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TCP and Stream Control Transmission Protocol (SCTP) RTO Restart

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 7765.
Authors Per Hurtig , Anna Brunstrom , Andreas Petlund , Michael Welzl
Last updated 2016-02-11 (Latest revision 2015-11-05)
Replaces draft-hurtig-tcpm-rtorestart
RFC stream Internet Engineering Task Force (IETF)
Additional resources Mailing list discussion
Stream WG state Submitted to IESG for Publication
Document shepherd Michael Scharf
Shepherd write-up Show Last changed 2015-07-20
IESG IESG state RFC 7765 (Experimental)
Consensus boilerplate Yes
Telechat date (None)
Responsible AD Martin Stiemerling
Send notices to (None)
IANA IANA review state Version Changed - Review Needed
IANA action state No IANA Actions
TCP Maintenance and Minor Extensions (tcpm)                    P. Hurtig
Internet-Draft                                              A. Brunstrom
Intended status: Experimental                        Karlstad University
Expires: May 8, 2016                                          A. Petlund
                                           Simula Research Laboratory AS
                                                                M. Welzl
                                                      University of Oslo
                                                        November 5, 2015

                        TCP and SCTP RTO Restart


   This document describes a modified sender-side 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, RTO Restart (RTOR), allows the
   transport to restart its retransmission timer using a smaller timeout
   duration, so that the effective RTO becomes more aggressive in
   situations where fast retransmit cannot be used.  This enables faster
   loss detection and recovery for connections that are short-lived or

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

   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 May 8, 2016.

Copyright Notice

   Copyright (c) 2015 IETF Trust and the persons identified as the
   document authors.  All rights reserved.

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   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.

1.  Introduction

   TCP and SCTP use two almost identical mechanisms to detect and
   recover from data loss, specified in [RFC6298][RFC5681] (for TCP) and
   [RFC4960] (for SCTP).  First, if transmitted data is not acknowledged
   within a certain amount of time, a retransmission timeout (RTO)
   occurs, and the data is retransmitted.  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 data are not mistaken as lost.  Second, when a sender
   receives duplicate acknowledgments, or similar information via
   selective acknowledgments, the fast retransmit algorithm suspects
   data loss and can trigger a retransmission.  Duplicate (and
   selective) acknowledgments are generated by a receiver when data
   arrives out-of-order.  As both data loss and data reordering cause
   out-of-order arrival, fast retransmit waits for three out-of-order
   notifications before considering the corresponding data as lost.  In
   some situations, however, the amount of outstanding data is not
   enough to trigger three such acknowledgments, and the sender must
   rely on lengthy RTOs for loss recovery.

   The amount of outstanding data 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

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

   (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.

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   While the reasons listed above are valid for any flow, the third
   reason is most common for applications that transmit short flows, or
   use a bursty transmission pattern.  A typical example of applications
   that produce short flows are web-based applications.  [RJ10] shows
   that 70% of all web objects, found at the top 500 sites, are too
   small for fast retransmit to work.  [FDT13] shows that about 77% of
   all retransmissions sent by a major web service are sent after RTO
   expiry.  Applications with bursty transmission patterns often send
   data in response to actions, or as a reaction to real life events.
   Typical examples of such applications are stock trading systems,
   remote computer operations, online games, and web-based applications
   using persistent connections.  What is special about this class of
   applications is that they often are time-dependant, and extra latency
   can reduce the application service level [P09].

   The RTO Restart (RTOR) mechanism described in this document makes the
   effective RTO slightly more aggressive when the amount of outstanding
   data is too small for fast retransmit to work, in an attempt to
   enable faster loss recovery while being robust to reordering.  While
   RTOR still conforms to the requirement for when a segment can be
   retransmitted, specified in [RFC6298] (for TCP) and [RFC4960] (for
   SCTP) it could increase the risk of spurious timeouts.  To determine
   whether this modification is safe to deploy and enable by default,
   further experimentation is required.  Section 5 discusses experiments
   still needed, including evaluations in environments where the risk of
   spurious retransmissions are increased e.g. mobile networks with
   highly varying RTTs.

