Internet DRAFT - draft-hurtig-tcpm-rtorestart
draft-hurtig-tcpm-rtorestart
TCP Maintenance and Minor Extensions P. Hurtig
(tcpm) Karlstad University
Internet-Draft A. Petlund
Intended status: Experimental Simula Research Laboratory AS
Expires: May 1, 2012 M. Welzl
University of Oslo
October 29, 2011
TCP and SCTP RTO Restart
draft-hurtig-tcpm-rtorestart-01
Abstract
This document describes a modified algorithm for managing the TCP and
SCTP retransmission timers, that provides faster loss recovery when a
connection's amount of outstanding data is small. The modification
allows the transport to restart its retransmission timer more
aggressively in situations where fast retransmit can not be used.
This enables faster loss detection, and recovery, for connections
that are short-lived or application-limited.
Status of this Memo
This Internet-Draft is submitted in full conformance with the
provisions of BCP 78 and BCP 79.
Internet-Drafts are working documents of the Internet Engineering
Task Force (IETF). Note that other groups may also distribute
working documents as Internet-Drafts. The list of current Internet-
Drafts is at http://datatracker.ietf.org/drafts/current/.
Internet-Drafts are draft documents valid for a maximum of six months
and may be updated, replaced, or obsoleted by other documents at any
time. It is inappropriate to use Internet-Drafts as reference
material or to cite them other than as "work in progress."
This Internet-Draft will expire on May 1, 2012.
Copyright Notice
Copyright (c) 2011 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
Hurtig, et al. Expires May 1, 2012 [Page 1]
Internet-Draft TCP and SCTP RTO Restart October 2011
carefully, as they describe your rights and restrictions with respect
to this document. Code Components extracted from this document must
include Simplified BSD License text as described in Section 4.e of
the Trust Legal Provisions and are provided without warranty as
described in the Simplified BSD License.
1. Introduction
TCP uses two mechanisms to detect segment loss. First, if a segment
is not acknowledged within a certain amount of time, a retransmission
timeout (RTO) occurs, and the segment is retransmitted [RFC6298].
While the RTO is based on measured round-trip times (RTTs) between
the sender and receiver, it also has a conservative lower bound of 1
second to ensure that delayed segments are not mistaken as lost.
Second, when a sender receives duplicate acknowledgments, the fast
retransmit algorithm infers segment loss and triggers a
retransmission. Duplicate acknowledgments are generated by a
receiver when out-of-order segments arrive. As both segment loss and
segment reordering cause out-of-order arrival, fast retransmit waits
for three duplicate acknowledgments before considering the segment as
lost. In some situations, however, the number of outstanding
segments is not enough to trigger three duplicate acknowledgments,
and the sender must rely on lengthy RTOs for loss recovery.
The amount of outstanding segments can be small for several reasons:
(1) The connection is limited by the congestion control when the
path has a low total capacity (bandwidth-delay product) or the
connection's share of the capacity is small. It is also limited
by the congestion control in the first RTTs of a connection 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 when the total
amount of data is small (e.g. web traffic) or if the available
capacity of the path is not fully utilized (e.g. interactive
applications).
The first two situations can occur for any flow, as external factors
at the network and/or host level cause them. However, the third
situation only affects flows that are short or have a low
transmission rate. Typical examples of applications that produce
short flows are web servers. [RJ10] shows that 70% of all web
objects, found at the top 500 sites, are too small for fast
retransmit to work. [BPS98] shows that about 56% of all
Hurtig, et al. Expires May 1, 2012 [Page 2]
Internet-Draft TCP and SCTP RTO Restart October 2011
retransmissions sent by a busy web server are sent after RTO expiry.
While the experiments were not conducted using SACK [RFC2018], only
4% of the RTO-based retransmissions could have been avoided.
Applications have a low transmission rate when data is sent in
response to actions, or as a reaction to real life events. Typical
examples of such applications are stock trading systems, remote
computer operations and online games. What is special about this
class of applications is that they are time-dependant, and extra
latency can reduce the application service level [P09]. Although
such applications may represent a small amount of data sent on the
network, a considerable number of flows have such properties and the
importance of low latency is high.
To enable timely loss recovery in the above situations a number of
proposals have been made. The limited transmit mechanism [RFC3042]
allows a TCP sender to transmit a previously unsent segment for each
of the first two duplicate acknowledgments. By transmitting new
segments, the sender attempts to generate additional duplicate
acknowledgments to enable fast retransmit. However, the limited
transmit algorithm does not help if no previously unset data is ready
for transmission or if the receiver is out of buffer space.
[RFC5827] specifies an early retransmit algorithm to enable fast loss
recovery in such situations. By dynamically lowering the amount of
duplicate acknowledgments needed for fast retransmit (dupthresh),
based on the number of outstanding segments, a smaller number of
duplicate acknowledgments are needed to trigger a retransmission. In
some situations, however, the algorithm is of no use or might not
work properly. First, if a single segment is outstanding, and lost,
it is impossible to use early retransmit. Second, if the network
path reorders segments, the algorithm might cause more unnecessary
retransmissions than fast retransmit.
