Internet DRAFT - draft-eggert-tsvwg-udp-guidelines
draft-eggert-tsvwg-udp-guidelines
Transport Area Working Group L. Eggert
Internet-Draft Nokia
Intended status: Best Current April 4, 2007
Practice
Expires: October 6, 2007
UDP Usage Guidelines for Application Designers
draft-eggert-tsvwg-udp-guidelines-02
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Copyright (C) The IETF Trust (2007).
Abstract
The User Datagram Protocol (UDP) provides a minimal, message-passing
transport that has no inherent congestion control mechanisms.
Because congestion control is critical to the stable operation of the
Internet, applications and upper-layer protocols that choose to use
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UDP as an Internet transport must employ mechanisms to prevent
congestion collapse and establish some degree of fairness with
concurrent traffic. This document provides guidelines on the use of
UDP for the designers of such applications and upper-layer protocols
that cover congestion-control and other topics, including message
sizes and reliability.
Table of Contents
1. Introduction . . . . . . . . . . . . . . . . . . . . . . . . . 3
2. Terminology . . . . . . . . . . . . . . . . . . . . . . . . . 4
3. UDP Usage Guidelines . . . . . . . . . . . . . . . . . . . . . 4
3.1. Congestion Control Guidelines . . . . . . . . . . . . . . 5
3.2. Message Size Guidelines . . . . . . . . . . . . . . . . . 7
3.3. Reliability Guidelines . . . . . . . . . . . . . . . . . . 7
4. Security Considerations . . . . . . . . . . . . . . . . . . . 8
5. IANA Considerations . . . . . . . . . . . . . . . . . . . . . 8
6. Acknowledgments . . . . . . . . . . . . . . . . . . . . . . . 8
7. References . . . . . . . . . . . . . . . . . . . . . . . . . . 8
7.1. Normative References . . . . . . . . . . . . . . . . . . . 8
7.2. Informative References . . . . . . . . . . . . . . . . . . 9
Author's Address . . . . . . . . . . . . . . . . . . . . . . . . . 11
Intellectual Property and Copyright Statements . . . . . . . . . . 12
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1. Introduction
The User Datagram Protocol (UDP) [RFC0768] provides a minimal,
unreliable, message-passing transport to applications and upper-layer
protocols (both simply called "applications" in the remainder of this
document). Compared to other transport protocols, UDP is unique in
that it does not establish end-to-end connections between
communicating end systems. UDP communication consequently does not
incur connection establishment and teardown overheads and there is no
associated end system state. Because of these characteristics, UDP
can offer a very efficient communication transport to some
applications.
A second unique characteristic of UDP is that it provides no inherent
congestion control mechanisms. [RFC2914] describes the best current
practice for congestion control in the Internet. It identifies two
major reasons why congestion control mechanisms are critical for the
stable operation of the Internet:
1. The prevention of congestion collapse, i.e., a state where an
increase in network load results in a decrease in useful work
done by the network.
2. The establishment of a degree of fairness, i.e., allowing
multiple flows to share the capacity of a path reasonably
equitably.
Because UDP itself provides no congestion control mechanisms, it is
up to the applications that use UDP for Internet communication to
employ suitable mechanisms to prevent congestion collapse and
establish a degree of fairness. [RFC2309] discusses the dangers of
congestion-unresponsive flows and states that "all UDP-based
streaming applications should incorporate effective congestion
avoidance mechanisms." This is an important requirement, even for
applications that do not use UDP for streaming. For example, an
application that generates five 1500-byte UDP packets in one second
can already exceed the capacity of a 56 Kb/s path. For applications
that can operate at higher, potentially unbounded data rates,
congestion control becomes vital. Section 3 describes a number of
simple guidelines for the designers of such applications.
A UDP message is carried in a single IP packet and is hence limited
to a maximum payload of 65,487 bytes. The transmission of large IP
packets frequently requires IP fragmentation, which decreases
communication reliability and efficiency and should be avoided
[I-D.heffner-frag-harmful]. Some of the guidelines in Section 3
describe how applications should determine appropriate message sizes.
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This document provides guidelines to designers of applications that
use UDP for unicast transmission. A special class of applications
uses UDP for IP multicast transmissions. Congestion control, flow
control or reliability for multicast transmissions is more difficult
to establish than for unicast transmissions, because a single sender
may transmit to multiple receivers across potentially very
heterogeneous paths at the same time. Designing multicast
applications requires expertise that goes beyond the simple
guidelines given in this document. The IETF has defined a reliable
multicast framework [RFC3048] and several building blocks to aid the
designers of multicast applications, such as [RFC3738] or [RFC4654].
