Internet Engineering Task Force S. Dawkins
INTERNET DRAFT G. Montenegro
M. Kojo
V. Magret
May 3, 2001
End-to-end Performance Implications of Slow Links
draft-ietf-pilc-slow-06.txt
Status of This Memo
This document is an Internet-Draft and is in full conformance
with all provisions of Section 10 of RFC 2026.
Comments should be submitted to the PILC mailing list at
pilc@grc.nasa.gov.
Distribution of this memo is unlimited.
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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
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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 RFC 2689,
"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.
Changes since last draft:
o some minor edits and some clarifications throughout the document
o added some MTU selection tradeoff material to section 2.3
o added more explanation in section 2.4 about the packet drop
behavior during initial slow start and full queues problem
o deleted Limited Transmit text in section 4.0 as a topic for
future work, as it is now a proposed standard
o deleted the Appendix
o added a new section on Limited Transmit as it is now a proposed
standard
o added section on Security Considerations
o updated contact information for Gab Montenegro
o added section on IANA Considerations
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Table of Contents
1.0 Introduction . . . . . . . . . . . . . . . . . . . . . . . . . . 4
2.0 Description of Optimizations . . . . . . . . . . . . . . . . . . 4
2.1 Header Compression Alternatives . . . . . . . . . . . . . . 4
2.2 Payload Compression Alternatives . . . . . . . . . . . . . 6
2.3 Choosing MTU sizes . . . . . . . . . . . . . . . . . . . . 6
2.4 Interactions with TCP Congestion Control [RFC2581] . . . . 7
2.5 TCP Buffer Auto-tuning . . . . . . . . . . . . . . . . . . 10
2.6 Small Window Effects . . . . . . . . . . . . . . . . . . . . . . 11
3.0 Summary of Recommended Optimizations . . . . . . . . . . . . . . 11
4.0 Topics For Further Work . . . . . . . . . . . . . . . . . . . . 13
5.0 Security Considerations . . . . . . . . . . . . . . . . . . . . 13
6.0 IANA Considerations . . . . . . . . . . . . . . . . . . . . . . 13
7.0 Acknowledgements . . . . . . . . . . . . . . . . . . . . . . . . 13
8.0 References . . . . . . . . . . . . . . . . . . . . . . . . . . . 13
Authors' addresses . . . . . . . . . . . . . . . . . . . . . . . . . 15
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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] provide the following
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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.
[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].
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].
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[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 prevents header compression and because
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
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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.
Forth, 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
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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 occuring, 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
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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 capasity 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 accomodate many environments
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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-
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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
retranmission 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
threshhold 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.
[RFC1323] TCP timestamps must be turned off to allow header
compression.
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IP Payload Compression [RFC2393] should be implemented, although
compression at higher layers of the protocol stack (examples:
[RFC 2616, HTTP-NG]) 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
isssue, it does not need to be standardized.
Of the above mechanisms, only Header Compression (for IP and TCP)
ceases to work in the presence of end-to-end IPSEC.
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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. Since HTTP-NG isn't moving forward and HTTP/1.1 isn't
being enhanced to include a more compact HTTP header representation,
the Wireless Application Protocol (WAP) Forum has proposed
mechanisms like the Wireless Session Prococol [WSP], which does
a binary encoding of HTTP/1.1 functionality. It would be nice to
agree on a more compact HTTP header representation that will be used
by all WWW communities, not only the wireless WAN community.
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.
It would be nice to be able to use timestamps with header
compression.
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.
All other mechanisms except Header Compression (for IP and TCP) are
applicable to TCP connections protected by end-to-end IPSec.
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 benefitted 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.
[SMM98] Jeffrey Semke, Matthew Mathis, and Jamshid Mahdavi,
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"Automatic TCP Buffer Tuning", 1998. Available from
http://www.acm.org/sigcomm/sigcomm98/tp/abs_26.html.
