Transport Area Working Group L. Eggert
Internet-Draft Nokia
Intended status: Best Current G. Fairhurst
Practice University of Aberdeen
Expires: January 10, 2008 July 9, 2007
UDP Usage Guidelines for Application Designers
draft-ietf-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
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
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UDP for the designers of such applications and upper-layer protocols
that cover congestion-control and other topics, including message
sizes, reliability, checksums and middlebox traversal.
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 . . . . . . . . . . . . . . . . . . 8
3.4. Checksum Guidelines . . . . . . . . . . . . . . . . . . . 8
3.5. Middlebox Traversal Guidelines . . . . . . . . . . . . . . 10
4. Security Considerations . . . . . . . . . . . . . . . . . . . 11
5. IANA Considerations . . . . . . . . . . . . . . . . . . . . . 11
6. Acknowledgments . . . . . . . . . . . . . . . . . . . . . . . 11
7. References . . . . . . . . . . . . . . . . . . . . . . . . . . 11
7.1. Normative References . . . . . . . . . . . . . . . . . . . 11
7.2. Informative References . . . . . . . . . . . . . . . . . . 12
Authors' Addresses . . . . . . . . . . . . . . . . . . . . . . . . 14
Intellectual Property and Copyright Statements . . . . . . . . . . 15
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1. Introduction
The User Datagram Protocol (UDP) [RFC0768] provides a minimal,
unreliable, best-effort, 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 and its UDP-Lite variant [RFC3828] are unique in that they do 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 to prevent congestion collapse and
establish some degree of fairness. 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. One
reason for this decrease in reliability is because many NATs and
firewalls do not forward IP fragments; other reasons are documented
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in [I-D.heffner-frag-harmful]. Some of the guidelines in Section 3
describe how applications should determine appropriate message sizes.
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 awareness of the maximum transmission unit
supported by the path (PMTU) results in efficient data transmission
that quickly compensates for the initial connection setup delay, for
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transfers that exchange more than a few packets.
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 implement 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, because TFRC already addresses them, but
SHOULD still follow the remaining guidelines in the subsequent
subsections of Section 3.
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
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cannot be specified exactly, but is intended as an "order-of-
magnitude" comparison in timescale and throughput.
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
When applications that exchange only a small number of messages with
a destination at any time implement TFRC or one of the other
congestion control schemes in Section 3.1.1, the network sees little
benefit, because those mechanisms perform congestion control in a way
that is only effective for longer transmissions.
Applications that exchange only a small number of messages with a
destination at any time 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. TCP uses an
initial value of 3 seconds [RFC2988], which is also RECOMMENDED as an
initial value for UDP applications. SIP [RFC3261] and GIST
[I-D.ietf-nsis-ntlp] use an initial value of 500 ms, and initial
timeouts that are shorter than this are likely problematic in many
cases. It is also important to note that the initial timeout is not
the maximum possible timeout - the RECOMMENDED algorithm in [RFC2988]
yields timeout values after a series of losses that are much longer
than the initial value.
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. TCP uses an initial
value of 3 seconds [RFC2988], which is also RECOMMENDED as an initial
value for UDP applications. SIP [RFC3261] and GIST
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[I-D.ietf-nsis-ntlp] use an interval of 500 ms, and shorter values
are likely problematic in many cases. As in the previous case, note
that the initial timeout is not the maximum possible timeout.
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, and SHOULD consider if they can use an even less aggressive
rate when possible. 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 the initial
timeout interval must be more conservative than in the two previous
cases, 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
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 adapt the packet size 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.
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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
the receiver in an order different from the transmission order.
Applications that require ordered delivery SHOULD reestablish message
ordering themselves.
3.4. Checksum Guidelines
The UDP header includes an optional, 16-bit ones' complement checksum
that provides an integrity check. The UDP checksum provides
assurance that the payload was not corrupted in transit. It also
verifies that the datagram was delivered to the intended end point,
because it covers the IP addresses, port numbers and protocol number,
and it verifies that the datagram is not truncated or padded, because
it covers the size field. It therefore protects an application
against receiving corrupted payload data in place of, or in addition
to, the data that was sent.
