Transport Area Working Group L. Eggert
Internet-Draft Nokia
Intended status: BCP G. Fairhurst
Expires: February 23, 2009 University of Aberdeen
August 22, 2008
Unicast UDP Usage Guidelines for Application Designers
draft-ietf-tsvwg-udp-guidelines-10
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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
UDP for the designers of unicast applications and upper-layer
protocols. Congestion control guidelines are a primary focus, but
the document also provides guidance on other topics, including
message sizes, reliability, checksums and middlebox traversal.
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Table of Contents
1. Introduction . . . . . . . . . . . . . . . . . . . . . . . . . 3
2. Terminology . . . . . . . . . . . . . . . . . . . . . . . . . 5
3. UDP Usage Guidelines . . . . . . . . . . . . . . . . . . . . . 5
3.1. Congestion Control Guidelines . . . . . . . . . . . . . . 6
3.2. Message Size Guidelines . . . . . . . . . . . . . . . . . 11
3.3. Reliability Guidelines . . . . . . . . . . . . . . . . . . 12
3.4. Checksum Guidelines . . . . . . . . . . . . . . . . . . . 13
3.5. Middlebox Traversal Guidelines . . . . . . . . . . . . . . 15
3.6. Programming Guidelines . . . . . . . . . . . . . . . . . . 17
3.7. ICMP Guidelines . . . . . . . . . . . . . . . . . . . . . 18
4. Security Considerations . . . . . . . . . . . . . . . . . . . 18
5. Summary . . . . . . . . . . . . . . . . . . . . . . . . . . . 20
6. IANA Considerations . . . . . . . . . . . . . . . . . . . . . 21
7. Acknowledgments . . . . . . . . . . . . . . . . . . . . . . . 21
8. References . . . . . . . . . . . . . . . . . . . . . . . . . . 21
8.1. Normative References . . . . . . . . . . . . . . . . . . . 21
8.2. Informative References . . . . . . . . . . . . . . . . . . 22
Authors' Addresses . . . . . . . . . . . . . . . . . . . . . . . . 26
Intellectual Property and Copyright Statements . . . . . . . . . . 27
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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 minimal 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. On many platforms, applications can
send UDP datagrams at the line rate of the link interface, which is
often much greater than the available path capacity, and doing so
contributes to congestion along the path. [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. In addition,
congestion-controlled transmission is of benefit to an application
itself, because it can reduce self-induced packet loss, minimize
retransmissions and hence reduce delays. Congestion control is
essential even at relatively slow transmission rates. For example,
an application that generates five 1500-byte UDP datagrams 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.
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A UDP datagram is carried in a single IP packet and is hence limited
to a maximum payload of 65,507 bytes for IPv4 and 65,527 bytes for
IPv6. The transmission of large IP packets usually requires IP
fragmentation. Fragmentation decreases communication reliability and
efficiency and should be avoided. IPv6 allows the option of
transmitting large packets ("jumbograms") without fragmentation when
all link layers along the path support this [RFC2675]. Some of the
guidelines in Section 3 describe how applications should determine
appropriate message sizes. Other sections of this document provide
guidance on reliability, checksums and middlebox traversal.
This document provides guidelines and recommendations. Although most
unicast UDP applications are expected to follow these guidelines,
there do exist valid reasons why a specific application may decide
not to follow a given guideline. In such cases, it is RECOMMENDED
that the application designers document the rationale for their
design choice in the technical specification of their application or
protocol.
This document provides guidelines to designers of applications that
use UDP for unicast transmission, which is the most common case.
Specialized classes of applications use UDP for IP multicast
[RFC1112], broadcast [RFC0919], or anycast [RFC1546] transmissions.
The design of such specialized applications requires expertise that
goes beyond the simple, unicast-specific guidelines given in this
document. Multicast and broadcast senders may transmit to multiple
receivers across potentially very heterogeneous paths at the same
time, which significantly complicates congestion control, flow
control and reliability mechanisms. 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].
Anycast senders must be aware that successive messages sent to the
same anycast IP address may be delivered to different anycast nodes,
i.e., arrive at different locations in the topology. It is not
intended that the guidelines in this document apply to multicast,
broadcast or anycast applications that use UDP.
Finally, although this document specifically refers to unicast
applications that use UDP, the spirit of some of its guidelines also
applies to other message-passing applications and protocols
(specifically on the topics of congestion control, message sizes and
reliability). Examples include signaling or control applications
that choose to run directly over IP by registering their own IP
protocol number with IANA. This document may provide useful
background reading to the designers of such applications and
protocols.
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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
Internet paths can have widely varying characteristics, including
transmission delays, available bandwidths, congestion levels,
reordering probabilities, supported message sizes or loss rates.