2. Terminology
The key words "MUST", "MUST NOT", "REQUIRED", "SHALL", "SHALL NOT",
"SHOULD", "SHOULD NOT", "RECOMMENDED", "MAY", and "OPTIONAL" in this
document are to be interpreted as described in BCP 14, RFC 2119
[RFC2119].
3. UDP Usage Guidelines
The RECOMMENDED alternative to the UDP usage guidelines described in
this section is the use of a transport protocol that is congestion-
controlled, such as TCP [RFC0793], SCTP [RFC2960] or DCCP [RFC4340]
with its different congestion control types
[RFC4341][RFC4342][I-D.floyd-dccp-ccid4]. Congestion control
mechanisms are difficult to implement correctly, and for most
applications, the use of one of the existing, congestion-controlled
protocols is the simplest method of satisfying [RFC2914]. The same
is true for message size determination and reliability mechanisms.
If used correctly, congestion-controlled transport protocols are not
as "heavyweight" as often claimed. For example, TCP with SYN cookies
[I-D.ietf-tcpm-syn-flood], which are available on many platforms,
does not require a server to maintain per-connection state until the
connection is established. TCP also requires the end that closes a
connection to maintain the TIME-WAIT state that prevents delayed
segments from one connection instance to interfere with a later one.
Applications that are aware of this behavior can shift maintenance of
the TIME-WAIT state to conserve resources. Finally, TCP's built-in
capacity-probing and PMTU awareness results in efficient data
transmission that quickly compensates for the initial connection
setup delay, for transfers that exchange more than a few packets.
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3.1. Congestion Control Guidelines
If an application or upper-layer protocol chooses not to use a
congestion-controlled transport protocol, it SHOULD control the rate
at which it sends UDP messages to a destination host. It is
important to stress that an application SHOULD perform congestion
control over all UDP traffic it sends to a destination, independent
of how it generates this traffic. For example, an application that
forks multiple worker processes or otherwise uses multiple sockets to
generate UDP messages SHOULD perform congestion control over the
aggregate traffic. The remainder of this section discusses several
approaches for this purpose.
It is important to note that congestion control should not be viewed
as an add-on to a finished application. Many of the mechanisms
discussed in the guidelines below require application support to
operate correctly. Application designers need to consider congestion
control throughout the design of their application, similar to how
they consider security aspects throughout the design process.
3.1.1. Bulk Transfer Applications
Applications that perform bulk transmission of data to a peer over
UDP SHOULD consider implementing TCP-Friendly Rate Control (TFRC)
[RFC3448], window-based, TCP-like congestion control, or otherwise
ensure that the application complies with the congestion control
principles.
TFRC has been designed to provide both congestion control and
fairness in a way that is compatible with the IETF's other transport
protocols. TFRC is currently being updated
[I-D.ietf-dccp-rfc3448bis], and application designers SHOULD always
evaluate whether the latest published specification fits their needs.
If an application implements TFRC, it need not follow the remaining
guidelines in Section 3.1, but SHOULD still follow the guidelines on
message sizes in Section 3.2 and reliability in Section 3.2.
Bulk transfer applications that choose not to implement TFRC or TCP-
like windowing SHOULD implement a congestion control scheme that
results in bandwidth use that competes fairly with TCP within an
order of magnitude. [RFC3551] suggests that applications SHOULD
monitor the packet loss rate to ensure that it is within acceptable
parameters. Packet loss is considered acceptable if a TCP flow
across the same network path under the same network conditions would
achieve an average throughput, measured on a reasonable timescale,
that is not less than that of the UDP flow. The comparison to TCP
cannot be specified exactly, but is intended as an "order-of-
magnitude" comparison in timescale and throughput.
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Finally, some bulk transfer applications chose not to implement any
congestion control mechanism and instead rely on transmitting across
reserved path capacity. This might be an acceptable choice for a
subset of restricted networking environments, but is by no means a
safe practice for operation in the Internet. When the UDP traffic of
such applications leaks out on unprovisioned paths, results are
detrimental.
3.1.2. Low Data-Volume Applications
Applications that exchange only a small number of messages with a
destination at any time may not benefit from implementing TFRC or one
of the other congestion control schemes in Section 3.1.1. Such
applications SHOULD still control their transmission behavior by not
sending more than one UDP message per round-trip time (RTT) to a
destination. Similar to the recommendation in [RFC1536], an
application SHOULD maintain an estimate of the RTT for any
destination it communicates with. Applications SHOULD implement the
algorithm specified in [RFC2988] to compute a smoothed RTT (SRTT)
estimate. A lost response from the peer SHOULD be treated as a very
large RTT sample, instead of being ignored, in order to cause a
sufficiently large (exponential) back-off. When implementing this
scheme, applications need to choose a sensible initial value for the
RTT. This value SHOULD generally be as conservative as possible for
the given application. For example, SIP [RFC3261] and GIST
[I-D.ietf-nsis-ntlp] use an initial value of 500 ms, and shorter
values are likely problematic in many cases.
