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Internet Engineering Task Force (IETF) L. Eggert Request for Comments: 8085 NetApp BCP: 145 G. Fairhurst Obsoletes: 5405 University of Aberdeen Category: Best Current Practice G. Shepherd ISSN: 2070-1721 Cisco Systems

                                                            March 2017
                        UDP Usage Guidelines

Abstract

 The User Datagram Protocol (UDP) provides a minimal message-passing
 transport that has no inherent congestion control mechanisms.  This
 document provides guidelines on the use of UDP for the designers of
 applications, tunnels, and other protocols that use UDP.  Congestion
 control guidelines are a primary focus, but the document also
 provides guidance on other topics, including message sizes,
 reliability, checksums, middlebox traversal, the use of Explicit
 Congestion Notification (ECN), Differentiated Services Code Points
 (DSCPs), and ports.
 Because congestion control is critical to the stable operation of the
 Internet, applications and other 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.  They may also need to implement additional mechanisms,
 depending on how they use UDP.
 Some guidance is also applicable to the design of other protocols
 (e.g., protocols layered directly on IP or via IP-based tunnels),
 especially when these protocols do not themselves provide congestion
 control.
 This document obsoletes RFC 5405 and adds guidelines for multicast
 UDP usage.

Eggert, et al. Best Current Practice [Page 1] RFC 8085 UDP Usage Guidelines March 2017

Status of This Memo

 This memo documents an Internet Best Current Practice.
 This document is a product of the Internet Engineering Task Force
 (IETF).  It represents the consensus of the IETF community.  It has
 received public review and has been approved for publication by the
 Internet Engineering Steering Group (IESG).  Further information on
 BCPs is available in Section 2 of RFC 7841.
 Information about the current status of this document, any errata,
 and how to provide feedback on it may be obtained at
 http://www.rfc-editor.org/info/rfc8085.

Copyright Notice

 Copyright (c) 2017 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.  Code Components extracted from this document must
 include Simplified BSD License text as described in Section 4.e of
 the Trust Legal Provisions and are provided without warranty as
 described in the Simplified BSD License.

Eggert, et al. Best Current Practice [Page 2] RFC 8085 UDP Usage Guidelines March 2017

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 ...................................19
    3.3. Reliability Guidelines ....................................21
    3.4. Checksum Guidelines .......................................22
    3.5. Middlebox Traversal Guidelines ............................25
    3.6. Limited Applicability and Controlled Environments .........27
 4. Multicast UDP Usage Guidelines .................................28
    4.1. Multicast Congestion Control Guidelines ...................30
    4.2. Message Size Guidelines for Multicast .....................32
 5. Programming Guidelines .........................................32
    5.1. Using UDP Ports ...........................................34
    5.2. ICMP Guidelines ...........................................37
 6. Security Considerations ........................................38
 7. Summary ........................................................40
 8. References .....................................................42
    8.1. Normative References ......................................42
    8.2. Informative References ....................................43
 Appendix A. .......................................................53
 Acknowledgments ...................................................55
 Authors' Addresses ................................................55

1. Introduction

 The User Datagram Protocol (UDP) [RFC768] provides a minimal,
 unreliable, best-effort, message-passing transport to applications
 and other protocols (such as tunnels) that wish to operate over IP.
 Both are 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 platform's link interface,
 which is often much greater than the available end-to-end path
 capacity, and doing so contributes to congestion along the path.
 [RFC2914] describes the best current practice for congestion control

Eggert, et al. Best Current Practice [Page 3] RFC 8085 UDP Usage Guidelines March 2017

 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."  [RFC7567] reaffirms this statement.  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.
 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, middlebox traversal and use of
 multicast.
 This document provides guidelines and recommendations.  Although most
 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

Eggert, et al. Best Current Practice [Page 4] RFC 8085 UDP Usage Guidelines March 2017

 application designers cite the respective section(s) of this document
 in the technical specification of their application or protocol and
 explain their rationale for their design choice.
 [RFC5405] was scoped to provide guidelines for unicast applications
 only, whereas this document also provides guidelines for UDP flows
 that use IP anycast, multicast, broadcast, and applications that use
 UDP tunnels to support IP flows.
 Finally, although this document specifically refers to usage of 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, tunnel or control applications that choose to run
 directly over IP by registering their own IP protocol number with
 IANA.  This document is expected to provide useful background reading
 to the designers of such applications and protocols.

2. Terminology

 The key words "MUST", "MUST NOT", "REQUIRED", "SHALL", "SHALL NOT",
 "SHOULD", "SHOULD NOT", "RECOMMENDED", "NOT RECOMMENDED", "MAY", and
 "OPTIONAL" in this document are to be interpreted as described in
 [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.  Doing
 so also avoids issues that protocols using a new IP protocol number
 face when being deployed over the Internet, where middleboxes that
 only support TCP and UDP are sometimes present.  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

Eggert, et al. Best Current Practice [Page 5] RFC 8085 UDP Usage Guidelines March 2017

 [RFC793], 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][RFC5622], or transport
 protocols specified by the IETF in the future.  (UDP-encapsulated
 SCTP [RFC6951] and DCCP [RFC6773] can offer support for traversing
 firewalls and other middleboxes where the native protocols are not
 supported.)
 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 they
 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
 [RFC1122] 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
 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 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.

Eggert, et al. Best Current Practice [Page 6] RFC 8085 UDP Usage Guidelines March 2017

 Several approaches to perform congestion control are discussed in the
 remainder of this section.  This section describes generic topics
 with an intended emphasis on unicast and anycast [RFC1546] usage.
 Not all approaches discussed below are appropriate for all UDP-
 transmitting applications.  Section 3.1.2 discusses congestion
 control options for applications that perform bulk transfers over
 UDP.  Such applications can employ schemes that sample the path over
 several subsequent round-trips during which data is exchanged 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 3.1.3 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 3.1.11 discusses congestion control
 considerations when UDP is used as a tunneling protocol.  Section 4
 provides additional recommendations for broadcast and multicast
 usage.
 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 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. Protocol Timer Guidelines

 Understanding the latency between communicating endpoints is usually
 a crucial part of effective congestion control implementations for
 protocols and applications.  Latency estimation can be used in a
 number of protocol functions, such as calculating a congestion-
 controlled transmission rate, triggering retransmission, and
 detecting packet loss.  Additional protocol functions, for example,
 determining an interval for probing a path, determining an interval
 between keep-alive messages, determining an interval for measuring
 the quality of experience, or determining if a remote endpoint has

Eggert, et al. Best Current Practice [Page 7] RFC 8085 UDP Usage Guidelines March 2017

 responded to a request to perform an action, typically operate over
 longer timescales than congestion control and therefore are not
 covered in this section.
 The general recommendation in this document is that applications
 SHOULD leverage existing congestion control techniques and the
 latency estimators specified therein (see next subsection).  The
 following guidelines are provided for applications that need to
 design their own latency estimation mechanisms.
 The guidelines are framed in terms of "latency" and not "round-trip
 time" because some situations require characterizing only the
 network-based latency (e.g., TCP-Friendly Rate Control (TFRC)
 [RFC5348]), while other cases necessitate inclusion of the time
 required by the remote endpoint to provide feedback (e.g., developing
 an understanding of when to retransmit a message).
 The latency between endpoints is generally a dynamic property.
 Therefore, estimates SHOULD represent some sort of averaging of
 multiple recent measurement samples to account for variance.
 Leveraging an Exponentially Weighted Moving Average (EWMA) has proven
 useful for this purpose (e.g., in TCP [RFC6298] and TFRC [RFC5348]).
 Independent latency estimates SHOULD be maintained for each
 destination with which an endpoint communicates.
 Latency samples MUST NOT be derived from ambiguous transactions.  The
 canonical example is in a protocol that retransmits data, but
 subsequently cannot determine which copy is being acknowledged.  This
 ambiguity makes correct computation of the latency problematic.  See
 the discussion of Karn's algorithm in [RFC6298].  This requirement
 ensures a sender establishes a sound estimate of the latency without
 relying on misleading measurements.
 When a latency estimate is used to arm a timer that provides loss
 detection -- with or without retransmission -- expiry of the timer
 MUST be interpreted as an indication of congestion in the network,
 causing the sending rate to be adapted to a safe conservative rate
 (e.g., TCP collapses the congestion window to one segment [RFC5681]).
 Some applications require an initial latency estimate before the
 latency between endpoints can be empirically sampled.  For instance,
 when arming a retransmission timer, an initial value is needed to
 protect the messages sent before the endpoints sample the latency.
 This initial latency estimate SHOULD generally be as conservative
 (large) as possible for the given application.  For instance, in the
 absence of any knowledge about the latency of a path, TCP requires
 the initial Retransmission Timeout (RTO) to be set to no less than 1

Eggert, et al. Best Current Practice [Page 8] RFC 8085 UDP Usage Guidelines March 2017

 second [RFC6298].  UDP applications SHOULD similarly use an initial
 latency estimate of 1 second.  Values shorter than 1 second can be
 problematic (see the data analysis in the appendix of [RFC6298]).

3.1.2. Bulk-Transfer Applications

 Applications that perform bulk transmission of data to a peer over
 UDP, i.e., applications that exchange more than a few UDP datagrams
 per RTT, SHOULD implement 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 3.1.2, because TFRC already addresses
 them, but it 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 (capacity) 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.  The recommendations for
 managing timers specified in Section 3.1.1 also apply.
 Finally, some bulk-transfer applications may choose not to implement
 any congestion control mechanism and instead rely on transmitting
 across reserved path capacity (see Section 3.1.9).  This might be an
 acceptable choice for a subset of restricted networking environments,
 but is by no means a safe practice for operation over the wider
 Internet.  When the UDP traffic of such applications leaks out into
 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, and they SHOULD verify that sufficient path capacity has
 been reserved for them.

Eggert, et al. Best Current Practice [Page 9] RFC 8085 UDP Usage Guidelines March 2017

3.1.3. Low Data-Volume Applications

 When applications that at any time exchange only a few UDP datagrams
 with a destination implement TFRC or one of the other congestion
 control schemes in Section 3.1.2, 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 few UDP datagrams with
 a destination SHOULD still control their transmission behavior by not
 sending on average more than one UDP datagram per 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 using the methods specified in
 Section 3.1.1.
 Some applications cannot maintain a reliable RTT estimate for a
 destination.  These applications do not need to or are unable to use
 protocol timers to measure the RTT (Section 3.1.1).  Two cases can be
 identified:
 1.  The first case is that of applications that exchange too few UDP
     datagrams with a peer to establish a statistically accurate RTT
     estimate but that can monitor the reliability of transmission
     (Section 3.3).  Such applications MAY use a predetermined
     transmission interval that is exponentially backed off when
     packets are deemed lost.  TCP specifies an initial value of 1
     second [RFC6298], which is also RECOMMENDED as an initial value
     for UDP applications.  Some low data-volume applications, e.g.,
     SIP [RFC3261] and General Internet Signaling Transport (GIST)
     [RFC5971] 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, see
     Section 3.1.1.
 2.  A second case 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.  Shorter values are likely problematic in
     many cases.  Note that the sending rate in this case must be more
     conservative than in the previous cases, because the lack of
     return traffic prevents the detection of packet loss, i.e.,
     congestion, and the application therefore cannot perform
     exponential back off to reduce load.

Eggert, et al. Best Current Practice [Page 10] RFC 8085 UDP Usage Guidelines March 2017

3.1.4. Applications Supporting Bidirectional Communications

 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 directions 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.

