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TCP and SCTP RTO Restart
draft-ietf-tcpm-rtorestart-01

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This is an older version of an Internet-Draft that was ultimately published as RFC 7765.
Authors Per Hurtig , Anna Brunstrom , Andreas Petlund , Michael Welzl
Last updated 2013-09-17
Replaces draft-hurtig-tcpm-rtorestart
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draft-ietf-tcpm-rtorestart-01
TCP Maintenance and Minor Extensions (tcpm)                    P. Hurtig
Internet-Draft                                              A. Brunstrom
Intended status: Experimental                        Karlstad University
Expires: March 21, 2014                                       A. Petlund
                                           Simula Research Laboratory AS
                                                                M. Welzl
                                                      University of Oslo
                                                      September 17, 2013

                        TCP and SCTP RTO Restart
                     draft-ietf-tcpm-rtorestart-01

Abstract

   This document describes a modified algorithm for managing the TCP and
   SCTP retransmission timers that provides faster loss recovery when
   there is a small amount of outstanding data for a connection.  The
   modification allows the transport to restart its retransmission timer
   more aggressively in situations where fast retransmit cannot be used.
   This enables faster loss detection and recovery for connections that
   are short-lived or application-limited.

Status of This Memo

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

   Internet-Drafts are working documents of the Internet Engineering
   Task Force (IETF).  Note that other groups may also distribute
   working documents as Internet-Drafts.  The list of current Internet-
   Drafts is at http://datatracker.ietf.org/drafts/current/.

   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."

   This Internet-Draft will expire on March 21, 2014.

Copyright Notice

   Copyright (c) 2013 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
   (http://trustee.ietf.org/license-info) in effect on the date of

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   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.

1.  Introduction

   TCP uses two mechanisms to detect segment loss.  First, if a segment
   is not acknowledged within a certain amount of time, a retransmission
   timeout (RTO) occurs, and the segment is retransmitted [RFC6298].
   While the RTO is based on measured round-trip times (RTTs) between
   the sender and receiver, it also has a conservative lower bound of 1
   second to ensure that delayed segments are not mistaken as lost.
   Second, when a sender receives duplicate acknowledgments, the fast
   retransmit algorithm infers segment loss and triggers a
   retransmission [RFC5681].  Duplicate acknowledgments are generated by
   a receiver when out-of-order segments arrive.  As both segment loss
   and segment reordering cause out-of-order arrival, fast retransmit
   waits for three duplicate acknowledgments before considering the
   segment as lost.  In some situations, however, the number of
   outstanding segments is not enough to trigger three duplicate
   acknowledgments, and the sender must rely on lengthy RTOs for loss
   recovery.

   The number of outstanding segments can be small for several reasons:

   (1)  The connection is limited by the congestion control when the
      path has a low total capacity (bandwidth-delay product) or the
      connection's share of the capacity is small.  It is also limited
      by the congestion control in the first few RTTs of a connection or
      after an RTO when the available capacity is probed using slow-
      start.

   (2)  The connection is limited by the receiver's available buffer
      space.

   (3)  The connection is limited by the application if the available
      capacity of the path is not fully utilized (e.g. interactive
      applications), or at the end of a transfer.

   While the reasons listed above are valid for any flow, the third
   reason is common for applications that transmit short flows, or use a
   low transmission rate.  Typical examples of applications that produce
   short flows are web servers.  [RJ10] shows that 70% of all web
   objects, found at the top 500 sites, are too small for fast
   retransmit to work.  [BPS98] shows that about 56% of all

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   retransmissions sent by a busy web server are sent after RTO expiry.
   While the experiments were not conducted using SACK [RFC2018], only
   4% of the RTO-based retransmissions could have been avoided.
   Applications have a low transmission rate when data is sent in
   response to actions, or as a reaction to real life events.  Typical
   examples of such applications are stock trading systems, remote
   computer operations and online games.  What is special about this
   class of applications is that they are time-dependant, and extra
   latency can reduce the application service level [P09].  Although
   such applications may represent a small amount of data sent on the
   network, a considerable number of flows have such properties and the
   importance of low latency is high.

