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Versions: (draft-hurtig-tcpm-rtorestart) 00 01 02 03 04 05 06 07 08 09 10 RFC 7765

TCP Maintenance and Minor Extensions                           P. Hurtig
(tcpm)                                                      A. Brunstrom
Internet-Draft                                       Karlstad University
Intended status: Experimental                                 A. Petlund
Expires: August 20, 2013                   Simula Research Laboratory AS
                                                                M. Welzl
                                                      University of Oslo
                                                       February 16, 2013


                        TCP and SCTP RTO Restart
                     draft-ietf-tcpm-rtorestart-00

Abstract

   This document describes a modified algorithm for managing the TCP and
   SCTP retransmission timers that provides faster loss recovery when a
   connection's amount of outstanding data is small.  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 August 20, 2013.

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 amount 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 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, which is frequent if
        the total amount of data is small (e.g. web traffic).

   The first two situations can occur for any flow, as external factors
   at the network and/or host level cause them.  The third situation
   primarily affects flows that are short or have a low transmission
   rate.  Typical examples of applications that produce short flows are



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   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 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 chance for a spurious timeout, which could degrade
   performance when the congestion window (cwnd) is large -- for
   example, when an application sends enough data to reach a cwnd
   covering 100 segments and then stops.  The likelihood and potential
   impact of this problem as well as possible mitigation strategies are
   currently under investigation.

   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



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   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".

   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.  The 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,
   and 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."

   There are only a few situations where timeouts are appropriate, or



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   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 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 the following update of 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 or the
                receiver's advertised window does not permit
                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.



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   Linux TCP) already track the transmission times of all segments.


4.  Discussion

   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].  Applications that have strict timing requirements
   (e.g. telephony signaling and gaming) rather than throughput
   requirements may want to use a lower RTOmin than the standard of 1
   second [RFC4166].  For such applications the modified restart
   approach could be important as the RTT and also the delayed ACK timer
   of receivers will be 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 with up to 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.

   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
   is out of buffer space.  [RFC5827] specifies an early retransmit
   algorithm to enable fast loss recovery in such situations.  By
   dynamically lowering the amount 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.




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   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 at least 2 RTTs, and only scheduled in
   case the RTO is farther than the PTO.  A spurious PTO is less risky
   than a spurious RTO, as it would not have the same negative effects
   (clearing the scoreboard and restarting with slow-start).  In
   contrast, RTO restart is trying to make the RTO more appropriate in
   cases where there is no need to be overly cautious.

   TLP could kick in 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 4 and no new segments are available for
   transmission.  The shorter RTO from RTO restart also reduces the
   probability that TLP is activated because PTO might be farther than
   RTO.


5.  IANA Considerations

   This memo includes no request to IANA.


6.  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.


7.  References

7.1.  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



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              TCP's Loss Recovery Using Limited Transmit", RFC 3042,
              January 2001.

   [RFC4166]  Coene, L. and J. Pastor-Balbas, "Telephony Signalling
              Transport over Stream Control Transmission Protocol (SCTP)
              Applicability Statement", RFC 4166, February 2006.

   [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.

   [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.

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

7.2.  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.




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   [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", draft-dukkipati-tcpm-tcp-loss-probe-00.txt
              (work in progress), July 2012.


Authors' Addresses

   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







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   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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