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Versions: (draft-eggert-tsvwg-udp-guidelines) 00 01 02 03 04 05 06 07 08 09 10 11 RFC 5405

Transport Area Working Group                                   L. Eggert
Internet-Draft                                                     Nokia
Intended status: BCP                                        G. Fairhurst
Expires: April 13, 2009                           University of Aberdeen
                                                        October 10, 2008


         Unicast UDP Usage Guidelines for Application Designers
                   draft-ietf-tsvwg-udp-guidelines-11

Status of this Memo

   By submitting this Internet-Draft, each author represents that any
   applicable patent or other IPR claims of which he or she is aware
   have been or will be disclosed, and any of which he or she becomes
   aware will be disclosed, in accordance with Section 6 of BCP 79.

   Internet-Drafts are working documents of the Internet Engineering
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   This Internet-Draft will expire on April 13, 2009.

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 . . . . . . . . . . . . . . 14
     3.6.  Programming Guidelines . . . . . . . . . . . . . . . . . . 16
     3.7.  ICMP Guidelines  . . . . . . . . . . . . . . . . . . . . . 18
   4.  Security Considerations  . . . . . . . . . . . . . . . . . . . 19
   5.  Summary  . . . . . . . . . . . . . . . . . . . . . . . . . . . 20
   6.  IANA Considerations  . . . . . . . . . . . . . . . . . . . . . 21
   7.  Acknowledgments  . . . . . . . . . . . . . . . . . . . . . . . 22
   8.  References . . . . . . . . . . . . . . . . . . . . . . . . . . 22
     8.1.  Normative References . . . . . . . . . . . . . . . . . . . 22
     8.2.  Informative References . . . . . . . . . . . . . . . . . . 23
   Authors' Addresses . . . . . . . . . . . . . . . . . . . . . . . . 27
   Intellectual Property and Copyright Statements . . . . . . . . . . 28






























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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)
   [RFC5348], 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.  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.

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



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   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
   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.




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   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
   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



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   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-
       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



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       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
   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



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   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
   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



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   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 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
   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.




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   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).

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



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   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.

   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]



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   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.

3.6.  Programming Guidelines

   The de facto standard application programming interface (API) for
   TCP/IP applications is the "sockets" interface [POSIX].  Some
   platforms also offer applications the ability to directly assemble
   and transmit IP packets through "raw sockets" or similar facilities.
   This is a second, more cumbersome method of using UDP.  The
   guidelines in this document cover all such methods through which an



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   application may use UDP.  Because the sockets API is by far the most
   common method, the remainder of this section discusses it in more
   detail.

   Although the sockets 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.

   The Internet can provide service differentiation to applications
   based on IP-layer packet markings [RFC2475].  This facility can be
   used for UDP traffic.  Different operating systems provide different
   interfaces for marking packets to applications.  Differentiated
   services require support from the network, and application deployers
   need to discuss the provisioning of this functionality with their
   network operator.

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



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   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
   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.



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   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.

   The Generalized TTL Security Mechanism (GTSM) [RFC3682] may be used
   with UDP applications (especially when the intended endpoint is on
   the same link as the sender).  This is a lightweight mechanism that
   allows a receiver to filter unwanted packets.

   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 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.


6.  IANA Considerations

   This document raises no IANA considerations.

   (Note to the RFC Editor: Please remove this section upon
   publication.)






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7.  Acknowledgments

   Thanks to Paul Aitken, Mark Allman, Francois Audet, Iljitsch van
   Beijnum, Stewart Bryant, Remi Denis-Courmont, Lisa Dusseault, 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, Dave Ward 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.

   [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.

   [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.

   [RFC5348]  Floyd, S., Handley, M., Padhye, J., and J. Widmer, "TCP
              Friendly Rate Control (TFRC): Protocol Specification",
              RFC 5348, September 2008.

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-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-19 (work in progress),
              September 2008.

   [I-D.ietf-nsis-ntlp]



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              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.

   [RFC2475]  Blake, S., Black, D., Carlson, M., Davies, E., Wang, Z.,
              and W. Weiss, "An Architecture for Differentiated
              Services", RFC 2475, December 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",



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              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.

   [RFC3682]  Gill, V., Heasley, J., and D. Meyer, "The Generalized TTL
              Security Mechanism (GTSM)", RFC 3682, February 2004.

   [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.

   [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.



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   [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
              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/.



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   [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

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