   The RTO restart approach outlined in this document makes the RTO
   slightly more aggressive when the number of outstanding segments is
   small, in an attempt to enable faster loss recovery for all segments
   while being robust to reordering.  While it still conforms to the
   requirement in [RFC6298] that segments must not be retransmitted
   earlier than RTO seconds after their original transmission, it could
   increase the risk of spurious timeout.  Spurious timeouts typically
   degrade the performance of flows with multiple bursts of data, as a
   burst following a spurious timeout might not fit within the reduced
   congestion window (cwnd).

   While this document focuses on TCP, the described changes are also
   valid for the Stream Control Transmission Protocol (SCTP) [RFC4960]
   which has similar loss recovery and congestion control algorithms.

1.1.  Requirements Language

   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 RFC 2119 [RFC2119].

2.  RTO Restart Overview

   The RTO management algorithm described in [RFC6298] recommends that
   the retransmission timer is restarted when an acknowledgment (ACK)
   that acknowledges new data is received and there is still outstanding
   data.  The restart is conducted to guarantee that unacknowledged
   segments will be retransmitted after approximately RTO seconds.
   However, by restarting the timer on each incoming acknowledgment,
   retransmissions are not typically triggered RTO seconds after their
   previous transmission but rather RTO seconds after the last ACK
   arrived.  The duration of this extra delay depends on several factors
   but is in most cases approximately one RTT.  Hence, in most
   situations the time before a retransmission is triggered is equal to
   "RTO + RTT".

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   The extra delay can be significant, especially for applications that
   use a lower RTOmin than the standard of 1 second and/or in
   environments with high RTTs, e.g. mobile networks.  The restart
   approach is illustrated in Figure 1 where a TCP sender transmits
   three segments to a receiver.  The arrival of the first and second
   segment triggers a delayed ACK [RFC1122], which restarts the RTO
   timer at the sender.  RTO restart is performed approximately one RTT
   after the transmission of the third segment.  Thus, if the third
   segment is lost, as indicated in Figure 1, the effective loss
   detection time is "RTO + RTT" seconds.  In some situations, the
   effective loss detection time becomes even longer.  Consider a
   scenario where only two segments are outstanding.  If the second
   segment is lost, the time to expire the delayed ACK timer will also
   be included in the effective loss detection time.

            Sender                               Receiver
                          ...
            DATA [SEG 1] ----------------------> (ack delayed)
            DATA [SEG 2] ----------------------> (send ack)
            DATA [SEG 3] ----X         /-------- ACK
            (restart RTO)  <----------/
                          ...
            (RTO expiry)
            DATA [SEG 3] ---------------------->

                       Figure 1: RTO restart example

   During normal TCP bulk transfer the current RTO restart approach is
   not a problem.  Actually, as long as enough segments arrive at a
   receiver to enable fast retransmit, RTO-based loss recovery should be
   avoided.  RTOs should only be used as a last resort, as they
   drastically lower the congestion window compared to fast retransmit.
   The current approach can therefore be beneficial -- it is described
   in [EL04] to act as a "safety margin" that compensates for some of
   the problems that the authors have identified with the standard RTO
   calculation.  Notably, the authors of [EL04] also state that "this
   safety margin does not exist for highly interactive applications
   where often only a single packet is in flight."

   Although fast retransmit is preferrable there are situations where
   timeouts are appropriate, or the only choice.  For example, if the
   network is severely congested and no segments arrive, RTO-based
   recovery should be used.  In this situation, the time to recover from
   the loss(es) will not be the performance bottleneck.  Furthermore,
   for connections that do not utilize enough capacity to enable fast

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   retransmit, RTO is the only choice.  The time needed for loss
   detection in such scenarios can become a serious performance
   bottleneck.

3.  RTO Restart Algorithm

   To enable faster loss recovery for connections that are unable to use
   fast retransmit, an alternative RTO restart can be used.  By
   resetting the timer to "RTO - T_earliest", where T_earliest is the
   time elapsed since the earliest outstanding segment was transmitted,
   retransmissions will always occur after exactly RTO seconds.  This
   approach makes the RTO more aggressive than the standardized approach
   in [RFC6298] but still conforms to the requirement in [RFC6298] that
   segments must not be retransmitted earlier than RTO seconds after
   their original transmission.

