Network Working Group S. Dawkins
Request for Comments: 3150 G. Montenegro
BCP: 48 M . Kojo
Category: Best Current Practice V. Magret
July 2001

End-to-end Performance Implications of Slow Links

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 makes performance-related recommendations for users of
network paths that traverse "very low bit-rate" links.

"Very low bit-rate" implies "slower than we would like". This
recommendation may be useful in any network where hosts can saturate
available bandwidth, but the design space for this recommendation
eXPlicitly includes connections that traverse 56 Kb/second modem
links or 4.8 Kb/second wireless Access links - both of which are
widely deployed.

This document discusses general-purpose mechanisms. Where
application-specific mechanisms can outperform the relevant general-
purpose mechanism, we point this out and explain why.

This document has some recommendations in common with RFC2689,
"Providing integrated services over low-bitrate links", especially in
areas like header compression. This document focuses more on
traditional data applications for which "best-effort delivery" is
appropriate.

Table of Contents

1.0 IntrodUCtion ................................................. 2
2.0 Description of Optimizations ................................. 3
2.1 Header Compression Alternatives ...................... 3
2.2 Payload Compression Alternatives ..................... 5
2.3 Choosing MTU sizes ................................... 5
2.4 Interactions with TCP Congestion Control [RFC2581] ... 6
2.5 TCP Buffer Auto-tuning ............................... 9
2.6 Small Window Effects ................................. 10
3.0 Summary of Recommended Optimizations ......................... 10
4.0 Topics For Further Work ...................................... 12
5.0 Security Considerations ...................................... 12
6.0 IANA Considerations .......................................... 13
7.0 Acknowledgements ............................................. 13
8.0 References ................................................... 13
Authors' Addresses ............................................... 16
Full Copyright Statement ......................................... 17

1.0 Introduction

The Internet protocol stack was designed to operate in a wide range
of link speeds, and has met this design goal with only a limited
number of enhancements (for example, the use of TCP window scaling as
described in "TCP Extensions for High Performance" [RFC1323] for
very-high-bandwidth connections).

Pre-World Wide Web application protocols tended to be either
interactive applications sending very little data (e.g., Telnet) or
bulk transfer applications that did not require interactive response
(e.g., File Transfer Protocol, Network News). The World Wide Web has
given us traffic that is both interactive and often "bulky",
including images, sound, and video.

The World Wide Web has also popularized the Internet, so that there
is significant interest in accessing the Internet over link speeds
that are much "slower" than typical Office network speeds. In fact,
a significant proportion of the current Internet users is connected
to the Internet over a relatively slow last-hop link. In future, the
number of such users is likely to increase rapidly as various mobile
devices are foreseen to to be attached to the Internet over slow
wireless links.

In order to provide the best interactive response for these "bulky"
transfers, implementors may wish to minimize the number of bits
actually transmitted over these "slow" connections. There are two

areas that can be considered - compressing the bits that make up the
overhead associated with the connection, and compressing the bits
that make up the payload being transported over the connection.

In addition, implementors may wish to consider TCP receive window
settings and queuing mechanisms as techniques to improve performance
over low-speed links. While these techniques do not involve protocol
changes, they are included in this document for completeness.

2.0 Description of Optimizations

This section describes optimizations which have been suggested for
use in situations where hosts can saturate their links. The next
section summarizes recommendations about the use of these
optimizations.

2.1 Header Compression Alternatives

Mechanisms for TCP and IP header compression defined in [RFC1144,
RFC2507, RFC2508, RFC2509, RFC3095] provide the following benefits:

- Improve interactive response time

- Decrease header overhead (for a typical dialup MTU of 296
bytes, the overhead of TCP/IP headers can decrease from about
13 percent with typical 40-byte headers to 1-1.5 percent with
with 3-5 byte compressed headers, for most packets). This
enables use of small packets for delay-sensitive low data-rate
traffic and good line efficiency for bulk data even with small
segment sizes (for reasons to use a small MTU on slow links,
see section 2.3)

- Many slow links today are wireless and tend to be significantly
lossy. Header compression reduces packet loss rate over lossy
links (simply because shorter transmission times expose packets
to fewer events that cause loss).

