2004-11-09)",
           November 2004.

   [Jac88] Jacobson, V., "Congestion Avoidance and Control", Computer
           Communication Review, vol. 18, no. 4, pp. 314-329, Aug.  1988.

Expires: September 14, 2011                                     [Page 6]
draft-paxson-tcpm-rfc2988bis-02.txt                           March 2011

   [JK88]  Jacobson, V. and M. Karels, "Congestion Avoidance and

   [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
   1947 Center Street
   Suite 600
   Berkeley, CA 94704-1198

   Phone: 510-666-2882

   Mark Allman
   1947 Center Street
   Suite 600
   Berkeley, CA 94704-1198

   Phone: 440-235-1792

   H.K. Jerry Chu
   Google, Inc.
   1600 Amphitheatre Parkway
   Mountain View, CA 94043
   Phone: 650-253-3010

   Matt Sargent
   Case Western Reserve University Olin Building
   10900 Euclid Avenue
   Room 505
   Cleveland, OH 44106

   Phone: 440-223-5932

Appendix A

Expires: September 14, 2011                                     [Page 7]
draft-paxson-tcpm-rfc2988bis-02.txt                           March 2011

    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 

    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.

Expires: September 14, 2011                                     [Page 8]
draft-paxson-tcpm-rfc2988bis-02.txt                           March 2011

      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

    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.

Expires: September 14, 2011                                     [Page 9]