UDP Usage Guidelines
draft-ietf-tsvwg-rfc5405bis-02

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Last updated 2015-04-08
Replaces draft-eggert-tsvwg-rfc5405bis, draft-tsvwg-rfc5405bis
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Transport Area Working Group                                   L. Eggert
Internet-Draft                                                    NetApp
Obsoletes: 5405 (if approved)                               G. Fairhurst
Intended status: Best Current Practice            University of Aberdeen
Expires: October 10, 2015                                    G. Shepherd
                                                           Cisco Systems
                                                           April 8, 2015

                          UDP Usage Guidelines
                     draft-ietf-tsvwg-rfc5405bis-02

Abstract

   The User Datagram Protocol (UDP) provides a minimal message-passing
   transport that has no inherent congestion control mechanisms.
   Because congestion control is critical to the stable operation of the
   Internet, applications and 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.

   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 ECN, DSCPs,
   and ports.

   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.

   If published as an RFC, this document will obsolete RFC5405.

Status of This Memo

   This Internet-Draft is submitted in full conformance with the
   provisions of BCP 78 and BCP 79.

   Internet-Drafts are working documents of the Internet Engineering
   Task Force (IETF).  Note that other groups may also distribute
   working documents as Internet-Drafts.  The list of current Internet-
   Drafts is at http://datatracker.ietf.org/drafts/current/.

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   Internet-Drafts are draft documents valid for a maximum of six months
   and may be updated, replaced, or obsoleted by other documents at any
   time.  It is inappropriate to use Internet-Drafts as reference
   material or to cite them other than as "work in progress."

   This Internet-Draft will expire on October 10, 2015.

Copyright Notice

   Copyright (c) 2015 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.

Table of Contents

   1.  Introduction  . . . . . . . . . . . . . . . . . . . . . . . .   3
   2.  Terminology . . . . . . . . . . . . . . . . . . . . . . . . .   4
   3.  UDP Usage Guidelines  . . . . . . . . . . . . . . . . . . . .   4
     3.1.  Congestion Control Guidelines . . . . . . . . . . . . . .   6
     3.2.  Message Size Guidelines . . . . . . . . . . . . . . . . .  14
     3.3.  Reliability Guidelines  . . . . . . . . . . . . . . . . .  15
     3.4.  Checksum Guidelines . . . . . . . . . . . . . . . . . . .  16
     3.5.  Middlebox Traversal Guidelines  . . . . . . . . . . . . .  19
   4.  Multicast UDP Usage Guidelines  . . . . . . . . . . . . . . .  21
     4.1.  Multicast Congestion Control Guidelines . . . . . . . . .  22
     4.2.  Message Size Guidelines for Multicast . . . . . . . . . .  24
   5.  Programming Guidelines  . . . . . . . . . . . . . . . . . . .  24
     5.1.  Using UDP Ports . . . . . . . . . . . . . . . . . . . . .  26
     5.2.  ICMP Guidelines . . . . . . . . . . . . . . . . . . . . .  28
   6.  Security Considerations . . . . . . . . . . . . . . . . . . .  28
   7.  Summary . . . . . . . . . . . . . . . . . . . . . . . . . . .  29
   8.  IANA Considerations . . . . . . . . . . . . . . . . . . . . .  31
   9.  Acknowledgments . . . . . . . . . . . . . . . . . . . . . . .  31
   10. References  . . . . . . . . . . . . . . . . . . . . . . . . .  31
     10.1.  Normative References . . . . . . . . . . . . . . . . . .  31
     10.2.  Informative References . . . . . . . . . . . . . . . . .  32
   Appendix A.  Case Study of the Use of IPv6 UDP Zero-Checksum Mode  39
   Appendix B.  Revision Notes . . . . . . . . . . . . . . . . . . .  40
   Authors' Addresses  . . . . . . . . . . . . . . . . . . . . . . .  42

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1.  Introduction

   The User Datagram Protocol (UDP) [RFC0768] provides a minimal,
   unreliable, best-effort, message-passing transport to applications
   and other protocols (such as tunnels) that desire to operate over UDP
   (both simply called "applications" in the remainder of this
   document).  Compared to other transport protocols, UDP and its UDP-
   Lite variant [RFC3828] are unique in that they do not establish end-
   to-end connections between communicating end systems.  UDP
   communication consequently does not incur connection establishment
   and teardown overheads, and there is minimal associated end system
   state.  Because of these characteristics, UDP can offer a very
   efficient communication transport to some applications.

   A second unique characteristic of UDP is that it provides no inherent
   congestion control mechanisms.  On many platforms, applications can
   send UDP datagrams at the line rate of the link interface, which is
   often much greater than the available path capacity, and doing so
   contributes to congestion along the path.  [RFC2914] describes the
   best current practice for congestion control in the Internet.  It
   identifies two major reasons why congestion control mechanisms are
   critical for the stable operation of the Internet:

   1.  The prevention of congestion collapse, i.e., a state where an
       increase in network load results in a decrease in useful work
       done by the network.

