Transport Area Working Group                                   L. Eggert
Internet-Draft                                                     Nokia
Intended status: Best Current                               G. Fairhurst
Practice                                          University of Aberdeen
Expires: August 8, 2008                                 February 5, 2008

             UDP Usage Guidelines for Application Designers

Status of this Memo

   By submitting this Internet-Draft, each author represents that any
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Copyright Notice

   Copyright (C) The IETF Trust (2008).


   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.
   Congestion control guidelines are a primary focus, but the document
   also provides guidance on 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  . . . . . . . . . . . . . . . . . 10
     3.3.  Reliability Guidelines . . . . . . . . . . . . . . . . . . 11
     3.4.  Checksum Guidelines  . . . . . . . . . . . . . . . . . . . 12
     3.5.  Middlebox Traversal Guidelines . . . . . . . . . . . . . . 13
     3.6.  Programming Guidelines . . . . . . . . . . . . . . . . . . 15
     3.7.  ICMP Guidelines  . . . . . . . . . . . . . . . . . . . . . 17
   4.  Security Considerations  . . . . . . . . . . . . . . . . . . . 17
   5.  Summary  . . . . . . . . . . . . . . . . . . . . . . . . . . . 18
   6.  IANA Considerations  . . . . . . . . . . . . . . . . . . . . . 19
   7.  Acknowledgments  . . . . . . . . . . . . . . . . . . . . . . . 19
   8.  References . . . . . . . . . . . . . . . . . . . . . . . . . . 20
     8.1.  Normative References . . . . . . . . . . . . . . . . . . . 20
     8.2.  Informative References . . . . . . . . . . . . . . . . . . 21
   Authors' Addresses . . . . . . . . . . . . . . . . . . . . . . . . 24
   Intellectual Property and Copyright Statements . . . . . . . . . . 26

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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 messages 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

   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 messages 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 65,507 bytes for IPv4 and 65,527 bytes for
   IPv6.  The transmission of large IP packets usually requires IP

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   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.

   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",
   document are to be interpreted as described in BCP 14, RFC 2119

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 Internet path 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],

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   or DCCP [RFC4340] with its different congestion control types

   If used correctly, these more fully-featured transport protocols are
   not as "heavyweight" as often claimed.  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 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

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, 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 messages 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 messages 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.  Finally, Section 3.1.3 discusses congestion

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   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.

   Finally, in the past, the IETF has investigated integrated congestion
   control mechanisms that act on the traffic aggregate between two
   hosts, i.e., across all communication sessions active at a given time
   independent of specific transport protocols, such as the Congestion
   Manager [RFC3124].  Such mechanisms have failed to see deployment,
   but would otherwise also fulfill the congestion control requirements
   in [RFC2914], because they provide congestion control for UDP

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
   messages 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

   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
   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 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.

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 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 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 applications that exchange too few
   messages 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]

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   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

   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 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.

3.1.3.  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 messages and transmits them to another tunnel endpoint, which
   decapsulates the UDP messages 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.

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   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
   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.  Tunnel generates UDP traffic at a volume that corresponds to the
       volume of payload traffic, and the payload traffic is IP-based
       and hence assumed to be 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.  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-

       This can be 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

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       the intermediate routers are concerned, the guidelines in
       Section 3.1.1 apply.

   3.  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-
       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 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

   Due to these issues, 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

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   Applications that choose to not adapt their transmit message size
   SHOULD NOT send UDP messages that would result in IP datagrams that
   exceed the effective PMTU.  In the absence of actual knowledge of the
   minimum MTU along the path, the effective PMTU depends upon the IP
   version used for transmission.  It is the smaller of 576 bytes and
   the first-hop MTU for IPv4 [RFC1122] and 1280 bytes for IPv6
   [RFC2460].  The effective PMTU for a directly connected destination
   (with no routers on the path) is the configured interface MTU, which
   could be less than the maximum link payload size.  Transmission of
   minimum-sized messages 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 fragmentation even when
   applications send messages that do not exceed the effective PMTU, so
   implementing PMTU discovery will still be beneficial in some cases.

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 message duplication, i.e., an
   application may receive multiple copies of the same message.
   Application designers SHOULD verify that their application handles
   message duplication gracefully, and may consequently need to
   implement mechanisms to detect duplicates.  Even if message reception
   triggers idempotent operations, applications may want to suppress
   duplicate messages 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 message reordering, where
   UDP messages arrive at the receiver in an order different from the
   transmission order.  Applications that require ordered delivery MUST
   reestablish message ordering themselves.

   Finally, it is important to note that delay spikes can be very large.

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   This can cause reordered packets to arrive many seconds after they
   were sent.  [RFC0793] defines the the maximum delay a TCP segment
   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.

   Applications that require 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 a coding point of view [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 assurance 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.

   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 message is received that was originally sent
   to a different destination or is otherwise corrupted.  The use of the
   UDP checksum is MANDATORY when applications transmit UDP over IPv6

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
   User Datagram Protocol (UDP-Lite) [RFC3828] variant of UDP instead of

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   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.

   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".
   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 message 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.  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 messages
   being treated the same as UDP messages, i.e., result in packet loss.
   Use of IP fragmentation can also prevent special treatment for UDP-
   Lite messages, and is another reason why applications SHOULD avoid IP
   fragmentation Section 3.2.

3.5.  Middlebox Traversal Guidelines

   Network address translators (NATs) and firewalls are examples of
   intermediary devices ("middleboxes") that can exist along an end-to-

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   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 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.