Applications SHOULD enable UDP checksums, although [RFC0793] permits
the option to disable their use. Applications that choose to disable
UDP checksums when transmitting over IPv4 therefore MUST NOT make
assumptions regarding the correctness of received data and MUST
behave correctly when a packet is received that was originally sent
to a different end point or is otherwise corrupted. The use of the
UDP checksum is MANDATORY when applications transmit UDP over IPv6
[RFC2460] and applications consequently MUST NOT disable their use.
(The IPv6 header does not have a separate checksum field to protect
the IP addressing information.)
The UDP checksum provides relatively weak protection from a coding
point of view [RFC3819] and, where data integrity is important,
application developers SHOULD provide additional checks, e.g.,
through a CRC included with the data to verify the integrity of an
entire object/file sent over UDP service.
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3.4.1. UDP-Lite
A special class of applications derive benefit from having partially
damaged payloads delivered rather than discarded when using paths
that include error-prone links. Such applications can tolerate
payload corruption and MAY choose to use the Lightweight User
Datagram Protocol (UDP-Lite) [RFC3828] variant of UDP instead of
basic UDP. Applications that choose to use UDP-Lite instead of UDP
MUST still follow the congestion control and other guidelines
described for use with UDP in Section 3.1.
UDP-Lite changes the semantics of the UDP "payload length" field to
that of a "checksum coverage length" field. Otherwise, UDP-Lite is
semantically identical to UDP. The interface of UDP-Lite differs
from that of UDP by the addition of a single (socket) option that
communicates a checksum coverage length value: at the sender, this
specifies the intended datagram checksum coverage, with the remaining
unprotected part of the payload called the "error insensitive part".
If required, an application may dynamically modify this length value,
e.g., to offer greater protection to some packets. UDP-Lite always
verifies that a datagram was delivered to the intended end point,
i.e., always verifies the header fields. Errors in the insensitive
part will not cause a packet to be discarded by the receiving end
host. Applications using UDP-Lite therefore MUST NOT make
assumptions regarding the correctness of the data received in the
insensitive part of the UDP-Lite payload.
The sending application SHOULD select the minimum checksum coverage
to include all sensitive protocol headers (e.g., the RTP header),
and, where appropriate, MUST also introduce their own appropriate
validity checks for protocol information carried in the insensitive
part of the UDP-Lite payload (e.g., internal CRCs).
The receiver MUST set a minimum coverage threshold for incoming
datagrams that is not smaller than the smallest coverage used by the
sender. This may be a fixed value, or may be negotiated by an
application. UDP-Lite does not provide mechanisms to negotiate the
checksum coverage between the sender and receiver.
Applications may still experience packet loss, rather than
corruption, when using UDP-Lite. The enhancements offered by UDP-
Lite rely upon a link being able to intercept the UDP-Lite header to
correctly identify the partial-coverage required. When tunnels
and/or encryption are used, this can result in UDP-Lite packets being
treated the same as UDP packets, i.e., result in packet loss. Use of
IP fragmentation can also prevent special treatment for UDP-Lite
packets, and is another reason why applications SHOULD avoid IP
fragmentation Section 3.2.
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3.5. Middlebox Traversal Guidelines
Network address translators (NATs) and firewalls are examples of
intermediary devices ("middleboxes") that can exist along an end-to-
end path. A middlebox typically performs a function that requires it
to maintain per-flow state. For connection-oriented protocols, such
as TCP, middleboxes snoop and parse the connection-management traffic
and create and destroy per-flow state accordingly. For a
connectionless protocol such as UDP, this approach is not possible.
Consequently, middleboxes may create per-flow state when they see a
packet that indicates a new flow, and destroy the state after some
period of time during which no packets belonging to the same flow
have arrived.