   This document specifies a sender-only modification to TCP and SCTP
   which updates step 5.3 in Section 5 of [RFC6298] (and a similar
   update in Section 6.3.2 of [RFC4960] for SCTP):

      When an ACK is received that acknowledges new data:

      (1)  Set T_earliest = 0.

      (2)  If the following two conditions hold:

            (a)  The number of outstanding segments is less than four.

            (b)  There is no unsent data ready for transmission.

            set T_earliest to the time elapsed since the earliest
            outstanding segment was sent.

      (3)  Restart the retransmission timer so that it will expire after
            "RTO - T_earliest" seconds (for the current value of RTO).

   The update requires TCP implementations to track the time elapsed
   since the transmission of the earliest outstanding segment
   (T_earliest).  As the alternative restart is used only when the
   number of outstanding segments is less than four only four segments
   need to be tracked.  Furthermore, some implementations of TCP (e.g.
   Linux TCP) already track the transmission times of all segments.

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4.  Discussion

   In this section, we discuss the applicability and a number of issues
   surrounding the modified RTO restart.

4.1.  Applicability

   The currently standardized algorithm has been shown to add at least
   one RTT to the loss recovery process in TCP [LS00] and SCTP
   [HB08][PBP09].  For applications that have strict timing requirements
   (e.g. interactive web and gaming) rather than throughput
   requirements, the modified restart approach could be important
   because the RTT and also the delayed ACK timer of receivers are often
   large components of the effective loss recovery time.  Measurements
   in [HB08] have shown that the total transfer time of a lost segment
   (including the original transmission time and the loss recovery time)
   can be reduced by 35% using the suggested approach.  These results
   match those presented in [PGH06][PBP09], where the modified restart
   approach is shown to significantly reduce retransmission latency.

4.2.  Spurious Timeouts

   This document describes a modified RTO restart behavior that, in some
   situations, reduces the loss detection time and thereby increases the
   risk of spurious timeouts.  In theory, the retransmission timer has a
   lower bound of 1 second [RFC6298], which limits the risk of having
   spurious timeouts.  However, in practice most implementations use a
   significantly lower value.  Initial measurements, conducted by the
   authors, show slight increases in the number of spurious timeouts
   when such lower values are used.  However, further experiments, in
   different environments and with different types of traffic, are
   encouraged to quantify such increases more reliably.

   Does a slightly increased risk matter?  Generally, spurious timeouts
   have a negative effect on TCP/SCTP performance as the congestion
   window is reduced to one segment [RFC5681], limiting an application's
   ability to transmit large amounts of data instantaneously.  However,
   with respect to RTO restart spurious timeouts are only a problem for
   applications transmitting multiple bursts of data within a single
   flow.  Other types of flows, e.g. long-lived bulk flows, are not
   affected as the algorithm is only applied when the amount of
   outstanding segments is less than four and no previously unsent data
   is available.  Furthermore, short-lived and application-limited flows
   are typically not affected as they are too short to experience the
   effect of congestion control or have a transmission rate that is
   quickly attainable.

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   While a slight increase in spurious timeouts has been observed using
   the modified RTO restart approach, it is not clear whether the
   effects of this increase mandate any future algorithmic changes or
   not -- especially since most modern operating systems already include
   mechanisms to detect [RFC3522][RFC3708][RFC5682] and resolve
   [RFC4015] possible problems with spurious retransmissions.  Further
   experimentation is needed to determine this and thereby move this
   specification from experimental to proposed standard.