3.1.5. Implications of RTT and Loss Measurements on Congestion Control

 Transports such as TCP, SCTP, and DCCP provide timely detection of
 congestion that results in an immediate reduction of their maximum
 sending rate when congestion is experienced.  This reaction is
 typically completed 1-2 RTTs after loss/congestion is encountered.
 Applications using UDP SHOULD implement a congestion control scheme
 that provides a prompt reaction to signals indicating congestion
 (e.g., by reducing the rate within the next RTT following a
 congestion signal).
 The operation of a UDP congestion control algorithm can be very
 different from the way TCP operates.  This includes congestion
 controls that respond on timescales that fit applications that cannot
 usefully work within the "change rate every RTT" model of TCP.
 Applications that experience a low or varying RTT are particularly
 vulnerable to sampling errors (e.g., due to measurement noise or
 timer accuracy).  This suggests the need to average loss/congestion
 and RTT measurements over a longer interval; however, this also can
 contribute additional delay in detecting congestion.  Some
 applications may not react by reducing their sending rate immediately
 for various reasons, including the following: RTT and loss
 measurements are only made periodically (e.g., using RTCP),
 additional time is required to filter information, or the application
 is only able to change its sending rate at predetermined interval
 (e.g., some video codecs).
 When designing a congestion control algorithm, the designer therefore
 needs to consider the total time taken to reduce the load following a
 lack of feedback or a congestion event.  An application where the
 most recent RTT measurement is smaller than the actual RTT or the
 measured loss rate is smaller than the current rate, can result in
 over estimating the available capacity.  Such over-estimation can

Eggert, et al. Best Current Practice [Page 11] RFC 8085 UDP Usage Guidelines March 2017

 result in a sending rate that creates congestion to the application
 or other flows sharing the path capacity, and can contribute to
 congestion collapse -- both of these need to be avoided.
 A congestion control designed for UDP SHOULD respond as quickly as
 possible when it experiences congestion, and it SHOULD take into
 account both the loss rate and the response time when choosing a new
 rate.  The implemented congestion control scheme SHOULD result in
 bandwidth (capacity) use that is comparable to that of TCP within an
 order of magnitude, so that it does not starve other flows sharing a
 common bottleneck.

3.1.6. Burst Mitigation and Pacing

 UDP applications SHOULD provide mechanisms to regulate the bursts of
 transmission that the application may send to the network.  Many TCP
 and SCTP implementations provide mechanisms that prevent a sender
 from generating long bursts at line-rate, since these are known to
 induce early loss to applications sharing a common network
 bottleneck.  The use of pacing with TCP [ALLMAN] has also been shown
 to improve the coexistence of TCP flows with other flows.  The need
 to avoid excessive transmission bursts is also noted in
 specifications for applications (e.g., [RFC7143]).
 Even low data-volume UDP flows may benefit from packet pacing, e.g.,
 an application that sends three copies of a packet to improve
 robustness to loss is RECOMMENDED to pace out those three packets
 over several RTTs, to reduce the probability that all three packets
 will be lost due to the same congestion event (or other event, such
 as burst corruption).

3.1.7. Explicit Congestion Notification

 Internet applications can use Explicit Congestion Notification (ECN)
 [RFC3168] to gain benefits for the services they support [RFC8087].
 Internet transports, such as TCP, provide a set of mechanisms that
 are needed to utilize ECN.  ECN operates by setting an ECN-capable
 codepoint (ECT(0) or ECT(1)) in the IP header of packets that are
 sent.  This indicates to ECN-capable network devices (routers and
 other devices) that they may mark (set the congestion experienced,
 Congestion Experience (CE) codepoint) rather than drop the IP packet
 as a signal of incipient congestion.
 UDP applications can also benefit from enabling ECN, providing that
 the API supports ECN and that they implement the required protocol
 mechanisms to support ECN.

Eggert, et al. Best Current Practice [Page 12] RFC 8085 UDP Usage Guidelines March 2017

 The set of mechanisms required for an application to use ECN over UDP
 are:
 o  A sender MUST provide a method to determine (e.g., negotiate) that
    the corresponding application is able to provide ECN feedback
    using a compatible ECN method.
 o  A receiver that enables the use of ECN for a UDP port MUST check
    the ECN field at the receiver for each UDP datagram that it
    receives on this port.
 o  The receiving application needs to provide feedback of congestion
    information to the sending application.  This MUST report the
    presence of datagrams received with a CE-mark by providing a
    mechanism to feed this congestion information back to the sending
    application.  The feedback MAY also report the presence of ECT(1)
    and ECT(0)/Not-ECT packets [RFC7560].  ([RFC3168] and [RFC7560]
    specify methods for TCP.)
 o  An application sending ECN-capable datagrams MUST provide an
    appropriate congestion reaction when it receives feedback
    indicating that congestion has been experienced.  This ought to
    result in reduction of the sending rate by the UDP congestion
    control method (see Section 3.1) that is not less than the
    reaction of TCP under equivalent conditions.
 o  A sender SHOULD detect network paths that do not support the ECN
    field correctly.  When detected, they need to either
    conservatively react to congestion or even fall back to not using
    ECN [RFC8087].  This method needs to be robust to changes within
    the network path that may occur over the lifetime of a session.
 o  A sender is encouraged to provide a mechanism to detect and react
    appropriately to misbehaving receivers that fail to report
    CE-marked packets [RFC8087].
 [RFC6679] provides guidance and an example of this support, by
 describing a method to allow ECN to be used for UDP-based
 applications using the Real-Time Protocol (RTP).  Applications that
 cannot provide this set of mechanisms, but wish to gain the benefits
 of using ECN, are encouraged to use a transport protocol that already
 supports ECN (such as TCP).

3.1.8. Differentiated Services Model

 An application using UDP can use the differentiated services
 (DiffServ) Quality of Service (QoS) framework.  To enable
 differentiated services processing, a UDP sender sets the

Eggert, et al. Best Current Practice [Page 13] RFC 8085 UDP Usage Guidelines March 2017

 Differentiated Services Code Point (DSCP) field [RFC2475] in packets
 sent to the network.  Normally, a UDP source/destination port pair
 will set a single DSCP value for all packets belonging to a flow, but
 multiple DSCPs can be used as described later in this section.  A
 DSCP may be chosen from a small set of fixed values (the class
 selector code points), or from a set of recommended values defined in
 the Per Hop Behavior (PHB) specifications, or from values that have
 purely local meanings to a specific network that supports DiffServ.
 In general, packets may be forwarded across multiple networks between
 source and destination.
 In setting a non-default DSCP value, an application must be aware
 that DSCP markings may be changed or removed between the traffic
 source and destination.  This has implications on the design of
 applications that use DSCPs.  Specifically, applications SHOULD be
 designed not to rely on implementation of a specific network
 treatment; they need instead to implement congestion control methods
 to determine if their current sending rate is inducing congestion in
 the network.
 [RFC7657] describes the implications of using DSCPs and provides
 recommendations on using multiple DSCPs within a single network five-
 tuple (source and destination addresses, source and destination
 ports, and the transport protocol used, in this case, UDP or
 UDP-Lite), and particularly the expected impact on transport protocol
 interactions, with congestion control or reliability functionality
 (e.g., retransmission, reordering).  Use of multiple DSCPs can result
 in reordering by increasing the set of network forwarding resources
 used by a sender.  It can also increase exposure to resource
 depletion or failure.

3.1.9. QoS, Pre-Provisioned, or Reserved Capacity

 The IETF usually specifies protocols for use within the Best Effort
 General Internet.  Sometimes it is relevant to specify protocols with
 a different applicability.  An application using UDP can use the
 integrated services QoS framework.  This framework is usually made
 available within controlled environments (e.g., within a single
 administrative domain or bilaterally agreed connection between
 domains).  Applications intended for the Internet SHOULD NOT assume
 that QoS mechanisms are supported by the networks they use, and
 therefore need to provide congestion control, error recovery, etc.,
 in case the actual network path does not provide provisioned service.
 Some UDP applications are only expected to be deployed over network
 paths that use pre-provisioned capacity or capacity reserved using
 dynamic provisioning, e.g., through the Resource Reservation Protocol
 (RSVP).  Multicast applications are also used with pre-provisioned

Eggert, et al. Best Current Practice [Page 14] RFC 8085 UDP Usage Guidelines March 2017

 capacity (e.g., IPTV deployments within access networks).  These
 applications MAY choose not to implement any congestion control
 mechanism and instead rely on transmitting only on paths where the
 capacity is provisioned and reserved for this use.  This might be an
 acceptable choice for a subset of restricted networking environments,
 but is by no means a safe practice for operation over the wider
 Internet.  Applications that choose this option SHOULD carefully and
 in detail describe the provisioning and management procedures that
 result in the desired containment.
 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.
 Applications designed for use within a controlled environment (see
 Section 3.6) may be able to exploit network management functions to
 detect whether they are causing congestion, and react accordingly.
 If the traffic of such applications leaks out into unprovisioned
 Internet paths, it can significantly degrade the performance of other
 traffic sharing the path and even result in congestion collapse.
 Protocols designed for such networks SHOULD provide mechanisms at the
 network edge to prevent leakage of traffic into unprovisioned
 Internet paths (e.g., [RFC7510]).  To protect other applications
 sharing the same path, applications SHOULD also deploy an appropriate
 circuit breaker, as described in Section 3.1.10.
 An IETF specification targeting a controlled environment is expected
 to provide an applicability statement that restricts the application
 to the controlled environment (see Section 3.6).

3.1.10. Circuit Breaker Mechanisms

 A transport circuit breaker is an automatic mechanism that is used to
 estimate the congestion caused by a flow, and to terminate (or
 significantly reduce the rate of) the flow when excessive congestion
 is detected [RFC8084].  This is a safety measure to prevent
 congestion collapse (starvation of resources available to other
 flows), essential for an Internet that is heterogeneous and for
 traffic that is hard to predict in advance.
 A circuit breaker is intended as a protection mechanism of last
 resort.  Under normal circumstances, a circuit breaker should not be
 triggered; it is designed to protect things when there is severe
 overload.  The goal is usually to limit the maximum transmission rate
 that reflects the available capacity of a network path.  Circuit
 breakers can operate on individual UDP flows or traffic aggregates,
 e.g., traffic sent using a network tunnel.

Eggert, et al. Best Current Practice [Page 15] RFC 8085 UDP Usage Guidelines March 2017

 [RFC8084] provides guidance and examples on the use of circuit
 breakers.  The use of a circuit breaker in RTP is specified in
 [RFC8083].
 Applications used in the general Internet SHOULD implement a
 transport circuit breaker if they do not implement congestion control
 or operate a low data-volume service (see Section 3.6).  All
 applications MAY implement a transport circuit breaker [RFC8084] and
 are encouraged to consider implementing at least a slow-acting
 transport circuit breaker to provide a protection of last resort for
 their network traffic.

3.1.11. UDP Tunnels

 One increasingly popular use of UDP is as a tunneling protocol
 [INT-TUNNELS], 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.  One example of such a
 protocol is Teredo [RFC4380].  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 UDP-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; and 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 unicast traffic is generally assumed to be congestion
 controlled, i.e., it is assumed that the transport protocols
 generating IP-based unicast traffic at the sender already employ
 mechanisms that are sufficient to address congestion on the path.
 Consequently, a tunnel carrying IP-based unicast traffic should

Eggert, et al. Best Current Practice [Page 16] RFC 8085 UDP Usage Guidelines March 2017

 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 not to be
 congestion controlled, additional measures are RECOMMENDED to limit
 the impact of the tunneled traffic on other traffic sharing the path.
 For the specific case of a tunnel that carries IP multicast traffic,
 see Section 4.1.
 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.  A circuit breaker
     mechanism may provide benefit by controlling the envelope of the
     aggregated 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 or
     circuit breaker mechanism designed for the traffic it carries.
     Because tunnels are usually bulk-transfer applications as far as
     the intermediate routers are concerned, the guidelines in
     Section 3.1.2 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.

