GENWiki

Premier IT Outsourcing and Support Services within the UK

User Tools

Site Tools


rfc:rfc5405

Network Working Group L. Eggert Request for Comments: 5405 Nokia BCP: 145 G. Fairhurst Category: Best Current Practice University of Aberdeen

                                                         November 2008
       Unicast UDP Usage Guidelines for Application Designers

Status of This Memo

 This document specifies an Internet Best Current Practices for the
 Internet Community, and requests discussion and suggestions for
 improvements.  Distribution of this memo is unlimited.

Copyright Notice

 Copyright (c) 2008 IETF Trust and the persons identified as the
 document authors.  All rights reserved.
 This document is subject to BCP 78 and the IETF Trust's Legal
 Provisions Relating to IETF Documents (http://trustee.ietf.org/
 license-info) in effect on the date of publication of this document.
 Please review these documents carefully, as they describe your rights
 and restrictions with respect to this document.

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 to 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.

Eggert & Fairhurst Best Current Practice [Page 1] RFC 5405 Unicast UDP Usage Guidelines November 2008

Table of Contents

 1.  Introduction . . . . . . . . . . . . . . . . . . . . . . . . .  3
 2.  Terminology  . . . . . . . . . . . . . . . . . . . . . . . . .  5
 3.  UDP Usage Guidelines . . . . . . . . . . . . . . . . . . . . .  5
   3.1.  Congestion Control Guidelines  . . . . . . . . . . . . . .  6
   3.2.  Message Size Guidelines  . . . . . . . . . . . . . . . . . 11
   3.3.  Reliability Guidelines . . . . . . . . . . . . . . . . . . 12
   3.4.  Checksum Guidelines  . . . . . . . . . . . . . . . . . . . 13
   3.5.  Middlebox Traversal Guidelines . . . . . . . . . . . . . . 15
   3.6.  Programming Guidelines . . . . . . . . . . . . . . . . . . 17
   3.7.  ICMP Guidelines  . . . . . . . . . . . . . . . . . . . . . 18
 4.  Security Considerations  . . . . . . . . . . . . . . . . . . . 19
 5.  Summary  . . . . . . . . . . . . . . . . . . . . . . . . . . . 20
 6.  Acknowledgments  . . . . . . . . . . . . . . . . . . . . . . . 22
 7.  References . . . . . . . . . . . . . . . . . . . . . . . . . . 22
   7.1.  Normative References . . . . . . . . . . . . . . . . . . . 22
   7.2.  Informative References . . . . . . . . . . . . . . . . . . 23

Eggert & Fairhurst Best Current Practice [Page 2] RFC 5405 Unicast UDP Usage Guidelines November 2008

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.

Eggert & Fairhurst Best Current Practice [Page 3] RFC 5405 Unicast UDP Usage Guidelines November 2008

 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.

Eggert & Fairhurst Best Current Practice [Page 4] RFC 5405 Unicast UDP Usage Guidelines November 2008

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], Stream Control Transmission Protocol
 (SCTP) [RFC4960], and SCTP Partial Reliability Extension (SCTP-PR)
 [RFC3758], or Datagram Congestion Control Protocol (DCCP) [RFC4340]
 with its different congestion control types
 [RFC4341][RFC4342][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 from interfering with a

Eggert & Fairhurst Best Current Practice [Page 5] RFC 5405 Unicast UDP Usage Guidelines November 2008

 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, in the case of 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.
 Several approaches to perform congestion control are discussed in the
 remainder of this section.  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],

Eggert & Fairhurst Best Current Practice [Page 6] RFC 5405 Unicast UDP Usage Guidelines November 2008

 where active sessions may share current congestion information in a
 way that is independent of the transport protocol.  Such mechanisms
 have currently failed to see deployment, but would otherwise simplify
 the 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.