Furthermore, the same Internet path can have very different
conditions over time. Consequently, applications that may be used on
the Internet MUST NOT make assumptions about specific path
characteristics. They MUST instead use mechanisms that let them
operate safely under very different path conditions. Typically, this
requires conservatively probing the current conditions of the
Internet path they communicate over to establish a transmission
behavior that it can sustain and that is reasonably fair to other
traffic sharing the path.
These mechanisms are difficult to implement correctly. For most
applications, the use of one of the existing IETF transport protocols
is the simplest method of acquiring the required mechanisms.
Consequently, the RECOMMENDED alternative to the UDP usage described
in the remainder of this section is the use of an IETF transport
protocol such as TCP [RFC0793], SCTP [RFC4960] and SCTP-PR [RFC3758],
or DCCP [RFC4340] with its different congestion control types
[RFC4341][RFC4342][I-D.ietf-dccp-ccid4].
If used correctly, these more fully-featured transport protocols are
not as "heavyweight" as often claimed. For example, the TCP
algorithms have been continuously improved over decades, and have
reached a level of efficiency and correctness that custom
application-layer mechanisms will struggle to easily duplicate. In
addition, many TCP implementations allow connections to be tuned by
an application to its purposes. For example, TCP's "Nagle" algorithm
[RFC0896] can be disabled, improving communication latency at the
expense of more frequent - but still congestion-controlled - packet
transmissions. Another example is the TCP SYN Cookie mechanism
[RFC4987], which is available on many platforms. TCP with SYN
Cookies 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 and designed for this
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behavior can shift maintenance of the TIME-WAIT state to conserve
resources by controlling which end closes a TCP connection [FABER].
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 transfers that exchange more than a few segments.
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 datagrams to a destination host, in order to
fulfill the requirements of [RFC2914]. It is important to stress
that an application SHOULD perform congestion control over all UDP
traffic it sends to a destination, independently from how it
generates this traffic. For example, an application that forks
multiple worker processes or otherwise uses multiple sockets to
generate UDP datagrams SHOULD perform congestion control over the
aggregate traffic.
The remainder of this section discusses several approaches for this
purpose. Not all approaches discussed below are appropriate for all
UDP-transmitting applications. Section 3.1.1 discusses congestion
control options for applications that perform bulk transfers over
UDP. Such applications can employ schemes that sample the path over
several subsequent RTTs during which data is exchanged, in order to
determine a sending rate that the path at its current load can
support. Other applications only exchange a few UDP datagrams with a
destination. Section 3.1.2 discusses congestion control options for
such "low data-volume" applications. Because they typically do not
transmit enough data to iteratively sample the path to determine a
safe sending rate, they need to employ different kinds of congestion
control mechanisms. Section 3.1.3 discusses congestion control
considerations when UDP is used as a tunneling protocol.
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.
In the past, the IETF has also investigated integrated congestion
control mechanisms that act on the traffic aggregate between two
hosts, i.e., a framework such as the Congestion Manager [RFC3124],
where active sessions may share current congestion information in a
way that is independent of the transport protocol. Such mechanisms
have failed to see deployment, but would otherwise simplify the
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design of congestion control mechanisms for UDP sessions, so that
they fulfill the requirements in [RFC2914].
3.1.1. Bulk Transfer Applications
Applications that perform bulk transmission of data to a peer over
UDP, i.e., applications that exchange more than a small number of UDP
datagrams per RTT, 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 3.1.1, 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. Section 2 of [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.
Finally, some bulk transfer applications may choose 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 Internet
paths, it can significantly degrade the performance of other traffic
sharing the path and even result in congestion collapse.
Applications that support an uncontrolled or unadaptive transmission
behavior SHOULD NOT do so by default and SHOULD instead require users
to explicitly enable this mode of operation.
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3.1.2. Low Data-Volume Applications
When applications that exchange only a small number of UDP datagrams
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 UDP datagrams with
a destination at any time SHOULD still control their transmission
behavior by not sending on average more than one UDP datagram 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 with which it communicates.
Applications SHOULD implement the algorithm specified in [RFC2988] to
compute a smoothed RTT (SRTT) estimate. They SHOULD also detect
packet loss and exponentially back-off their retransmission timer
when a loss event occurs. 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 that of applications that exchange
too few UDP datagrams with a peer to establish a statistically
accurate RTT estimate. Such applications MAY use a pre-determined
transmission interval that is exponentially backed-off when packets
are lost. 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 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 datagram every 3
seconds, and SHOULD 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 sending rate in this
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case 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 either independently detect
and respond to congestion along both directions, or limit new and
retransmitted requests based on acknowledged responses across the
entire round trip path.
3.1.3. UDP Tunnels
One increasingly popular use of UDP is as a tunneling protocol, where
a tunnel endpoint encapsulates the packets of another protocol inside
UDP datagrams and transmits them to another tunnel endpoint, which
decapsulates the UDP datagrams and forwards the original packets
contained in the payload. Tunnels establish virtual links that
appear to directly connect locations that are distant in the physical
Internet topology, and can be used to create virtual (private)
networks. Using UDP as a tunneling protocol is attractive when the
payload protocol is not supported by middleboxes that may exist along
the path, because many middleboxes support UDP transmissions.