Eggert, et al. Best Current Practice [Page 17] RFC 8085 UDP Usage Guidelines March 2017

     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 unconstrained 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 and SHOULD consider use of a circuit
     breaker mechanism.
 The type of encapsulated payload might be identified by a UDP port;
 identified by an Ethernet Type or IP protocol number.  A tunnel
 SHOULD provide mechanisms to restrict the types of flows that may be
 carried by the tunnel.  For instance, a UDP tunnel designed to carry
 IP needs to filter out non-IP traffic at the ingress.  This is
 particularly important when a generic tunnel encapsulation is used
 (e.g., one that encapsulates using an EtherType value).  Such tunnels
 SHOULD provide a mechanism to restrict the types of traffic that are
 allowed to be encapsulated for a given deployment (see
 [INT-TUNNELS]).
 Designing a tunneling mechanism requires significantly more expertise
 than needed for many other UDP applications, because tunnels are
 usually intended to be transparent to the endpoints transmitting over
 them, so they need to correctly emulate the behavior of an IP link
 [INT-TUNNELS], for example:
 o  Requirements for tunnels that carry or encapsulate using ECN code
    points [RFC6040].
 o  Usage of the IP DSCP field by tunnel endpoints [RFC2983].
 o  Encapsulation considerations in the design of tunnels [ENCAP].
 o  Usage of ICMP messages [INT-TUNNELS].
 o  Handling of fragmentation and packet size for tunnels
    [INT-TUNNELS].
 o  Source port usage for tunnels designed to support equal cost
    multipath (ECMP) routing (see Section 5.1.1).
 o  Guidance on the need to protect headers [INT-TUNNELS] and the use
    of checksums for IPv6 tunnels (see Section 3.4.1).
 o  Support for operations and maintenance [INT-TUNNELS].

Eggert, et al. Best Current Practice [Page 18] RFC 8085 UDP Usage Guidelines March 2017

 At the same time, the tunneled traffic is application traffic like
 any other from the perspective of the networks the tunnel transmits
 over.  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; instead, they 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 Maximum Transmission Unit
 (MTU) along 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 (PMTUD) itself [RFC1191]
 [RFC1981] [RFC4821] to determine whether the path to a destination
 will support its desired message size without fragmentation.
 However, the ICMP messages that enable path MTU discovery are being
 increasingly filtered by middleboxes (including Firewalls) [RFC4890].
 When the path includes a tunnel, some devices acting as a tunnel
 ingress discard ICMP messages that originate from network devices
 over which the tunnel passes, preventing these from reaching the UDP
 endpoint.
 Packetization Layer Path MTU Discovery (PLPMTUD) [RFC4821] does not
 rely upon network support for ICMP messages and is therefore
 considered more robust than standard PMTUD.  It is not susceptible to
 "black holing" of ICMP messages.  To operate, PLPMTUD requires
 changes to the way the transport is used: both to transmit probe
 packets and to account for the loss or success of these probes.  This
 not only updates the PMTU algorithm, it also impacts loss recovery,
 congestion control, etc.  These updated mechanisms can be implemented

Eggert, et al. Best Current Practice [Page 19] RFC 8085 UDP Usage Guidelines March 2017

 within a connection-oriented transport (e.g., TCP, SCTP, DCCP), but
 they are not a part of UDP; this type of feedback is not typically
 present for unidirectional applications.
 Therefore, PLPMTUD places additional design requirements on a UDP
 application that wishes to use this method.  This is especially true
 for UDP tunnels, because the overhead of sending probe packets needs
 to be accounted for and may require adding a congestion control
 mechanism to the tunnel (see Section 3.1.11) as well as complicating
 the data path at a tunnel decapsulator.
 Applications that do not follow the recommendation to do PMTU/PLPMTUD
 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
 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 Maximum
 Segment Size (MSS), 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 [INT-TUNNELS], 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.

Eggert, et al. Best Current Practice [Page 20] RFC 8085 UDP Usage Guidelines March 2017

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 protocol.
 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,
 with some duplicates arriving potentially much later than the first.
 Application designers SHOULD handle such datagram duplication
 gracefully, and they may consequently need to implement mechanisms to
 detect duplicates.  Even if UDP datagram reception triggers only
 idempotent operations, applications may want to suppress duplicate
 datagrams to reduce load.
 Applications that require ordered delivery MUST reestablish datagram
 ordering themselves.  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 use multiple transport ports need to be robust to
 reordering between sessions.  Load-balancing techniques within the
 network, such as Equal Cost Multipath (ECMP) forwarding can also
 result in a lack of ordering between different transport sessions,
 even between the same two network endpoints.
 It is important to note that the time by which packets are reordered
 or after which duplicates can still arrive can be very large.  Even
 more importantly, there is no well-defined upper boundary here.
 [RFC793] defines 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.  The MSL
 value defined for TCP is conservative enough that it SHOULD be used
 by other protocols, including UDP.  Therefore, applications SHOULD be
 robust to the reception of delayed or duplicate packets that are
 received within this 2-minute interval.
 Retransmission of lost packets or messages is a common reliability
 mechanism.  Such retransmissions can increase network load in
 response to congestion, worsening that congestion.  Any application
 that uses retransmission is responsible for congestion control of its
 retransmissions (as well as the application's original traffic);
 hence, it is subject to the Congestion Control guidelines in

Eggert, et al. Best Current Practice [Page 21] RFC 8085 UDP Usage Guidelines March 2017

 Section 3.1.  Guidance on the appropriate measurement of RTT in
 Section 3.1.1 also applies for timers used for retransmission packet-
 loss detection.
 Instead of implementing these relatively complex reliability
 mechanisms by itself, an application that requires reliable and
 ordered message delivery SHOULD whenever possible choose an IETF
 standard transport protocol that provides these features.

3.4. Checksum Guidelines

 The UDP header includes an optional, 16-bit one's complement checksum
 that provides an integrity check.  These checks are not strong from a
 coding or cryptographic perspective and are not designed to detect
 physical-layer errors or malicious modification of the datagram
 [RFC3819].  Application developers SHOULD implement additional checks
 where data integrity is important, e.g., through a Cyclic Redundancy
 Check (CRC) or keyed or non-keyed cryptographic hash 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.  Therefore, it protects an application against receiving
 corrupted payload data in place of, or in addition to, the data that
 was sent.  More description of the set of checks performed using the
 checksum field is provided in Section 3.1 of [RFC6396].
 Applications SHOULD enable UDP checksums [RFC1122].  For IPv4,
 [RFC768] permits an option to disable their use, by setting a zero
 checksum value.  An application is permitted to optionally discard
 UDP datagrams with a zero checksum [RFC1122].
 When UDP is used over IPv6, the UDP checksum is relied upon to
 protect both the IPv6 and UDP headers from corruption (because IPv6
 lacks a checksum) and MUST be used as specified in [RFC2460].  Under
 specific conditions, a UDP application is allowed to use a zero UDP
 zero-checksum mode with a tunnel protocol (see Section 3.4.1).
 Applications that choose to disable UDP checksums 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.

Eggert, et al. Best Current Practice [Page 22] RFC 8085 UDP Usage Guidelines March 2017

3.4.1. IPv6 Zero UDP Checksum

 [RFC6935] defines a method that enables use of a zero UDP zero-
 checksum mode with a tunnel protocol, providing that the method
 satisfies the requirements in [RFC6936].  The application MUST
 implement mechanisms and/or usage restrictions when enabling this
 mode.  This includes defining the scope for usage and measures to
 prevent leakage of traffic to other UDP applications (see Appendix A
 and Section 3.6).  These additional design requirements for using a
 zero IPv6 UDP checksum are not present for IPv4, since the IPv4
 header validates information that is not protected in an IPv6 packet.
 Key requirements are:
 o  Use of the UDP checksum with IPv6 MUST be the default
    configuration for all implementations [RFC6935].  The receiving
    endpoint MUST only allow the use of UDP zero-checksum mode for
    IPv6 on a UDP destination port that is specifically enabled.
 o  An application that supports a checksum different than that in
    [RFC2460] MUST comply with all implementation requirements
    specified in Section 4 of [RFC6936] and with the usage
    requirements specified in Section 5 of [RFC6936].
 o  A UDP application MUST check that the source and destination IPv6
    addresses are valid for any packets with a UDP zero-checksum and
    MUST discard any packet for which this check fails.  To protect
    from misdelivery, new encapsulation designs SHOULD include an
    integrity check at the transport layer that includes at least the
    IPv6 header, the UDP header and the shim header for the
    encapsulation, if any [RFC6936].
 o  One way to help satisfy the requirements of [RFC6936] may be to
    limit the usage of such tunnels, e.g., to constrain traffic to an
    operator network, as discussed in Section 3.6.  The encapsulation
    defined for MPLS in UDP [RFC7510] chooses this approach.
 As in IPv4, IPv6 applications that choose to disable UDP checksums
 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.
 IPv6 datagrams with a zero UDP checksum will not be passed by any
 middlebox that validates the checksum based on [RFC2460] or that
 updates the UDP checksum field, such as NATs or firewalls.  Changing
 this behavior would require such middleboxes to be updated to
 correctly handle datagrams with zero UDP checksums.  To ensure end-
 to-end robustness, applications that may be deployed in the general
 Internet MUST provide a mechanism to safely fall back to using a

Eggert, et al. Best Current Practice [Page 23] RFC 8085 UDP Usage Guidelines March 2017

 checksum when a path change occurs that redirects a zero UDP checksum
 flow over a path that includes a middlebox that discards IPv6
 datagrams with a zero UDP checksum.

3.4.2. 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.
 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 the checksum coverage length: 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.  Therefore, applications using UDP-Lite MUST NOT make
 assumptions regarding the correctness of the data received in the
 insensitive part of the UDP-Lite payload.
 A UDP-Lite sender SHOULD select the minimum checksum coverage to
 include all sensitive payload information.  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).
 A UDP-Lite 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 it may be negotiated
 by an application.  UDP-Lite does not provide mechanisms to negotiate
 the checksum coverage between the sender and receiver.  Therefore,
 this needs to be performed by the application.

Eggert, et al. Best Current Practice [Page 24] RFC 8085 UDP Usage Guidelines March 2017

 Applications can still experience packet loss 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).
 UDP-Lite is supported in some endpoint protocol stacks.  Current
 support for middlebox traversal using UDP-Lite is poor, because UDP-
 Lite uses a different IPv4 protocol number or IPv6 "next header"
 value than that used for UDP; therefore, few middleboxes are
 currently able to interpret UDP-Lite and take appropriate actions
 when forwarding the packet.  This makes UDP-Lite less suited for
 applications needing general Internet support, until such time as
 UDP-Lite has achieved better support in middleboxes.

3.5. Middlebox Traversal Guidelines

 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 information, and create and
 destroy per-flow state accordingly.  For a connectionless protocol
 such as UDP, this approach is not possible.  Consequently,
 middleboxes can create per-flow state when they see a packet that --
 according to some local criteria -- indicates a new flow, and destroy
 the state after some 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.

Eggert, et al. Best Current Practice [Page 25] RFC 8085 UDP Usage Guidelines March 2017

 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 (e.g., [RFC7675]).  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-alive messages 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 [RFC5245].  When an application is
 deployed in a controlled environment, the deployer 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 the Port
 Control Protocol (PCP) [RFC6887], Middlebox Communications (MIDCOM)
 [RFC3303], Next Steps in Signaling (NSIS) [RFC5973], 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 [RFC7675].