Eggert & Fairhurst Best Current Practice [Page 7] RFC 5405 Unicast UDP Usage Guidelines November 2008

3.1.2. Low Data-Volume Applications

 When applications that at any time exchange only a small number of
 UDP datagrams with a destination 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 at any time exchange only a small number of UDP
 datagrams with a destination 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 [GIST] 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 predetermined
 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 [GIST] 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

Eggert & Fairhurst Best Current Practice [Page 8] RFC 5405 Unicast UDP Usage Guidelines November 2008

 case must be more conservative than in the two previous cases,
 because the lack of return traffic prevents the detection of packet
 loss, i.e., congestion events, and the application therefore cannot
 perform exponential back-off to reduce load.
 Applications that communicate bidirectionally SHOULD employ
 congestion control for both directions of the communication.  For
 example, for a client-server, request-response-style application,
 clients SHOULD congestion-control their request transmission to a
 server, and the server SHOULD congestion-control its responses to the
 clients.  Congestion in the forward and reverse direction is
 uncorrelated, and an application SHOULD either independently detect
 and respond to congestion along both directions, or limit new and
 retransmitted requests based on acknowledged responses across the
 entire round-trip path.

3.1.3. UDP Tunnels

 One increasingly popular use of UDP is as a tunneling protocol, where
 a tunnel endpoint encapsulates the packets of another protocol inside
 UDP datagrams and transmits them to another tunnel endpoint, which
 decapsulates the UDP datagrams and forwards the original packets
 contained in the payload.  Tunnels establish virtual links that
 appear to directly connect locations that are distant in the physical
 Internet topology and can be used to create virtual (private)
 networks.  Using UDP as a tunneling protocol is attractive when the
 payload protocol is not supported by middleboxes that may exist along
 the path, because many middleboxes support transmission using UDP.
 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 payload traffic
 is IP-based; second, whether the tunneling scheme generates UDP
 traffic at a volume that corresponds to the volume of payload traffic
 carried within the tunnel.
 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

Eggert & Fairhurst Best Current Practice [Page 9] RFC 5405 Unicast UDP Usage Guidelines November 2008

 IP-based traffic should already interact appropriately with other
 traffic sharing the path, and specific congestion control mechanisms
 for the tunnel are not necessary.
 However, if the IP traffic in the tunnel is known to not be
 congestion-controlled, additional measures are RECOMMENDED in order
 to limit the impact of the tunneled traffic on other traffic sharing
 the path.
 The following guidelines define these possible cases in more detail:
 1.  A tunnel generates UDP traffic at a volume that corresponds to
     the volume of payload traffic, and the payload traffic is IP-
     based and congestion-controlled.
     This is arguably the most common case for Internet tunnels.  In
     this case, the UDP tunnel SHOULD NOT employ its own congestion
     control mechanism, because congestion losses of tunneled traffic
     will already trigger an appropriate congestion response at the
     original senders of the tunneled traffic.
     Note that this guideline is built on the assumption that most IP-
     based communication is congestion-controlled.  If a UDP tunnel is
     used for IP-based traffic that is known to not be congestion-
     controlled, the next set of guidelines applies.
 2.  A tunnel generates UDP traffic at a volume that corresponds to
     the volume of payload traffic, and the payload traffic is not
     known to be IP-based, or is known to be IP-based but not
     congestion-controlled.
     This can be the case, for example, when some link-layer protocols
     are encapsulated within UDP (but not all link-layer protocols;
     some are congestion-controlled).  Because it is not known that
     congestion losses of tunneled non-IP traffic will trigger an
     appropriate congestion response at the senders, the UDP tunnel
     SHOULD employ an appropriate congestion control mechanism.
     Because tunnels are usually bulk-transfer applications as far as
     the intermediate routers are concerned, the guidelines in
     Section 3.1.1 apply.
 3.  A tunnel generates UDP traffic at a volume that does not
     correspond to the volume of payload traffic, independent of
     whether the payload traffic is IP-based or congestion-controlled.
     Examples of this class include UDP tunnels that send at a
     constant rate, increase their transmission rates under loss, for
     example, due to increasing redundancy when Forward Error