   An application that needs to employ keep-alives to deliver useful
   service 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

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   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].  Applications that operate in more
   controlled network environments SHOULD investigate whether the
   environment they operate in allows them 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], STUN
   [I-D.wing-behave-nat-control-stun-usage] or UPnP [UPNP].

   It is important to note that sending keep-alives is not a substitute
   for implementing 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

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 messages 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

   If a client/server application executes on a host with more than one
   IP interface, the application SHOULD send any UDP responses in reply
   to arriving UDP datagrams with an IP source address that matches the
   IP destination address of the datagram that carried the request.
   Many middleboxes expect this transmission behavior and drop replies

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   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

   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 IP source routing.  On some stacks, a bound socket also
   allows an application to be notified when ICMP error messages are
   received for its transmissions [RFC1122].

   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 data when the application issues them too frequently
   (i.e., when no new data has arrived) or not frequently enough (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 robust to 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.

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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 respond to 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

4.  Security Considerations

   UDP does not provide communications security.  Applications that need
   to protect their communications against eavesdropping, tampering, or
   message forgery SHOULD employ end-to-end security services provided
   by other IETF protocols.

   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

   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

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   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.  These include IETF security frameworks such as GSS-API GSS-
   API [RFC2743], SASL [RFC4422] and EAP [RFC3748], which are designed
   to provide security services for network protocols.  For some
   applications, S/MIME [RFC3851] or PGP [RFC4880] might provide a
   better solution, because they provide protection for larger
   standalone objects such as files or messages.  However, they
   generally involve public-key operations on an entire object, which
   can have performance implications.  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
   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.

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   | Recommendation                                          | Section |
   | MUST accommodate 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 1 message/RTT           |         |
   | else, SHOULD send at most 1 message every 3 seconds     |         |
   |                                                         |         |
   | SHOULD NOT send messages that exceed the PMTU, i.e.,    | 3.2     |
   | SHOULD discover PMTU or send messages < minimum PMTU    |         |
   |                                                         |         |
   | SHOULD handle message 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.

6.  IANA Considerations

   This document raises no IANA considerations.

7.  Acknowledgments

   Thanks to Paul Aitken, Mark Allman, Francois Audet, Iljitsch van
   Beijnum, Stewart Bryant, Remi Denis-Courmont, Wesley Eddy, Sally

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   Floyd, Jeffrey Hutzelman, Cullen Jennings, Tero Kivinen, 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 [] 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

8.  References

8.1.  Normative References

   [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.

   [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.

   [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

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              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.

              Ford, B., "Application Design Guidelines for Traversal
              through Network Address  Translators",
              draft-ford-behave-app-05 (work in progress), March 2007.

              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-ccid4-01 (work in progress),
              November 2007.

              Handley, M., Floyd, S., Padhye, J., and J. Widmer, "TCP
              Friendly Rate Control (TFRC): Protocol Specification",
              draft-ietf-dccp-rfc3448bis-04 (work in progress),
              January 2008.

              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.


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              Stiemerling, M., Tschofenig, H., Aoun, C., and E. Davies,
              "NAT/Firewall NSIS Signaling Layer Protocol (NSLP)",
              draft-ietf-nsis-nslp-natfw-17 (work in progress),
              January 2008.

              Schulzrinne, H. and R. Hancock, "GIST: General Internet
              Signalling Transport", draft-ietf-nsis-ntlp-15 (work in
              progress), February 2008.

              Wing, D., Rosenberg, J., and H. Tschofenig, "Discovering,
              Querying, and Controlling Firewalls and NATs",
              draft-wing-behave-nat-control-stun-usage-05 (work in
              progress), October 2007.

   [RFC0896]  Nagle, J., "Congestion control in IP/TCP internetworks",
              RFC 896, January 1984.

   [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.

   [RFC2675]  Borman, D., Deering, S., and R. Hinden, "IPv6 Jumbograms",
              RFC 2675, August 1999.

   [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.

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   [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.

   [RFC3738]  Luby, M. and V. Goyal, "Wave and Equation Based Rate
              Control (WEBRC) Building Block", RFC 3738, April 2004.

   [RFC3748]  Aboba, B., Blunk, L., Vollbrecht, J., Carlson, J., and H.
              Levkowetz, "Extensible Authentication Protocol (EAP)",
              RFC 3748, June 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.

   [RFC3851]  Ramsdell, B., "Secure/Multipurpose Internet Mail
              Extensions (S/MIME) Version 3.1 Message Specification",
              RFC 3851, 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.

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   [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.

   [RFC4422]  Melnikov, A. and K. Zeilenga, "Simple Authentication and
              Security Layer (SASL)", RFC 4422, June 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
              Mitigations", RFC 4987, August 2007.

   [STEVENS]  Stevens, W., Fenner, B., and A. Rudoff, "UNIX Network
              Programming, The sockets Networking API",  Addison-Wesley,

   [TRILOGY]  "Trilogy Project",

   [UPNP]     UPnP Forum, "Internet Gateway Device (IGD) Standardized
              Device Control Protocol V 1.0", November 2001.

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Authors' Addresses

   Lars Eggert
   Nokia Research Center
   P.O. Box 407
   Nokia Group  00045

   Phone: +358 50 48 24461

   Godred Fairhurst
   University of Aberdeen
   Department of Engineering
   Fraser Noble Building
   Aberdeen  AB24 3UE


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Full Copyright Statement

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