Eggert, et al. Best Current Practice [Page 26] RFC 8085 UDP Usage Guidelines March 2017

 Sending keep-alive messages is not a substitute for implementing a
 mechanism to recover from broken sessions.  Like all UDP datagrams,
 keep-alive messages 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-alive messages.  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-alive messages 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-alive messages
 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 [RFC5245], does succeed less often
 with TCP than with UDP.  The trade-offs between different transport
 protocols -- especially when it comes to middlebox traversal --
 deserve careful analysis.
 UDP applications that could be deployed in the Internet need to be
 designed understanding that there are many variants of middlebox
 behavior, and although UDP is connectionless, middleboxes often
 maintain state for each UDP flow.  Using multiple UDP flows can
 consume available state space and also can lead to changes in the way
 the middlebox handles subsequent packets (either to protect its
 internal resources, or to prevent perceived misuse).  The probability
 of path failure can increase when applications use multiple UDP flows
 in parallel (see Section 5.1.2 for recommendations on usage of
 multiple ports).

3.6. Limited Applicability and Controlled Environments

 Two different types of applicability have been identified for the
 specification of IETF applications that utilize UDP:
 General Internet.  By default, IETF specifications target deployment
    on the general Internet.  Experience has shown that successful
    protocols developed in one specific context or for a particular
    application tend to become used in a wider range of contexts.  For
    example, a protocol with an initial deployment within a local area

Eggert, et al. Best Current Practice [Page 27] RFC 8085 UDP Usage Guidelines March 2017

    network may subsequently be used over a virtual network that
    traverses the Internet, or in the Internet in general.
    Applications designed for general Internet use may experience a
    range of network device behaviors and, in particular, should
    consider whether applications need to operate over paths that may
    include middleboxes.
 Controlled Environment.  A protocol/encapsulation/tunnel could be
    designed to be used only within a controlled environment.  For
    example, an application designed for use by a network operator
    might only be deployed within the network of that single network
    operator or on networks of an adjacent set of cooperating network
    operators.  The application traffic may then be managed to avoid
    congestion, rather than relying on built-in mechanisms, which are
    required when operating over the general Internet.  Applications
    that target a limited applicability use case may be able to take
    advantage of specific hardware (e.g., carrier-grade equipment) or
    underlying protocol features of the subnetwork over which they are
    used.
 Specifications addressing a limited applicability use case or a
 controlled environment SHOULD identify how, in their restricted
 deployment, a level of safety is provided that is equivalent to that
 of a protocol designed for operation over the general Internet (e.g.,
 a design based on extensive experience with deployments of particular
 methods that provide features that cannot be expected in general
 Internet equipment and the robustness of the design of MPLS to
 corruption of headers both helped justify use of an alternate UDP
 integrity check [RFC7510]).
 An IETF specification targeting a controlled environment is expected
 to provide an applicability statement that restricts the application
 traffic to the controlled environment, and it would be expected to
 describe how methods can be provided to discourage or prevent escape
 of corrupted packets from the environment (for example, Section 5 of
 [RFC7510]).

4. Multicast UDP Usage Guidelines

 This section complements Section 3 by providing additional guidelines
 that are applicable to multicast and broadcast usage of UDP.
 Multicast and broadcast transmission [RFC1112] usually employ the UDP
 transport protocol, although they may be used with other transport
 protocols (e.g., UDP-Lite).

Eggert, et al. Best Current Practice [Page 28] RFC 8085 UDP Usage Guidelines March 2017

 There are currently two models of multicast delivery: the Any-Source
 Multicast (ASM) model as defined in [RFC1112] and the Source-Specific
 Multicast (SSM) model as defined in [RFC4607].  ASM group members
 will receive all data sent to the group by any source, while SSM
 constrains the distribution tree to only one single source.
 Specialized classes of applications also use UDP for IP multicast or
 broadcast [RFC919].  The design of such specialized applications
 requires expertise that goes beyond simple, unicast-specific
 guidelines, since these senders may transmit to potentially very many
 receivers across potentially very heterogeneous paths at the same
 time, which significantly complicates congestion control, flow
 control, and reliability mechanisms.
 This section provides guidance on multicast and broadcast UDP usage.
 Use of broadcast by an application is normally constrained by routers
 to the local subnetwork.  However, use of tunneling techniques and
 proxies can and does result in some broadcast traffic traversing
 Internet paths.  These guidelines therefore also apply to broadcast
 traffic.
 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].
 Senders to anycast destinations 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.
 Most UDP tunnels that carry IP multicast traffic use a tunnel
 encapsulation with a unicast destination address, such as Automatic
 Multicast Tunneling [RFC7450].  These MUST follow the same
 requirements as a tunnel carrying unicast data (see Section 3.1.11).
 There are deployment cases and solutions where the outer header of a
 UDP tunnel contains a multicast destination address, such as
 [RFC6513].  These cases are primarily deployed in controlled
 environments over reserved capacity, often operating within a single
 administrative domain, or between two domains over a bilaterally
 agreed upon path with reserved capacity, and so congestion control is
 OPTIONAL, but circuit breaker techniques are still RECOMMENDED in
 order to restore some degree of service should the offered load
 exceed the reserved capacity (e.g., due to misconfiguration).

Eggert, et al. Best Current Practice [Page 29] RFC 8085 UDP Usage Guidelines March 2017

4.1. Multicast Congestion Control Guidelines

 Unicast congestion-controlled transport mechanisms are often not
 applicable to multicast distribution services, or simply do not scale
 to large multicast trees, since they require bidirectional
 communication and adapt the sending rate to accommodate the network
 conditions to a single receiver.  In contrast, multicast distribution
 trees may fan out to massive numbers of receivers, which limits the
 scalability of an in-band return channel to control the sending rate,
 and the one-to-many nature of multicast distribution trees prevents
 adapting the rate to the requirements of an individual receiver.  For
 this reason, generating TCP-compatible aggregate flow rates for
 Internet multicast data, either native or tunneled, is the
 responsibility of the application implementing the congestion
 control.
 Applications using multicast SHOULD provide appropriate congestion
 control.  Multicast congestion control needs to be designed using
 mechanisms that are robust to the potential heterogeneity of both the
 multicast distribution tree and the receivers belonging to a group.
 Heterogeneity may manifest itself in some receivers experiencing more
 loss that others, higher delay, and/or less ability to respond to
 network conditions.  Congestion control is particularly important for
 any multicast session where all or part of the multicast distribution
 tree spans an access network (e.g., a home gateway).  Two styles of
 congestion control have been defined in the RFC Series:
 o  Feedback-based congestion control, in which the sender receives
    multicast or unicast UDP messages from the receivers allowing it
    to assess the level of congestion and then adjust the sender
    rate(s) (e.g., [RFC5740],[RFC4654]).  Multicast methods may
    operate on longer timescales than for unicast (e.g., due to the
    higher group RTT of a heterogeneous group).  A control method
    could decide not to reduce the rate of the entire multicast group
    in response to a control message received from a single receiver
    (e.g., a sender could set a minimum rate and decide to request a
    congested receiver to leave the multicast group and could also
    decide to distribute content to these congested receivers at a
    lower rate using unicast congestion control).
 o  Receiver-driven congestion control, which does not require a
    receiver to send explicit UDP control messages for congestion
    control (e.g., [RFC3738], [RFC5775]).  Instead, the sender
    distributes the data across multiple IP multicast groups (e.g.,
    using a set of {S,G} channels).  Each receiver determines its own
    level of congestion and controls its reception rate using only
    multicast join/leave messages sent in the network control plane.
    This method scales to arbitrary large groups of receivers.

Eggert, et al. Best Current Practice [Page 30] RFC 8085 UDP Usage Guidelines March 2017

 Any multicast-enabled receiver may attempt to join and receive
 traffic from any group.  This may imply the need for rate limits on
 individual receivers or the aggregate multicast service.  Note, at
 the transport layer, there is no way to prevent a join message
 propagating to the next-hop router.
 Some classes of multicast applications support applications that can
 monitor the user-level quality of the transfer at the receiver.
 Applications that can detect a significant reduction in user quality
 SHOULD regard this as a congestion signal (e.g., to leave a group
 using layered multicast encoding); if not, they SHOULD use this
 signal to provide a circuit breaker to terminate the flow by leaving
 the multicast group.

4.1.1. Bulk-Transfer Multicast Applications

 Applications that perform bulk transmission of data over a multicast
 distribution tree, i.e., applications that exchange more than a few
 UDP datagrams per RTT, SHOULD implement a method for congestion
 control.  The currently RECOMMENDED IETF methods are as follows:
 Asynchronous Layered Coding (ALC) [RFC5775], TCP-Friendly Multicast
 Congestion Control (TFMCC) [RFC4654], Wave and Equation Based Rate
 Control (WEBRC) [RFC3738], NACK-Oriented Reliable Multicast (NORM)
 transport protocol [RFC5740], File Delivery over Unidirectional
 Transport (FLUTE) [RFC6726], Real Time Protocol/Control Protocol
 (RTP/RTCP) [RFC3550].
 An application can alternatively implement another congestion control
 scheme following the guidelines of [RFC2887] and utilizing the
 framework of [RFC3048].  Bulk-transfer applications that choose not
 to implement [RFC4654], [RFC5775], [RFC3738], [RFC5740], [RFC6726],
 or [RFC3550] 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] states that multimedia 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.

4.1.2. Low Data-Volume Multicast Applications

 All the recommendations in Section 3.1.3 are also applicable to low
 data-volume multicast applications.

Eggert, et al. Best Current Practice [Page 31] RFC 8085 UDP Usage Guidelines March 2017

4.2. Message Size Guidelines for Multicast

 A multicast application SHOULD NOT send UDP datagrams that result in
 IP packets that exceed the effective MTU as described in Section 3 of
 [RFC6807].  Consequently, an application SHOULD either use the
 effective MTU information provided by the "Population Count
 Extensions to Protocol Independent Multicast (PIM)" [RFC6807] or
 implement path MTU discovery itself (see Section 3.2) to determine
 whether the path to each destination will support its desired message
 size without fragmentation.

5. 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 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, 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.
 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

Eggert, et al. Best Current Practice [Page 32] RFC 8085 UDP Usage Guidelines March 2017

 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].
 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.
 UDP provides no flow-control, i.e., the sender at any given time does
 not know whether the receiver is able to handle incoming
 transmissions.  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 at 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 such as TCP.
 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 to reordering and delay.
 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.

Eggert, et al. Best Current Practice [Page 33] RFC 8085 UDP Usage Guidelines March 2017

5.1. Using UDP Ports

 The rules and procedures for the management of the "Service Name and
 Transport Protocol Port Number Registry" are specified in [RFC6335].
 Recommendations for use of UDP ports are provided in [RFC7605].
 A UDP sender SHOULD NOT use a source port value of zero.  A source
 port number that cannot be easily determined from the address or
 payload type provides protection at the receiver from data injection
 attacks by off-path devices.  A UDP receiver SHOULD NOT bind to port
 zero.
 Applications SHOULD implement receiver port and address checks at the
 application layer or explicitly request that the operating system
 filter the received packets to prevent receiving packets with an
 arbitrary port.  This measure is designed to provide additional
 protection from data injection attacks from an off-path source (where
 the port values may not be known).
 Applications SHOULD provide a check that protects from off-path data
 injection, avoiding an application receiving packets that were
 created by an unauthorized third party.  TCP stacks commonly use a
 randomized source port to provide this protection [RFC6056]; UDP
 applications should follow the same technique.  Middleboxes and end
 systems often make assumptions about the system ports or user ports;
 hence, it is recommended to use randomized ports in the Dynamic and/
 or Private Port range.  Setting a "randomized" source port also
 provides greater assurance that reported ICMP errors originate from
 network systems on the path used by a particular flow.  Some UDP
 applications choose to use a predetermined value for the source port
 (including some multicast applications), these applications need to
 therefore employ a different technique.  Protection from off-path
 data attacks can also be provided by randomizing the initial value of
 another protocol field within the datagram payload, and checking the
 validity of this field at the receiver (e.g., RTP has random initial
 sequence number and random media timestamp offsets [RFC3550]).
 When using multicast, IP routers perform a reverse-path forwarding
 (RPF) check for each multicast packet.  This provides protection from
 off-path data injection, restricting opportunities to forge a
 packet's source address.  When a receiver joins a multicast group and
 filters based on the source address the filter verifies the sender's
 IP address.  This is always the case when using an SSM {S,G} channel.

