Internet Engineering Task Force S. Dawkins
INTERNET DRAFT G. Montenegro
M. Kojo
V. Magret
October 21, 1999
End-to-end Performance Implications of Slow Links
draft-ietf-pilc-slow-02.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:
Rewrite of Abstract to say less about history and more about
technical motivation.
Addition of considerations about MTU sizes.
Clarification about whether TCP timestamps are actually
recommended(!).
Clarify discussion of "Interactions with TCP Congestion
Avoidance", and add discussion of "Buffer Auto-Tuning".
Other editorial changes and corrections.
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Table of Contents
1.0 Introduction . . . . . . . . . . . . . . . . . . . . . . . . . . 3
2.0 Description of Optimizations . . . . . . . . . . . . . . . . . . 3
2.1 Header Compression Alternatives . . . . . . . . . . . . . . 3
2.2 Payload Compression Alternatives . . . . . . . . . . . . . 6
2.3 Interactions with TCP Congestion Avoidance [RFC2581] . . . 6
2.4 Choosing MTU sizes . . . . . . . . . . . . . . . . . . . . 8
2.5 Small Window Effects (Experimental) . . . . . . . . . . . . 8
2.6 TCP Buffer Auto-tuning . . . . . . . . . . . . . . . . . . 9
3.0 Summary of Recommended Optimizations . . . . . . . . . . . . . . 9
4.0 Acknowledgements . . . . . . . . . . . . . . . . . . . . . . . . 11
5.0 References . . . . . . . . . . . . . . . . . . . . . . . . . . . 11
Authors' addresses . . . . . . . . . . . . . . . . . . . . . . . . . 12
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1.0 Introduction
The Internet protocol stack was designed to span 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
"bulky", including images, sound, and video.
The World Wide Web has also popularized the Internet, so that there
is significant interest in accessing the World Wide Web over link
speeds that are much "slower" than typical desktop host speeds.
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 don't 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
benefits:
- Improve interactive response time
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- Allow using small packets for bulk data with good line
efficiency
- Allow using small packets for delay sensitive low data-rate
traffic
- 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)
- Reduce packet loss rate over lossy links (simply because
shorter transmission times expose packets to fewer events
that cause loss).
Van Jacobson (VJ) header compression [RFC1144] describes a
Proposed Standard for TCP Header compression that is widely
deployed. It uses TCP timeouts to detect a loss of
synchronization between the compressor and decompressor. 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).
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].
[RFC1144] header compression should only be enabled when operating
over reliable "slow" links, because even a single bit error may
result in dropping a full TCP window, waiting for a full RTO, and
performing slow-start unnecessarily.
[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. For these reasons, and because connections
traversing "slow" links do not require protection against TCP
sequence-number wrapping, use of TCP Timestamps is not recommended
for use with these connections.
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2.2 Payload Compression Alternatives
Compression of IP payloads is also desirable on "slow" network
lists. "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 FTPed), 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.
The most recent HTTP-NG proposal [HTTP-NG] replaces the text-based
HTTP header representation with a binary representation for
compactness.
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.
All these compression techniques will reduce the need for IPComp,
especially for WWW users.
Recommendation: IPComp may optionally be implemented. Track
HTTP-NG standardization (or any proposed mechanism that will
compress HTTP headers).
2.3 Interactions with TCP Congestion Avoidance [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,
with an HTTP server on a high-speed LAN "behind" the terminal
server.
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The HTTP server may be able to place packets on a directly-attached
high-speed LAN at a higher rate than the terminal server can forward
them on the low-speed link. The consequence of this action is that
the terminal server will be unable to buffer unlimited traffic
intended for the low-speed link, and will begin to "drop" the
excess packets. The self-clocking nature of TCP's slow start and
congestion avoidance algorithms prevent this buffer overrun from
continuing, but these algorithms also 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 a receive window that is
unrealistically large for the low-speed link. The peer host
continues to probe for available bandwidth, trying to fill the
receive window, until packet loss occurs.
Hosts that are directly connected to low-speed links should
limit the receive windows they advertise. This recommendation
takes two forms:
- Modern operating systems are using increasingly larger default
TCP receive buffers, in order to maximize throughput on
high-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 without
requiring hand-tuning by experienced network engineers).
- Application developers should rely on the system default,
instead of increasing the receive buffer in use (typically via
a socket option), to accomodate users connecting via low-speed
links. If an application does manage the receiver buffer in
use, this should still be under the user's control, as previously
suggested.
For example - in the case (described in [RFC2416]) where a modem
has only three buffers, whenever the HTTP server returns four
back-to-back packets, one will be dropped. If this bottleneck link
causes the TCP window to be less than four to five segments, it will
not be possible to receive three duplicate acknowledgements, so
Fast Retransmit/Fast Recovery will never happen, and TCP recovery
will take place with full RTO and slow start.