Well-implemented tunnels are generally invisible to the endpoints
that happen to transmit over a path that includes tunneled links. On
the other hand, to the routers along the path of a UDP tunnel, i.e.,
the routers between the two tunnel endpoints, the traffic that a UDP
tunnel generates is a regular UDP flow, and the encapsulator and
decapsulator appear as regular UDP-sending and -receiving
applications. Because other flows can share the path with one or
more UDP tunnels, congestion control needs to be considered.
Two factors determine whether a UDP tunnel needs to employ specific
congestion control mechanisms. First, whether the tunneling scheme
generates UDP traffic at a volume that corresponds to the volume of
payload traffic carried within the tunnel. Second, whether the
payload traffic is IP-based.
IP-based traffic is generally assumed to be congestion-controlled,
i.e., it is assumed that the transport protocols generating IP-based
traffic at the sender already employ mechanisms that are sufficient
to address congestion on the path. Consequently, a tunnel carrying
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IP-based traffic should already interact appropriately with other
traffic sharing the path, and specific congestion control mechanism
for the tunnel are not necessary.
However, if the IP traffic in the tunnel is known to not be
congestion-controlled, additional measures are RECOMMENDED in order
to limit the impact of the tunneled traffic on other traffic sharing
the path.
The following guidelines define these possible cases in more detail:
1. A tunnel generates UDP traffic at a volume that corresponds to
the volume of payload traffic, and the payload traffic is IP-
based and congestion-controlled.
This is arguably the most common case for Internet tunnels. In
this case, the UDP tunnel SHOULD NOT employ its own congestion
control mechanism, because congestion losses of tunneled traffic
will already trigger an appropriate congestion response at the
original senders of the tunneled traffic.
Note that this guideline is built on the assumption that most IP-
based communication is congestion-controlled. If a UDP tunnel is
used for IP-based traffic that is known to not be congestion-
controlled, the next set of guidelines applies:
2. A tunnel generates UDP traffic at a volume that corresponds to
the volume of payload traffic, and the payload traffic is not
known to be IP-based, or is known to be IP-based but not
congestion-controlled.
This can be the case, for example, when some link-layer protocols
are encapsulated within UDP (but not all link-layer protocols;
some are congestion-controlled.) Because it is not known that
congestion losses of tunneled non-IP traffic will trigger an
appropriate congestion response at the senders, the UDP tunnel
SHOULD employ an appropriate congestion control mechanism.
Because tunnels are usually bulk-transfer applications as far as
the intermediate routers are concerned, the guidelines in
Section 3.1.1 apply.
3. A tunnel generates UDP traffic at a volume that does not
correspond to the volume of payload traffic, independent of
whether the payload traffic is IP-based or congestion-controlled.
Examples of this class include UDP tunnels that send at a
constant rate, increase their transmission rates under loss, for
example, due to increasing redundancy when forward-error-
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correction is used, or are otherwise constrained in their
transmission behavior. These specialized uses of UDP for
tunneling go beyond the scope of the general guidelines given in
this document. The implementer of such specialized tunnels
SHOULD carefully consider congestion control in the design of
their tunneling mechanism.
Designing a tunneling mechanism requires significantly more expertise
than needed for many other UDP applications, because tunnels
virtualize lower-layer components of the Internet, and the
virtualized components need to correctly interact with the
infrastructure at that layer. This document only touches upon the
congestion control considerations for implementing UDP tunnels; a
discussion of other required tunneling behavior is out of scope.
3.2. Message Size Guidelines
IP fragmentation lowers the efficiency and reliability of Internet
communication. The loss of a single fragment results in the loss of
an entire fragmented packet, because even if all other fragments are
received correctly, the original packet cannot be reassembled and
delivered. This fundamental issue with fragmentation exists for both
IPv4 and IPv6. In addition, some NATs and firewalls drop IP
fragments. The network address translation performed by a NAT only
operates on complete IP packets, and some firewall policies also
require inspection of complete IP packets. Even with these being the
case, some NATs and firewalls simply do not implement the necessary
reassembly functionality, and instead choose to drop all fragments.
Finally, [RFC4963] documents other issues specific to IPv4
fragmentation.
Due to these issues, an application SHOULD NOT send UDP datagrams
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 do not follow this recommendation to do PMTU
discovery SHOULD still avoid sending UDP datagrams that would result
in IP packets that exceed the path MTU. Because the actual path MTU
is unknown, such applications SHOULD fall back to sending messages
that are shorter that the default effective MTU for sending (EMTU_S
in [RFC1122]). For IPv4, EMTU_S is the smaller of 576 bytes and the
first-hop MTU [RFC1122]. For IPv6, EMTU_S is 1280 bytes [RFC2460].