Eggert, et al. Best Current Practice [Page 34] RFC 8085 UDP Usage Guidelines March 2017

5.1.1. Usage of UDP for Source Port Entropy and the IPv6 Flow Label

 Some applications use the UDP datagram header as a source of entropy
 for network devices that implement ECMP [RFC6438].  A UDP tunnel
 application targeting this usage encapsulates an inner packet using
 UDP, where the UDP source port value forms a part of the entropy that
 can be used to balance forwarding of network traffic by the devices
 that use ECMP.  A sending tunnel endpoint selects a source port value
 in the UDP datagram header that is computed from the inner flow
 information (e.g., the encapsulated packet headers).  To provide
 sufficient entropy, the sending tunnel endpoint maps the encapsulated
 traffic to one of a range of UDP source values.  The value SHOULD be
 within the ephemeral port range, i.e., 49152 to 65535, where the high
 order two bits of the port are set to one.  The available source port
 entropy of 14 bits (using the ephemeral port range) plus the outer IP
 addresses seems sufficient for entropy for most ECMP applications
 [ENCAP].
 To avoid reordering within an IP flow, the same UDP source port value
 SHOULD be used for all packets assigned to an encapsulated flow
 (e.g., using a hash of the relevant headers).  The entropy mapping
 for a flow MAY change over the lifetime of the encapsulated flow
 [ENCAP].  For instance, this could be changed as a Denial of Service
 (DOS) mitigation, or as a means to effect routing through the ECMP
 network.  However, the source port selected for a flow SHOULD NOT
 change more than once in every thirty seconds (e.g., as in
 [RFC8086]).
 The use of the source port field for entropy has several side effects
 that need to be considered, including:
 o  It can increase the probability of misdelivery of corrupted
    packets, which increases the need for checksum computation or an
    equivalent mechanism to protect other UDP applications from
    misdelivery errors Section 3.4.
 o  It is expected to reduce the probability of successful middlebox
    traversal Section 3.5.  This use of the source port field will
    often not be suitable for applications targeting deployment in the
    general Internet.
 o  It can prevent the field being usable to protect from off-path
    attacks (described in Section 5.1).  Designers therefore need to
    consider other mechanisms to provide equivalent protection (e.g.,
    to restrict use to a controlled environment [RFC7510]
    Section 3.6).

Eggert, et al. Best Current Practice [Page 35] RFC 8085 UDP Usage Guidelines March 2017

 The UDP source port number field has also been leveraged to produce
 entropy with IPv6.  However, in the case of IPv6, the "flow label"
 [RFC6437] may also alternatively be used to provide entropy for load
 balancing [RFC6438].  This use of the flow label for load balancing
 is consistent with the definition of the field, although further
 clarity was needed to ensure the field can be consistently used for
 this purpose.  Therefore, an updated IPv6 flow label [RFC6437] and
 ECMP routing [RFC6438] usage was specified.
 To ensure future opportunities to use the flow label, UDP
 applications SHOULD set the flow label field, even when an entropy
 value is also set in the source port field (e.g., An IPv6 tunnel
 endpoint could copy the source port flow entropy value to the IPv6
 flow label field [RFC8086]).  Router vendors are encouraged to start
 using the IPv6 flow label as a part of the flow hash, providing
 support for IP-level ECMP without requiring use of UDP.  The end-to-
 end use of flow labels for load balancing is a long-term solution.
 Even if the usage of the flow label has been clarified, there will be
 a transition time before a significant proportion of endpoints start
 to assign a good quality flow label to the flows that they originate.
 The use of load balancing using the transport header fields will
 likely continue until widespread deployment is finally achieved.

5.1.2. Applications Using Multiple UDP Ports

 A single application may exchange several types of data.  In some
 cases, this may require multiple UDP flows (e.g., multiple sets of
 flows, identified by different five-tuples).  [RFC6335] recommends
 application developers not to apply to IANA to be assigned multiple
 well-known ports (user or system).  It does not discuss the
 implications of using multiple flows with the same well-known port or
 pairs of dynamic ports (e.g., identified by a service name or
 signaling protocol).
 Use of multiple flows can affect the network in several ways:
 o  Starting a series of successive connections can increase the
    number of state bindings in middleboxes (e.g., NAPT or Firewall)
    along the network path.  UDP-based middlebox traversal usually
    relies on timeouts to remove old state, since middleboxes are
    unaware when a particular flow ceases to be used by an
    application.
 o  Using several flows at the same time may result in seeing
    different network characteristics for each flow.  It cannot be
    assumed both follow the same path (e.g., when ECMP is used,
    traffic is intentionally hashed onto different parallel paths
    based on the port numbers).

Eggert, et al. Best Current Practice [Page 36] RFC 8085 UDP Usage Guidelines March 2017

 o  Using several flows can also increase the occupancy of a binding
    or lookup table in a middlebox (e.g., NAPT or Firewall), which may
    cause the device to change the way it manages the flow state.
 o  Further, using excessive numbers of flows can degrade the ability
    of a unicast congestion control to react to congestion events,
    unless the congestion state is shared between all flows in a
    session.  A receiver-driven multicast congestion control requires
    the sending application to distribute its data over a set of IP
    multicast groups, each receiver is therefore expected to receive
    data from a modest number of simultaneously active UDP ports.
 Therefore, applications MUST NOT assume consistent behavior of
 middleboxes when multiple UDP flows are used; many devices respond
 differently as the number of used ports increases.  Using multiple
 flows with different QoS requirements requires applications to verify
 that the expected performance is achieved using each individual flow
 (five-tuple), see Section 3.1.9.

5.2. ICMP Guidelines

 Applications can utilize information about ICMP error messages that
 the UDP layer passes up for a variety of purposes [RFC1122].
 Applications SHOULD appropriately validate the payload of ICMP
 messages to ensure these are received in response to transmitted
 traffic (i.e., a reported error condition that corresponds to a UDP
 datagram actually sent by the application).  This 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.  Note that not all platforms have the necessary APIs to
 support this validation, and some platforms 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 (sometimes called "soft errors"), e.g.,
 transient ICMP "unreachable" messages ought to not normally cause a
 communication abort.
 ICMP messages are being increasingly filtered by middleboxes.  A UDP
 application therefore SHOULD NOT rely on their delivery for correct
 and safe operation.

Eggert, et al. Best Current Practice [Page 37] RFC 8085 UDP Usage Guidelines March 2017

6. 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.
 UDP applications SHOULD provide protection from off-path data
 injection attacks using a randomized source port or equivalent
 technique (see Section 5.1).
 Applications that respond to short requests with potentially large
 responses are a potential vector for amplification attacks, and
 SHOULD take steps to minimize their potential for being abused as
 part of a DoS attack.  That could mean authenticating the sender
 before responding; noting that the source IP address of a request is
 not a useful authenticator, because it can easily be spoofed.  Or it
 may mean otherwise limiting the cases where short unauthenticated
 requests produce large responses.  Applications MAY also want to
 offer ways to limit the number of requests they respond to in a time
 interval, in order to cap the bandwidth they consume.
 One option for 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)
 [RFC7296] 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, and the
 network path must permit IKE and IPsec traffic.  This may become more
 common with IPv6 deployments [RFC6092].
 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 for securing
 UDP communications is through Datagram Transport Layer Security
 (DTLS) [RFC6347][RFC7525].  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, et al. Best Current Practice [Page 38] RFC 8085 UDP Usage Guidelines March 2017

 Many other options for authenticating or encrypting UDP payloads
 exist.  For example, the GSS-API security framework [RFC2743] or
 Cryptographic Message Syntax (CMS) [RFC5652] could be used to protect
 UDP payloads.  There exist a number of security options for RTP
 [RFC3550] over UDP, especially to accomplish key-management, see
 [RFC7201].  These options covers many usages, including point-to-
 point, centralized group communication as well as multicast.  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 [RFC5652], S/MIME [RFC5751] 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 when the intended endpoint is on the same link
 as the sender.  This lightweight mechanism 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.  [RFC8084]
 describes methods that can be used to set a performance envelope that
 can assist in preventing congestion collapse in the absence of
 congestion control or when the congestion control fails to react to
 congestion events.  This document provides guidelines to designers of
 UDP-based applications to congestion-control their transmissions, and
 does not raise any additional security concerns.
 Some network operators have experienced surges of UDP attack traffic
 that are multiple orders of magnitude above the baseline traffic rate
 for UDP.  This can motivate operators to limit the data rate or
 packet rate of UDP traffic.  This may in turn limit the throughput
 that an application can achieve using UDP and could also result in
 higher packet loss for UDP traffic that would not be experienced if
 other transport protocols had been used.
 A UDP application with a long-lived association between the sender
 and receiver, ought to be designed so that the sender periodically
 checks that the receiver still wants ("consents") to receive traffic
 and need to be designed to stop if there is no explicit confirmation
 of this [RFC7675].  Applications that require communications in two
 directions to implement protocol functions (such as reliability or

Eggert, et al. Best Current Practice [Page 39] RFC 8085 UDP Usage Guidelines March 2017

 congestion control) will need to independently check both directions
 of communication, and may have to exchange keep-alive messages to
 traverse middleboxes (see Section 3.5).

7. Summary

 This section summarizes the key guidelines made in Sections 3 - 6 in
 a tabular format (Table 1) for easy referencing.
 +---------------------------------------------------------+---------+
 | Recommendation                                          | Section |
 +---------------------------------------------------------+---------+
 | MUST tolerate a wide range of Internet path conditions  | 3       |
 | SHOULD use a full-featured transport (e.g., TCP)        |         |
 |                                                         |         |
 | SHOULD control rate of transmission                     | 3.1     |
 | SHOULD perform congestion control over all traffic      |         |
 |                                                         |         |
 | for bulk transfers,                                     | 3.1.2   |
 | SHOULD consider implementing TFRC                       |         |
 | else, SHOULD in other ways use bandwidth similar to TCP |         |
 |                                                         |         |
 | for non-bulk transfers,                                 | 3.1.3   |
 | SHOULD measure RTT and transmit max. 1 datagram/RTT     | 3.1.1   |
 | else, SHOULD send at most 1 datagram every 3 seconds    |         |
 | SHOULD back-off retransmission timers following loss    |         |
 |                                                         |         |
 | SHOULD provide mechanisms to regulate the bursts of     | 3.1.6   |
 | transmission                                            |         |
 |                                                         |         |
 | MAY implement ECN; a specific set of application        | 3.1.7   |
 | mechanisms are REQUIRED if ECN is used.                 |         |
 |                                                         |         |
 | for DiffServ, SHOULD NOT rely on implementation of PHBs | 3.1.8   |
 |                                                         |         |
 | for QoS-enabled paths, MAY choose not to use CC         | 3.1.9   |
 |                                                         |         |
 | SHOULD NOT rely solely on QoS for their capacity        | 3.1.10  |
 | non-CC controlled flows SHOULD implement a transport    |         |
 | circuit breaker                                         |         |
 | MAY implement a circuit breaker for other applications  |         |
 |                                                         |         |
 | for tunnels carrying IP traffic,                        | 3.1.11  |
 | SHOULD NOT perform congestion control                   |         |
 | MUST correctly process the IP ECN field                 |         |
 |                                                         |         |