   2.  The establishment of a degree of fairness, i.e., allowing
       multiple flows to share the capacity of a path reasonably
       equitably.

   Because UDP itself provides no congestion control mechanisms, it is
   up to the applications that use UDP for Internet communication to
   employ suitable mechanisms to prevent congestion collapse and
   establish a degree of fairness.  [RFC2309] discusses the dangers of
   congestion-unresponsive flows and states that "all UDP-based
   streaming applications should incorporate effective congestion
   avoidance mechanisms."  This is an important requirement, even for
   applications that do not use UDP for streaming.  In addition,
   congestion-controlled transmission is of benefit to an application
   itself, because it can reduce self-induced packet loss, minimize
   retransmissions, and hence reduce delays.  Congestion control is
   essential even at relatively slow transmission rates.  For example,
   an application that generates five 1500-byte UDP datagrams in one
   second can already exceed the capacity of a 56 Kb/s path.  For
   applications that can operate at higher, potentially unbounded data
   rates, congestion control becomes vital to prevent congestion

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   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, and middlebox traversal.

   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
   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 applications
   that use UDP, the spirit of some of its guidelines also applies to
   other message-passing applications and protocols (specifically on the
   topics of congestion control, message sizes, and reliability).
   Examples include signaling or control applications that choose to run
   directly over IP by registering their own IP protocol number with
   IANA.  This document may provide useful background reading to the
   designers of such applications and protocols.

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.

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   Furthermore, the same Internet path can have very different
   conditions over time.  Consequently, applications that may be used on
   the Internet MUST NOT make assumptions about specific path
   characteristics.  They MUST instead use mechanisms that let them
   operate safely under very different path conditions.  Typically, this
   requires conservatively probing the current conditions of the
   Internet path they communicate over to establish a transmission
   behavior that it can sustain and that is reasonably fair to other
   traffic sharing the path.

   These mechanisms are difficult to implement correctly.  For most
   applications, the use of one of the existing IETF transport protocols
   is the simplest method of acquiring the required mechanisms.
   Consequently, the RECOMMENDED alternative to the UDP usage described
   in the remainder of this section is the use of an IETF transport
   protocol such as TCP [RFC0793], Stream Control Transmission Protocol
   (SCTP) [RFC4960], and SCTP Partial Reliability Extension (SCTP-PR)
   [RFC3758], or Datagram Congestion Control Protocol (DCCP) [RFC4340]
   with its different congestion control types
   [RFC4341][RFC4342][RFC5622].

   If used correctly, these more fully-featured transport protocols are
   not as "heavyweight" as often claimed.  For example, the TCP
   algorithms have been continuously improved over decades, and have
   reached a level of efficiency and correctness that custom
   application-layer mechanisms will struggle to easily duplicate.  In
   addition, many TCP implementations allow connections to be tuned by
   an application to its purposes.  For example, TCP's "Nagle" algorithm
   [RFC0896] can be disabled, improving communication latency at the
   expense of more frequent -- but still congestion-controlled -- packet
   transmissions.  Another example is the TCP SYN cookie mechanism
   [RFC4987], which is available on many platforms.  TCP with SYN
   cookies does not require a server to maintain per-connection state
   until the connection is established.  TCP also requires the end that
   closes a connection to maintain the TIME-WAIT state that prevents
   delayed segments from one connection instance from interfering with a
   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.

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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.

   Several approaches to perform congestion control are discussed in the
   remainder of this section.  The 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 3.1.1 discusses congestion control options for
   applications that perform bulk transfers over UDP.  Such applications
   can employ schemes that sample the path over several subsequent RTTs
   during which data is exchanged, in order to determine a sending rate
   that the path at its current load can support.  Other applications
   only exchange a few UDP datagrams with a destination.  Section 3.1.2
   discusses congestion control options for such "low data-volume"
   applications.  Because they typically do not transmit enough data to
   iteratively sample the path to determine a safe sending rate, they
   need to employ different kinds of congestion control mechanisms.
   Section 3.1.7 discusses congestion control considerations when UDP is
   used as a tunneling protocol.  Section 4 provides additional
   recommendations for broadcast and multicast usage.

   UDP applications may gain the benefits of using Explicit Congestion
   Notification (ECN), providing that the application programming
   interface can support ECN and the congestion control can
   appropriately react to ECN-marked packets
   [I-D.ietf-aqm-ecn-benefits].  [RFC6679] provides guidance on how to
   use ECN for UDP-based applications using the Real-Time Protocol
   (RTP).

   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

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   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.  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 TCP-Friendly Rate Control (TFRC) [RFC5348],
   window-based TCP-like congestion control, or otherwise ensure that
   the application complies with the congestion control principles.

   TFRC has been designed to provide both congestion control and
   fairness in a way that is compatible with the IETF's other transport
   protocols.  If an application implements TFRC, it need not follow the
   remaining guidelines in Section 3.1.1, because TFRC already addresses
   them, but SHOULD still follow the remaining guidelines in the
   subsequent subsections of Section 3.