The effective PMTU for a directly connected destination (with no
routers on the path) is the configured interface MTU, which could be
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less than the maximum link payload size. Transmission of minimum-
sized UDP datagrams is inefficient over paths that support a larger
PMTU, which is a second reason to implement PMTU discovery.
To determine an appropriate UDP payload size, applications MUST
subtract the size of the IP header (which includes any IPv4 optional
headers or IPv6 extension headers) as well as the length of the UDP
header (8 bytes) from the PMTU size. This size, known as the MMS_S,
can be obtained from the TCP/IP stack [RFC1122].
Applications that do not send messages that exceed the effective PMTU
of IPv4 or IPv6 need not implement any of the above mechanisms. Note
that the presence of tunnels can cause an additional reduction of the
effective PMTU, so implementing PMTU discovery will still be
beneficial in some cases.
Applications that fragment an application-layer message into multiple
UDP datagrams SHOULD perform this fragmentation so that each datagram
can be received independently, and be independently retransmitted in
the case where an application implements its own reliability
mechanisms.
3.3. Reliability Guidelines
Application designers are generally aware that UDP does not provide
any reliability, e.g., it does not retransmit any lost packets.
Often, this is a main reason to consider UDP as a transport.
Applications that do require reliable message delivery MUST implement
an appropriate mechanism themselves.
UDP also does not protect against datagram duplication, i.e., an
application may receive multiple copies of the same UDP datagram.
Application designers SHOULD verify that their application handles
datagram duplication gracefully, and may consequently need to
implement mechanisms to detect duplicates. Even if UDP datagram
reception triggers idempotent operations, applications may want to
suppress duplicate datagrams to reduce load.
In addition, the Internet can significantly delay some packets with
respect to others, e.g., due to routing transients, intermittent
connectivity, or mobility. This can cause reordering, where UDP
datagrams arrive at the receiver in an order different from the
transmission order. Applications that require ordered delivery MUST
reestablish datagram ordering themselves.
Finally, it is important to note that delay spikes can be very large.
This can cause reordered packets to arrive many seconds after they
were sent. [RFC0793] defines the the maximum delay a TCP segment
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should experience - the Maximum Segment Lifetime (MSL) - as 2
minutes. No other RFC defines an MSL for other transport protocols
or IP itself. This document clarifies that the MSL value to be used
for UDP SHOULD be the same 2 minutes as for TCP. Applications SHOULD
be robust to the reception of delayed or duplicate packets that are
received within this 2-minute interval.
An application that requires reliable and ordered message delivery
SHOULD choose an IETF standard transport protocol that provides these
features. If this is not possible, it will need to implement a set
of appropriate mechanisms itself.
3.4. Checksum Guidelines
The UDP header includes an optional, 16-bit ones-complement checksum
that provides an integrity check. This results in a relatively weak
protection from in terms of coding theory [RFC3819] and application
developers SHOULD implement additional checks where data integrity is
important, e.g., through a Cyclic Redundancy Check (CRC) included
with the data to verify the integrity of an entire object/file sent
over UDP service.
The UDP checksum provides a statistical guarantee that the payload
was not corrupted in transit. It also allows the receiver to verify
that it was the intended destination of the packet, because it covers
the IP addresses, port numbers and protocol number, and it verifies
that the packet 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. This check is not strong from a coding or cryptographic
perspective, and is not designed to detect physical-layer errors or
malicious modification of the datagram [RFC3819].
Applications SHOULD enable UDP checksums, although [RFC0768] 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 UDP datagram is received that was originally
sent to a different destination or is otherwise corrupted. The use
of the UDP checksum is REQUIRED when applications transmit UDP over
IPv6 [RFC2460].
3.4.1. UDP-Lite
A special class of applications can 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
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User Datagram Protocol (UDP-Lite) [RFC3828] variant of UDP instead of
basic UDP. Applications that choose to use UDP-Lite instead of UDP
should still follow the congestion control and other guidelines
described for use with UDP in Section 3.
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 checksum coverage, with the remaining
unprotected part of the payload called the "error insensitive part".
By default, the UDP-Lite checksum coverage extends across the entire
datagram. If required, an application may dynamically modify this
length value, e.g., to offer greater protection to some messages.
UDP-Lite always verifies that a packet was delivered to the intended
destination, i.e., always verifies the header fields. Errors in the
insensitive part will not cause a UDP datagram to be discarded by the
destination. 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. For example, applications
that use the Real-Time Protocol (RTP) [RFC3550] will likely want to
protect the RTP header against corruption. Applications, 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
packets that is not smaller than the smallest coverage used by the
sender [RFC3828]. The receiver SHOULD select a threshold that is
sufficiently large to block packets with an inappropriately short
coverage field. 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 datagrams
being treated the same as UDP datagrams, i.e., result in packet loss.
Use of IP fragmentation can also prevent special treatment for UDP-
Lite datagrams, 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 the middlebox.