   The remainder of this document describes RTOR and its implementation
   for TCP only, to make the document easier to read.  However, the RTOR
   algorithm described in Section 4 is applicable also for SCTP.
   Furthermore, Section 7 details the SCTP socket API needed to control

2.  Terminology

   The key words "MUST", "MUST NOT", "REQUIRED", "SHALL", "SHALL NOT",
   document are to be interpreted as described in RFC 2119 [RFC2119].

   This document introduces the following variables:

   The number of previously unsent segments (prevunsnt): The number of
   segments that a sender has queued for transmission, but has not yet

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   RTO Restart threshold (rrthresh): RTOR is enabled whenever the sum of
   the number of outstanding and previously unsent segments (prevunsnt)
   is below this threshold.

3.  RTO Overview and Rationale for RTOR

   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.  The
   standardized RTO timer management 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 (delACK)
   [RFC1122], which restarts the RTO timer at the sender.  The RTO is
   restarted 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 become "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
   delACK timer will also be included in the effective loss detection

            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

   For bulk traffic the current approach is 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".  In
   general, however, 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.

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   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.  However, for
   connections that do not utilize enough capacity to enable fast
   retransmit, RTO-based loss detection is the only choice and the time
   required for this can become a performance bottleneck.

4.  RTOR Algorithm

   To enable faster loss recovery for connections that are unable to use
   fast retransmit, RTOR can be used.  This section specifies the
   modifications required to use RTOR.  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 document specifies an OPTIONAL 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).  A sender
   that implements this method MUST follow the algorithm below:

      When an ACK is received that acknowledges new data:

      (1)  Set T_earliest = 0.

      (2)  If the sum of the number of outstanding and previously unsent
           segments (prevunsnt) is less than an RTOR threshold
           (rrthresh), 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
           (for the current value of RTO):

           (a)  RTO - T_earliest, if RTO - T_earliest > 0.

           (b)  RTO, otherwise.

   The RECOMMENDED value of rrthresh is four, as this value will ensure
   that RTOR is only used when fast retransmit cannot be triggered.
   With this update, TCP implementations MUST track the time elapsed
   since the transmission of the earliest outstanding segment
   (T_earliest).  As RTOR is only used when the amount of outstanding
   and previously unsent data is less than rrthresh segments, TCP
   implementations also need to track whether the amount of outstanding
   and previously unsent data is more, equal, or less than rrthresh
   segments.  Although some packet-based TCP implementations (e.g.

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   Linux TCP) already track both the transmission times of all segments
   and also the number of outstanding segments, not all implementations
   do.  Section 5.3 describes how to implement segment tracking for a
   general TCP implementation.  To use RTOR, the calculated expiration
   time MUST be positive (step 3(a) in the list above); this is required
   to ensure that RTOR does not trigger retransmissions prematurely when
   previously retransmitted segments are acknowledged.

5.  Discussion

   Although RTOR conforms to the requirement in [RFC6298] that segments
   must not be retransmitted earlier than RTO seconds after their
   original transmission, RTOR makes the effective RTO more aggressive.
   In this section, we discuss the applicability and the issues related
   to RTOR.

5.1.  Applicability

   The currently standardized algorithm has been shown to add at least
   one RTT to the loss recovery process in TCP [LS00] and SCTP
   [HB11][PBP09].  For applications that have strict timing requirements
   (e.g.  interactive web) rather than throughput requirements, using
   RTOR could be beneficial because the RTT and also the delACK timer of
   receivers are often large components of the effective loss recovery
   time.  Measurements in [HB11] 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 RTOR.  These results
   match those presented in [PGH06][PBP09], where RTOR is shown to
   significantly reduce retransmission latency.