[RFC1144] header compression is a Proposed Standard for TCP Header
compression that is widely deployed. Unfortunately it is vulnerable
on lossy links, because even a single bit error results in loss of
synchronization between the compressor and decompressor. It uses TCP
timeouts to detect a loss of such synchronization, but these errors
result in loss of data (up to a full TCP window), delay of a full
RTO, and unnecessary slow-start.

A more recent header compression proposal [RFC2507] includes an
explicit request for retransmission of an uncompressed packet to
allow resynchronization without waiting for a TCP timeout (and
executing congestion avoidance procedures). This works much better
on links with lossy characteristics.

The above scheme ceases to perform well under conditions as extreme
as those of many cellular links (error conditions of 1e-3 or 1e-2 and
round trip times over 100 ms.). For these cases, the 'Robust Header
Compression' working group has developed ROHC [RFC3095]. Extensions
of ROHC to support compression of TCP headers are also under
development.

[RFC1323] defines a "TCP Timestamp" option, used to prevent
"wrapping" of the TCP sequence number space on high-speed links, and
to improve TCP RTT estimates by providing unambiguous TCP roundtrip
timings. Use of TCP timestamps prevents header compression, because
the timestamps are sent as TCP options. This means that each
timestamped header has TCP options that differ from the previous
header, and headers with changed TCP options are always sent
uncompressed. In addition, timestamps do not seem to have much of an
impact on RTO estimation [AlPa99].

Nevertheless, the ROHC working group is developing schemes to
compress TCP headers, including options such as timestamps and
selective acknowledgements.

Recommendation: Implement [RFC2507], in particular as it relates to
IPv4 tunnels and Minimal Encapsulation for Mobile IP, as well as TCP
header compression for lossy links and links that reorder packets.
PPP capable devices should implement "IP Header Compression over PPP"
[RFC2509]. Robust Header Compression [RFC3095] is recommended for
extremely slow links with very high error rates (see above), but
implementors should judge if its complexity is justified (perhaps by
the cost of the radio frequency resources).

[RFC1144] header compression should only be enabled when operating
over reliable "slow" links.

Use of TCP Timestamps [RFC1323] is not recommended with these
connections, because it complicates header compression. Even though
the Robust Header Compression (ROHC) working group is developing
specifications to remedy this, those mechanisms are not yet fully
developed nor deployed, and may not be generally justifiable.
Furthermore, connections traversing "slow" links do not require
protection against TCP sequence-number wrapping.

2.2 Payload Compression Alternatives

Compression of IP payloads is also desirable on "slow" network links.
"IP Payload Compression Protocol (IPComp)" [RFC2393] defines a
framework where common compression algorithms can be applied to
arbitrary IP segment payloads.

IP payload compression is something of a niche optimization. It is
necessary because IP-level security converts IP payloads to random
bitstreams, defeating commonly-deployed link-layer compression
mechanisms which are faced with payloads that have no redundant
"information" that can be more compactly represented.

However, many IP payloads are already compressed (images, audio,
video, "zipped" files being transferred), or are already encrypted
above the IP layer (e.g., SSL [SSL]/TLS [RFC2246]). These payloads
will not "compress" further, limiting the benefit of this
optimization.

For uncompressed HTTP payload types, HTTP/1.1 [RFC2616] also includes
Content-Encoding and Accept-Encoding headers, supporting a variety of
compression algorithms for common compressible MIME types like
text/plain. This leaves only the HTTP headers themselves
uncompressed.

In general, application-level compression can often outperform
IPComp, because of the opportunity to use compression dictionaries
based on knowledge of the specific data being compressed.

Extensive use of application-level compression techniques will reduce
the need for IPComp, especially for WWW users.

Recommendation: IPComp may optionally be implemented.

2.3 Choosing MTU Sizes

There are several points to keep in mind when choosing an MTU for
low-speed links.

