| Rfc | 3155 |
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| Title | End-to-end Performance Implications of Links with Errors |
|---|
| Author | S.
Dawkins, G. Montenegro, M. Kojo, V. Magret, N. Vaidya |
|---|
| Date | August 2001 |
|---|
| Format: | TXT, HTML |
|---|
| Also | BCP0050 |
|---|
| Status: | BEST CURRENT PRACTICE |
|---|
|
Network Working Group S. Dawkins
Request for Comments: 3155 G. Montenegro
BCP: 50 M. Kojo
Category: Best Current Practice V. Magret
N. Vaidya
August 2001
End-to-end Performance Implications of Links with Errors
Status of this Memo
This document specifies an Internet Best Current Practices for the
Internet Community, and requests discussion and suggestions for
improvements. Distribution of this memo is unlimited.
Copyright Notice
Copyright (C) The Internet Society (2001). All Rights Reserved.
Abstract
This document discusses the specific TCP mechanisms that are
problematic in environments with high uncorrected error rates, and
discusses what can be done to mitigate the problems without
introducing intermediate devices into the connection.
Table of Contents
1.0 Introduction ............................................. 2
1.1 Should you be reading this recommendation? ........... 3
1.2 Relationship of this recommendation to PEPs ........... 4
1.3 Relationship of this recommendation to Link Layer
Mechanisms............................................. 4
2.0 Errors and Interactions with TCP Mechanisms .............. 5
2.1 Slow Start and Congestion Avoidance [RFC2581] ......... 5
2.2 Fast Retransmit and Fast Recovery [RFC2581] ........... 6
2.3 Selective Acknowledgements [RFC2018, RFC2883] ......... 7
3.0 Summary of Recommendations ............................... 8
4.0 Topics For Further Work .................................. 9
4.1 Achieving, and maintaining, large windows ............. 10
5.0 Security Considerations .................................. 11
6.0 IANA Considerations ...................................... 11
7.0 Acknowledgements ......................................... 11
References ................................................... 11
Authors' Addresses ........................................... 14
Full Copyright Statement ..................................... 16
1.0 Introduction
The rapidly-growing Internet is being accessed by an increasingly
wide range of devices over an increasingly wide variety of links. At
least some of these links do not provide the degree of reliability
that hosts expect, and this expansion into unreliable links causes
some Internet protocols, especially TCP [RFC793], to perform poorly.
Specifically, TCP congestion control [RFC2581], while appropriate for
connections that lose traffic primarily because of congestion and
buffer exhaustion, interacts badly with uncorrected errors when TCP
connections traverse links with high uncorrected error rates. The
result is that sending TCPs may spend an excessive amount of time
waiting for acknowledgement that do not arrive, and then, although
these losses are not due to congestion-related buffer exhaustion, the
sending TCP transmits at substantially reduced traffic levels as it
probes the network to determine "safe" traffic levels.
This document does not address issues with other transport protocols,
for example, UDP.
Congestion avoidance in the Internet is based on an assumption that
most packet losses are due to congestion. TCP's congestion avoidance
strategy treats the absence of acknowledgement as a congestion
signal. This has worked well since it was introduced in 1988 [VJ-
DCAC], because most links and subnets have relatively low error rates
in normal operation, and congestion is the primary cause of loss in
these environments. However, links and subnets that do not enjoy low
uncorrected error rates are becoming more prevalent in parts of the
Internet. In particular, these include terrestrial and satellite
wireless links. Users relying on traffic traversing these links may
see poor performance because their TCP connections are spending
excessive time in congestion avoidance and/or slow start procedures
triggered by packet losses due to transmission errors.
The recommendations in this document aim at improving utilization of
available path capacity over such high error-rate links in ways that
do not threaten the stability of the Internet.
Applications use TCP in very different ways, and these have
interactions with TCP's behavior [RFC2861]. Nevertheless, it is
possible to make some basic assumptions about TCP flows.