   There are also traffic types that do not benefit from RTOR.  One
   example of such traffic is bulk transmission.  The reason why bulk
   traffic does not benefit from RTOR is that such traffic flows mostly
   have four or more segments outstanding, allowing loss recovery by
   fast retransmit.  However, there is no harm in using RTOR for such
   traffic as the algorithm only is active when the amount of
   outstanding and unsent segments are less than rrthresh (default 4).

   Given that RTOR is a mostly conservative algorithm, it is suitable
   for experimentation as a system-wide default for TCP traffic.

5.2.  Spurious Timeouts

   RTOR can in some situations reduce the loss detection time and
   thereby increase 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

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   measurements show slight increases in the number of spurious timeouts
   when such lower values are used [RHB15].  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 the network as segments are transmitted
   needlessly.  However, recent experiments do not show a significant
   increase in network load for a number of realistic scenarios [RHB15].
   Another problem with spurious retransmissions is related to the
   performance of TCP/SCTP, as the congestion window is reduced to one
   segment when timeouts occur [RFC5681].  This could be a potential
   problem for applications transmitting multiple bursts of data within
   a single flow, e.g. web-based HTTP/1.1 and HTTP/2.0 applications.
   However, results from recent experiments involving persistent web
   traffic [RHB15] revealed a net gain of using RTOR.  Other types of
   flows, e.g.  long-lived bulk flows, are not affected as the algorithm
   is only applied when the amount of outstanding and unsent segments is
   less than rrthresh.  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.

   While a slight increase in spurious timeouts has been observed using
   RTOR, 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 the standards track.  For instance, RTOR has not been evaluated in
   the context of mobile networks.  Mobile networks often incur highly
   variable RTTs (delay spikes), due to e.g. handovers, and would
   therefore be a useful scenario for further experimentation.

5.3.  Tracking Outstanding and Previously Unsent Segments

   The method of tracking outstanding and previously unsent segments
   will probably differ depending on the actual TCP implementation.  For
   packet-based TCP implementations, tracking outstanding segments is
   often straightforward and can be implemented using a simple counter.
   For byte-based TCP stacks it is a more complex task.  Section 3.2 of
   [RFC5827] outlines a general method of tracking the number of
   outstanding segments.  The same method can be used for RTOR.  The
   implementation will have to track segment boundaries to form an
   understanding as to how many actual segments have been transmitted,
   but not acknowledged.  This can be done by the sender tracking the
   boundaries of the rrthresh segments on the right side of the current

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   window (which involves tracking rrthresh + 1 sequence numbers in
   TCP).  This could be done by keeping a circular list of the segment
   boundaries, for instance.  Cumulative ACKs that do not fall within
   this region indicate that at least rrthresh segments are outstanding,
   and therefore RTOR is not enabled.  When the outstanding window
   becomes small enough that RTOR can be invoked, a full understanding
   of the number of outstanding segments will be available from the
   rrthresh + 1 sequence numbers retained.  (Note: the implicit sequence
   number consumed by the TCP FIN bit can also be included in the
   tracking of segment boundaries.)

   Tracking the number of previously unsent segments depends on the
   segmentation strategy used by the TCP implementation, not whether it
   is packet-based or byte-based.  In the case segments are formed
   directly on socket writes, the process of determining the number of
   previously unsent segments should be trivial.  In the case that
   unsent data can be segmented (or re-segmented) as long as it still is
   unsent, a straightforward strategy could be to divide the amount of
   unsent data (in bytes) with the SMSS to obtain an estimate.  In some
   cases, such an estimation could be too simplistic, depending on the
   segmentation strategy of the TCP implementation.  However, this
   estimation is not critical to RTOR.  The tracking of prevunsnt is
   only made to optimize a corner case in which RTOR was unnecessarily
   disabled.  Implementations can use a simplified method by setting
   prevunsnt to rrthresh whenever previously unsent data is available,
   and set prevunsnt to zero when no new data is available.  This will
   disable RTOR in the presence of unsent data and only use the number
   of outstanding segments to enable/disable RTOR.