The RTO restart approach outlined in this document makes the RTO
slightly more aggressive when the number of outstanding segments is
small, in an attempt to enable faster loss recovery for all segments
while being robust to reordering.
While this document focuses on TCP, the described changes are also
valid for the Stream Control Transmission Protocol (SCTP) [RFC4960]
which has similar loss recovery and congestion control algorithms.
1.1. Requirements Language
The key words "MUST", "MUST NOT", "REQUIRED", "SHALL", "SHALL NOT",
"SHOULD", "SHOULD NOT", "RECOMMENDED", "MAY", and "OPTIONAL" in this
document are to be interpreted as described in RFC 2119 [RFC2119].
Hurtig, et al. Expires May 1, 2012 [Page 3]
Internet-Draft TCP and SCTP RTO Restart October 2011
2. RTO Restart Overview
The RTO management algorithm described in [RFC6298] recommends that
the retransmission timer is restarted when an acknowledgment (ACK)
that acknowledges new data is received and there is still outstanding
data. The restart is conducted to guarantee that unacknowledged
segments will be retransmitted after approximately RTO seconds.
However, by restarting the timer on each incoming acknowledgment,
retransmissions are not typically triggered RTO seconds after their
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 needed to trigger an RTO is equal to "RTO + RTT".
The restart approach is illustrated in Figure 1 where a TCP sender
transmits three segments to a receiver. The arrival of the first and
second segment triggers a delayed ACK [RFC1122], which restarts the
RTO timer at the sender. The RTO restart is performed approximately
one RTT after the transmission of the third segment. Thus, if the
third segment is lost, as indicated in Figure 1, the effective loss
detection time is "RTO + RTT" seconds. In some situations, the
effective loss detection time becomes even longer. Consider a
scenario where only two segments are outstanding. If the second
segment is lost, the time to expire the delayed ACK timer will also
be included in the effective loss detection time.
Sender Receiver
...
DATA [SEG 1] ----------------------> (ack delayed)
DATA [SEG 2] ----------------------> (send ack)
DATA [SEG 3] ----X /------- ACK
(restart RTO) <----------/
...
(RTO expiry)
DATA [SEG 3] ---------------------->
Figure 1: RTO restart example
During normal TCP bulk transfer the current RTO restart approach is
not a problem. Actually, as long as enough segments arrive at a
receiver to enable fast retransmit, RTO-based loss recovery should be
avoided. RTOs should only be used as a last resort, as they
drastically lower the congestion window compared to fast retransmit.
There are only a few situations where timeouts are appropriate, or
the only choice. For example, if the network is severely congested
and no segments arrive, RTO-based recovery should be used. In this
Hurtig, et al. Expires May 1, 2012 [Page 4]
Internet-Draft TCP and SCTP RTO Restart October 2011
situation, the time to recover from the loss(es) will not be the
performance bottleneck. Furthermore, for connections that do not
utilize enough capacity to enable fast retransmit, RTO is the only
choice. The time needed for loss detection in such scenarios can
become a serious performance bottleneck.
3. RTO Restart Algorithm
To enable faster loss recovery for connections that are unable to use
fast retransmit, an alternative RTO restart can be used. By
resetting the timer to "RTO - T_earliest", where T_earliest is the
time elapsed since the earliest outstanding segment was transmitted,
retransmissions will always occur after exactly RTO seconds. This
approach makes the RTO more aggressive than the standardized approach
in [RFC6298] but still conforms to the requirement in [RFC6298] that
segments must not be retransmitted earlier than RTO seconds after
their original transmission. Furthermore, the possible negative
impacts of a more aggressive RTO are less severe when the amount of
outstanding data is small. For example, if a spurious RTO is
performed the resulting congestion window will not be smaller than if
fast retransmit was used, as the window is already small. For larger
congestion windows the RTT sometimes increases within a congestion
window, due to the increased queueing demand of the flow. Making the
timer more aggressive could trigger spurious retransmissions in this
situation. However, as the restart is only conducted for small
congestion windows, it is less likely that such RTT fluctuations
would occur.
This document specifies the following update of step 5.3 in Section 5
of [RFC6298] (and a similar update in Section 6.3.2 of [RFC4960] for
SCTP):
When an ACK is received that acknowledges new data:
(1) Set T_earliest = 0.
(2) If the following two conditions hold:
(a) The number of outstanding segments is less than four.
(b) There is no unsent data ready for transmission or the
receiver's advertised window does not permit
transmission.
set T_earliest to the time elapsed since the earliest
outstanding segment was sent.
Hurtig, et al. Expires May 1, 2012 [Page 5]
Internet-Draft TCP and SCTP RTO Restart October 2011
(3) Restart the retransmission timer so that it will expire after
"RTO - T_earliest" seconds (for the current value of RTO).
The update requires TCP implementations to track the time elapsed
since the transmission of the earliest outstanding segment
(T_earliest). In practice, this requires the implementation to store
the transmission time of each unacknowledged segment. In modern
implementations like Linux, this is already done to easily conduct
RTT measurements.