Accordingly, the mechanisms discussed here are applicable to all uses
of TCP, albeit in varying degrees according to different scenarios
(as noted where appropriate).
This recommendation is based on the explicit assumption that major
changes to the entire installed base of routers and hosts are not a
practical possibility. This constrains any changes to hosts that are
directly affected by errored links.
1.1 Should you be reading this recommendation?
All known subnetwork technologies provide an "imperfect" subnetwork
service - the bit error rate is non-zero. But there's no obvious way
for end stations to tell the difference between packets discarded due
to congestion and losses due to transmission errors.
If a directly-attached subnetwork is reporting transmission errors to
a host, these reports matter, but we can't rely on explicit
transmission error reports to both hosts.
Another way of deciding if a subnetwork should be considered to have
a "high error rate" is by appealing to mathematics.
An approximate formula for the TCP Reno response function is given in
[PFTK98]:
s
T = --------------------------------------------------
RTT*sqrt(2p/3) + tRTO*(3*sqrt(3p/8))*p*(1 + 32p**2)
where
T = the sending rate in bytes per second
s = the packet size in bytes
RTT = round-trip time in seconds
tRTO = TCP retransmit timeout value in seconds
p = steady-state packet loss rate
If one plugs in an observed packet loss rate, does the math and then
sees predicted bandwidth utilization that is greater than the link
speed, the connection will not benefit from recommendations in this
document, because the level of packet losses being encountered won't
affect the ability of TCP to utilize the link. If, however, the
predicted bandwidth is less than the link speed, packet losses are
affecting the ability of TCP to utilize the link.
If further investigation reveals a subnetwork with significant
transmission error rates, the recommendations in this document will
improve the ability of TCP to utilize the link.
A few caveats are in order, when doing this calculation:
(1) the RTT is the end-to-end RTT, not the link RTT.
(2) Max(1.0, 4*RTT) can be substituted as a simplification for
tRTO.
(3) losses may be bursty - a loss rate measured over an interval
that includes multiple bursty loss events may understate the
impact of these loss events on the sending rate.
1.2 Relationship of this recommendation to PEPs
This document discusses end-to-end mechanisms that do not require
TCP-level awareness by intermediate nodes. This places severe
limitations on what the end nodes can know about the nature of losses
that are occurring between the end nodes. Attempts to apply
heuristics to distinguish between congestion and transmission error
have not been successful [BV97, BV98, BV98a]. This restriction is
relaxed in an informational document on Performance Enhancing Proxies
(PEPs) [RFC3135]. Because PEPs can be placed on boundaries where
network characteristics change dramatically, PEPs have an additional
opportunity to improve performance over links with uncorrected
errors.
However, generalized use of PEPs contravenes the end-to-end principle
and is highly undesirable given their deleterious implications, which
include the following: lack of fate sharing (a PEP adds a third point
of failure besides the endpoints themselves), end-to-end reliability
and diagnostics, preventing end-to-end security (particularly network
layer security such as IPsec), mobility (handoffs are much more
complex because state must be transferred), asymmetric routing (PEPs
typically require being on both the forward and reverse paths of a
connection), scalability (PEPs add more state to maintain), QoS
transparency and guarantees.
Not every type of PEP has all the drawbacks listed above.
Nevertheless, the use of PEPs may have very serious consequences
which must be weighed carefully.
1.3 Relationship of this recommendation to Link Layer Mechanisms
This recommendation is for use with TCP over subnetwork technologies
(link layers) that have already been deployed. Subnetworks that are
intended to carry Internet protocols, but have not been completely
specified are the subject of a best common practices (BCP) document
which has been developed or is under development by the Performance
Implications of Link Characteristics WG (PILC) [PILC-WEB]. This last
document is aimed at designers who still have the opportunity to
reduce the number of uncorrected errors TCP will encounter.
