TCP Maintenance and Minor Extensions (tcpm) P. Hurtig
Internet-Draft A. Brunstrom
Intended status: Experimental Karlstad University
Expires: September 10, 2015 A. Petlund
Simula Research Laboratory AS
M. Welzl
University of Oslo
March 9, 2015
TCP and SCTP RTO Restart
draft-ietf-tcpm-rtorestart-05
Abstract
This document describes a modified algorithm for managing the TCP and
SCTP retransmission timers that provides faster loss recovery when
there is a small amount of outstanding data for a connection. The
modification, RTO Restart (RTOR), allows the transport to restart its
retransmission timer more aggressively in situations where fast
retransmit cannot be used. This enables faster loss detection and
recovery for connections that are short-lived or application-limited.
Status of This Memo
This Internet-Draft is submitted in full conformance with the
provisions of BCP 78 and BCP 79.
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This Internet-Draft will expire on September 10, 2015.
Copyright Notice
Copyright (c) 2015 IETF Trust and the persons identified as the
document authors. All rights reserved.
This document is subject to BCP 78 and the IETF Trust's Legal
Provisions Relating to IETF Documents
(http://trustee.ietf.org/license-info) in effect on the date of
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1. Introduction
TCP uses two mechanisms to detect segment loss. First, if a segment
is not acknowledged within a certain amount of time, a retransmission
timeout (RTO) occurs, and the segment is retransmitted [RFC6298].
While the RTO is based on measured round-trip times (RTTs) between
the sender and receiver, it also has a conservative lower bound of 1
second to ensure that delayed segments are not mistaken as lost.
Second, when a sender receives dupACKs, the fast retransmit algorithm
infers segment loss and triggers a retransmission [RFC5681].
Duplicate acknowledgments are generated by a receiver when out-of-
order segments arrive. As both segment loss and segment reordering
cause out-of-order arrival, fast retransmit waits for three dupACKs
before considering the segment as lost. In some situations, however,
the number of outstanding segments is not enough to trigger three
dupACKs, and the sender must rely on lengthy RTOs for loss recovery.
The number of outstanding segments can be small for several reasons:
(1) The connection is limited by the congestion control when the
path has a low total capacity (bandwidth-delay product) or the
connection's share of the capacity is small. It is also limited
by the congestion control in the first few RTTs of a connection
or after an RTO when the available capacity is probed using
slow-start.
(2) The connection is limited by the receiver's available buffer
space.
(3) The connection is limited by the application if the available
capacity of the path is not fully utilized (e.g. interactive
applications), or at the end of a transfer.
While the reasons listed above are valid for any flow, the third
reason is 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
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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
RTO slightly more aggressive when the number of outstanding segments
is too small for fast retransmit to work, in an attempt to enable
faster loss recovery for all segments while being robust to
reordering. While RTOR still conforms to the requirement in
[RFC6298] that segments must not be retransmitted earlier than RTO
seconds after their original transmission, it could increase the risk
of spurious timeout. Spurious timeouts can degrade the performance
of flows with multiple bursts of data, as a burst following a
spurious timeout might not fit within the reduced congestion window
(cwnd). There are, however, several techniques to mitigate the
effects of such unnecessary retransmissions (cf. [RFC4015]).
While this document focuses on TCP, the described changes are also
valid for the Stream Control Transmission Protocol (SCTP) [RFC4960]
which has similar loss recovery and congestion control algorithms.
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:
The number of previously unsent segments (prevunsnt): The number of
segments that a sender has queued for transmission, but has not yet
sent.
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 Restart Overview
The RTO management algorithm described in [RFC6298] recommends that
the retransmission timer is restarted when an acknowledgment (ACK)
that acknowledges new data is received and there is still outstanding
data. The restart is conducted to guarantee that unacknowledged
segments will be retransmitted after approximately RTO seconds.
However, by restarting the timer on each incoming ACK,
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retransmissions are not typically triggered RTO seconds after their
previous transmission but rather RTO seconds after the last ACK
arrived. The duration of this extra delay depends on several factors
but is in most cases approximately one RTT. Hence, in most
situations, the time before a retransmission is triggered is equal to
"RTO + RTT".
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 is "RTO + RTT" seconds.
In some situations, the effective loss detection time becomes even
longer. Consider a scenario where only two segments are outstanding.
If the second segment is lost, the time to expire the 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
During normal TCP bulk transfer the current RTO restart approach is
not a problem. Actually, as long as enough segments arrive at a
receiver to enable fast retransmit, RTO-based loss recovery should be
avoided. RTOs should only be used as a last resort, as they
drastically lower the congestion window compared to fast retransmit.
The current approach can therefore be beneficial -- it is described
in [EL04] to act as a "safety margin" that compensates for some of
the problems that the authors have identified with the standard RTO
calculation. Notably, the authors of [EL04] also state that "this
safety margin does not exist for highly interactive applications
where often only a single packet is in flight."