First, if a full-length MTU occupies a link for longer than the
delayed ACK timeout (typically 200 milliseconds, but may be up to 500
milliseconds), this timeout will cause an ACK to be generated for
every segment, rather than every second segment, as occurs with most
implementations of the TCP delayed ACK algorithm.

Second, "relatively large" MTUs, which take human-perceptible amounts
of time to be transmitted into the network, create human-perceptible
delays in other flows using the same link. [RFC1144] considers
100-200 millisecond delays as human-perceptible. The convention of
choosing 296-byte MTUs (with header compression enabled) for dialup
access is a compromise that limits the maximum link occupancy delay
with full-length MTUs close to 200 milliseconds on 9.6 Kb/second
links.

Third, on last-hop links using a larger link MTU size, and therefore
larger MSS, would allow a TCP sender to increase its congestion
window faster in bytes than when using a smaller MTU size (and a
smaller MSS). However, with a smaller MTU size, and a smaller MSS
size, the congestion window, when measured in segments, increases
more quickly than it would with a larger MSS size. Connections using
smaller MSS sizes are more likely to be able to send enough segments
to generate three duplicate acknowledgements, triggering fast
retransmit/fast recovery when packet losses are encountered. Hence,
a smaller MTU size is useful for slow links with lossy
characteristics.

Fourth, using a smaller MTU size also decreases the queuing delay of
a TCP flow (and thereby RTT) compared to use of larger MTU size with
the same number of packets in a queue. This means that a TCP flow
using a smaller segment size and traversing a slow link is able to
inflate the congestion window (in number of segments) to a larger
value while experiencing the same queuing delay.

Finally, some networks charge for traffic on a per-packet basis, not
on a per-kilobyte basis. In these cases, connections using a larger
MTU may be charged less than connections transferring the same number
of bytes using a smaller MTU.

Recommendation: If it is possible to do so, MTUs should be chosen
that do not monopolize network interfaces for human-perceptible
amounts of time, and implementors should not chose MTUs that will
occupy a network interface for significantly more than 100-200
milliseconds.

2.4 Interactions with TCP Congestion Control [RFC2581]

In many cases, TCP connections that traverse slow links have the slow
link as an "access" link, with higher-speed links in use for most of
the connection path. One common configuration might be a laptop
computer using dialup access to a terminal server (a last-hop
router), with an HTTP server on a high-speed LAN "behind" the
terminal server.

In this case, the HTTP server may be able to place packets on its
directly-attached high-speed LAN at a higher rate than the last-hop
router can forward them on the low-speed link. When the last-hop
router falls behind, it will be unable to buffer the traffic intended
for the low-speed link, and will become a point of congestion and
begin to drop the excess packets. In particular, several packets may
be dropped in a single transmission window when initial slow start
overshoots the last-hop router buffer.

Although packet loss is occurring, it isn't detected at the TCP
sender until one RTT time after the router buffer space is exhausted
and the first packet is dropped. This late congestion signal allows
the congestion window to increase up to double the size it was at the
time the first packet was dropped at the router.

If the link MTU is large enough to take more than the delayed ACK
timeout interval to transmit a packet, an ACK is sent for every
segment and the congestion window is doubled in a single RTT. If a
smaller link MTU is in use and delayed ACKs can be utilized, the
congestion window increases by a factor of 1.5 in one RTT. In both
cases the sender continues transmitting packets well beyond the
congestion point of the last-hop router, resulting in multiple packet
losses in a single window.

The self-clocking nature of TCP's slow start and congestion avoidance
algorithms prevent this buffer overrun from continuing. In addition,
these algorithms allow senders to "probe" for available bandwidth -
cycling through an increasing rate of transmission until loss occurs,
followed by a dramatic (50-percent) drop in transmission rate. This
happens when a host directly connected to a low-speed link offers an
advertised window that is unrealistically large for the low-speed
link. During the congestion avoidance phase the peer host continues
to probe for available bandwidth, trying to fill the advertised
window, until packet loss occurs.