2.0 Errors and Interactions with TCP Mechanisms
A TCP sender adapts its use of network path capacity based on
feedback from the TCP receiver. As TCP is not able to distinguish
between losses due to congestion and losses due to uncorrected
errors, it is not able to accurately determine available path
capacity in the presence of significant uncorrected errors.
2.1 Slow Start and Congestion Avoidance [RFC2581]
Slow Start and Congestion Avoidance [RFC2581] are essential to the
current stability of the Internet. These mechanisms were designed to
accommodate networks that do not provide explicit congestion
notification. Although experimental mechanisms such as [RFC2481] are
moving in the direction of explicit congestion notification, the
effect of ECN on ECN-aware TCPs is essentially the same as the effect
of implicit congestion notification through congestion-related loss,
except that ECN provides this notification before packets are lost,
and must then be retransmitted.
TCP connections experiencing high error rates on their paths interact
badly with Slow Start and with Congestion Avoidance, because high
error rates make the interpretation of losses ambiguous - the sender
cannot know whether detected losses are due to congestion or to data
corruption. TCP makes the "safe" choice and assumes that the losses
are due to congestion.
- Whenever sending TCPs receive three out-of-order
acknowledgement, they assume the network is mildly congested
and invoke fast retransmit/fast recovery (described below).
- Whenever TCP's retransmission timer expires, the sender assumes
that the network is congested and invokes slow start.
- Less-reliable link layers often use small link MTUs. This
slows the rate of increase in the sender's window size during
slow start, because the sender's window is increased in units
of segments. Small link MTUs alone don't improve reliability.
Path MTU discovery [RFC1191] must also be used to prevent
fragmentation. Path MTU discovery allows the most rapid
opening of the sender's window size during slow start, but a
number of round trips may still be required to open the window
completely.
Recommendation: Any standards-conformant TCP will implement Slow
Start and Congestion Avoidance, which are MUSTs in STD 3 [RFC1122].
Recommendations in this document will not interfere with these
mechanisms.
2.2 Fast Retransmit and Fast Recovery [RFC2581]
TCP provides reliable delivery of data as a byte-stream to an
application, so that when a segment is lost (whether due to either
congestion or transmission loss), the receiver TCP implementation
must wait to deliver data to the receiving application until the
missing data is received. The receiver TCP implementation detects
missing segments by segments arriving with out-of-order sequence
numbers.
TCPs should immediately send an acknowledgement when data is received
out-of-order [RFC2581], providing the next expected sequence number
with no delay, so that the sender can retransmit the required data as
quickly as possible and the receiver can resume delivery of data to
the receiving application. When an acknowledgement carries the same
expected sequence number as an acknowledgement that has already been
sent for the last in-order segment received, these acknowledgement
are called "duplicate ACKs".
Because IP networks are allowed to reorder packets, the receiver may
send duplicate acknowledgments for segments that arrive out of order
due to routing changes, link-level retransmission, etc. When a TCP
sender receives three duplicate ACKs, fast retransmit [RFC2581]
allows it to infer that a segment was lost. The sender retransmits
what it considers to be this lost segment without waiting for the
full retransmission timeout, thus saving time.
After a fast retransmit, a sender halves its congestion window and
invokes the fast recovery [RFC2581] algorithm, whereby it invokes
congestion avoidance from a halved congestion window, but does not
invoke slow start from a one-segment congestion window as it would do
after a retransmission timeout. As the sender is still receiving
dupacks, it knows the receiver is receiving packets sent, so the full
reduction after a timeout when no communication has been received is
not called for. This relatively safe optimization also saves time.
It is important to be realistic about the maximum throughput that TCP
can have over a connection that traverses a high error-rate link. In
general, TCP will increase its congestion window beyond the delay-
bandwidth product. TCP's congestion avoidance strategy is additive-
increase, multiplicative-decrease, which means that if additional
errors are encountered before the congestion window recovers
completely from a 50-percent reduction, the effect can be a "downward
spiral" of the congestion window due to additional 50-percent
reductions. Even using Fast Retransmit/Fast Recovery, the sender
will halve the congestion window each time a window contains one or
more segments that are lost, and will re-open the window by one
additional segment for each congestion window's worth of
acknowledgement received.