Although fast retransmit is preferrable there are situations where
timeouts are appropriate, or the only choice. For example, if the
network is severely congested and no segments arrive RTO-based
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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. By resetting the timer to "RTO -
T_earliest", where T_earliest is the time elapsed since the earliest
outstanding segment was transmitted, retransmissions will always
occur after exactly RTO seconds. This approach makes the RTO more
aggressive than the standardized approach in [RFC6298] but still
conforms to the requirement in [RFC6298] that segments must not be
retransmitted earlier than RTO seconds after their original
transmission.
This document specifies 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 it will prevent RTOR
from being more aggressive and potentially causing RTOs instead of
fast retransmits. This update needs TCP implementations to 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
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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.
5. Discussion
In this section, we discuss the applicability and a number of issues
surrounding 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
measurements, show slight increases in the number of spurious
timeouts when such lower values are used [RHB15]. However, further
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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] only 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 proposed standard. 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
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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. For instance, 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. This strategy was used in an earlier version of
the algorithm and works well. The addition of tracking prevunsnt was
only made to optimize a corner case in which RTOR was unnecessarily
disabled.
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 unnecessary 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
retransmit.
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. Acknowledgements
The authors wish to thank 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 (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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8. IANA Considerations
This memo includes no request to IANA.
9. Security Considerations
This document discusses 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.
10. Changes from Previous Versions
RFC-Editor note: please remove this section prior to publication.
10.1. Changes from draft-ietf-...-04 to -05
o Introduced variable to track the number of previously unsent
segments.
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
RTOR.
o Improved wording throughout the document.
10.2. 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.
10.3. 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.
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o Added a section to generalize the tracking of outstanding
segments.
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.
10.4. 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.
10.5. 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.
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11. References
11.1. Normative References
[RFC1122] Braden, R., "Requirements for Internet Hosts -
Communication Layers", STD 3, RFC 1122, October 1989.
[RFC2119] Bradner, S., "Key words for use in RFCs to Indicate
Requirement Levels", BCP 14, RFC 2119, March 1997.
[RFC3042] Allman, M., Balakrishnan, H., and S. Floyd, "Enhancing
TCP's Loss Recovery Using Limited Transmit", RFC 3042,
January 2001.
[RFC3522] Ludwig, R. and M. Meyer, "The Eifel Detection Algorithm
for TCP", RFC 3522, April 2003.
[RFC3708] Blanton, E. and M. Allman, "Using TCP Duplicate Selective
Acknowledgement (DSACKs) and Stream Control Transmission
Protocol (SCTP) Duplicate Transmission Sequence Numbers
(TSNs) to Detect Spurious Retransmissions", RFC 3708,
February 2004.
[RFC4015] Ludwig, R. and A. Gurtov, "The Eifel Response Algorithm
for TCP", RFC 4015, February 2005.
[RFC4960] Stewart, R., "Stream Control Transmission Protocol", RFC
4960, September 2007.
[RFC5681] Allman, M., Paxson, V., and E. Blanton, "TCP Congestion
Control", RFC 5681, September 2009.
[RFC5682] Sarolahti, P., Kojo, M., Yamamoto, K., and M. Hata,
"Forward RTO-Recovery (F-RTO): An Algorithm for Detecting
Spurious Retransmission Timeouts with TCP", RFC 5682,
September 2009.
[RFC5827] Allman, M., Avrachenkov, K., Ayesta, U., Blanton, J., and
P. Hurtig, "Early Retransmit for TCP and Stream Control
Transmission Protocol (SCTP)", RFC 5827, May 2010.
[RFC6298] Paxson, V., Allman, M., Chu, J., and M. Sargent,
"Computing TCP's Retransmission Timer", RFC 6298, June
2011.
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11.2. 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.
[LS00] Ludwig, R. and K. Sklower, "The Eifel retransmission
timer", ACM SIGCOMM Comput. Commun. Rev., 30(3), July
2000.
[P09] Petlund, A., "Improving latency for interactive, thin-
stream applications over reliable transport", Unipub PhD
Thesis, Oct 2009.
[PBP09] Petlund, A., Beskow, P., Pedersen, J., Paaby, E., Griwodz,
C., and P. Halvorsen, "Improving SCTP Retransmission
Delays for Time-Dependent Thin Streams", Springer
Multimedia Tools and Applications, 45(1-3), 2009.
[PGH06] Pedersen, J., Griwodz, C., and P. Halvorsen,
"Considerations of SCTP Retransmission Delays for Thin
Streams", IEEE LCN 2006, November 2006.
[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
http://code.google.com/speed/articles/web-metrics.html,
May 2010.
[TLP] Dukkipati, N., Cardwell, N., Cheng, Y., and M. Mathis,
"TCP Loss Probe (TLP): An Algorithm for Fast Recovery of
Tail Losses", Internet-draft draft-dukkipati-tcpm-tcp-
loss-probe-01.txt, February 2013.
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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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