The same problems may also exist when a sending host is directly
connected to a slow link as most slow links have some local buffer in
the link interface. This link interface buffer is subject to
overflow exactly in the same way as the last-hop router buffer.

When a last-hop router with a small number of buffers per outbound
link is used, the first buffer overflow occurs earlier than it would
if the router had a larger number of buffers. Subsequently with a
smaller number of buffers the periodic packet losses occur more
frequently during congestion avoidance, when the sender probes for
available bandwidth.

The most important responsibility of router buffers is to absorb
bursts. Too few buffers (for example, only three buffers per
outbound link as described in [RFC2416]) means that routers will
overflow their buffer pools very easily and are unlikely to absorb
even a very small burst. When a larger number of router buffers are
allocated per outbound link, the buffer space does not overflow as
quickly but the buffers are still likely to become full due to TCP's
default behavior. A larger number of router buffers leads to longer
queuing delays and a longer RTT.

If router queues become full before congestion is signaled or remain
full for long periods of time, this is likely to result in "lock-
out", where a single connection or a few connections occupy the
router queue space, preventing other connections from using the link
[RFC2309], especially when a tail drop queue management discipline is
being used.

Therefore, it is essential to have a large enough number of buffers
in routers to be able to absorb data bursts, but keep the queues
normally small. In order to achieve this it has been recommended in
[RFC2309] that an active queue management mechanism, like Random
Early Detection (RED) [RED93], should be implemented in all Internet
routers, including the last-hop routers in front of a slow link. It
should also be noted that RED requires a sufficiently large number of
router buffers to work properly. In addition, the appropriate
parameters of RED on a last-hop router connected to a slow link will
likely deviate from the defaults recommended.

Active queue management mechanism do not eliminate packet drops but,
instead, drop packets at earlier stage to solve the full-queue
problem for flows that are responsive to packet drops as congestion
signal. Hosts that are directly connected to low-speed links may
limit the receive windows they advertise in order to lower or
eliminate the number of packet drops in a last-hop router. When
doing so one should, however, take care that the advertised window is
large enough to allow full utilization of the last-hop link capacity
and to allow triggering fast retransmit, when a packet loss is
encountered. This recommendation takes two forms:

- Modern operating systems use relatively large default TCP receive
buffers compared to what is required to fully utilize the link
capacity of low-speed links. Users should be able to choose the
default receive window size in use - typically a system-wide
parameter. (This "choice" may be as simple as "dial-up access/LAN
access" on a dialog box - this would accommodate many environments
without requiring hand-tuning by experienced network engineers.)

- Application developers should not attempt to manually manage
network bandwidth using socket buffer sizes. Only in very rare
circumstances will an application actually know both the bandwidth
and delay of a path and be able to choose a suitably low (or high)
value for the socket buffer size to oBTain good network
performance.

This recommendation is not a general solution for any network path
that might involve a slow link. Instead, this recommendation is
applicable in environments where the host "knows" it is always
connected to other hosts via "slow links". For hosts that may
connect to other host over a variety of links (e.g., dial-up laptop
computers with LAN-connected docking stations), buffer auto-tuning
for the receive buffer is a more reasonable recommendation, and is
discussed below.

2.5 TCP Buffer Auto-tuning

[SMM98] recognizes a tension between the desire to allocate "large"
TCP buffers, so that network paths are fully utilized, and a desire
to limit the amount of memory dedicated to TCP buffers, in order to
efficiently support large numbers of connections to hosts over
network paths that may vary by six orders of magnitude.

The technique proposed is to dynamically allocate TCP buffers, based
on the current congestion window, rather than attempting to
preallocate TCP buffers without any knowledge of the network path.

This proposal results in receive buffers that are appropriate for the
window sizes in use, and send buffers large enough to contain two
windows of segments, so that SACK and fast recovery can recover
losses without forcing the connection to use lengthy retransmission
timeouts.

