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RFC 7765

Experimental
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TCP and Stream Control Transmission Protocol (SCTP) RTO Restart

 


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Internet Engineering Task Force (IETF)                         P. Hurtig
Request for Comments: 7765                                  A. Brunstrom
Category: Experimental                               Karlstad University
ISSN: 2070-1721                                               A. Petlund
                                           Simula Research Laboratory AS
                                                                M. Welzl
                                                      University of Oslo
                                                           February 2016


    TCP and Stream Control Transmission Protocol (SCTP) RTO Restart

Abstract

   This document describes a modified sender-side algorithm for managing
   the TCP and Stream Control Transmission Protocol (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 retransmission timeout (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
   application limited.

Status of This Memo

   This document is not an Internet Standards Track specification; it is
   published for examination, experimental implementation, and
   evaluation.

   This document defines an Experimental Protocol for the Internet
   community.  This document is a product of the Internet Engineering
   Task Force (IETF).  It represents the consensus of the IETF
   community.  It has received public review and has been approved for
   publication by the Internet Engineering Steering Group (IESG).  Not
   all documents approved by the IESG are a candidate for any level of
   Internet Standard; see Section 2 of RFC 5741.

   Information about the current status of this document, any errata,
   and how to provide feedback on it may be obtained at
   http://www.rfc-editor.org/info/rfc7765.

Page 2 
Copyright Notice

   Copyright (c) 2016 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
   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.

Table of Contents

   1.  Introduction  . . . . . . . . . . . . . . . . . . . . . . . .   3
   2.  Terminology . . . . . . . . . . . . . . . . . . . . . . . . .   4
   3.  RTO Overview and Rationale for RTOR . . . . . . . . . . . . .   4
   4.  RTOR Algorithm  . . . . . . . . . . . . . . . . . . . . . . .   6
   5.  Discussion  . . . . . . . . . . . . . . . . . . . . . . . . .   7
     5.1.  Applicability . . . . . . . . . . . . . . . . . . . . . .   7
     5.2.  Spurious Timeouts . . . . . . . . . . . . . . . . . . . .   7
     5.3.  Tracking Outstanding and Previously Unsent Segments . . .   8
   6.  Related Work  . . . . . . . . . . . . . . . . . . . . . . . .   9
   7.  SCTP Socket API Considerations  . . . . . . . . . . . . . . .  10
     7.1.  Data Types  . . . . . . . . . . . . . . . . . . . . . . .  10
     7.2.  Socket Option for Controlling the RTO Restart Support
           (SCTP_RTO_RESTART)  . . . . . . . . . . . . . . . . . . .  10
   8.  Security Considerations . . . . . . . . . . . . . . . . . . .  11
   9.  References  . . . . . . . . . . . . . . . . . . . . . . . . .  11
     9.1.  Normative References  . . . . . . . . . . . . . . . . . .  11
     9.2.  Informative References  . . . . . . . . . . . . . . . . .  13
   Acknowledgements  . . . . . . . . . . . . . . . . . . . . . . . .  14
   Authors' Addresses  . . . . . . . . . . . . . . . . . . . . . . .  15

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1.  Introduction

   TCP and SCTP use two almost identical mechanisms to detect and
   recover from data loss, specified in [RFC6298] and [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 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 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 is at the end of a transfer.

   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

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   using persistent connections.  What is special about this class of
   applications is that they are often time dependent, 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
   RTOR.

2.  Terminology

   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].

   This document introduces the following variables:

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

   o  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 be 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

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   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 becomes "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
   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

   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 as compared to fast retransmit.

   Although fast retransmit is preferable, there are situations where
   timeouts are appropriate or are 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.

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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.,
   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.

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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 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] and [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 is only 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
   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

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   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 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 or not the effects of this increase
   mandate any future algorithmic changes -- 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
   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

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   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 where 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 resegmented) as long as it is still
   unsent, a straightforward strategy could be to divide the amount of
   unsent data (in bytes) with the Sender Maximum Segment Size (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
   duplicate acknowledgements (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 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 it is possible to adapt to
   allow tighter integration with other experimental algorithms such as
   early retransmit.

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   Tail Loss Probe [TLP] is a proposal to send up to two "probe
   segments" when a timer fires that is set to a value smaller than the
   RTO.  A "probe segment" is a new segment if new data is available,
   else it is 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 further reduce loss recovery times.
   However, it also noticeably increases the amount of spurious
   retransmissions, 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.9945]: uintN_t means an
   unsigned integer of exactly N bits (e.g., uint16_t).  This is the
   same as in [RFC6458].