If a connection's path traverses a link that loses one or more
segments during this recovery period, the one-half reduction takes
place again, this time on a reduced congestion window - and this
downward spiral will continue to hold the congestion window below
path capacity until the connection is able to recover completely by
additive increase without experiencing loss.
Of course, no downward spiral occurs if the error rate is constantly
high and the congestion window always remains small; the
multiplicative-increase "slow start" will be exited early, and the
congestion window remains low for the duration of the TCP connection.
In links with high error rates, the TCP window may remain rather
small for long periods of time.
Not all causes of small windows are related to errors. For example,
HTTP/1.0 commonly closes TCP connections to indicate boundaries
between requested resources. This means that these applications are
constantly closing "trained" TCP connections and opening "untrained"
TCP connections which will execute slow start, beginning with one or
two segments. This can happen even with HTTP/1.1, if webmasters
configure their HTTP/1.1 servers to close connections instead of
waiting to see if the connection will be useful again.
A small window - especially a window of less than four segments -
effectively prevents the sender from taking advantage of Fast
Retransmits. Moreover, efficient recovery from multiple losses
within a single window requires adoption of new proposals (NewReno
[RFC2582]).
Recommendation: Implement Fast Retransmit and Fast Recovery at this
time. This is a widely-implemented optimization and is currently at
Proposed Standard level. [RFC2488] recommends implementation of Fast
Retransmit/Fast Recovery in satellite environments.
2.3 Selective Acknowledgements [RFC2018, RFC2883]
Selective Acknowledgements [RFC2018] allow the repair of multiple
segment losses per window without requiring one (or more) round-trips
per loss.
[RFC2883] proposes a minor extension to SACK that allows receiving
TCPs to provide more information about the order of delivery of
segments, allowing "more robust operation in an environment of
reordered packets, ACK loss, packet replication, and/or early
retransmit timeouts". Unless explicitly stated otherwise, in this
document, "Selective Acknowledgements" (or "SACK") refers to the
combination of [RFC2018] and [RFC2883].
Selective acknowledgments are most useful in LFNs ("Long Fat
Networks") because of the long round trip times that may be
encountered in these environments, according to Section 1.1 of
[RFC1323], and are especially useful if large windows are required,
because there is a higher probability of multiple segment losses per
window.
On the other hand, if error rates are generally low but occasionally
higher due to channel conditions, TCP will have the opportunity to
increase its window to larger values during periods of improved
channel conditions between bursts of errors. When bursts of errors
occur, multiple losses within a window are likely to occur. In this
case, SACK would provide benefits in speeding the recovery and
preventing unnecessary reduction of the window size.
Recommendation: Implement SACK as specified in [RFC2018] and updated
by [RFC2883], both Proposed Standards. In cases where SACK cannot be
enabled for both sides of a connection, TCP senders may use NewReno
[RFC2582] to better handle partial ACKs and multiple losses within a
single window.
3.0 Summary of Recommendations
The Internet does not provide a widely-available loss feedback
mechanism that allows TCP to distinguish between congestion loss and
transmission error. Because congestion affects all traffic on a path
while transmission loss affects only the specific traffic
encountering uncorrected errors, avoiding congestion has to take
precedence over quickly repairing transmission errors. This means
that the best that can be achieved without new feedback mechanisms is
minimizing the amount of time that is spent unnecessarily in
congestion avoidance.
The Fast Retransmit/Fast Recovery mechanism allows quick repair of
loss without giving up the safety of congestion avoidance. In order
for Fast Retransmit/Fast Recovery to work, the window size must be
large enough to force the receiver to send three duplicate
acknowledgments before the retransmission timeout interval expires,
forcing full TCP slow-start.