For example, NATs and firewalls typically define the partial path on
one side of them to be interior to the domain they serve, 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 that use UDP for communication operate across
middleboxes without needing to employ additional mechanisms. One
example is the Domain Name System (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
application-layer sessions and state.
For some applications, such as media transmissions, this re-
synchronization is highly undesirable, because it can cause user-
perceivable playback artifacts. Such specialized applications MAY
send periodic keep-alive messages to attempt to refresh middlebox
state. It is important to note that keep-alive messages are NOT
RECOMMENDED for general use - they are unnecessary for many
applications and can consume significant amounts of system and
network resources.
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An application that needs to employ keep-alives to deliver useful
service over UDP in the presence of middleboxes SHOULD NOT transmit
them more frequently than once every 15 seconds and SHOULD use longer
intervals when possible. 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. However, empirical
evidence suggests that a significant fraction of the deployed
middleboxes unfortunately uses shorter timeouts. The timeout of 15
seconds originates with the Interactive Connectivity Establishment
(ICE) protocol [I-D.ietf-mmusic-ice]. When applications are deployed
in more controlled network environments, the deployers SHOULD
investigate whether the target environment allows applications to use
longer intervals, or whether it offers mechanisms to explicitly
control middlebox state timeout durations, for example, using MIDCOM
[RFC3303], NSIS [I-D.ietf-nsis-nslp-natfw] or UPnP [UPNP]. It is
RECOMMENDED that applications apply slight random variations
("jitter") to the timing of keep-alive transmissions, in order to
reduce the potential for persistent synchronization between keep-
alive transmissions from different hosts.
Sending keep-alives is not a substitute for implementing robust
connection handling. Like all UDP datagrams, 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.
Keep-alive messages are NOT RECOMMENDED for general use. They are
unnecessary for many applications and can consume significant amounts
of system and network resources. For example, on battery-powered
devices, if an application needs to maintain connectivity for long
periods with little traffic, the frequency at which keep-alives are
sent can become the determining factor that governs power
consumption, depending on the underlying network technology. Because
many middleboxes are designed to require keep-alives for TCP
connections at a frequency that is much lower than that needed for
UDP, this difference alone can often be sufficient to prefer TCP over
UDP for these deployments. On the other hand, there is anecdotal
evidence that suggests that direct communication through middleboxes,
e.g., by using ICE [I-D.ietf-mmusic-ice], does succeed less often
with TCP than with UDP. The tradeoffs between different transport
protocols - especially when it comes to middlebox traversal - deserve
careful analysis.
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3.6. Programming Guidelines
The de facto standard application programming interface (API) for
TCP/IP applications is the "sockets" interface [POSIX]. Although
this API was developed for UNIX in the early 1980s, a wide variety of
non-UNIX operating systems also implements it. The sockets API
supports both IPv4 and IPv6 [RFC3493]. The UDP sockets API differs
from that for TCP in several key ways. Because application
programmers are typically more familiar with the TCP sockets API, the
remainder of this section discusses these differences. [STEVENS]
provides usage examples of the UDP sockets API.
UDP datagrams may be directly sent and received, without any
connection setup. Using the sockets API, applications can receive
packets from more than one IP source address on a single UDP socket.
Some servers use this to exchange data with more than one remote host
through a single UDP socket at the same time. When applications need
to ensure that they receive packets from a particular source address,
they MUST implement corresponding checks at the application layer or
explicitly request that the operating system filter the received
packets.
If a client/server application executes on a host with more than one
IP interface, the application SHOULD send any UDP responses with an
IP source address that matches the IP destination address of the UDP
datagram that carried the request (see [RFC1122], Section 4.1.3.5).
Many middleboxes expect this transmission behavior and drop replies
that are sent from a different IP address, as explained in
Section 3.5.
A UDP receiver can receive a valid UDP datagram with a zero-length
payload. Note that this is different from a return value of zero
from a read() socket call, which for TCP indicates the end of the
connection.
Many operating systems also allow a UDP socket to be connected, i.e.,
to bind a UDP socket to a specific pair of addresses and ports. This
is similar to the corresponding TCP sockets API functionality.
However, for UDP, this is only a local operation that serves to
simplify the local send/receive functions and to filter the traffic
for the specified addresses and ports. Binding a UDP socket does not
establish a connection - UDP does not notify the remote end when a
local UDP socket is bound. Binding a socket also allows configuring
options that affect the UDP or IP layers, for example, use of the UDP
checksum or the IP Time Stamp Option. On some stacks, a bound socket
also allows an application to be notified when ICMP error messages
are received for its transmissions [RFC1122].
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UDP provides no flow-control. This is another reason why UDP-based
applications need to be robust in the presence of packet loss. This
loss can also occur within the sending host, when an application
sends data faster than the line rate of the outbound network
interface. It can also occur on the destination, where receive calls
fail to return all the data that was sent when the application issues
them too infrequently (i.e., such that the receive buffer overflows).