While most of the motivation for this proposal is given from a
server's perspective, hosts that connect using multiple interfaces
with markedly-different link speeds may also find this kind of
technique useful. This is true in particular with slow links, which
are likely to dominate the end-to-end RTT. If the host is connected
only via a single slow link interface at a time, it is fairly easy to
(dynamically) adjust the receive window (and thus the advertised
window) to a value appropriate for the slow last-hop link with known
bandwidth and delay characteristics.

Recommendation: If a host is sometimes connected via a slow link but
the host is also connected using other interfaces with markedly-
different link speeds, it may use receive buffer auto-tuning to
adjust the advertised window to an appropriate value.

2.6 Small Window Effects

If a TCP connection stabilizes with a congestion window of only a few
segments (as could be expected on a "slow" link), the sender isn't
sending enough segments to generate three duplicate acknowledgements,
triggering fast retransmit and fast recovery. This means that a
retransmission timeout is required to repair the loss - dropping the
TCP connection to a congestion window with only one segment.

[TCPB98] and [TCPF98] observe that (in studies of network trace
datasets) it is relatively common for TCP retransmission timeouts to
occur even when some duplicate acknowledgements are being sent. The
challenge is to use these duplicate acknowledgements to trigger fast
retransmit/fast recovery without injecting traffic into the network
unnecessarily - and especially not injecting traffic in ways that
will result in instability.

The "Limited Transmit" algorithm [RFC3042] suggests sending a new
segment when the first and second duplicate acknowledgements are
received, so that the receiver is more likely to be able to continue
to generate duplicate acknowledgements until the TCP retransmit
threshold is reached, triggering fast retransmit and fast recovery.
When the congestion window is small, this is very useful in assisting
fast retransmit and fast recovery to recover from a packet loss
without using a retransmission timeout. We note that a maximum of
two additional new segments will be sent before the receiver sends
either a new acknowledgement advancing the window or two additional
duplicate acknowledgements, triggering fast retransmit/fast recovery,
and that these new segments will be acknowledgement-clocked, not
back-to-back.

Recommendation: Limited Transmit should be implemented in all hosts.

3.0 Summary of Recommended Optimizations

This section summarizes our recommendations regarding the previous
standards-track mechanisms, for end nodes that are connected via a
slow link.

Header compression should be implemented. [RFC1144] header
compression can be enabled over robust network links. [RFC2507]
should be used over network connections that are expected to
experience loss due to corruption as well as loss due to congestion.
For extremely lossy and slow links, implementors should evaluate ROHC
[RFC3095] as a potential solution. [RFC1323] TCP timestamps must be
turned off because (1) their protection against TCP sequence number
wrapping is unjustified for slow links, and (2) they complicate TCP
header compression.

IP Payload Compression [RFC2393] should be implemented, although
compression at higher layers of the protocol stack (for example [RFC
2616]) may make this mechanism less useful.

For HTTP/1.1 environments, [RFC2616] payload compression should be
implemented and should be used for payloads that are not already
compressed.

Implementors should choose MTUs that don't monopolize network
interfaces for more than 100-200 milliseconds, in order to limit the
impact of a single connection on all other connections sharing the
network interface.

Use of active queue management is recommended on last-hop routers
that provide Internet access to host behind a slow link. In
addition, number of router buffers per slow link should be large
enough to absorb concurrent data bursts from more than a single flow.
To absorb concurrent data bursts from two or three TCP senders with a
typical data burst of three back-to-back segments per sender, at
least six (6) or nine (9) buffers are needed. Effective use of
active queue management is likely to require even larger number of
buffers.

Implementors should consider the possibility that a host will be
directly connected to a low-speed link when choosing default TCP
receive window sizes.

Application developers should not attempt to manually manage network
bandwidth using socket buffer sizes as only in very rare
circumstances an application will be able to choose a suitable value
for the socket buffer size to obtain good network performance.

Limited Transmit [RFC3042] should be implemented in all end hosts as
it assists in triggering fast retransmit when congestion window is
small.