Selective Acknowledgements (SACK) extend the benefit of Fast
Retransmit/Fast Recovery to situations where multiple segment losses
in the window need to be repaired more quickly than can be
accomplished by executing Fast Retransmit for each segment loss, only
to discover the next segment loss.
These mechanisms are not limited to wireless environments. They are
usable in all environments.
4.0 Topics For Further Work
"Limited Transmit" [RFC3042] has been specified as an optimization
extending Fast Retransmit/Fast Recovery for TCP connections with
small congestion windows that will not trigger three duplicate
acknowledgments. This specification is deemed safe, and it also
provides benefits for TCP connections that experience a large amount
of packet (data or ACK) loss. Implementors should evaluate this
standards track specification for TCP in loss environments.
Delayed Duplicate Acknowledgements [MV97, VMPM99] attempts to prevent
TCP-level retransmission when link-level retransmission is still in
progress, adding additional traffic to the network. This proposal is
worthy of additional study, but is not recommended at this time,
because we don't know how to calculate appropriate amounts of delay
for an arbitrary network topology.
It is not possible to use explicit congestion notification [RFC2481]
as a surrogate for explicit transmission error notification (no
matter how much we wish it was!). Some mechanism to provide explicit
notification of transmission error would be very helpful. This might
be more easily provided in a PEP environment, especially when the PEP
is the "first hop" in a connection path, because current checksum
mechanisms do not distinguish between transmission error to a payload
and transmission error to the header. Furthermore, if the header is
damaged, sending explicit transmission error notification to the
right endpoint is problematic.
Losses that take place on the ACK stream, especially while a TCP is
learning network characteristics, can make the data stream quite
bursty (resulting in losses on the data stream, as well). Several
ways of limiting this burstiness have been proposed, including TCP
transmit pacing at the sender and ACK rate control within the
network.
"Appropriate Byte Counting" (ABC) [ALL99], has been proposed as a way
of opening the congestion window based on the number of bytes that
have been successfully transfered to the receiver, giving more
appropriate behavior for application protocols that initiate
connections with relatively short packets. For SMTP [RFC2821], for
instance, the client might send a short HELO packet, a short MAIL
packet, one or more short RCPT packets, and a short DATA packet -
followed by the entire mail body sent as maximum-length packets. An
ABC TCP sender would not use ACKs for each of these short packets to
increase the congestion window to allow additional full-length
packets. ABC is worthy of additional study, but is not recommended
at this time, because ABC can lead to increased burstiness when
acknowledgments are lost.
4.1 Achieving, and maintaining, large windows
The recommendations described in this document will aid TCPs in
injecting packets into ERRORed connections as fast as possible
without destabilizing the Internet, and so optimizing the use of
available bandwidth.
In addition to these TCP-level recommendations, there is still
additional work to do at the application level, especially with the
dominant application protocol on the World Wide Web, HTTP.
HTTP/1.0 (and earlier versions) closes TCP connections to signal a
receiver that all of a requested resource had been transmitted.
Because WWW objects tend to be small in size [MOGUL], TCPs carrying
HTTP/1.0 traffic experience difficulty in "training" on available
path capacity (a substantial portion of the transfer has already
happened by the time TCP exits slow start).
Several HTTP modifications have been introduced to improve this
interaction with TCP ("persistent connections" in HTTP/1.0, with
improvements in HTTP/1.1 [RFC2616]). For a variety of reasons, many
HTTP interactions are still HTTP/1.0-style - relatively short-lived.
Proposals which reuse TCP congestion information across connections,
like TCP Control Block Interdependence [RFC2140], or the more recent
Congestion Manager [BS00] proposal, will have the effect of making
multiple parallel connections impact the network as if they were a
single connection, "trained" after a single startup transient. These
proposals are critical to the long-term stability of the Internet,
because today's users always have the choice of clicking on the
"reload" button in their browsers and cutting off TCP's exponential
backoff - replacing connections which are building knowledge of the
available bandwidth with connections with no knowledge at all.