Robust flow control mechanisms are difficult to implement, which is
why applications that need this functionality SHOULD consider using a
full-featured transport protocol.
When an application closes a TCP, SCTP or DCCP socket, the transport
protocol on the receiving host is required to maintain TIME-WAIT
state. This prevents delayed packets from the closed connection
instance from being mistakenly associated with a later connection
instance that happens to reuse the same IP address and port pairs.
The UDP protocol does not implement such a mechanism. Therefore,
UDP-based applications need to be robust in this case. One
application may close a socket or terminate, followed in time by
another application receiving on the same port. This later
application may then receive packets intended for the first
application that were delayed in the network.
3.7. ICMP Guidelines
Applications can utilize information about ICMP error messages that
the UDP layer passes up for a variety of purposes [RFC1122].
Applications SHOULD validate that the information in the ICMP message
payload, e.g., a reported error condition, corresponds to a UDP
datagram that the application actually sent. Note that not all APIs
have the necessary functions to support this validation, and some
APIs already perform this validation internally before passing ICMP
information to the application.
Any application response to ICMP error messages SHOULD be robust to
temporary routing failures, i.e., transient ICMP "unreachable"
messages should not normally cause a communication abort.
Applications SHOULD appropriately process ICMP messages generated in
response to transmitted traffic. A correct response often requires
context, such as local state about communication instances to each
destination, that although readily available in connection-oriented
transport protocols is not always maintained by UDP-based
applications.
4. Security Considerations
UDP does not provide communications security. Applications that need
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to protect their communications against eavesdropping, tampering, or
message forgery SHOULD employ end-to-end security services provided
by other IETF protocols. Applications that respond to short requests
with potentially large responses are vulnerable to amplification
attacks, and SHOULD authenticate the sender before responding. The
source IP address of a request is not a useful authenticator, because
it can be spoofed.
One option of securing UDP communications is with IPsec [RFC4301],
which can provide authentication for flows of IP packets through the
Authentication Header (AH) [RFC4302] and encryption and/or
authentication through the Encapsulating Security Payload (ESP)
[RFC4303]. Applications use the Internet Key Exchange (IKE)
[RFC4306] to configure IPsec for their sessions. Depending on how
IPsec is configured for a flow, it can authenticate or encrypt the
UDP headers as well as UDP payloads. If an application only requires
authentication, ESP with no encryption but with authentication is
often a better option than AH, because ESP can operate across
middleboxes. In order to be able to use IPsec, an application must
execute on an operating system that implements the IPsec protocol
suite.
Although it is possible to use IPsec to secure UDP communications,
not all operating systems support IPsec or allow applications to
easily configure it for their flows. A second option of securing UDP
communications is through Datagram Transport Layer Security (DTLS)
[RFC4347]. DTLS provides communication privacy by encrypting UDP
payloads. It does not protect the UDP headers. Applications can
implement DTLS without relying on support from the operating system.
Many other options for authenticating or encrypting UDP payloads
exist. For example, the GSS-API security framework [RFC2743] or
Cryptographic Message Syntax (CMS) [RFC3852] could be used to protect
UDP payloads. The IETF standard for securing RTP [RFC3550] realtime
communication sessions over UDP is SRTP [RFC3711]. In some
applications, a better solution is to protect larger standalone
objects, such as files or messages, instead of individual UDP
payloads. In these situations, CMS [RFC3852], S/MIME [RFC3851] or
OpenPGP [RFC4880] could be used. In addition, there are many non-
IETF protocols in this area.
Like congestion control mechanisms, security mechanisms are difficult
to design and implement correctly. It is hence RECOMMENDED that
applications employ well-known standard security mechanisms such as
DTLS or IPsec, rather than inventing their own.
In terms of congestion control, [RFC2309] and [RFC2914] discuss the
dangers of congestion-unresponsive flows to the Internet. This
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document provides guidelines to designers of UDP-based applications
to congestion-control their transmissions, and does not raise any
additional security concerns.
5. Summary
This section summarizes the guidelines made in Section 3 and
Section 4 in a tabular format in Table 1 for easy referencing.
