Internet Engineering Task Force S. Dawkins
INTERNET DRAFT G. Montenegro
M. Kojo
V. Magret
November 24, 2000
End-to-end Performance Implications of Slow Links
draft-ietf-pilc-slow-05.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.
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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:
Slightly restructured the document in part.
Editorial changes and corrections and added some new text.
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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 at times "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
benefits:
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- 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
using 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).
Van Jacobson (VJ) header compression [RFC1144] describes 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. This may, however, result in dropping a full TCP
window, waiting for a full RTO, and performing slow-start
unnecessarily. In addition, VJ header compression does work in
presence of IPsec [RFC2401].
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 either [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, using a link MTU that takes more than delayed ACK timeout
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(typically 200 milliseconds) to transmit will cause an ACK to be
generated for every segment, rather than one per two segments as is
normal 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
296-byte MTUs for dialup access was chosen to limit the impact of a
single MTU size to be not significantly longer than 100-200
milliseconds on 9.6 Kb/second links [RFC1144].
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. However,
with a larger MTU size the congestion window increases slower in
segments than with a smaller MTU size. This, in turn, means that with
larger MTU sizes the congestion window size would stay in only a few
segments for much longer time and the sender is more likely to be
unable to send enough segments to generate three duplicate
acknowledgements and thus triggering fast retransmit/fast recovery,
when a packet loss is encountered. Therefore, a smaller MTU size is
better choice, especially with slow links with lossy characteristics.
Finally, using a smaller MTU size also decreases the queuing delay of
a TCP flow compared to use of larger MTU size with the same number of
packets in a queue. This means that a TCP flow using smaller segment
size and traversing a slow link is more likely to be able to inflate
the congestion window (in number of segments) while experiencing the
same queuing delay.
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.
The HTTP server may be able to place packets on a directly-attached
high-speed LAN at a higher rate than the last-hop router can forward
them on the low-speed link. The consequence of this action is that
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the last-hop router will be unable to buffer unlimited traffic
intended for the low-speed link, and will become a point of
congestion and thus 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 an
advertised window that is unrealistically large for the low-speed
link. The peer host continues to probe for available bandwidth,
trying to fill the advertised window, until packet loss occurs. The
same problem also exists 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.
Smaller the number of buffers in the last-hop router is, earlier it
becomes congested even with a single TCP flow. In addition, the most
important responsibility of the router buffers is to absorb data
bursts. Too small number of buffers, like only three buffers per
outbound link as described in [RFC2416], become full very easily and
are unlikely to absorb even very a small burst. On the other hand, a
larger number of router buffers leads to longer queuing delays but
the buffers are still likely to become (almost) full due to nature of
TCP behavior. Therefore, it is essential to have normally-small
queues in routers. In order to achieve this it is recommended
[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 reasonable number of router
buffers to work properly. (In addition, on a last-hop router towards
a slow link the appropriate parameters of RED are likely to 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-queues
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. This
recommendation takes two forms:
- Modern operating systems use relatively large 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
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accomodate 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 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 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.
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3.0 Summary of Recommended Optimizations
This section summarizes our recommendations regarding the previous
standards-track mechanisms, for end nodes that are capable of
saturating available bandwidth.
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 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 queuue management is recommended on last-hop routers
that provode Internet access to host behind a slow link. In addition,
number of router buffers per slow link should be large enough.
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 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).
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.
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In addition, researchers may wish to experiment with injecting new
traffic into the network when duplicate acknowledgements are being
received to stimulate fast retransmit when not enough segments are in
the pipe, 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.
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.
Very slow link speeds usually mean that TCP connections are likely to
have small congestion windows, interacting badly with Fast
Retransmit/Fast Recovery. TCPs recover without full RTO timeouts when
connections experience losses, as long as the window is large enough
to generate three duplicate acknowledgements. More aggressive
introduction of new segments when duplicate acknowledgements are
being received may provide faster recovery when the congestion window
sizes are small. (See Appendix A for details on existing proposals
like [TCPB98] and [TCPF98].)
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 Acknowledgements
This recommendation has grown out of "Long Thin Networks" [RFC2757],
which in turn benefitted from work done in the IETF TCPSAT working
group.
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6.0 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,
"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
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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
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)
[SACK-EXT] Sally Floyd, Jamshid Mahdavi, Matt Mathis, Matthew
Podolsky, Allyn Romanow, "An Extension to the Selective
Acknowledgement (SACK) Option for TCP", August 1999. Work in
progress, available at
http://www.ietf.org/internet-drafts/draft-floyd-sack-00.txt
[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
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http://www1.wapforum.org/tech/documents/SPEC-WSP-19991105.pdf
Authors' addresses
Questions about this document may be directed to:
Spencer Dawkins
Fujitsu Network Communications
2801 Telecom Parkway
Richardson, Texas 75082
Voice: +1-972-479-3782
E-Mail: spencer.dawkins@fnc.fujitsu.com
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
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
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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
Appendix A Small Window Effects (Experimental)
If a TCP connection stabilizes with a congestion 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.
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
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delivery to the far-end TCP [PAX97].
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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 Choosing MTU sizes . . . . . . . . . . . . . . . . . . . . 6
2.4 Interactions with TCP Congestion Control [RFC2581] . . . . 7
2.5 TCP Buffer Auto-tuning . . . . . . . . . . . . . . . . . . 9
3.0 Summary of Recommended Optimizations . . . . . . . . . . . . . . 10
4.0 Topics For Further Work . . . . . . . . . . . . . . . . . . . . 11
5.0 Acknowledgements . . . . . . . . . . . . . . . . . . . . . . . . 11
6.0 References . . . . . . . . . . . . . . . . . . . . . . . . . . . 12
Authors' addresses . . . . . . . . . . . . . . . . . . . . . . . . . 14
Appendix A Small Window Effects (Experimental) . . . . . . . . . . . 15
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