6.  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
   dupACKs.  By transmitting new segments, the sender attempts to
   generate additional dupACKs to enable fast retransmit.  However,
   limited transmit does not help if no previously unsent data is ready
   for transmission.  [RFC5827] specifies an early retransmit algorithm
   to enable fast loss recovery in such situations.  By dynamically
   lowering the number of dupACKs needed for fast retransmit
   (dupthresh), based on the number of outstanding segments, a smaller
   number of dupACKs is 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, early

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   retransmit cannot help.  Third, if the network path reorders
   segments, the algorithm might cause more spurious retransmissions
   than fast retransmit.  The recommended value of RTOR's rrthresh
   variable is based on the dupthresh, but is possible to adapt to allow
   tighter integration with other experimental algorithms such as early

   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.
   Furthermore, TLP also tries to circumvent the congestion window reset
   to one segment by instead enabling fast recovery.  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-start).  TLP is a
   more advanced mechanism than RTOR, requiring e.g.  SACK to work, and
   is often able to reduce loss recovery times more.  However, it also
   increases the amount of spurious retransmissions noticeably, as
   compared to RTOR [RHB15].

   TLP is applicable in situations where RTOR does not apply, and it
   could overrule (yielding a similar general behavior, but with a lower
   timeout) RTOR 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 problem of restarting the timer on an
   incoming ACK, and could be combined with a strategy similar to RTOR's
   to offer more consistent timeouts.

7.  SCTP Socket API Considerations

   This section describes how the socket API for SCTP defined in
   [RFC6458] is extended to control the usage of RTO restart for SCTP.

   Please note that this section is informational only.

7.1.  Data Types

   This section uses data types from [IEEE.1003-1G.1997]: uintN_t means
   an unsigned integer of exactly N bits (e.g., uint16_t).  This is the
   same as in [RFC6458].

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7.2.  Socket Option for Controlling the RTO Restart Support

   This socket option allows the enabling or disabling of RTO Restart
   for SCTP associations.

   Whether RTO Restart is enabled or not per default is implementation

   This socket option uses IPPROTO_SCTP as its level and
   SCTP_RTO_RESTART as its name.  It can be used with getsockopt() and
   setsockopt().  The socket option value uses the following structure
   defined in [RFC6458]:

   struct sctp_assoc_value {
     sctp_assoc_t assoc_id;
     uint32_t assoc_value;

   assoc_id:  This parameter is ignored for one-to-one style sockets.
      For one-to-many style sockets, this parameter indicates upon which
      association the user is performing an action.  The special
      sctp_assoc_t SCTP_{FUTURE|CURRENT|ALL}_ASSOC can also be used in
      assoc_id for setsockopt().  For getsockopt(), the special value
      SCTP_FUTURE_ASSOC can be used in assoc_id, but it is an error to
      use SCTP_{CURRENT|ALL}_ASSOC in assoc_id.

   assoc_value:  A non-zero value encodes the enabling of RTO restart
      whereas a value of 0 encodes the disabling of RTO restart.

   sctp_opt_info() needs to be extended to support SCTP_RTO_RESTART.

8.  IANA Considerations

   This memo includes no request to IANA.

9.  Security Considerations

   This document specifies an experimental sender-only modification to
   TCP and SCTP.  The modification introduces a change in how to set the
   retransmission timer's value when restarted.  Therefore, the security
   considerations found in [RFC6298] apply to this document.  No
   additional security problems have been identified with RTO Restart at
   this time.

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10.  Acknowledgements

   The authors wish to thank Michael Tuexen for contributing the SCTP
   Socket API considerations and Godred Fairhurst, Yuchung Cheng, Mark
   Allman, Anantha Ramaiah, Richard Scheffenegger, Nicolas Kuhn,
   Alexander Zimmermann, and Michael Scharf for commenting on the draft
   and the ideas behind it.