+---------------------------------------------------------+---------+
| Recommendation | Section |
+---------------------------------------------------------+---------+
| MUST tolerate wide range of Internet path conditions | 3 |
| SHOULD use a full-featured transport (TCP, SCTP, DCCP) | |
| | |
| SHOULD control rate of transmission | 3.1 |
| SHOULD perform congestion control over all traffic | |
| | |
| for bulk transfers, | 3.1.1 |
| SHOULD consider implementing TFRC | |
| else, SHOULD otherwise use bandwidth similar to TCP | |
| | |
| for non-bulk transfers, | 3.1.2 |
| SHOULD measure RTT and transmit max. 1 datagram/RTT | |
| else, SHOULD send at most 1 datagram every 3 seconds | |
| | |
| SHOULD NOT send datagrams that exceed the PMTU, i.e., | 3.2 |
| SHOULD discover PMTU or send datagrams < minimum PMTU | |
| | |
| SHOULD handle datagram loss, duplication, reordering | 3.3 |
| SHOULD be robust to delivery delays up to 2 minutes | |
| | |
| SHOULD enable UDP checksum | 3.4 |
| else, MAY use UDP-Lite with suitable checksum coverage | 3.4.1 |
| | |
| SHOULD NOT always send middlebox keep-alives | 3.5 |
| MAY use keep-alives when needed (min. interval 15 sec) | |
| | |
| MUST check IP source address | 3.6 |
| and, for client/server applications | |
| SHOULD send responses from src address matching request | |
| | |
| SHOULD use standard IETF security protocols when needed | 4 |
+---------------------------------------------------------+---------+
Table 1: Summary of recommendations.
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6. IANA Considerations
This document raises no IANA considerations.
(Note to the RFC Editor: Please remove this section upon
publication.)
7. Acknowledgments
Thanks to Paul Aitken, Mark Allman, Francois Audet, Iljitsch van
Beijnum, Stewart Bryant, Remi Denis-Courmont, Wesley Eddy, Pasi
Eronen, Sally Floyd, Robert Hancock, Jeffrey Hutzelman, Cullen
Jennings, Tero Kivinen, Peter Koch, Jukka Manner, Philip Matthews,
Joerg Ott, Colin Perkins, Tom Petch, Carlos Pignataro, Pasi
Sarolahti, Pascal Thubert, Joe Touch 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.
Lars Eggert is partly funded by [TRILOGY], a research project
supported by the European Commission under its Seventh Framework
Program.
8. References
8.1. 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.
[RFC1122] Braden, R., "Requirements for Internet Hosts -
Communication Layers", STD 3, RFC 1122, October 1989.
[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.
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[RFC2460] Deering, S. and R. Hinden, "Internet Protocol, Version 6
(IPv6) Specification", RFC 2460, December 1998.
[RFC2914] Floyd, S., "Congestion Control Principles", BCP 41,
RFC 2914, September 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.
[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.
[RFC4787] Audet, F. and C. Jennings, "Network Address Translation
(NAT) Behavioral Requirements for Unicast UDP", BCP 127,
RFC 4787, January 2007.
[RFC4821] Mathis, M. and J. Heffner, "Packetization Layer Path MTU
Discovery", RFC 4821, March 2007.
8.2. Informative References
[FABER] Faber, T., Touch, J., and W. Yue, "The TIME-WAIT State in
TCP and Its Effect on Busy Servers", Proc. IEEE Infocom,
March 1999.
[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.
[I-D.ietf-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-ietf-dccp-ccid4-02 (work in progress),
February 2008.
[I-D.ietf-dccp-rfc3448bis]
Handley, M., Floyd, S., Padhye, J., and J. Widmer, "TCP
Friendly Rate Control (TFRC): Protocol Specification",
draft-ietf-dccp-rfc3448bis-06 (work in progress),
April 2008.
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[I-D.ietf-mmusic-ice]
Rosenberg, J., "Interactive Connectivity Establishment
(ICE): A Protocol for Network Address Translator (NAT)
Traversal for Offer/Answer Protocols",
draft-ietf-mmusic-ice-19 (work in progress), October 2007.
[I-D.ietf-nsis-nslp-natfw]
Stiemerling, M., Tschofenig, H., Aoun, C., and E. Davies,
"NAT/Firewall NSIS Signaling Layer Protocol (NSLP)",
draft-ietf-nsis-nslp-natfw-18 (work in progress),
February 2008.
[I-D.ietf-nsis-ntlp]
Schulzrinne, H. and R. Hancock, "GIST: General Internet
Signalling Transport", draft-ietf-nsis-ntlp-16 (work in
progress), July 2008.
[POSIX] IEEE Std. 1003.1-2001, "Standard for Information
Technology - Portable Operating System Interface (POSIX)",
Open Group Technical Standard: Base Specifications Issue
6, ISO/IEC 9945:2002, December 2001.
[RFC0896] Nagle, J., "Congestion control in IP/TCP internetworks",
RFC 896, January 1984.
[RFC0919] Mogul, J., "Broadcasting Internet Datagrams", STD 5,
RFC 919, October 1984.
[RFC1112] Deering, S., "Host extensions for IP multicasting", STD 5,
RFC 1112, August 1989.
[RFC1536] Kumar, A., Postel, J., Neuman, C., Danzig, P., and S.
Miller, "Common DNS Implementation Errors and Suggested
Fixes", RFC 1536, October 1993.
[RFC1546] Partridge, C., Mendez, T., and W. Milliken, "Host
Anycasting Service", RFC 1546, November 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.
[RFC2675] Borman, D., Deering, S., and R. Hinden, "IPv6 Jumbograms",
RFC 2675, August 1999.