Some applications cannot maintain a reliable RTT estimate for a
destination. The first case is applications that exchange too few
messages with a peer to establish a statistically accurate RTT
estimate. Such applications MAY use a fixed transmission interval
that is exponentially backed-off during loss. For example, SIP
[RFC3261] and GIST [I-D.ietf-nsis-ntlp] use an interval of 500 ms,
and shorter values are likely problematic in many cases.
A second class of applications cannot maintain an RTT estimate for a
destination, because the destination does not send return traffic.
Such applications SHOULD NOT send more than one UDP message every 3
seconds. The 3-second interval was chosen based on TCP's
retransmission timeout when the RTT is unknown [RFC2988], and shorter
values are likely problematic in many cases. Note that this interval
must be more conservative than above, because the lack of return
traffic prevents the detection of packet loss, i.e., congestion
events, and the application therefore cannot perform exponential
back-off to reduce load.
Applications that communicate bidirectionally SHOULD employ
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congestion control for both directions of the communication. For
example, for a client-server, request-response-style application,
clients SHOULD congestion control their request transmission to a
server, and the server SHOULD congestion control its responses to the
clients. Congestion in the forward and reverse direction is
uncorrelated and an application SHOULD independently detect and
respond to congestion along both directions.
3.2. Message Size Guidelines
Because IP fragmentation lowers the efficiency and reliability of
Internet communication [I-D.heffner-frag-harmful], an application
SHOULD NOT send UDP messages that result in IP packets that exceed
the MTU of the path to the destination. Consequently, an application
SHOULD either use the path MTU information provided by the IP layer
or implement path MTU discovery itself [RFC1191][RFC1981][RFC4821] to
determine whether the path to a destination will support its desired
message size without fragmentation.
Applications that choose not to do so SHOULD NOT send UDP messages
that exceed the minimum PMTU. The minimum PMTU depends on the IP
version used for transmission, and is the lesser of 576 bytes and the
first-hop MTU for IPv4 [RFC1122] and 1280 bytes for IPv6 [RFC2460].
To determine an appropriate UDP payload size, applications must
subtract IP header and option lengths as well as the length of the
UDP header from the PMTU size. Transmission of minimum-sized
messages is inefficient over paths that support a larger PMTU, which
is a second reason to implement PMTU discovery.
Applications that do not send messages that exceed the minimum PMTU
of IPv4 or IPv6 need not implement any of the above mechanisms.
3.3. Reliability Guidelines
Application designers are generally aware that UDP does not provide
any reliability. Often, this is a main reason to consider UDP as a
transport. Applications that do require reliable message delivery
SHOULD implement an appropriate mechanism themselves.
UDP also does not protect against message duplication, i.e., an
application may receive multiple copies of the same message.
Application designers SHOULD consider whether their application
handles message duplication gracefully, and may need to implement
mechanisms to detect duplicates. Even if message reception triggers
idempotent operations, applications may want to suppress duplicate
messages to reduce load.
Finally, UDP messages may be reordered in the network and arrive at
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the receiver in an order different from the send order. Applications
that require ordered delivery SHOULD reestablish message ordering
themselves.
4. Security Considerations
[RFC2309] and [RFC2914] discuss the dangers of congestion-
unresponsive flows to the Internet. This document provides
guidelines to designers of UDP-based applications to congestion-
control to their transmissions. As such, it does not raise any
additional security concerns.
5. IANA Considerations
This document raises no IANA considerations.
6. Acknowledgments
Thanks to Mark Allman, Gorry Fairhurst, Sally Floyd, Joerg Ott, Colin
Perkins, Pasi Sarolahti and Magnus Westerlund for their comments on
this document.
7. References
7.1. Normative References
[RFC0768] Postel, J., "User Datagram Protocol", STD 6, RFC 768,
August 1980.
[RFC0793] Postel, J., "Transmission Control Protocol", STD 7,
RFC 793, September 1981.
[RFC1191] Mogul, J. and S. Deering, "Path MTU discovery", RFC 1191,
November 1990.
[RFC1981] McCann, J., Deering, S., and J. Mogul, "Path MTU Discovery
for IP version 6", RFC 1981, August 1996.