Eggert, et al. Best Current Practice [Page 40] RFC 8085 UDP Usage Guidelines March 2017

 | for non-IP tunnels or rate not determined by traffic,   |         |
 | SHOULD perform CC or use circuit breaker                | 3.1.11  |
 | SHOULD restrict types of traffic transported by the     |         |
 | tunnel                                                  |         |
 |                                                         |         |
 | SHOULD NOT send datagrams that exceed the PMTU, i.e.,   | 3.2     |
 | SHOULD discover PMTU or send datagrams < minimum PMTU;  |         |
 | Specific application mechanisms are REQUIRED if PLPMTUD |         |
 | is used.                                                |         |
 |                                                         |         |
 | 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     |
 | SHOULD enable IPv6 UDP checksum; Specific application   | 3.4.1   |
 | mechanisms are REQUIRED if a zero IPv6 UDP checksum is  |         |
 | used.                                                   |         |
 |                                                         |         |
 | SHOULD provide protection from off-path attacks         | 5.1     |
 | else, MAY use UDP-Lite with suitable checksum coverage  | 3.4.2   |
 |                                                         |         |
 | SHOULD NOT always send middlebox keep-alive messages    | 3.5     |
 | MAY use keep-alives when needed (min. interval 15 sec)  |         |
 |                                                         |         |
 | Applications specified for use in limited use (or       | 3.6     |
 | controlled environments) SHOULD identify equivalent     |         |
 | mechanisms and describe their use case.                 |         |
 |                                                         |         |
 | Bulk-multicast apps SHOULD implement congestion control | 4.1.1   |
 |                                                         |         |
 | Low volume multicast apps SHOULD implement congestion   | 4.1.2   |
 | control                                                 |         |
 |                                                         |         |
 | Multicast apps SHOULD use a safe PMTU                   | 4.2     |
 |                                                         |         |
 | SHOULD avoid using multiple ports                       | 5.1.2   |
 | MUST check received IP source address                   |         |
 |                                                         |         |
 | SHOULD validate payload in ICMP messages                | 5.2     |
 |                                                         |         |
 | SHOULD use a randomized source port or equivalent       | 6       |
 | technique, and, for client/server applications, SHOULD  |         |
 | send responses from source address matching request     |         |
 | 5.1                                                     |         |
 | SHOULD use standard IETF security protocols when needed | 6       |
 +---------------------------------------------------------+---------+
                  Table 1: Summary of Recommendations

Eggert, et al. Best Current Practice [Page 41] RFC 8085 UDP Usage Guidelines March 2017

8. References

8.1. Normative References

 [RFC768]   Postel, J., "User Datagram Protocol", STD 6, RFC 768,
            DOI 10.17487/RFC0768, August 1980,
            <http://www.rfc-editor.org/info/rfc768>.
 [RFC793]   Postel, J., "Transmission Control Protocol", STD 7,
            RFC 793, DOI 10.17487/RFC0793, September 1981,
            <http://www.rfc-editor.org/info/rfc793>.
 [RFC1122]  Braden, R., Ed., "Requirements for Internet Hosts -
            Communication Layers", STD 3, RFC 1122,
            DOI 10.17487/RFC1122, October 1989,
            <http://www.rfc-editor.org/info/rfc1122>.
 [RFC1191]  Mogul, J. and S. Deering, "Path MTU discovery", RFC 1191,
            DOI 10.17487/RFC1191, November 1990,
            <http://www.rfc-editor.org/info/rfc1191>.
 [RFC1981]  McCann, J., Deering, S., and J. Mogul, "Path MTU Discovery
            for IP version 6", RFC 1981, DOI 10.17487/RFC1981, August
            1996, <http://www.rfc-editor.org/info/rfc1981>.
 [RFC2119]  Bradner, S., "Key words for use in RFCs to Indicate
            Requirement Levels", BCP 14, RFC 2119,
            DOI 10.17487/RFC2119, March 1997,
            <http://www.rfc-editor.org/info/rfc2119>.
 [RFC2460]  Deering, S. and R. Hinden, "Internet Protocol, Version 6
            (IPv6) Specification", RFC 2460, DOI 10.17487/RFC2460,
            December 1998, <http://www.rfc-editor.org/info/rfc2460>.
 [RFC2914]  Floyd, S., "Congestion Control Principles", BCP 41,
            RFC 2914, DOI 10.17487/RFC2914, September 2000,
            <http://www.rfc-editor.org/info/rfc2914>.
 [RFC3828]  Larzon, L-A., Degermark, M., Pink, S., Jonsson, L-E., Ed.,
            and G. Fairhurst, Ed., "The Lightweight User Datagram
            Protocol (UDP-Lite)", RFC 3828, DOI 10.17487/RFC3828, July
            2004, <http://www.rfc-editor.org/info/rfc3828>.
 [RFC4787]  Audet, F., Ed. and C. Jennings, "Network Address
            Translation (NAT) Behavioral Requirements for Unicast
            UDP", BCP 127, RFC 4787, DOI 10.17487/RFC4787, January
            2007, <http://www.rfc-editor.org/info/rfc4787>.

Eggert, et al. Best Current Practice [Page 42] RFC 8085 UDP Usage Guidelines March 2017

 [RFC4821]  Mathis, M. and J. Heffner, "Packetization Layer Path MTU
            Discovery", RFC 4821, DOI 10.17487/RFC4821, March 2007,
            <http://www.rfc-editor.org/info/rfc4821>.
 [RFC5348]  Floyd, S., Handley, M., Padhye, J., and J. Widmer, "TCP
            Friendly Rate Control (TFRC): Protocol Specification",
            RFC 5348, DOI 10.17487/RFC5348, September 2008,
            <http://www.rfc-editor.org/info/rfc5348>.
 [RFC5405]  Eggert, L. and G. Fairhurst, "Unicast UDP Usage Guidelines
            for Application Designers", BCP 145, RFC 5405,
            DOI 10.17487/RFC5405, November 2008,
            <http://www.rfc-editor.org/info/rfc5405>.
 [RFC6040]  Briscoe, B., "Tunnelling of Explicit Congestion
            Notification", RFC 6040, DOI 10.17487/RFC6040, November
            2010, <http://www.rfc-editor.org/info/rfc6040>.
 [RFC6298]  Paxson, V., Allman, M., Chu, J., and M. Sargent,
            "Computing TCP's Retransmission Timer", RFC 6298,
            DOI 10.17487/RFC6298, June 2011,
            <http://www.rfc-editor.org/info/rfc6298>.
 [RFC8084]  Fairhurst, G., "Network Transport Circuit Breakers",
            BCP 208, RFC 8084, DOI 10.17487/RFC8084, March 2017,
            <http://www.rfc-editor.org/info/rfc8084>.

8.2. Informative References

 [ALLMAN]   Allman, M. and E. Blanton, "Notes on burst mitigation for
            transport protocols", March 2005.
 [BEHAVE-APP]
            Ford, B., "Application Design Guidelines for Traversal
            through Network Address Translators", Work in Progress,
            draft-ford-behave-app-05, March 2007.
 [ENCAP]    Nordmark, E., Ed., Tian, A., Gross, J., Hudson, J.,
            Kreeger, L., Garg, P., Thaler, P., and T. Herbert,
            "Encapsulation Considerations", Work in Progress,
            draft-ietf-rtgwg-dt-encap-02, October 2016.
 [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.

Eggert, et al. Best Current Practice [Page 43] RFC 8085 UDP Usage Guidelines March 2017

 [INT-TUNNELS]
            Touch, J. and W. Townsley, "IP Tunnels in the Internet
            Architecture", Work in Progress,
            draft-ietf-intarea-tunnels-03, July 2016.
 [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.
 [RFC919]   Mogul, J., "Broadcasting Internet Datagrams", STD 5,
            RFC 919, DOI 10.17487/RFC0919, October 1984,
            <http://www.rfc-editor.org/info/rfc919>.
 [RFC1112]  Deering, S., "Host extensions for IP multicasting", STD 5,
            RFC 1112, DOI 10.17487/RFC1112, August 1989,
            <http://www.rfc-editor.org/info/rfc1112>.
 [RFC1536]  Kumar, A., Postel, J., Neuman, C., Danzig, P., and S.
            Miller, "Common DNS Implementation Errors and Suggested
            Fixes", RFC 1536, DOI 10.17487/RFC1536, October 1993,
            <http://www.rfc-editor.org/info/rfc1536>.
 [RFC1546]  Partridge, C., Mendez, T., and W. Milliken, "Host
            Anycasting Service", RFC 1546, DOI 10.17487/RFC1546,
            November 1993, <http://www.rfc-editor.org/info/rfc1546>.
 [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, DOI 10.17487/RFC2309, April 1998,
            <http://www.rfc-editor.org/info/rfc2309>.
 [RFC2475]  Blake, S., Black, D., Carlson, M., Davies, E., Wang, Z.,
            and W. Weiss, "An Architecture for Differentiated
            Services", RFC 2475, DOI 10.17487/RFC2475, December 1998,
            <http://www.rfc-editor.org/info/rfc2475>.
 [RFC2675]  Borman, D., Deering, S., and R. Hinden, "IPv6 Jumbograms",
            RFC 2675, DOI 10.17487/RFC2675, August 1999,
            <http://www.rfc-editor.org/info/rfc2675>.
 [RFC2743]  Linn, J., "Generic Security Service Application Program
            Interface Version 2, Update 1", RFC 2743,
            DOI 10.17487/RFC2743, January 2000,
            <http://www.rfc-editor.org/info/rfc2743>.

Eggert, et al. Best Current Practice [Page 44] RFC 8085 UDP Usage Guidelines March 2017

 [RFC2887]  Handley, M., Floyd, S., Whetten, B., Kermode, R.,
            Vicisano, L., and M. Luby, "The Reliable Multicast Design
            Space for Bulk Data Transfer", RFC 2887,
            DOI 10.17487/RFC2887, August 2000,
            <http://www.rfc-editor.org/info/rfc2887>.
 [RFC2983]  Black, D., "Differentiated Services and Tunnels",
            RFC 2983, DOI 10.17487/RFC2983, October 2000,
            <http://www.rfc-editor.org/info/rfc2983>.
 [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, DOI 10.17487/RFC3048, January 2001,
            <http://www.rfc-editor.org/info/rfc3048>.
 [RFC3124]  Balakrishnan, H. and S. Seshan, "The Congestion Manager",
            RFC 3124, DOI 10.17487/RFC3124, June 2001,
            <http://www.rfc-editor.org/info/rfc3124>.
 [RFC3168]  Ramakrishnan, K., Floyd, S., and D. Black, "The Addition
            of Explicit Congestion Notification (ECN) to IP",
            RFC 3168, DOI 10.17487/RFC3168, September 2001,
            <http://www.rfc-editor.org/info/rfc3168>.
 [RFC3261]  Rosenberg, J., Schulzrinne, H., Camarillo, G., Johnston,
            A., Peterson, J., Sparks, R., Handley, M., and E.
            Schooler, "SIP: Session Initiation Protocol", RFC 3261,
            DOI 10.17487/RFC3261, June 2002,
            <http://www.rfc-editor.org/info/rfc3261>.
 [RFC3303]  Srisuresh, P., Kuthan, J., Rosenberg, J., Molitor, A., and
            A. Rayhan, "Middlebox communication architecture and
            framework", RFC 3303, DOI 10.17487/RFC3303, August 2002,
            <http://www.rfc-editor.org/info/rfc3303>.
 [RFC3493]  Gilligan, R., Thomson, S., Bound, J., McCann, J., and W.
            Stevens, "Basic Socket Interface Extensions for IPv6",
            RFC 3493, DOI 10.17487/RFC3493, February 2003,
            <http://www.rfc-editor.org/info/rfc3493>.
 [RFC3550]  Schulzrinne, H., Casner, S., Frederick, R., and V.
            Jacobson, "RTP: A Transport Protocol for Real-Time
            Applications", STD 64, RFC 3550, DOI 10.17487/RFC3550,
            July 2003, <http://www.rfc-editor.org/info/rfc3550>.