Depending on the specific function that the middlebox performs, this
behavior can introduce a time-dependency that restricts the kinds of
UDP traffic exchanges that will be successful across it. For
example, NATs and firewalls typically define the partial path on one
side of them to be interior to the domain they control, whereas the
partial path on their other side is defined to be exterior to that
domain. Per-flow state is typically created when the first packet
crosses from the interior to the exterior, and while the state is
present, NATs and firewalls will forward return traffic. Return
traffic arriving after the per-flow state has timed out is dropped,
as is other traffic arriving from the exterior.
Many applications use UDP for communication operate across
middleboxes without needing to employ additional mechanisms. One
example is the DNS, which has a strict request-response communication
pattern that typically completes within seconds.
Other applications may experience communication failures when
middleboxes destroy the per-flow state associated with an application
session during periods when the application does not exchange any UDP
traffic. Applications SHOULD be able to gracefully handle such
communication failures and implement mechanisms to re-establish their
UDP sessions.
Applications MAY in addition send periodic keep-alive messages to
attempt to refresh middlebox state. Unfortunately, no common timeout
has been specified for per-flow UDP state for arbitrary middleboxes.
For NATs, [RFC4787] requires a state timeout of 2 minutes or longer,
and it is likely that other types of middleboxes use timeouts of
similar timescales. Consequently, if applications choose to send
periodic keep-alives, they SHOULD NOT send them more frequently than
once every two minutes. (Not that some deployed middleboxes use a
shorter timeout value than 2 minutes, violating [RFC4787].)
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It is important to note that sending keep-alives is not a substitute
for implementing a robust connection handling. Like all UDP
messages, keep-alives can be delayed or dropped, causing middlebox
state to time out. In addition, the congestion control guidelines in
Section 3.1 cover all UDP transmissions by an application, including
the transmission of middlebox keep-alives. Congestion control may
thus lead to delays or temporary suspension of keep-alive
transmission.
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 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, Sally Floyd, Philip Matthews, Joerg Ott, Colin
Perkins, Pasi Sarolahti and Magnus Westerlund for their comments on
this document.
The middlebox traversal guidelines in Section 3.5 incorporate ideas
from Section 5 of [I-D.ford-behave-app] by Bryan Ford, Pyda Srisuresh
and Dan Kegel.
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.
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[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.
[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.
[RFC3819] Karn, P., Bormann, C., Fairhurst, G., Grossman, D.,
Ludwig, R., Mahdavi, J., Montenegro, G., Touch, J., and L.
Wood, "Advice for Internet Subnetwork Designers", BCP 89,
RFC 3819, July 2004.
[RFC3828] Larzon, L-A., Degermark, M., Pink, S., Jonsson, L-E., and
G. Fairhurst, "The Lightweight User Datagram Protocol
(UDP-Lite)", RFC 3828, July 2004.
[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-ccid4-01 (work in progress), July 2007.
[I-D.ford-behave-app]
Ford, B., "Application Design Guidelines for Traversal
through Network Address Translators",
draft-ford-behave-app-05 (work in progress), March 2007.
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[I-D.heffner-frag-harmful]
Heffner, J., "IPv4 Reassembly Errors at High Data Rates",
draft-heffner-frag-harmful-05 (work in progress),
May 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-05 (work in
progress), May 2007.
[RFC1122] Braden, R., "Requirements for Internet Hosts -
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
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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.
[RFC4787] Audet, F. and C. Jennings, "Network Address Translation
(NAT) Behavioral Requirements for Unicast UDP", BCP 127,
RFC 4787, January 2007.
Authors' Addresses
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/
Godred Fairhurst
University of Aberdeen
Department of Engineering
Fraser Noble Building
Aberdeen AB24 3UE
Scotland
Email: gorry@erg.abdn.ac.uk
URI: http://www.erg.abdn.ac.uk/
Eggert & Fairhurst Expires January 10, 2008 [Page 14]
Internet-Draft UDP Usage Guidelines July 2007
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Eggert & Fairhurst Expires January 10, 2008 [Page 15]