7.2.  Socket Option for Controlling the RTO Restart Support
      (SCTP_RTO_RESTART)

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

   Whether or not RTO restart is enabled per default is implementation
   specific.

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   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.  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.

9.  References

9.1.  Normative References

   [RFC1122]  Braden, R., Ed., "Requirements for Internet Hosts -
              Communication Layers", STD 3, RFC 1122,
              DOI 10.17487/RFC1122, October 1989,
              <http://www.rfc-editor.org/info/rfc1122>.

   [RFC2119]  Bradner, S., "Key words for use in RFCs to Indicate
              Requirement Levels", BCP 14, RFC 2119,
              DOI 10.17487/RFC2119, March 1997,
              <http://www.rfc-editor.org/info/rfc2119>.

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   [RFC3042]  Allman, M., Balakrishnan, H., and S. Floyd, "Enhancing
              TCP's Loss Recovery Using Limited Transmit", RFC 3042,
              DOI 10.17487/RFC3042, January 2001,
              <http://www.rfc-editor.org/info/rfc3042>.

   [RFC3522]  Ludwig, R. and M. Meyer, "The Eifel Detection Algorithm
              for TCP", RFC 3522, DOI 10.17487/RFC3522, April 2003,
              <http://www.rfc-editor.org/info/rfc3522>.

   [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,
              <http://www.rfc-editor.org/info/rfc3708>.

   [RFC4015]  Ludwig, R. and A. Gurtov, "The Eifel Response Algorithm
              for TCP", RFC 4015, DOI 10.17487/RFC4015, February 2005,
              <http://www.rfc-editor.org/info/rfc4015>.

   [RFC4960]  Stewart, R., Ed., "Stream Control Transmission Protocol",
              RFC 4960, DOI 10.17487/RFC4960, September 2007,
              <http://www.rfc-editor.org/info/rfc4960>.

   [RFC5681]  Allman, M., Paxson, V., and E. Blanton, "TCP Congestion
              Control", RFC 5681, DOI 10.17487/RFC5681, September 2009,
              <http://www.rfc-editor.org/info/rfc5681>.

   [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,
              <http://www.rfc-editor.org/info/rfc5682>.

   [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,
              <http://www.rfc-editor.org/info/rfc5827>.

   [RFC6298]  Paxson, V., Allman, M., Chu, J., and M. Sargent,
              "Computing TCP's Retransmission Timer", RFC 6298,
              DOI 10.17487/RFC6298, June 2011,
              <http://www.rfc-editor.org/info/rfc6298>.

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9.2.  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,
              DOI 10.1109/INFCOM.2004.1354671, 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.,
              DOI 10.1145/2486001.2486014, August 2013.

   [HB11]     Hurtig, P. and A. Brunstrom, "SCTP: designed for timely
              message delivery?", Springer Telecommunication Systems 47
              (3-4), DOI 10.1007/s11235-010-9321-3, August 2011.

   [IEEE.9945]
              IEEE/ISO/IEC, "International Standard - Information
              technology Portable Operating System Interface (POSIX)
              Base Specifications, Issue 7", IEEE 9945-2009,
              <http://standards.ieee.org/findstds/
              standard/9945-2009.html>.

   [LS00]     Ludwig, R. and K. Sklower, "The Eifel retransmission
              timer", ACM SIGCOMM Comput. Commun. Rev., 30(3),
              DOI 10.1145/382179.383014, 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),
              DOI 10.1007/s11042-009-0286-8, October 2009.

   [PGH06]    Pedersen, J., Griwodz, C., and P. Halvorsen,
              "Considerations of SCTP Retransmission Delays for Thin
              Streams", IEEE LCN 2006, DOI 10.1109/LCN.2006.322082,
              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,
              <http://www.rfc-editor.org/info/rfc6458>.

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   [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),
              DOI 10.1145/2717646.2717648, January 2015.

   [RJ10]     Ramachandran, S., "Web metrics: Size and number of
              resources", May 2010, <https://goo.gl/0a6Q9A>.

   [TLP]      Dukkipati, N., Cardwell, N., Cheng, Y., and M. Mathis,
              "Tail Loss Probe (TLP): An Algorithm for Fast Recovery of
              Tail Losses", Work in Progress, draft-dukkipati-tcpm-tcp-
              loss-probe-01, February 2013.

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
   document and the ideas behind it.

   All the authors are supported by RITE (http://riteproject.eu/), 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.

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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


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

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