Eggert, et al. Best Current Practice [Page 45] RFC 8085 UDP Usage Guidelines March 2017

 [RFC3551]  Schulzrinne, H. and S. Casner, "RTP Profile for Audio and
            Video Conferences with Minimal Control", STD 65, RFC 3551,
            DOI 10.17487/RFC3551, July 2003,
            <http://www.rfc-editor.org/info/rfc3551>.
 [RFC3738]  Luby, M. and V. Goyal, "Wave and Equation Based Rate
            Control (WEBRC) Building Block", RFC 3738,
            DOI 10.17487/RFC3738, April 2004,
            <http://www.rfc-editor.org/info/rfc3738>.
 [RFC3758]  Stewart, R., Ramalho, M., Xie, Q., Tuexen, M., and P.
            Conrad, "Stream Control Transmission Protocol (SCTP)
            Partial Reliability Extension", RFC 3758,
            DOI 10.17487/RFC3758, May 2004,
            <http://www.rfc-editor.org/info/rfc3758>.
 [RFC3819]  Karn, P., Ed., 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, DOI 10.17487/RFC3819, July 2004,
            <http://www.rfc-editor.org/info/rfc3819>.
 [RFC4301]  Kent, S. and K. Seo, "Security Architecture for the
            Internet Protocol", RFC 4301, DOI 10.17487/RFC4301,
            December 2005, <http://www.rfc-editor.org/info/rfc4301>.
 [RFC4302]  Kent, S., "IP Authentication Header", RFC 4302,
            DOI 10.17487/RFC4302, December 2005,
            <http://www.rfc-editor.org/info/rfc4302>.
 [RFC4303]  Kent, S., "IP Encapsulating Security Payload (ESP)",
            RFC 4303, DOI 10.17487/RFC4303, December 2005,
            <http://www.rfc-editor.org/info/rfc4303>.
 [RFC4340]  Kohler, E., Handley, M., and S. Floyd, "Datagram
            Congestion Control Protocol (DCCP)", RFC 4340,
            DOI 10.17487/RFC4340, March 2006,
            <http://www.rfc-editor.org/info/rfc4340>.
 [RFC4341]  Floyd, S. and E. Kohler, "Profile for Datagram Congestion
            Control Protocol (DCCP) Congestion Control ID 2: TCP-like
            Congestion Control", RFC 4341, DOI 10.17487/RFC4341, March
            2006, <http://www.rfc-editor.org/info/rfc4341>.

Eggert, et al. Best Current Practice [Page 46] RFC 8085 UDP Usage Guidelines March 2017

 [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,
            DOI 10.17487/RFC4342, March 2006,
            <http://www.rfc-editor.org/info/rfc4342>.
 [RFC4380]  Huitema, C., "Teredo: Tunneling IPv6 over UDP through
            Network Address Translations (NATs)", RFC 4380,
            DOI 10.17487/RFC4380, February 2006,
            <http://www.rfc-editor.org/info/rfc4380>.
 [RFC4607]  Holbrook, H. and B. Cain, "Source-Specific Multicast for
            IP", RFC 4607, DOI 10.17487/RFC4607, August 2006,
            <http://www.rfc-editor.org/info/rfc4607>.
 [RFC4654]  Widmer, J. and M. Handley, "TCP-Friendly Multicast
            Congestion Control (TFMCC): Protocol Specification",
            RFC 4654, DOI 10.17487/RFC4654, August 2006,
            <http://www.rfc-editor.org/info/rfc4654>.
 [RFC4880]  Callas, J., Donnerhacke, L., Finney, H., Shaw, D., and R.
            Thayer, "OpenPGP Message Format", RFC 4880,
            DOI 10.17487/RFC4880, November 2007,
            <http://www.rfc-editor.org/info/rfc4880>.
 [RFC4890]  Davies, E. and J. Mohacsi, "Recommendations for Filtering
            ICMPv6 Messages in Firewalls", RFC 4890,
            DOI 10.17487/RFC4890, May 2007,
            <http://www.rfc-editor.org/info/rfc4890>.
 [RFC4960]  Stewart, R., Ed., "Stream Control Transmission Protocol",
            RFC 4960, DOI 10.17487/RFC4960, September 2007,
            <http://www.rfc-editor.org/info/rfc4960>.
 [RFC4963]  Heffner, J., Mathis, M., and B. Chandler, "IPv4 Reassembly
            Errors at High Data Rates", RFC 4963,
            DOI 10.17487/RFC4963, July 2007,
            <http://www.rfc-editor.org/info/rfc4963>.
 [RFC4987]  Eddy, W., "TCP SYN Flooding Attacks and Common
            Mitigations", RFC 4987, DOI 10.17487/RFC4987, August 2007,
            <http://www.rfc-editor.org/info/rfc4987>.
 [RFC5082]  Gill, V., Heasley, J., Meyer, D., Savola, P., Ed., and C.
            Pignataro, "The Generalized TTL Security Mechanism
            (GTSM)", RFC 5082, DOI 10.17487/RFC5082, October 2007,
            <http://www.rfc-editor.org/info/rfc5082>.

Eggert, et al. Best Current Practice [Page 47] RFC 8085 UDP Usage Guidelines March 2017

 [RFC5245]  Rosenberg, J., "Interactive Connectivity Establishment
            (ICE): A Protocol for Network Address Translator (NAT)
            Traversal for Offer/Answer Protocols", RFC 5245,
            DOI 10.17487/RFC5245, April 2010,
            <http://www.rfc-editor.org/info/rfc5245>.
 [RFC5622]  Floyd, S. and E. Kohler, "Profile for Datagram Congestion
            Control Protocol (DCCP) Congestion ID 4: TCP-Friendly Rate
            Control for Small Packets (TFRC-SP)", RFC 5622,
            DOI 10.17487/RFC5622, August 2009,
            <http://www.rfc-editor.org/info/rfc5622>.
 [RFC5652]  Housley, R., "Cryptographic Message Syntax (CMS)", STD 70,
            RFC 5652, DOI 10.17487/RFC5652, September 2009,
            <http://www.rfc-editor.org/info/rfc5652>.
 [RFC5681]  Allman, M., Paxson, V., and E. Blanton, "TCP Congestion
            Control", RFC 5681, DOI 10.17487/RFC5681, September 2009,
            <http://www.rfc-editor.org/info/rfc5681>.
 [RFC5740]  Adamson, B., Bormann, C., Handley, M., and J. Macker,
            "NACK-Oriented Reliable Multicast (NORM) Transport
            Protocol", RFC 5740, DOI 10.17487/RFC5740, November 2009,
            <http://www.rfc-editor.org/info/rfc5740>.
 [RFC5751]  Ramsdell, B. and S. Turner, "Secure/Multipurpose Internet
            Mail Extensions (S/MIME) Version 3.2 Message
            Specification", RFC 5751, DOI 10.17487/RFC5751, January
            2010, <http://www.rfc-editor.org/info/rfc5751>.
 [RFC5775]  Luby, M., Watson, M., and L. Vicisano, "Asynchronous
            Layered Coding (ALC) Protocol Instantiation", RFC 5775,
            DOI 10.17487/RFC5775, April 2010,
            <http://www.rfc-editor.org/info/rfc5775>.
 [RFC5971]  Schulzrinne, H. and R. Hancock, "GIST: General Internet
            Signalling Transport", RFC 5971, DOI 10.17487/RFC5971,
            October 2010, <http://www.rfc-editor.org/info/rfc5971>.
 [RFC5973]  Stiemerling, M., Tschofenig, H., Aoun, C., and E. Davies,
            "NAT/Firewall NSIS Signaling Layer Protocol (NSLP)",
            RFC 5973, DOI 10.17487/RFC5973, October 2010,
            <http://www.rfc-editor.org/info/rfc5973>.
 [RFC6056]  Larsen, M. and F. Gont, "Recommendations for Transport-
            Protocol Port Randomization", BCP 156, RFC 6056,
            DOI 10.17487/RFC6056, January 2011,
            <http://www.rfc-editor.org/info/rfc6056>.

Eggert, et al. Best Current Practice [Page 48] RFC 8085 UDP Usage Guidelines March 2017

 [RFC6092]  Woodyatt, J., Ed., "Recommended Simple Security
            Capabilities in Customer Premises Equipment (CPE) for
            Providing Residential IPv6 Internet Service", RFC 6092,
            DOI 10.17487/RFC6092, January 2011,
            <http://www.rfc-editor.org/info/rfc6092>.
 [RFC6335]  Cotton, M., Eggert, L., Touch, J., Westerlund, M., and S.
            Cheshire, "Internet Assigned Numbers Authority (IANA)
            Procedures for the Management of the Service Name and
            Transport Protocol Port Number Registry", BCP 165,
            RFC 6335, DOI 10.17487/RFC6335, August 2011,
            <http://www.rfc-editor.org/info/rfc6335>.
 [RFC6347]  Rescorla, E. and N. Modadugu, "Datagram Transport Layer
            Security Version 1.2", RFC 6347, DOI 10.17487/RFC6347,
            January 2012, <http://www.rfc-editor.org/info/rfc6347>.
 [RFC6396]  Blunk, L., Karir, M., and C. Labovitz, "Multi-Threaded
            Routing Toolkit (MRT) Routing Information Export Format",
            RFC 6396, DOI 10.17487/RFC6396, October 2011,
            <http://www.rfc-editor.org/info/rfc6396>.
 [RFC6437]  Amante, S., Carpenter, B., Jiang, S., and J. Rajahalme,
            "IPv6 Flow Label Specification", RFC 6437,
            DOI 10.17487/RFC6437, November 2011,
            <http://www.rfc-editor.org/info/rfc6437>.
 [RFC6438]  Carpenter, B. and S. Amante, "Using the IPv6 Flow Label
            for Equal Cost Multipath Routing and Link Aggregation in
            Tunnels", RFC 6438, DOI 10.17487/RFC6438, November 2011,
            <http://www.rfc-editor.org/info/rfc6438>.
 [RFC6513]  Rosen, E., Ed. and R. Aggarwal, Ed., "Multicast in MPLS/
            BGP IP VPNs", RFC 6513, DOI 10.17487/RFC6513, February
            2012, <http://www.rfc-editor.org/info/rfc6513>.
 [RFC6679]  Westerlund, M., Johansson, I., Perkins, C., O'Hanlon, P.,
            and K. Carlberg, "Explicit Congestion Notification (ECN)
            for RTP over UDP", RFC 6679, DOI 10.17487/RFC6679, August
            2012, <http://www.rfc-editor.org/info/rfc6679>.
 [RFC6726]  Paila, T., Walsh, R., Luby, M., Roca, V., and R. Lehtonen,
            "FLUTE - File Delivery over Unidirectional Transport",
            RFC 6726, DOI 10.17487/RFC6726, November 2012,
            <http://www.rfc-editor.org/info/rfc6726>.