   All the authors are supported by RITE ( ), a
   research project (ICT-317700) funded by the European Community under
   its Seventh Framework Program.  The views expressed here are those of
   the author(s) only.  The European Commission is not liable for any
   use that may be made of the information in this document.

11.  Changes from Previous Versions

   RFC-Editor note: please remove this section prior to publication.

11.1.  Changed from draft-ietf-...-09 to -10

   o  Changed wording in abstract, from "delay" to "timeout duration".

11.2.  Changed from draft-ietf-...-08 to -09

   o  Clarified, in the abstract, that the modified restart causes a
      smaller retransmission delay in total.

   o  Clarified, in the introduction, that the fast retransmit algorithm
      may cause retransmissions upon receiving duplicate
      acknowledgments, not that it unconditionally does so.

   o  Changed wording from "to proposed standard" to "to the standards

   o  Changed algorithm description so that a TCP sender MUST track the
      time elapsed since the transmission of the earliest outstanding
      segment.  This was not explicitly stated in previous versions of
      the draft.

11.3.  Changes from draft-ietf-...-07 to -08

   o  Clarified, at multiple places in the document, that the
      modification only causes the effective RTO to be more aggressive,
      not the actual RTO.

   o  Removed information in the introduction that was too detailed,
      i.e., material that is hard to understand without knowing details
      of the algorithm.

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   o  Changed the name of Section 3 to more correctly capture the actual
      contents of the section.

   o  Re-arranged the text in Section 3 to have a more logical

   o  Moved text from the algorithm description (Section 4) to the
      introduction of the discussion section (Section 5).  The text was
      discussing the possible effects of the algorithm more than
      describing the actual algorithm.

   o  Clarified why the RECOMMENDED value of rrthresh is four.

   o  Reworked the introduction to be suitable for both TCP and SCTP.

11.4.  Changes from draft-ietf-...-06 to -07

   o  Clarified, at multiple places in the document, that the
      modification is sender-only.

   o  Added an explanation (in the introduction) to why the mechanism is
      experimental and what experiments are missing.

   o  Added a sentence in Section 4 to clarify that the section is the
      one describing the actual modification.

11.5.  Changes from draft-ietf-...-05 to -06

   o  Added socket API considerations, after discussing the draft in

11.6.  Changes from draft-ietf-...-04 to -05

   o  Introduced variable to track the number of previously unsent

   o  Clarified many concepts, e.g. extended the description on how to
      track outstanding and previously unsent segments.

   o  Added a reference to initial measurements on the effects of using

   o  Improved wording throughout the document.

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11.7.  Changes from draft-ietf-...-03 to -04

   o  Changed the algorithm to allow RTOR when there is unsent data
      available, but the cwnd does not allow transmission.

   o  Changed the algorithm to not trigger if RTOR <= 0.

   o  Made minor adjustments throughout the document to adjust for the
      algorithmic change.

   o  Improved the wording throughout the document.

11.8.  Changes from draft-ietf-...-02 to -03

   o  Updated the document to use "RTOR" instead of "RTO Restart" when
      refering to the modified algorithm.

   o  Moved document terminology to a section of its own.

   o  Introduced the rrthresh variable in the terminology section.

   o  Added a section to generalize the tracking of outstanding

   o  Updated the algorithm to work when the number of outstanding
      segments is less than four and one segment is ready for
      transmission, by restarting the timer when new data has been sent.

   o  Clarified the relationship between fast retransmit and RTOR.

   o  Improved the wording throughout the document.

11.9.  Changes from draft-ietf-...-01 to -02

   o  Changed the algorithm description in Section 3 to use formal RFC
      2119 language.

   o  Changed last paragraph of Section 3 to clarify why the RTO restart
      algorithm is active when less than four segments are outstanding.

   o  Added two paragraphs in Section 4.1 to clarify why the algorithm
      can be turned on for all TCP traffic without having any negative
      effects on traffic patterns that do not benefit from a modified
      timer restart.

   o  Improved the wording throughout the document.

   o  Replaced and updated some references.