[HTTP-NG] H. Frystyk Nielsen, Mike Spreitzer, Bill Janssen, Jim
Gettys, "HTTP-NG Overview", draft-frystyk-httpng-overview-00.txt,
November 17, 1998, expired, but also available from
http://www.w3.org/Protocols/HTTP-NG/1998/11/.
[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," RFC 1144, February 1990. (Proposed
Standard)
[RFC1323] Jacobson, V., Braden, R., Borman, D., "TCP Extensions
for High Performance", RFC 1323, May 1992. (Proposed Standard)
[RFC2246] T. Dierks, C. Allen, "The TLS Protocol: Version 1.0",
RFC 2246, January 1999. (Proposed Standard)
[RFC2309] B. Braden et. al., "Recommendations on Queue Management
and Congestion Avoidance in the Internet," RFC 2309, April
1998. (Informational)
[RFC2393] A. Shacham, R. Monsour, R. Pereira, M. Thomas, "IP
Payload Compression Protocol (IPComp)," RFC 2393, December
1998. (Proposed Standard)
[RFC2401] S. Kent, R. Atkinson, "Security Architecture for the
Internet Protocol," RFC 2401, November 1998.
[RFC2416] T. Shepard, C. Partridge, "When TCP Starts Up With
Four Packets Into Only Three Buffers", RFC 2416, September 1998.
[RFC2507] Mikael Degermark, Bjorn Nordgren, Stephen Pink. "IP
Header Compression," RFC 2507, February 1999. (Proposed
Standard)
[RFC2508] S. Casner, V. Jacobson. "Compressing IP/UDP/RTP
Headers for Low-Speed Serial Links," RFC 2508, February 1999.
(Proposed Standard)
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[RFC2509] Mathias Engan, S. Casner, C. Bormann. "IP Header
Compression over PPP," RFC 2509, February 1999. (Proposed
Standard)
[RFC2581] M. Allman, V. Paxson, W. Stevens, "TCP Congestion
Control, RFC 2581, April 1999. (Proposed Standard)
[RFC2616] R. Fielding, J. Gettys, J. Mogul, H. Frystyk, Masinter,
P. Leach, T. Berners-Lee. "Hypertext Transfer Protocol -- HTTP/1.1",
RFC 2616, June 1999. (Draft Standard)
[RFC2757] G. Montenegro, S. Dawkins, M. Kojo, V. Magret, N. Vaidya,
"Long Thin Networks", RFC 2757, January 2000. (Informational)
[RFC3042] M. Allman, H. Balakrishnan, S. Floyd, "Enhancing TCP's
Loss Recovery Using Limited Transmit", RFC 3042, January 2001.
(Proposed Standard)
[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, "Wireless Session
Protocol Specification", Version 5, November 1999. Available from
http://www1.wapforum.org/tech/documents/SPEC-WSP-19991105.pdf
Authors' addresses
Questions about this document may be directed to:
Expires November 3, 2001 [Page 15]
INTERNET DRAFT PILC - Slow Links February 2001
Spencer Dawkins
Fujitsu Network Communications
2801 Telecom Parkway
Richardson, Texas 75082
Voice: +1-972-479-3782
E-Mail: spencer.dawkins@fnc.fujitsu.com
Gabriel Montenegro
Sun Microsystems Laboratories, Europe
29, chemin du Vieux Chene
38240 Meylan, FRANCE
Voice: +33 476 18 80 45
E-Mail: gab@sun.com
Markku Kojo
Department of Computer Science
University of Helsinki
P.O. Box 26 (Teollisuuskatu 23)
FIN-00014 HELSINKI
Finland
Voice: +358-9-1914-4179
Fax: +358-9-1914-4441
E-Mail: kojo@cs.helsinki.fi
Vincent Magret
Corporate Research Center
Alcatel Network Systems, Inc
1201 Campbell
Mail stop 446-310
Richardson Texas 75081 USA
M/S 446-310
Voice: +1-972-996-2625
Fax: +1-972-996-5902
E-mail: vincent.magret@aud.alcatel.com
Expires November 3, 2001 [Page 16]