Eggert, et al. Best Current Practice [Page 49] RFC 8085 UDP Usage Guidelines March 2017

 [RFC6773]  Phelan, T., Fairhurst, G., and C. Perkins, "DCCP-UDP: A
            Datagram Congestion Control Protocol UDP Encapsulation for
            NAT Traversal", RFC 6773, DOI 10.17487/RFC6773, November
            2012, <http://www.rfc-editor.org/info/rfc6773>.
 [RFC6807]  Farinacci, D., Shepherd, G., Venaas, S., and Y. Cai,
            "Population Count Extensions to Protocol Independent
            Multicast (PIM)", RFC 6807, DOI 10.17487/RFC6807, December
            2012, <http://www.rfc-editor.org/info/rfc6807>.
 [RFC6887]  Wing, D., Ed., Cheshire, S., Boucadair, M., Penno, R., and
            P. Selkirk, "Port Control Protocol (PCP)", RFC 6887,
            DOI 10.17487/RFC6887, April 2013,
            <http://www.rfc-editor.org/info/rfc6887>.
 [RFC6935]  Eubanks, M., Chimento, P., and M. Westerlund, "IPv6 and
            UDP Checksums for Tunneled Packets", RFC 6935,
            DOI 10.17487/RFC6935, April 2013,
            <http://www.rfc-editor.org/info/rfc6935>.
 [RFC6936]  Fairhurst, G. and M. Westerlund, "Applicability Statement
            for the Use of IPv6 UDP Datagrams with Zero Checksums",
            RFC 6936, DOI 10.17487/RFC6936, April 2013,
            <http://www.rfc-editor.org/info/rfc6936>.
 [RFC6951]  Tuexen, M. and R. Stewart, "UDP Encapsulation of Stream
            Control Transmission Protocol (SCTP) Packets for End-Host
            to End-Host Communication", RFC 6951,
            DOI 10.17487/RFC6951, May 2013,
            <http://www.rfc-editor.org/info/rfc6951>.
 [RFC7143]  Chadalapaka, M., Satran, J., Meth, K., and D. Black,
            "Internet Small Computer System Interface (iSCSI) Protocol
            (Consolidated)", RFC 7143, DOI 10.17487/RFC7143, April
            2014, <http://www.rfc-editor.org/info/rfc7143>.
 [RFC7201]  Westerlund, M. and C. Perkins, "Options for Securing RTP
            Sessions", RFC 7201, DOI 10.17487/RFC7201, April 2014,
            <http://www.rfc-editor.org/info/rfc7201>.
 [RFC7296]  Kaufman, C., Hoffman, P., Nir, Y., Eronen, P., and T.
            Kivinen, "Internet Key Exchange Protocol Version 2
            (IKEv2)", STD 79, RFC 7296, DOI 10.17487/RFC7296, October
            2014, <http://www.rfc-editor.org/info/rfc7296>.
 [RFC7450]  Bumgardner, G., "Automatic Multicast Tunneling", RFC 7450,
            DOI 10.17487/RFC7450, February 2015,
            <http://www.rfc-editor.org/info/rfc7450>.

Eggert, et al. Best Current Practice [Page 50] RFC 8085 UDP Usage Guidelines March 2017

 [RFC7510]  Xu, X., Sheth, N., Yong, L., Callon, R., and D. Black,
            "Encapsulating MPLS in UDP", RFC 7510,
            DOI 10.17487/RFC7510, April 2015,
            <http://www.rfc-editor.org/info/rfc7510>.
 [RFC7525]  Sheffer, Y., Holz, R., and P. Saint-Andre,
            "Recommendations for Secure Use of Transport Layer
            Security (TLS) and Datagram Transport Layer Security
            (DTLS)", BCP 195, RFC 7525, DOI 10.17487/RFC7525, May
            2015, <http://www.rfc-editor.org/info/rfc7525>.
 [RFC7560]  Kuehlewind, M., Ed., Scheffenegger, R., and B. Briscoe,
            "Problem Statement and Requirements for Increased Accuracy
            in Explicit Congestion Notification (ECN) Feedback",
            RFC 7560, DOI 10.17487/RFC7560, August 2015,
            <http://www.rfc-editor.org/info/rfc7560>.
 [RFC7567]  Baker, F., Ed. and G. Fairhurst, Ed., "IETF
            Recommendations Regarding Active Queue Management",
            BCP 197, RFC 7567, DOI 10.17487/RFC7567, July 2015,
            <http://www.rfc-editor.org/info/rfc7567>.
 [RFC7605]  Touch, J., "Recommendations on Using Assigned Transport
            Port Numbers", BCP 165, RFC 7605, DOI 10.17487/RFC7605,
            August 2015, <http://www.rfc-editor.org/info/rfc7605>.
 [RFC7657]  Black, D., Ed. and P. Jones, "Differentiated Services
            (Diffserv) and Real-Time Communication", RFC 7657,
            DOI 10.17487/RFC7657, November 2015,
            <http://www.rfc-editor.org/info/rfc7657>.
 [RFC7675]  Perumal, M., Wing, D., Ravindranath, R., Reddy, T., and M.
            Thomson, "Session Traversal Utilities for NAT (STUN) Usage
            for Consent Freshness", RFC 7675, DOI 10.17487/RFC7675,
            October 2015, <http://www.rfc-editor.org/info/rfc7675>.
 [RFC8083]  Perkins, C. and V. Singh, "Multimedia Congestion Control:
            Circuit Breakers for Unicast RTP Sessions", RFC 8083,
            DOI 10.17487/RFC8083, March 2017,
            <http://www.rfc-editor.org/info/rfc8083>.
 [RFC8086]  Yong, L., Ed., Crabbe, E., Xu, X., and T. Herbert, "GRE-
            in-UDP Encapsulation", RFC 8086, DOI 10.17487/RFC8086,
            March 2017, <http://www.rfc-editor.org/info/rfc8086>.

Eggert, et al. Best Current Practice [Page 51] RFC 8085 UDP Usage Guidelines March 2017

 [RFC8087]  Fairhurst, G. and M. Welzl, "The Benefits of Using
            Explicit Congestion Notification (ECN)", RFC 8087,
            DOI 10.17487/RFC8087, March 2017,
            <http://www.rfc-editor.org/info/rfc8087>.
 [STEVENS]  Stevens, W., Fenner, B., and A. Rudoff, "UNIX Network
            Programming, The sockets Networking API", Addison-Wesley,
            2004.
 [UPnP]     UPnP Forum, , "Internet Gateway Device (IGD) Standardized
            Device Control Protocol V 1.0", November 2001.

Eggert, et al. Best Current Practice [Page 52] RFC 8085 UDP Usage Guidelines March 2017

Appendix A. Case Study of the Use of IPv6 UDP Zero-Checksum Mode

 This appendix provides a brief review of MPLS-in-UDP as an example of
 a UDP Tunnel Encapsulation that defines a UDP encapsulation.  The
 purpose of the appendix is to provide a concrete example of which
 mechanisms were required in order to safely use UDP zero-checksum
 mode for MPLS-in-UDP tunnels over IPv6.  By default, UDP requires a
 checksum for use with IPv6.  An option has been specified that
 permits a zero IPv6 UDP checksum when used in specific environments,
 specified in [RFC7510], and defines a set of operational constraints
 for use of this mode.  These are summarized below:
 A UDP tunnel or encapsulation using a zero-checksum mode with IPv6
 must only be deployed within a single network (with a single network
 operator) or networks of an adjacent set of cooperating network
 operators where traffic is managed to avoid congestion, rather than
 over the Internet where congestion control is required.  MPLS-in-UDP
 has been specified for networks under single administrative control
 (such as within a single operator's network) where it is known
 (perhaps through knowledge of equipment types and lower-layer checks)
 that packet corruption is exceptionally unlikely and where the
 operator is willing to take the risk of undetected packet corruption.
 The tunnel encapsulator SHOULD use different IPv6 addresses for each
 UDP tunnel that uses the UDP zero-checksum mode, regardless of the
 decapsulator, to strengthen the decapsulator's check of the IPv6
 source address (i.e., the same IPv6 source address SHOULD NOT be used
 with more than one IPv6 destination address, independent of whether
 that destination address is a unicast or multicast address).  Use of
 MPLS-in-UDP may be extended to networks within a set of closely
 cooperating network administrations (such as network operators who
 have agreed to work together to jointly provide specific services)
 [RFC7510].
 The requirement for MPLS-in-UDP endpoints to check the source IPv6
 address in addition to the destination IPv6 address, plus the strong
 recommendation against reuse of source IPv6 addresses among MPLS-in-
 UDP tunnels collectively provide some mitigation for the absence of
 UDP checksum coverage of the IPv6 header.  In addition, the MPLS data
 plane only forwards packets with valid labels (i.e., labels that have
 been distributed by the tunnel egress Label Switched Router, LSR),
 providing some additional opportunity to detect MPLS-in-UDP packet
 misdelivery when the misdelivered packet contains a label that is not
 valid for forwarding at the receiving LSR.  The expected result for
 IPv6 UDP zero-checksum mode for MPLS-in-UDP is that corruption of the
 destination IPv6 address will usually cause packet discard, as
 offsetting corruptions to the source IPv6 and/or MPLS top label are
 unlikely.

Eggert, et al. Best Current Practice [Page 53] RFC 8085 UDP Usage Guidelines March 2017

 Additional assurance is provided by the restrictions in the above
 exceptions that limit usage of IPv6 UDP zero-checksum mode to well-
 managed networks for which MPLS packet corruption has not been a
 problem in practice.  Hence, MPLS-in-UDP is suitable for transmission
 over lower layers in well-managed networks that are allowed by the
 exceptions stated above and the rate of corruption of the inner IP
 packet on such networks is not expected to increase by comparison to
 MPLS traffic that is not encapsulated in UDP.  For these reasons,
 MPLS-in-UDP does not provide an additional integrity check when UDP
 zero-checksum mode is used with IPv6, and this design is in
 accordance with requirements 2, 3, and 5 specified in Section 5 of
 [RFC6936].
 The MPLS-in-UDP encapsulation does not provide a mechanism to safely
 fall back to using a checksum when a path change occurs that
 redirects a tunnel over a path that includes a middlebox that
 discards IPv6 datagrams with a zero UDP checksum.  In this case, the
 MPLS-in-UDP tunnel will be black-holed by that middlebox.
 Recommended changes to allow firewalls, NATs and other middleboxes to
 support use of an IPv6 zero UDP checksum are described in Section 5
 of [RFC6936].  MPLS does not accumulate incorrect state as a
 consequence of label-stack corruption.  A corrupt MPLS label results
 in either packet discard or forwarding (and forgetting) of the packet
 without accumulation of MPLS protocol state.  Active monitoring of
 MPLS-in-UDP traffic for errors is REQUIRED because the occurrence of
 errors will result in some accumulation of error information outside
 the MPLS protocol for operational and management purposes.  This
 design is in accordance with requirement 4 specified in Section 5 of
 [RFC6936].  In addition, IPv6 traffic with a zero UDP checksum MUST
 be actively monitored for errors by the network operator.
 Operators SHOULD also deploy packet filters to prevent IPv6 packets
 with a zero UDP checksum from escaping from the network due to
 misconfiguration or packet errors.  In addition, IPv6 traffic with a
 zero UDP checksum MUST be actively monitored for errors by the
 network operator.

Eggert, et al. Best Current Practice [Page 54] RFC 8085 UDP Usage Guidelines March 2017

Acknowledgments

 The middlebox traversal guidelines in Section 3.5 incorporate ideas
 from Section 5 of [BEHAVE-APP] by Bryan Ford, Pyda Srisuresh, and Dan
 Kegel.  The protocol timer guidelines in Section 3.1.1 were largely
 contributed by Mark Allman.
 G.  Fairhurst received funding from the European Union's Horizon 2020
 research and innovation program 2014-2018 under grant agreement No.
 644334 (NEAT).  Lars Eggert has received funding from the European
 Union's Horizon 2020 research and innovation program 2014-2018 under
 grant agreement No. 644866 (SSICLOPS).  This document reflects only
 the authors' views and the European Commission is not responsible for
 any use that may be made of the information it contains.

Authors' Addresses

 Lars Eggert
 NetApp
 Sonnenallee 1
 Kirchheim  85551
 Germany
 Phone: +49 151 120 55791
 Email: lars@netapp.com
 URI:   https://eggert.org/
 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/
 Greg Shepherd
 Cisco Systems
 Tasman Drive
 San Jose
 United States of America
 Email: gjshep@gmail.com

Eggert, et al. Best Current Practice [Page 55]

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