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[RFC2743] Linn, J., "Generic Security Service Application Program
Interface Version 2, Update 1", RFC 2743, January 2000.
[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.
[RFC3124] Balakrishnan, H. and S. Seshan, "The Congestion Manager",
RFC 3124, June 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.
[RFC3303] Srisuresh, P., Kuthan, J., Rosenberg, J., Molitor, A., and
A. Rayhan, "Middlebox communication architecture and
framework", RFC 3303, August 2002.
[RFC3493] Gilligan, R., Thomson, S., Bound, J., McCann, J., and W.
Stevens, "Basic Socket Interface Extensions for IPv6",
RFC 3493, February 2003.
[RFC3550] Schulzrinne, H., Casner, S., Frederick, R., and V.
Jacobson, "RTP: A Transport Protocol for Real-Time
Applications", STD 64, RFC 3550, July 2003.
[RFC3551] Schulzrinne, H. and S. Casner, "RTP Profile for Audio and
Video Conferences with Minimal Control", STD 65, RFC 3551,
July 2003.
[RFC3711] Baugher, M., McGrew, D., Naslund, M., Carrara, E., and K.
Norrman, "The Secure Real-time Transport Protocol (SRTP)",
RFC 3711, March 2004.
[RFC3738] Luby, M. and V. Goyal, "Wave and Equation Based Rate
Control (WEBRC) Building Block", RFC 3738, April 2004.
[RFC3758] Stewart, R., Ramalho, M., Xie, Q., Tuexen, M., and P.
Conrad, "Stream Control Transmission Protocol (SCTP)
Partial Reliability Extension", RFC 3758, May 2004.
[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.
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[RFC3851] Ramsdell, B., "Secure/Multipurpose Internet Mail
Extensions (S/MIME) Version 3.1 Message Specification",
RFC 3851, July 2004.
[RFC3852] Housley, R., "Cryptographic Message Syntax (CMS)",
RFC 3852, July 2004.
[RFC4301] Kent, S. and K. Seo, "Security Architecture for the
Internet Protocol", RFC 4301, December 2005.
[RFC4302] Kent, S., "IP Authentication Header", RFC 4302,
December 2005.
[RFC4303] Kent, S., "IP Encapsulating Security Payload (ESP)",
RFC 4303, December 2005.
[RFC4306] Kaufman, C., "Internet Key Exchange (IKEv2) Protocol",
RFC 4306, December 2005.
[RFC4340] Kohler, E., Handley, M., and S. Floyd, "Datagram
Congestion Control Protocol (DCCP)", RFC 4340, March 2006.
[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.
[RFC4347] Rescorla, E. and N. Modadugu, "Datagram Transport Layer
Security", RFC 4347, April 2006.
[RFC4654] Widmer, J. and M. Handley, "TCP-Friendly Multicast
Congestion Control (TFMCC): Protocol Specification",
RFC 4654, August 2006.
[RFC4880] Callas, J., Donnerhacke, L., Finney, H., Shaw, D., and R.
Thayer, "OpenPGP Message Format", RFC 4880, November 2007.
[RFC4960] Stewart, R., "Stream Control Transmission Protocol",
RFC 4960, September 2007.
[RFC4963] Heffner, J., Mathis, M., and B. Chandler, "IPv4 Reassembly
Errors at High Data Rates", RFC 4963, July 2007.
[RFC4987] Eddy, W., "TCP SYN Flooding Attacks and Common
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Mitigations", RFC 4987, August 2007.
[STEVENS] Stevens, W., Fenner, B., and A. Rudoff, "UNIX Network
Programming, The sockets Networking API", Addison-Wesley,
2004.
[TRILOGY] "Trilogy Project", http://www.trilogy-project.org/.
[UPNP] UPnP Forum, "Internet Gateway Device (IGD) Standardized
Device Control Protocol V 1.0", November 2001.
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://people.nokia.net/~lars/
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/
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Full Copyright Statement
Copyright (C) The IETF Trust (2008).
This document is subject to the rights, licenses and restrictions
contained in BCP 78, and except as set forth therein, the authors
retain all their rights.
This document and the information contained herein are provided on an
"AS IS" basis and THE CONTRIBUTOR, THE ORGANIZATION HE/SHE REPRESENTS
OR IS SPONSORED BY (IF ANY), THE INTERNET SOCIETY, THE IETF TRUST AND
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OR IMPLIED, INCLUDING BUT NOT LIMITED TO ANY WARRANTY THAT THE USE OF
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Intellectual Property
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on the procedures with respect to rights in RFC documents can be
found in BCP 78 and BCP 79.
Copies of IPR disclosures made to the IETF Secretariat and any
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The IETF invites any interested party to bring to its attention any
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rights that may cover technology that may be required to implement
this standard. Please address the information to the IETF at
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Eggert & Fairhurst Expires February 23, 2009 [Page 27]