Eggert & Fairhurst Best Current Practice [Page 10] RFC 5405 Unicast UDP Usage Guidelines November 2008

     Correction is used, or are otherwise constrained in their
     transmission behavior.  These specialized uses of UDP for
     tunneling go beyond the scope of the general guidelines given in
     this document.  The implementer of such specialized tunnels
     SHOULD carefully consider congestion control in the design of
     their tunneling mechanism.
 Designing a tunneling mechanism requires significantly more expertise
 than needed for many other UDP applications, because tunnels
 virtualize lower-layer components of the Internet, and the
 virtualized components need to correctly interact with the
 infrastructure at that layer.  This document only touches upon the
 congestion control considerations for implementing UDP tunnels; a
 discussion of other required tunneling behavior is out of scope.

3.2. Message Size Guidelines

 IP fragmentation lowers the efficiency and reliability of Internet
 communication.  The loss of a single fragment results in the loss of
 an entire fragmented packet, because even if all other fragments are
 received correctly, the original packet cannot be reassembled and
 delivered.  This fundamental issue with fragmentation exists for both
 IPv4 and IPv6.  In addition, some network address translators (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 than 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

Eggert & Fairhurst Best Current Practice [Page 11] RFC 5405 Unicast UDP Usage Guidelines November 2008

 less than the maximum link payload size.  Transmission of minimum-
 sized UDP datagrams is inefficient over paths that support a larger
 PMTU, which is a second reason to implement PMTU discovery.
 To determine an appropriate UDP payload size, applications MUST
 subtract the size of the IP header (which includes any IPv4 optional
 headers or IPv6 extension headers) as well as the length of the UDP
 header (8 bytes) from the PMTU size.  This size, known as the MMS_S,
 can be obtained from the TCP/IP stack [RFC1122].
 Applications that do not send messages that exceed the effective PMTU
 of IPv4 or IPv6 need not implement any of the above mechanisms.  Note
 that the presence of tunnels can cause an additional reduction of the
 effective PMTU, so implementing PMTU discovery may be beneficial.
 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 maximum delay a TCP segment should
 experience -- the Maximum Segment Lifetime (MSL) -- as 2 minutes.  No

Eggert & Fairhurst Best Current Practice [Page 12] RFC 5405 Unicast UDP Usage Guidelines November 2008

 other RFC defines an MSL for other transport protocols or IP itself.
 This document clarifies that the MSL value to be used for UDP SHOULD
 be the same 2 minutes as for TCP.  Applications SHOULD be robust to
 the reception of delayed or duplicate packets that are received
 within this 2-minute interval.
 An application that requires reliable and ordered message delivery
 SHOULD choose an IETF standard transport protocol that provides these
 features.  If this is not possible, it will need to implement a set
 of appropriate mechanisms itself.

3.4. Checksum Guidelines

 The UDP header includes an optional, 16-bit one's 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 the 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

Eggert & Fairhurst Best Current Practice [Page 13] RFC 5405 Unicast UDP Usage Guidelines November 2008