All of the mechanisms described above are stable standards-track RFCs
(at Proposed Standard status, as of this writing).

In addition, implementors may wish to consider TCP buffer auto-
tuning, especially when the host system is likely to be used with a
wide variety of access link speeds. This is not a standards-track
TCP mechanism but, as it is an operating system implementation issue,
it does not need to be standardized.

Of the above mechanisms, only Header Compression (for IP and TCP) may
cease to work in the presence of end-to-end IPSEC. However,
[RFC3095] does allow compressing the ESP header.

4.0 Topics For Further Work

In addition to the standards-track mechanisms discussed above, there
are still opportunities to improve performance over low-speed links.

"Sending fewer bits" is an obvious response to slow link speeds. The
now-defunct HTTP-NG proposal [HTTP-NG] replaced the text-based HTTP
header representation with a binary representation for compactness.
However, HTTP-NG is not moving forward and HTTP/1.1 is not being
enhanced to include a more compact HTTP header representation.
Instead, the Wireless Application Protocol (WAP) Forum has opted for
the XML-based Wireless Session Protocol [WSP], which includes a
compact header encoding mechanism.

It would be nice to agree on a more compact header representation
that will be used by all WWW communities, not only the wireless WAN
community. Indeed, general XML content encodings have been proposed
[Millau], although they are not yet widely adopted.

We note that TCP options which change from segment to segment
effectively disable header compression schemes deployed today,
because there's no way to indicate that some fields in the header are
unchanged from the previous segment, while other fields are not. The
Robust Header Compression working group is developing such schemes
for TCP options such as timestamps and selective acknowledgements.
Hopefully, documents subsequent to [RFC3095] will define such
specifications.

Another effort worth following is that of 'Delta Encoding'. Here,
clients that request a slightly modified version of some previously
cached resource would receive a succinct description of the
differences, rather than the entire resource [HTTP-DELTA].

5.0 Security Considerations

All recommendations included in this document are stable standards-
track RFCs (at Proposed Standard status, as of this writing) or
otherwise do not suggest any changes to any protocol. With the
exception of Van Jacobson compression [RFC1144] and [RFC2507,
RFC2508, RFC2509], all other mechanisms are applicable to TCP
connections protected by end-to-end IPSec. This includes ROHC
[RFC3095], albeit partially, because even though it can compress the
outermost ESP header to some extent, encryption still renders any
payload data uncompressible (including any subsequent protocol
headers).

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 "Long Thin Networks" [RFC2757],
which in turn benefited from work done in the IETF TCPSAT working
group.

8.0 References

[AlPa99] Mark Allman and Vern Paxson, "On Estimating End-to-End
Network Path Properties", in ACM SIGCOMM 99 Proceedings,
1999.

[HTTP-DELTA] J. Mogul, et al., "Delta encoding in HTTP", Work in
Progress.

[HTTP-NG] Mike Spreitzer, Bill Janssen, "HTTP 'Next Generation'",
9th International WWW Conference, May, 2000. Also
available as: http://www.www9.org/w9cdrom/60/60.Html

[Millau] Marc Girardot, Neel Sundaresan, "Millau: an encoding
format for efficient representation and exchange of XML
over the Web", 9th International WWW Conference, May,
2000. Also available as:
http://www.www9.org/w9cdrom/154/154.html

[PAX97] Paxson, V., "End-to-End Internet Packet Dynamics", 1997,
in SIGCOMM 97 Proceedings, available as:
http://www.acm.org/sigcomm/ccr/archive/ccr-toc/ccr-toc-
97.html

[RED93] Floyd, S., and Jacobson, V., Random Early Detection
gateways for Congestion Avoidance, IEEE/ACM Transactions
on Networking, V.1 N.4, August 1993, pp. 397-413. Also
available from http://FTP.ee.lbl.gov/floyd/red.html.

[RFC1144] Jacobson, V., "Compressing TCP/IP Headers for Low-Speed
Serial Links", RFC1144, February 1990.