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11.10.  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.

   o  Added acknowledgments.

12.  References

12.1.  Normative References

   [RFC1122]  Braden, R., Ed., "Requirements for Internet Hosts -
              Communication Layers", STD 3, RFC 1122,
              DOI 10.17487/RFC1122, October 1989,

   [RFC2119]  Bradner, S., "Key words for use in RFCs to Indicate
              Requirement Levels", BCP 14, RFC 2119,
              DOI 10.17487/RFC2119, March 1997,

   [RFC3042]  Allman, M., Balakrishnan, H., and S. Floyd, "Enhancing
              TCP's Loss Recovery Using Limited Transmit", RFC 3042,
              DOI 10.17487/RFC3042, January 2001,

   [RFC3522]  Ludwig, R. and M. Meyer, "The Eifel Detection Algorithm
              for TCP", RFC 3522, DOI 10.17487/RFC3522, 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,
              DOI 10.17487/RFC3708, February 2004,

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   [RFC4015]  Ludwig, R. and A. Gurtov, "The Eifel Response Algorithm
              for TCP", RFC 4015, DOI 10.17487/RFC4015, February 2005,

   [RFC4960]  Stewart, R., Ed., "Stream Control Transmission Protocol",
              RFC 4960, DOI 10.17487/RFC4960, September 2007,

   [RFC5681]  Allman, M., Paxson, V., and E. Blanton, "TCP Congestion
              Control", RFC 5681, DOI 10.17487/RFC5681, 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,
              DOI 10.17487/RFC5682, 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,
              DOI 10.17487/RFC5827, May 2010,

   [RFC6298]  Paxson, V., Allman, M., Chu, J., and M. Sargent,
              "Computing TCP's Retransmission Timer", RFC 6298,
              DOI 10.17487/RFC6298, June 2011,

12.2.  Informative References

   [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.

   [FDT13]    Flach, T., Dukkipati, N., Terzis, A., Raghavan, B.,
              Cardwell, N., Cheng, Y., Jain, A., Hao, S., Katz-Bassett,
              E., and R. Govindan, "Reducing Web Latency: the Virtue of
              Gentle Aggression", Proc. ACM SIGCOMM Conf., August 2013.

   [HB11]     Hurtig, P. and A. Brunstrom, "SCTP: designed for timely
              message delivery?", Springer Telecommunication Systems 47
              (3-4), August 2011.

              Institute of Electrical and Electronics Engineers,
              "Protocol Independent Interfaces", IEEE Standard 1003.1G,
              March 1997.

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

   [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.

   [RFC6458]  Stewart, R., Tuexen, M., Poon, K., Lei, P., and V.
              Yasevich, "Sockets API Extensions for the Stream Control
              Transmission Protocol (SCTP)", RFC 6458,
              DOI 10.17487/RFC6458, December 2011,

   [RHB15]    Rajiullah, M., Hurtig, P., Brunstrom, A., Petlund, A., and
              M. Welzl, "An Evaluation of Tail Loss Recovery Mechanisms
              for TCP", ACM SIGCOMM CCR 45 (1), January 2015.

   [RJ10]     Ramachandran, S., "Web metrics: Size and number of
              resources", Google
              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-01.txt, February 2013.

Authors' Addresses

   Per Hurtig
   Karlstad University
   Universitetsgatan 2
   Karlstad  651 88

   Phone: +46 54 700 23 35

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   Anna Brunstrom
   Karlstad University
   Universitetsgatan 2
   Karlstad  651 88

   Phone: +46 54 700 17 95

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

   Phone: +47 67 82 82 00

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

   Phone: +47 22 85 24 20

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