 basic UDP.  Applications that choose to use UDP-Lite instead of UDP
 should still follow the congestion control and other guidelines
 described for use with UDP in Section 3.
 UDP-Lite changes the semantics of the UDP "payload length" field to
 that of a "checksum coverage length" field.  Otherwise, UDP-Lite is
 semantically identical to UDP.  The interface of UDP-Lite differs
 from that of UDP by the addition of a single (socket) option that
 communicates a checksum coverage length value: at the sender, this
 specifies the intended checksum coverage, with the remaining
 unprotected part of the payload called the "error-insensitive part".
 By default, the UDP-Lite checksum coverage extends across the entire
 datagram.  If required, an application may dynamically modify this
 length value, e.g., to offer greater protection to some messages.
 UDP-Lite always verifies that a packet was delivered to the intended
 destination, i.e., always verifies the header fields.  Errors in the
 insensitive part will not cause a UDP datagram to be discarded by the
 destination.  Applications using UDP-Lite therefore MUST NOT make
 assumptions regarding the correctness of the data received in the
 insensitive part of the UDP-Lite payload.
 The sending application SHOULD select the minimum checksum coverage
 to include all sensitive protocol headers.  For example, applications
 that use the Real-Time Protocol (RTP) [RFC3550] will likely want to
 protect the RTP header against corruption.  Applications, where
 appropriate, MUST also introduce their own appropriate validity
 checks for protocol information carried in the insensitive part of
 the UDP-Lite payload (e.g., internal CRCs).
 The receiver must set a minimum coverage threshold for incoming
 packets that is not smaller than the smallest coverage used by the
 sender [RFC3828].  The receiver SHOULD select a threshold that is
 sufficiently large to block packets with an inappropriately short
 coverage field.  This may be a fixed value, or may be negotiated by
 an application.  UDP-Lite does not provide mechanisms to negotiate
 the checksum coverage between the sender and receiver.
 Applications may still experience packet loss, rather than
 corruption, when using UDP-Lite.  The enhancements offered by UDP-
 Lite rely upon a link being able to intercept the UDP-Lite header to
 correctly identify the partial coverage required.  When tunnels
 and/or encryption are used, this can result in UDP-Lite datagrams
 being treated the same as UDP datagrams, i.e., result in packet loss.
 Use of IP fragmentation can also prevent special treatment for UDP-
 Lite datagrams, and this is another reason why applications SHOULD
 avoid IP fragmentation (Section 3.2).

Eggert & Fairhurst Best Current Practice [Page 14] RFC 5405 Unicast UDP Usage Guidelines November 2008

3.5. Middlebox Traversal Guidelines

 Network address translators (NATs) and firewalls are examples of
 intermediary devices ("middleboxes") that can exist along an end-to-
 end path.  A middlebox typically performs a function that requires it
 to maintain per-flow state.  For connection-oriented protocols, such
 as TCP, middleboxes snoop and parse the connection-management traffic
 and create and destroy per-flow state accordingly.  For a
 connectionless protocol such as UDP, this approach is not possible.
 Consequently, middleboxes may create per-flow state when they see a
 packet that indicates a new flow, and destroy the state after some
 period of time during which no packets belonging to the same flow
 have arrived.
 Depending on the specific function that the middlebox performs, this
 behavior can introduce a time-dependency that restricts the kinds of
 UDP traffic exchanges that will be successful across the middlebox.
 For example, NATs and firewalls typically define the partial path on
 one side of them to be interior to the domain they serve, whereas the
 partial path on their other side is defined to be exterior to that
 domain.  Per-flow state is typically created when the first packet
 crosses from the interior to the exterior, and while the state is
 present, NATs and firewalls will forward return traffic.  Return
 traffic that arrives after the per-flow state has timed out is
 dropped, as is other traffic that arrives 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.

Eggert & Fairhurst Best Current Practice [Page 15] RFC 5405 Unicast UDP Usage Guidelines November 2008

 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.  NATs require a state
 timeout of 2 minutes or longer [RFC4787].  However, empirical
 evidence suggests that a significant fraction of currently deployed
 middleboxes unfortunately use shorter timeouts.  The timeout of 15
 seconds originates with the Interactive Connectivity Establishment
 (ICE) protocol [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 Middlebox
 Communications (MIDCOM) [RFC3303], Next Steps in Signaling (NSIS)
 [NSLP], or Universal Plug and Play (UPnP) [UPnP].  It is RECOMMENDED
 that applications apply slight random variations ("jitter") to the
 timing of keep-alive transmissions, 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 may consume significant
 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 [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.