5.0 Security Considerations
A potential vulnerability introduced by Fast Retransmit/Fast Recovery
is (as pointed out in [RFC2581]) that an attacker may force TCP
connections to grind to a halt, or, more dangerously, behave more
aggressively. The latter possibility may lead to congestion
collapse, at least in some regions of the network.
Selective acknowledgments is believed to neither strengthen nor
weaken TCP's current security properties [RFC2018].
Given that the recommendations in this document are performed on an
end-to-end basis, they continue working even in the presence of end-
to-end IPsec. This is in direct contrast with mechanisms such as
PEP's which are implemented in intermediate nodes (section 1.2).
6.0 IANA Considerations
This document is a pointer to other, existing IETF standards. There
are no new IANA considerations.
7.0 Acknowledgements
This recommendation has grown out of RFC 2757, "Long Thin Networks",
which was in turn based on work done in the IETF TCPSAT working
group. The authors are indebted to the active members of the PILC
working group. In particular, Mark Allman and Lloyd Wood gave us
copious and insightful feedback, and Dan Grossman and Jamshid Mahdavi
provided text replacements.
References
[ALL99] M. Allman, "TCP Byte Counting Refinements," ACM Computer
Communication Review, Volume 29, Number 3, July 1999.
http://www.acm.org/sigcomm/ccr/archive/ccr-toc/ccr-toc-
99.html
[BS00] Balakrishnan, H. and S. Seshan, "The Congestion Manager",
RFC 3124, June 2001.
[BV97] S. Biaz and N. Vaidya, "Using End-to-end Statistics to
Distinguish Congestion and Corruption Losses: A Negative
Result," Texas A&M University, Technical Report 97-009,
August 18, 1997.
[BV98] S. Biaz and N. Vaidya, "Sender-Based heuristics for
Distinguishing Congestion Losses from Wireless
Transmission Losses," Texas A&M University, Technical
Report 98-013, June 1998.
[BV98a] S. Biaz and N. Vaidya, "Discriminating Congestion Losses
from Wireless Losses using Inter-Arrival Times at the
Receiver," Texas A&M University, Technical Report 98-014,
June 1998.
[MOGUL] "The Case for Persistent-Connection HTTP", J. C. Mogul,
Research Report 95/4, May 1995. Available as
http://www.research.digital.com/wrl/techreports/abstracts/
95.4.html
[MV97] M. Mehta and N. Vaidya, "Delayed Duplicate-
Acknowledgements: A Proposal to Improve Performance of
TCP on Wireless Links," Texas A&M University, December 24,
1997. Available at
http://www.cs.tamu.edu/faculty/vaidya/mobile.html
[PILC-WEB] http://pilc.grc.nasa.gov/
[PFTK98] Padhye, J., Firoiu, V., Towsley, D. and J.Kurose, "TCP
Throughput: A simple model and its empirical validation",
SIGCOMM Symposium on Communications Architectures and
Protocols, August 1998.
[RFC793] Postel, J., "Transmission Control Protocol", STD 7, RFC
793, September 1981.
[RFC2821] Klensin, J., Editor, "Simple Mail Transfer Protocol", RFC
2821, April 2001.
[RFC1122] Braden, R., "Requirements for Internet Hosts --
Communication Layers", STD 3, RFC 1122, October 1989.
[RFC1191] Mogul J., and S. Deering, "Path MTU Discovery", RFC 1191,
November 1990.
[RFC1323] Jacobson, V., Braden, R. and D. Borman. "TCP Extensions
for High Performance", RFC 1323, May 1992.
[RFC2018] Mathis, M., Mahdavi, J., Floyd, S. and A. Romanow "TCP
Selective Acknowledgment Options", RFC 2018, October 1996.
[RFC2140] Touch, J., "TCP Control Block Interdependence", RFC 2140,
April 1997.
