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Keywords: tcp, retransmission timer, rtor
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.
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RFC 7765 TCP and SCTP RTO Restart February 2016
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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RFC 7765 TCP and SCTP RTO Restart February 2016
[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>.
Hurtig, et al. Experimental [Page 13]
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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
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