Eggert & Fairhurst Best Current Practice [Page 16] RFC 5405 Unicast UDP Usage Guidelines November 2008

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
 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 implement this.  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.  Many applications need
 to ensure that they receive packets from a particular source address;
 these applications 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

Eggert & Fairhurst Best Current Practice [Page 17] RFC 5405 Unicast UDP Usage Guidelines November 2008

 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 Timestamp option.  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 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

Eggert & Fairhurst Best Current Practice [Page 18] RFC 5405 Unicast UDP Usage Guidelines November 2008

 have the necessary functions to support this validation, and some
 APIs already perform this validation internally before passing ICMP
 information to the application.
 Any application response to ICMP error messages SHOULD be robust to
 temporary routing failures, i.e., transient ICMP "unreachable"
 messages should not normally cause a communication abort.
 Applications SHOULD appropriately process ICMP messages generated in
 response to transmitted traffic.  A correct response often requires
 context, such as local state about communication instances to each
 destination, that although readily available in connection-oriented
 transport protocols is not always maintained by UDP-based
 applications.

4. Security Considerations

 UDP does not provide communications security.  Applications that need
 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.  An application that uses IPsec requires the support of
 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.

Eggert & Fairhurst Best Current Practice [Page 19] RFC 5405 Unicast UDP Usage Guidelines November 2008

 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]
 communication sessions over UDP is the Secure Real-time Transport
 Protocol (SRTP) [RFC3711].  In some applications, a better solution
 is to protect larger stand-alone objects, such as files or messages,
 instead of individual UDP payloads.  In these situations, CMS
 [RFC3852], S/MIME [RFC3851] or OpenPGP [RFC4880] could be used.  In
 addition, there are many non-IETF protocols in this area.
 Like congestion control mechanisms, security mechanisms are difficult
 to design and implement correctly.  It is hence RECOMMENDED that
 applications employ well-known standard security mechanisms such as
 DTLS or IPsec, rather than inventing their own.
 The Generalized TTL Security Mechanism (GTSM) [RFC5082] 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 Sections 3 and 4 in a
 tabular format (Table 1) for easy referencing.

Eggert & Fairhurst Best Current Practice [Page 20] RFC 5405 Unicast UDP Usage Guidelines November 2008

 +---------------------------------------------------------+---------+
 | Recommendation                                          | Section |
 +---------------------------------------------------------+---------+
 | MUST tolerate a 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 in other ways 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 back-off retransmission timers following loss    |         |
 |                                                         |         |
 | for tunnels carrying IP Traffic,                        | 3.1.3   |
 | SHOULD NOT perform congestion control                   |         |
 |                                                         |         |
 | for non-IP tunnels or rate not determined by traffic,   | 3.1.3   |
 | SHOULD perform congestion control                       |         |
 |                                                         |         |
 | 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 IPv4 UDP checksum                         | 3.4     |
 | MUST enable IPv6 UDP checksum                           |         |
 | 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

Eggert & Fairhurst Best Current Practice [Page 21] RFC 5405 Unicast UDP Usage Guidelines November 2008

6. 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 [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.  Gorry Fairhurst was partly funded by the EC SatNEx project.

7. References

7.1. Normative References

 [RFC0768]     Postel, J., "User Datagram Protocol", STD 6, RFC 768,
               August 1980.
 [RFC0793]     Postel, J., "Transmission Control Protocol", STD 7,
               RFC 793, September 1981.
 [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 Timer", RFC 2988, November 2000.

Eggert & Fairhurst Best Current Practice [Page 22] RFC 5405 Unicast UDP Usage Guidelines November 2008

 [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.