[RFC2309] Braden, B., Clark, D., Crowcrfot, J., Davie, B., Deering,
S., Estrin, D., Floyd, S., Jacobson, V., Minshall, G.,
Partridge, C., Peterson, L., Ramakrishnan, K., Shecker,
S., Wroclawski, J. and L, Zhang, "Recommendations on Queue
Management and Congestion Avoidance in the Internet", RFC
2309, April 1998.
[RFC2481] Ramakrishnan K. and S. Floyd, "A Proposal to add Explicit
Congestion Notification (ECN) to IP", RFC 2481, January
1999.
[RFC2488] Allman, M., Glover, D. and L. Sanchez. "Enhancing TCP Over
Satellite Channels using Standard Mechanisms", BCP 28, RFC
2488, January 1999.
[RFC2581] Allman, M., Paxson, V. and W. Stevens, "TCP Congestion
Control", RFC 2581, April 1999.
[RFC2582] Floyd, S. and T. Henderson, "The NewReno Modification to
TCP's Fast Recovery Algorithm", RFC 2582, April 1999.
[RFC2616] Fielding, R., Gettys, J., Mogul, J., Frystyk, H.,
Masinter, L., Leach P. and T. Berners-Lee, "Hypertext
Transfer Protocol -- HTTP/1.1", RFC 2616, June 1999.
[RFC2861] Handley, H., Padhye, J. and S., Floyd, "TCP Congestion
Window Validation", RFC 2861, June 2000.
[RFC2883] Floyd, S., Mahdavi, M., Mathis, M. and M. Podlosky, "An
Extension to the Selective Acknowledgement (SACK) Option
for TCP", RFC 2883, August 1999.
[RFC2923] Lahey, K., "TCP Problems with Path MTU Discovery", RFC
2923, September 2000.
[RFC3042] Allman, M., Balakrishnan, H. and S. Floyd, "Enhancing
TCP's Loss Recovery Using Limited Transmit", RFC 3042,
January, 2001.
[RFC3135] Border, J., Kojo, M., Griner, J., Montenegro, G. and Z.
Shelby, "Performance Enhancing Proxies Intended to
Mitigate Link-Related Degradations", RFC 3135, June 2001.
[VJ-DCAC] Jacobson, V., "Dynamic Congestion Avoidance / Control" e-
mail dated February 11, 1988, available from
http://www.kohala.com/~rstevens/vanj.88feb11.txt
[VMPM99] N. Vaidya, M. Mehta, C. Perkins, and G. Montenegro,
"Delayed Duplicate Acknowledgements: A TCP-Unaware
Approach to Improve Performance of TCP over Wireless,"
Technical Report 99-003, Computer Science Dept., Texas A&M
University, February 1999. Also, to appear in Journal of
Wireless Communications and Wireless Computing (Special
Issue on Reliable Transport Protocols for Mobile
Computing).
Authors' Addresses
Questions about this document may be directed to:
Spencer Dawkins
Fujitsu Network Communications
2801 Telecom Parkway
Richardson, Texas 75082
Phone: +1-972-479-3782
EMail: spencer.dawkins@fnc.fujitsu.com
Gabriel E. Montenegro
Sun Microsystems
Laboratories, Europe
29, chemin du Vieux Chene
38240 Meylan
FRANCE
Phone: +33 476 18 80 45
EMail: gab@sun.com
Markku Kojo
Department of Computer Science
University of Helsinki
P.O. Box 26 (Teollisuuskatu 23)
FIN-00014 HELSINKI
Finland
Phone: +358-9-1914-4179
EMail: kojo@cs.helsinki.fi
Vincent Magret
Alcatel Internetworking, Inc.
26801 W. Agoura road
Calabasas, CA, 91301
Phone: +1 818 878 4485
EMail: vincent.magret@alcatel.com
Nitin H. Vaidya
458 Coodinated Science Laboratory, MC-228
1308 West Main Street
Urbana, IL 61801
Phone: 217-265-5414
E-mail: nhv@crhc.uiuc.edu
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