5.  Related Work

   There are several proposals that address the problem of not having
   enough ACKs for loss recovery.  In what follows, we explain why the
   mechanism described here is complementary to these approaches:

   The limited transmit mechanism [RFC3042] allows a TCP sender to
   transmit a previously unsent segment for each of the first two
   duplicate acknowledgments.  By transmitting new segments, the sender
   attempts to generate additional duplicate acknowledgments to enable
   fast retransmit.  However, limited transmit does not help if no
   previously unsent data is ready for transmission or if the receiver
   has no buffer space.  [RFC5827] specifies an early retransmit
   algorithm to enable fast loss recovery in such situations.  By
   dynamically lowering the number of duplicate acknowledgments needed
   for fast retransmit (dupthresh), based on the number of outstanding
   segments, a smaller number of duplicate acknowledgments are needed to
   trigger a retransmission.  In some situations, however, the algorithm
   is of no use or might not work properly.  First, if a single segment
   is outstanding, and lost, it is impossible to use early retransmit.
   Second, if ACKs are lost, the early retransmit cannot help.  Third,
   if the network path reorders segments, the algorithm might cause more
   unnecessary retransmissions than fast retransmit.

   Following the fast retransmit mechanism standardized in [RFC5681]
   this draft assumes a value of 3 for dupthresh.  However, by
   considering a dynamic value for dupthresh a tighter integration with
   early retransmit (or other experimental algorithms) could also be
   possible.

   Tail Loss Probe [TLP] is a proposal to send up to two "probe
   segments" when a timer fires which is set to a value smaller than the
   RTO.  A "probe segment" is a new segment if new data is available,
   else a retransmission.  The intention is to compensate for sluggish
   RTO behavior in situations where the RTO greatly exceeds the RTT,
   which, according to measurements reported in [TLP], is not uncommon.
   The Probe timeout (PTO) is normally two RTTs, and a spurious PTO is
   less risky than a spurious RTO because it would not have the same
   negative effects (clearing the scoreboard and restarting with slow-

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   start).  In contrast, RTO restart is a small sender-only modification
   of the RTO management algorithm and does not require an additional
   timer or the use of SACK.

   TLP is applicable in situations where RTO restart does not apply, and
   it could overrule (yielding a similar general behavior, but with a
   lower timeout) RTO restart in cases where the number of outstanding
   segments is smaller than four and no new segments are available for
   transmission.  The PTO has the same inherent restart problems as the
   RTO timer and could be combined with the modified restart approach to
   offer more consistent timeouts.

6.  Acknowledgements

   The authors wish to thank Godred Fairhurst, Yuchung Cheng, Mark
   Allman, Anantha Ramaiah and Richard Scheffenegger for commenting the
   draft and the ideas behind it.

   All the authors are funded by the European Community under its
   Seventh Framework Programme through the Reducing Internet Transport
   Latency (RITE) project (ICT-317700).  The views expressed are solely
   those of the author(s).

7.  IANA Considerations

   This memo includes no request to IANA.

8.  Security Considerations

   This document discusses a change in how to set the retransmission
   timer's value when restarted.  This change does not raise any new
   security issues with TCP or SCTP.

9.  Changes from Previous Versions

9.1.  Changes from draft-ietf-...-00 to -01

   o  Improved the wording throughout the document.

   o  Removed the possibility for a connection limited by the receiver's
      advertised window to use RTO restart, decreasing the risk of
      spurious retransmission timeouts.

   o  Added a section that discusses the applicability of and problems
      related to the RTO restart mechanism.

   o  Updated the text describing the relationship to TLP to reflect
      updates made in this draft.

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   o  Added acknowledgments.

10.  References

10.1.  Normative References

   [RFC1122]  Braden, R., "Requirements for Internet Hosts -
              Communication Layers", STD 3, RFC 1122, October 1989.

   [RFC2018]  Mathis, M., Mahdavi, J., Floyd, S., and A. Romanow, "TCP
              Selective Acknowledgment Options", RFC 2018, October 1996.

   [RFC2119]  Bradner, S., "Key words for use in RFCs to Indicate
              Requirement Levels", BCP 14, RFC 2119, March 1997.

   [RFC3042]  Allman, M., Balakrishnan, H., and S. Floyd, "Enhancing
              TCP's Loss Recovery Using Limited Transmit", RFC 3042,
              January 2001.

   [RFC3522]  Ludwig, R. and M. Meyer, "The Eifel Detection Algorithm
              for TCP", RFC 3522, April 2003.