7.2. Informative References

 [BEHAVE-APP]  Ford, B., "Application Design Guidelines for Traversal
               through Network Address Translators", Work in Progress,
               March 2007.
 [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)",
               Work in Progress, February 2008.
 [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.
 [GIST]        Schulzrinne, H. and R. Hancock, "GIST: General Internet
               Signalling Transport", Work in Progress, July 2008.
 [ICE]         Rosenberg, J., "Interactive Connectivity Establishment
               (ICE): A Protocol for Network Address Translator (NAT)
               Traversal for Offer/Answer Protocols", Work
               in Progress, October 2007.
 [NSLP]        Stiemerling, M., Tschofenig, H., Aoun, C., and E.
               Davies, "NAT/Firewall NSIS Signaling Layer Protocol
               (NSLP)", Work in Progress, September 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.

Eggert & Fairhurst Best Current Practice [Page 23] RFC 5405 Unicast UDP Usage Guidelines November 2008

 [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", 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.

Eggert & Fairhurst Best Current Practice [Page 24] RFC 5405 Unicast UDP Usage Guidelines November 2008

 [RFC3303]     Srisuresh, P., Kuthan, J., Rosenberg, J., Molitor, A.,
               and A. Rayhan, "Middlebox communication architecture
               and framework", RFC 3303, August 2002.
 [RFC3493]     Gilligan, R., Thomson, S., Bound, J., McCann, J., and
               W. Stevens, "Basic Socket Interface Extensions for
               IPv6", RFC 3493, February 2003.
 [RFC3550]     Schulzrinne, H., Casner, S., Frederick, R., and V.
               Jacobson, "RTP: A Transport Protocol for Real-Time
               Applications", STD 64, RFC 3550, July 2003.
 [RFC3551]     Schulzrinne, H. and S. Casner, "RTP Profile for Audio
               and Video Conferences with Minimal Control", STD 65,
               RFC 3551, July 2003.
 [RFC3711]     Baugher, M., McGrew, D., Naslund, M., Carrara, E., and
               K. Norrman, "The Secure Real-time Transport Protocol
               (SRTP)", RFC 3711, March 2004.
 [RFC3738]     Luby, M. and V. Goyal, "Wave and Equation Based Rate
               Control (WEBRC) Building Block", RFC 3738, April 2004.
 [RFC3758]     Stewart, R., Ramalho, M., Xie, Q., Tuexen, M., and P.
               Conrad, "Stream Control Transmission Protocol (SCTP)
               Partial Reliability Extension", RFC 3758, May 2004.
 [RFC3819]     Karn, P., Bormann, C., Fairhurst, G., Grossman, D.,
               Ludwig, R., Mahdavi, J., Montenegro, G., Touch, J., and
               L. Wood, "Advice for Internet Subnetwork Designers",
               BCP 89, RFC 3819, July 2004.
 [RFC3851]     Ramsdell, B., "Secure/Multipurpose Internet Mail
               Extensions (S/MIME) Version 3.1 Message Specification",
               RFC 3851, July 2004.
 [RFC3852]     Housley, R., "Cryptographic Message Syntax (CMS)",
               RFC 3852, July 2004.
 [RFC4301]     Kent, S. and K. Seo, "Security Architecture for the
               Internet Protocol", RFC 4301, December 2005.
 [RFC4302]     Kent, S., "IP Authentication Header", RFC 4302,
               December 2005.
 [RFC4303]     Kent, S., "IP Encapsulating Security Payload (ESP)",
               RFC 4303, December 2005.

Eggert & Fairhurst Best Current Practice [Page 25] RFC 5405 Unicast UDP Usage Guidelines November 2008

 [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.
 [RFC5082]     Gill, V., Heasley, J., Meyer, D., Savola, P., and C.
               Pignataro, "The Generalized TTL Security Mechanism
               (GTSM)", RFC 5082, October 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>.

Eggert & Fairhurst Best Current Practice [Page 26] RFC 5405 Unicast UDP Usage Guidelines November 2008

 [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/

Eggert & Fairhurst Best Current Practice [Page 27]

/data/webs/external/dokuwiki/data/pages/rfc/rfc5405.txt · Last modified: 2008/11/17 23:51 by 127.0.0.1

Donate Powered by PHP Valid HTML5 Valid CSS Driven by DokuWiki