In this case, the common MTU of 296 bytes gives an MSS of 256
bytes, so an appropriate receive buffer size would be 768 bytes -
any value larger would allow unproductive probing for non-existent
bandwidth.
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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 is a more reasonable recommendation, and is
discussed below.
2.4 Choosing MTU Sizes
There are several points to keep in mind when choosing an MTU
for low-speed links.
First, using an MTU that takes more than 200 milliseconds to
transmit effectively turns off delayed acknowledgements, because
the receiver will never receive a second full-sized segment before
the delayed acknowledgement timer expires.
Second, "relatively large" MTUs (which take human-perceptible
amounts of time to be transmitted into the network) create human-
perceptible delays in other connections using the same network
interface. [RFC1144] considers 100-200 millisecond delays as
human-perceptible.
If it is possible to do so, MTUs should be chosen that do not
monopolize network interfaces for human-perceptible amounts of
time. The convention of 296-byte MTUs for dialup access was
chosen to limit the impact of a single MTU size to 100-200
milliseconds on 9.6 Kb/second links [RFC1144], and implementors
should not chose MTUs that will occupy a network interface for
more than 100-200 milliseconds.
2.5 Small Window Effects (Experimental)
If a TCP connection stabilizes with a window of only a few
segments (as would be expected on a "slow" link), the sender
isn't sending enough segments to generate three duplicate
acknowledgements, triggering fast retransmit/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.
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In these situations, it may be desireable to trigger fast
retransmit/fast recovery more aggressively. [TCPB98] and
[TCPF98] suggest sending a new segment when the first and second
duplicate acknowledgements are received, so that the receiver will
continue to generate duplicate acknowledgements until the TCP
retransmit threshhold is reached, triggering fast
retransmit/fast recovery.
We note that a maximum of two additional new segments will be
sent before the receiver sends either an 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.
The alternative, lowering the fast retransmit/fast recovery
threshold, is more likely to inject unnecessary retransmissions
when the duplicate acknowledgements are the result of out-of-order
delivery to the far-end TCP [PAX97].
2.6 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 effective window, rather than attempting to
preallocate TCP buffers based on anticipated window sizes that
may be achieved.
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 can recover losses without
"stalling" the connection.
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 technique
useful.
3.0 Summary of Recommended Optimizations
This section summarizes our recommendations regarding the previous
mechanisms, for end nodes that are capable of saturating available
bandwidth.
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Header compression should be implemented. [RFC1144] header
compression can be enabled over robust network connections.
[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.
IP Payload Compression [RFC2393] should be implemented, although
compression at higher layers of the protocol stack (examples:
[RFC 2068, HTTP-NG]) may make this mechanism less useful.
For HTTP/1.1 environments, [RFC2068] 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.
Implementors should consider the possibility that a host will be
directly connected to a low-speed link when choosing default TCP
receive window sizes, and, if the host is likely to be used with a
range of
Application developers should consider the possibility that an
application will be used on a host that is directly connected to a
low-speed link, before increasing the TCP receive window size beyond
the default for TCP connections used by this application.
All of the mechanisms described above are stable standards-track
RFCs (at Proposed Standard status, as of this writing), with the
exception of [HTTP-NG], which is included for completeness.
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.
In addition, researchers may wish to experiment with injecting
new traffic into the network when duplicate acknowledgements are
being received, as described in [TCPB98] and [TCPF98]. This is
not a standards-track TCP mechanism.
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 Acknowledgements
This recommendation has grown out of the Internet Draft "TCP Over
Long Thin Networks", which was in turn based on work done in the
IETF TCPSAT working group.
5.0 References
[SMM98] Jeffrey Semke, Matthew Mathis, and Jamshid Mahdavi,
"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
[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)
[RFC2393] A. Shacham, R. Monsour, R. Pereira, M. Thomas, "IP
Payload Compression Protocol (IPComp)," RFC 2393, December
1998. (Proposed Standard)
[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)
[RFC2509] Mathias Engan, S. Casner, C. Bormann. "IP Header
Compression over PPP," RFC 2509, February 1999. (Proposed
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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)
[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
Authors' addresses
Questions about this document may be directed to:
Spencer Dawkins
Nortel Networks
3 Crockett Ct
Allen, TX 75002
Voice: +1-972-684-4827
Fax: +1-972-685-3292
E-Mail: sdawkins@nortelnetworks.com
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Gabriel E. Montenegro
Sun Labs Networking and Security Group
Sun Microsystems, Inc.
901 San Antonio Road
Mailstop UMPK 15-214
Mountain View, California 94303
Voice: +1-650-786-6288
Fax: +1-650-786-6445
E-Mail: gab@sun.com
Markku Kojo
University of Helsinki/Department of Computer Science
P.O. Box 26 (Teollisuuskatu 23)
FIN-00014 HELSINKI
Finland
Voice: +358-9-7084-4179
Fax: +358-9-7084-4441
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
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