   [RFC3708]  Blanton, E. and M. Allman, "Using TCP Duplicate Selective
              Acknowledgement (DSACKs) and Stream Control Transmission
              Protocol (SCTP) Duplicate Transmission Sequence Numbers
              (TSNs) to Detect Spurious Retransmissions", RFC 3708,
              February 2004.

   [RFC4015]  Ludwig, R. and A. Gurtov, "The Eifel Response Algorithm
              for TCP", RFC 4015, February 2005.

   [RFC4960]  Stewart, R., "Stream Control Transmission Protocol", RFC
              4960, September 2007.

   [RFC5681]  Allman, M., Paxson, V., and E. Blanton, "TCP Congestion
              Control", RFC 5681, September 2009.

   [RFC5682]  Sarolahti, P., Kojo, M., Yamamoto, K., and M. Hata,
              "Forward RTO-Recovery (F-RTO): An Algorithm for Detecting
              Spurious Retransmission Timeouts with TCP", RFC 5682,
              September 2009.

   [RFC5827]  Allman, M., Avrachenkov, K., Ayesta, U., Blanton, J., and
              P. Hurtig, "Early Retransmit for TCP and Stream Control
              Transmission Protocol (SCTP)", RFC 5827, May 2010.

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   [RFC6298]  Paxson, V., Allman, M., Chu, J., and M. Sargent,
              "Computing TCP's Retransmission Timer", RFC 6298, June
              2011.

10.2.  Informative References

   [BPS98]    Balakrishnan, H., Padmanabhan, V., Seshan, S., Stemm, M.,
              and R. Katz, "TCP Behavior of a Busy Web Server: Analysis
              and Improvements", Proc. IEEE INFOCOM Conf., March 1998.

   [EL04]     Ekstroem, H. and R. Ludwig, "The Peak-Hopper: A New End-
              to-End Retransmission Timer for Reliable Unicast
              Transport", IEEE INFOCOM 2004, March 2004.

   [HB08]     Hurtig, P. and A. Brunstrom, "SCTP: designed for timely
              message delivery?", Springer Telecommunication Systems,
              May 2010.

   [LS00]     Ludwig, R. and K. Sklower, "The Eifel retransmission
              timer", ACM SIGCOMM Comput. Commun. Rev., 30(3), July
              2000.

   [P09]      Petlund, A., "Improving latency for interactive, thin-
              stream applications over reliable transport", Unipub PhD
              Thesis, Oct 2009.

   [PBP09]    Petlund, A., Beskow, P., Pedersen, J., Paaby, E., Griwodz,
              C., and P. Halvorsen, "Improving SCTP Retransmission
              Delays for Time-Dependent Thin Streams", Springer
              Multimedia Tools and Applications, 45(1-3), 2009.

   [PGH06]    Pedersen, J., Griwodz, C., and P. Halvorsen,
              "Considerations of SCTP Retransmission Delays for Thin
              Streams", IEEE LCN 2006, November 2006.

   [RJ10]     Ramachandran, S., "Web metrics: Size and number of
              resources", Google
              http://code.google.com/speed/articles/web-metrics.html,
              May 2010.

   [TLP]      Dukkipati, N., Cardwell, N., Cheng, Y., and M. Mathis,
              "TCP Loss Probe (TLP): An Algorithm for Fast Recovery of
              Tail Losses", Internet-draft draft-dukkipati-tcpm-tcp-
              loss-probe-00.txt, July 2012.

Authors' Addresses

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   Per Hurtig
   Karlstad University
   Universitetsgatan 2
   Karlstad  651 88
   Sweden

   Phone: +46 54 700 23 35
   Email: per.hurtig@kau.se

   Anna Brunstrom
   Karlstad University
   Universitetsgatan 2
   Karlstad  651 88
   Sweden

   Phone: +46 54 700 17 95
   Email: anna.brunstrom@kau.se

   Andreas Petlund
   Simula Research Laboratory AS
   P.O. Box 134
   Lysaker  1325
   Norway

   Phone: +47 67 82 82 00
   Email: apetlund@simula.no

   Michael Welzl
   University of Oslo
   PO Box 1080 Blindern
   Oslo  N-0316
   Norway

   Phone: +47 22 85 24 20
   Email: michawe@ifi.uio.no

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