There are several proposals that make use of a different dupthresh
than three. TCP-NCR [RFC4653] sets the dupthresh to three or more,
to better disambiguate reordered and lost segments. In addition, the
earlier mentioned early retransmit mechanism dynamically lowers the
dupthresh when the amount of outstanding data is small, to enable
faster loss recovery. The reasons why the RTO restart procedure
described in this document does not take dynamic dupthresh
considerations into account are twofold. First, if a larger
dupthresh is used, the RTO restart approach could be used when the
congestion window, and the amount of outstanding data, is larger.
However, in such situations the actual amount of outstanding data can
significantly impact the RTT of the connection, making it potentially
dangerous to be more aggressive. Second, if a smaller dupthresh is
used, the amount of outstanding data needed for a restart is smaller.
However, as the congestion window is already small, it does not
matter if a retransmission is due to a fast retransmit or an RTO.
The resulting congestion window will still be very small, and the
only difference is how quickly TCP infers segment loss.
4. Discussion
5. Related Work
The currently standardized algorithm has been shown to add at least
one RTT to the loss recovery process in TCP [LS00] and SCTP
[HB08][PBP09]. Applications that have strict timing requirements
(e.g. telephony signaling and gaming) rather than throughput
requirements may want to use a lower RTOmin than the standard of 1
second [RFC4166]. For such applications the modified restart
approach could be important as the RTT and also the delayed ACK timer
of receivers will be large components of the effective loss recovery
time. In [HB08] where an RTOmin of 100 ms was used, it was shown
that the total transfer time of a lost segment (including the
original transmission time and the loss recovery time) could be
reduced with up to 35% using the suggested approach.
Hurtig, et al. Expires May 1, 2012 [Page 6]
Internet-Draft TCP and SCTP RTO Restart October 2011
6. IANA Considerations
This memo includes no request to IANA.
7. Security Considerations
This document discusses a change in how to set the retransmission
timer's value when restarted. This change does not raise any new
security issues with TCP or SCTP.
8. References
8.1. Normative References
[RFC1122] Braden, R., "Requirements for Internet Hosts -
Communication Layers", STD 3, RFC 1122, October 1989.
[RFC2018] Mathis, M., Mahdavi, J., Floyd, S., and A. Romanow, "TCP
Selective Acknowledgment Options", RFC 2018, October 1996.
[RFC2119] Bradner, S., "Key words for use in RFCs to Indicate
Requirement Levels", BCP 14, RFC 2119, March 1997.
[RFC3042] Allman, M., Balakrishnan, H., and S. Floyd, "Enhancing
TCP's Loss Recovery Using Limited Transmit", RFC 3042,
January 2001.
[RFC4166] Coene, L. and J. Pastor-Balbas, "Telephony Signalling
Transport over Stream Control Transmission Protocol (SCTP)
Applicability Statement", RFC 4166, February 2006.
[RFC4653] Bhandarkar, S., Reddy, A., Allman, M., and E. Blanton,
"Improving the Robustness of TCP to Non-Congestion
Events", RFC 4653, August 2006.
[RFC4960] Stewart, R., "Stream Control Transmission Protocol",
RFC 4960, September 2007.
[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.
Hurtig, et al. Expires May 1, 2012 [Page 7]
Internet-Draft TCP and SCTP RTO Restart October 2011
8.2. Informative References
[BPS98] Balakrishnan, H., Padmanabhan, V., Seshan, S., Stemm, M.,
and R. Katz, "TCP Behavior of a Busy Web Server: Analysis
and Improvements", Proc. IEEE INFOCOM Conf., March 1998.
[EL04] Ekstroem, H. and R. Ludwig, "The Peak-Hopper: A New End-
to-End Retransmission Timer for Reliable Unicast
Transport", IEEE INFOCOM 2004, March 2004.
[HB08] Hurtig, P. and A. Brunstrom, "SCTP: designed for timely
message delivery?", Springer Telecommunication Systems,
May 2010.
[LS00] Ludwig, R. and K. Sklower, "The Eifel retransmission
timer", ACM SIGCOMM Comput. Commun. Rev., 30(3),
July 2000.
[P09] Petlund, A., "Improving latency for interactive, thin-
stream applications over reliable transport", Unipub PhD
Thesis, Oct 2009.
[PBP09] Petlund, A., Beskow, P., Pedersen, J., Paaby, E., Griwodz,
C., and P. Halvorsen, "Improving SCTP Retransmission
Delays for Time-Dependent Thin Streams",
Springer Multimedia Tools and Applications, 45(1-3), 2009.
[RJ10] Ramachandran, S., "Web metrics: Size and number of
resources", Google http://code.google.com/speed/articles/
web-metrics.html, May 2010.
Authors' Addresses
Per Hurtig
Karlstad University
Universitetsgatan 2
Karlstad, 651 88
Sweden
Phone: +46 54 700 23 35
Email: per.hurtig@kau.se
Hurtig, et al. Expires May 1, 2012 [Page 8]
Internet-Draft TCP and SCTP RTO Restart October 2011
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
Hurtig, et al. Expires May 1, 2012 [Page 9]