[RFC1323] Jacobson, V., Braden, R. and D. Borman, "TCP Extensions
for High Performance", RFC1323, May 1992.

[RFC2246] Dierks, T. and C. Allen, "The TLS Protocol: Version
1.0", RFC2246, January 1999.

[RFC2309] Braden, R., Clark, D., Crowcroft, J., Davie, B.,
Deering, S., Estrin, D., Floyd, S., Jacobson, V.,
Minshall, G., Partridge, C., Peterson, L., Ramakrishnan,
K., Shenker, S., Wroclawski, J. and L. Zhang,
"Recommendations on Queue Management and Congestion
Avoidance in the Internet", RFC2309, April 1998.

[RFC2393] Shacham, A., Monsour, R., Pereira, R. and M. Thomas, "IP
Payload Compression Protocol (IPComp)", RFC2393,
December 1998.

[RFC2401] Kent, S. and R. Atkinson, "Security Architecture for the
Internet Protocol", RFC2401, November 1998.

[RFC2416] Shepard, T. and C. Partridge, "When TCP Starts Up With
Four Packets Into Only Three Buffers", RFC2416,
September 1998.

[RFC2507] Degermark, M., Nordgren, B. and S. Pink, "IP Header
Compression", RFC2507, February 1999.

[RFC2508] Casner, S. and V. Jacobson. "Compressing IP/UDP/RTP
Headers for Low-Speed Serial Links", RFC2508, February
1999.

[RFC2509] Engan, M., Casner, S. and C. Bormann, "IP Header
Compression over PPP", RFC2509, February 1999.

[RFC2581] Allman, M., Paxson, V. and W. Stevens, "TCP Congestion
Control", RFC2581, 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", RFC2616, June 1999.

[RFC2757] Montenegro, G., Dawkins, S., Kojo, M., Magret, V., and
N. Vaidya, "Long Thin Networks", RFC2757, January 2000.

[RFC3042] Allman, M., Balakrishnan, H. and S. Floyd, "Enhancing
TCP's Loss Recovery Using Limited Transmit", RFC3042,
January 2001.

[RFC3095] Bormann, C., Burmeister, C., Degermark, M., Fukushima,
H., Hannu, H., Jonsson, L-E., Hakenberg, R., Koren, T.,
Le, K., Liu, Z., Martensson, A., Miyazaki, A., Svanbro,
K., Wiebke, T., Yoshimura, T. and H. Zheng, "RObust
Header Compression (ROHC): Framework and four Profiles:
RTP, UDP ESP and uncompressed", RFC3095, July 2001.

[SMM98] Jeffrey Semke, Matthew Mathis, and Jamshid Mahdavi,
"Automatic TCP Buffer Tuning", in ACM SIGCOMM 98
Proceedings 1998. Available from
http://www.acm.org/sigcomm/sigcomm98/tp/abs_26.html.

[SSL] Alan O. Freier, Philip Karlton, Paul C. Kocher, The SSL
Protocol: Version 3.0, March 1996. (Expired Internet-
Draft, available from
http://home.netscape.com/eng/ssl3/ssl-toc.html)

[TCPB98] Hari Balakrishnan, Venkata N. Padmanabhan, Srinivasan
Seshan, Mark Stemm, Randy H. Katz, "TCP Behavior of a
Busy Internet Server: Analysis and Improvements", IEEE
Infocom, March 1998. Available from:
http://www.cs.berkeley.edu/~hari/papers/infocom98.ps.gz

[TCPF98] Dong Lin and H.T. Kung, "TCP Fast Recovery Strategies:
Analysis and Improvements", IEEE Infocom, March 1998.
Available from:
http://www.eecs.harvard.edu/networking/papers/ infocom-
tcp-final-198.pdf

[WSP] Wireless Application Protocol Forum, "WAP Wireless
Session Protocol Specification", approved 4 May, 2000,
available from
http://www1.wapforum.org/tech/documents/WAP-203-WSP-
20000504-a.pdf. (informative reference).

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 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
Fax: +358-9-1914-4441
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

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