[RFC2119] Bradner, S., "Key words for use in RFCs to Indicate
Requirement Levels", BCP 14, RFC 2119, March 1997.
[RFC2914] Floyd, S., "Congestion Control Principles", BCP 41,
RFC 2914, September 2000.
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[RFC2960] Stewart, R., Xie, Q., Morneault, K., Sharp, C.,
Schwarzbauer, H., Taylor, T., Rytina, I., Kalla, M.,
Zhang, L., and V. Paxson, "Stream Control Transmission
Protocol", RFC 2960, October 2000.
[RFC2988] Paxson, V. and M. Allman, "Computing TCP's Retransmission
Timer", RFC 2988, November 2000.
[RFC3448] Handley, M., Floyd, S., Padhye, J., and J. Widmer, "TCP
Friendly Rate Control (TFRC): Protocol Specification",
RFC 3448, January 2003.
[RFC4340] Kohler, E., Handley, M., and S. Floyd, "Datagram
Congestion Control Protocol (DCCP)", RFC 4340, March 2006.
[RFC4821] Mathis, M. and J. Heffner, "Packetization Layer Path MTU
Discovery", RFC 4821, March 2007.
7.2. Informative References
[I-D.floyd-dccp-ccid4]
Floyd, S. and E. Kohler, "Profile for Datagram Congestion
Control Protocol (DCCP) Congestion ID 4: TCP-Friendly
Rate Control for Small Packets (TFRC-SP)",
draft-floyd-dccp-ccid4-00 (work in progress),
November 2006.
[I-D.heffner-frag-harmful]
Heffner, J., "IPv4 Reassembly Errors at High Data Rates",
draft-heffner-frag-harmful-04 (work in progress),
January 2007.
[I-D.ietf-dccp-rfc3448bis]
Handley, M., "TCP Friendly Rate Control (TFRC): Protocol
Specification", draft-ietf-dccp-rfc3448bis-01 (work in
progress), March 2007.
[I-D.ietf-nsis-ntlp]
Schulzrinne, H. and R. Hancock, "GIST: General Internet
Signalling Transport", draft-ietf-nsis-ntlp-13 (work in
progress), April 2007.
[I-D.ietf-tcpm-syn-flood]
Eddy, W., "TCP SYN Flooding Attacks and Common
Mitigations", draft-ietf-tcpm-syn-flood-02 (work in
progress), March 2007.
[RFC1122] Braden, R., "Requirements for Internet Hosts -
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Communication Layers", STD 3, RFC 1122, October 1989.
[RFC1536] Kumar, A., Postel, J., Neuman, C., Danzig, P., and S.
Miller, "Common DNS Implementation Errors and Suggested
Fixes", RFC 1536, October 1993.
[RFC2309] Braden, B., 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", RFC 2309, April 1998.
[RFC2460] Deering, S. and R. Hinden, "Internet Protocol, Version 6
(IPv6) Specification", RFC 2460, December 1998.
[RFC3048] Whetten, B., Vicisano, L., Kermode, R., Handley, M.,
Floyd, S., and M. Luby, "Reliable Multicast Transport
Building Blocks for One-to-Many Bulk-Data Transfer",
RFC 3048, January 2001.
[RFC3261] Rosenberg, J., Schulzrinne, H., Camarillo, G., Johnston,
A., Peterson, J., Sparks, R., Handley, M., and E.
Schooler, "SIP: Session Initiation Protocol", RFC 3261,
June 2002.
[RFC3551] Schulzrinne, H. and S. Casner, "RTP Profile for Audio and
Video Conferences with Minimal Control", STD 65, RFC 3551,
July 2003.
[RFC3738] Luby, M. and V. Goyal, "Wave and Equation Based Rate
Control (WEBRC) Building Block", RFC 3738, April 2004.
[RFC4341] Floyd, S. and E. Kohler, "Profile for Datagram Congestion
Control Protocol (DCCP) Congestion Control ID 2: TCP-like
Congestion Control", RFC 4341, March 2006.
[RFC4342] Floyd, S., Kohler, E., and J. Padhye, "Profile for
Datagram Congestion Control Protocol (DCCP) Congestion
Control ID 3: TCP-Friendly Rate Control (TFRC)", RFC 4342,
March 2006.
[RFC4654] Widmer, J. and M. Handley, "TCP-Friendly Multicast
Congestion Control (TFMCC): Protocol Specification",
RFC 4654, August 2006.
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Author's Address
Lars Eggert
Nokia Research Center
P.O. Box 407
Nokia Group 00045
Finland
Phone: +358 50 48 24461
Email: lars.eggert@nokia.com
URI: http://research.nokia.com/people/lars_eggert/
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