   Bulk transfer applications that choose not to implement TFRC or TCP-
   like windowing SHOULD implement a congestion control scheme that
   results in bandwidth use that competes fairly with TCP within an
   order of magnitude.

   Section 2 of [RFC3551] suggests that applications SHOULD monitor the
   packet loss rate to ensure that it is within acceptable parameters.
   Packet loss is considered acceptable if a TCP flow across the same
   network path under the same network conditions would achieve an
   average throughput, measured on a reasonable timescale, that is not
   less than that of the UDP flow.  The comparison to TCP cannot be
   specified exactly, but is intended as an "order-of-magnitude"
   comparison in timescale and throughput.

   Finally, some bulk transfer applications may choose not to implement
   any congestion control mechanism and instead rely on transmitting
   across reserved path capacity.  This might be an acceptable choice
   for a subset of restricted networking environments, but is by no
   means a safe practice for operation 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.

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3.1.2.  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 3.1.1, the network sees little benefit,
   because those mechanisms perform congestion control in a way that is
   only effective for longer transmissions.

   Applications that at any time exchange only a few UDP datagrams with
   a destination SHOULD still control their transmission behavior by not
   sending on average more than one UDP datagram per round-trip time
   (RTT) to a destination.  Similar to the recommendation in [RFC1536],
   an application SHOULD maintain an estimate of the RTT for any
   destination with which it communicates.  Applications SHOULD
   implement the algorithm specified in [RFC6298] to compute a smoothed
   RTT (SRTT) estimate.  They SHOULD also detect packet loss and
   exponentially back their retransmission timer off when a loss event
   occurs.  When implementing this scheme, applications need to choose a
   sensible initial value for the RTT.  This value SHOULD generally be
   as conservative as possible for the given application.  TCP specifies
   an initial value of 3 seconds [RFC6298], which is also RECOMMENDED as
   an initial value for UDP applications.  SIP [RFC3261] and GIST
   [RFC5971] use an initial value of 500 ms, and initial timeouts that
   are shorter than this are likely problematic in many cases.  It is
   also important to note that the initial timeout is not the maximum
   possible timeout -- the RECOMMENDED algorithm in [RFC6298] yields
   timeout values after a series of losses that are much longer than the
   initial value.

   Some applications cannot maintain a reliable RTT estimate for a
   destination.  The first case is that of applications that exchange
   too few UDP datagrams with a peer to establish a statistically
   accurate RTT estimate.  Such applications MAY use a predetermined
   transmission interval that is exponentially backed-off when packets
   are lost.  TCP uses an initial value of 3 seconds [RFC6298], which is
   also RECOMMENDED as an initial value for UDP applications.  SIP
   [RFC3261] and 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.

   A second class of applications cannot maintain an RTT estimate for a
   destination, because the destination does not send return traffic.
   Such applications SHOULD NOT send more than one UDP datagram every 3
   seconds, and SHOULD use an even less aggressive rate when possible.
   The 3-second interval was chosen based on TCP's retransmission
   timeout when the RTT is unknown [RFC6298], and shorter values are
   likely problematic in many cases.  Note that the sending rate in this

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   case must be more conservative than in the two previous cases,
   because the lack of return traffic prevents the detection of packet
   loss, i.e., congestion, and the application therefore cannot perform
   exponential back-off to reduce load.

   Applications that communicate bidirectionally SHOULD employ
   congestion control for both directions of the communication.  For
   example, for a client-server, request-response-style application,
   clients SHOULD congestion-control their request transmission to a
   server, and the server SHOULD congestion-control its responses to the
   clients.  Congestion in the forward and reverse direction is
   uncorrelated, and an application SHOULD either independently detect
   and respond to congestion along both directions, or limit new and
   retransmitted requests based on acknowledged responses across the
   entire round-trip path.

3.1.3.  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.

   Even low data-volume UDP flows may benefit from rate control, 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.

3.1.4.  Differentiated Services Model

   An application using UDP can use the differentiated services QoS
   framework.  To enable differentiated services processing, a UDP
   sender sets the 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.  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 the 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

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   source and destination.  This has implications on the design of
   applications that use DSCPs.  Specifically, applications SHOULD be
   designed to not 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.

   [I-D.ietf-dart-dscp-rtp] 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.

3.1.5.  QoS, Preprovisioned or Reserved Capacity

   An application using UDP can use the integrated services QoS
   framework.  These are usually 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 preprovisioned capacity or capacity reserved using
   dynamic provisioning, e.g., through the Resource Reservation Protocol
   (RSVP).  Multicast applications are also used with preprovisioned
   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.

   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.  For
   this reason, and to protect other applications sharing the same path,
   applications SHOULD deploy an appropriate circuit breaker, as
   described in Section 3.1.6.  Applications that support an
   uncontrolled or unadaptive transmission behavior SHOULD NOT do so by

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   default and SHOULD instead require users to explicitly enable this
   mode of operation.

   Applications used in networks within a controlled environment may be
   able to exploit network management functions to detect whether they
   are causing congestion, and react accordingly.

3.1.6.  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 [I-D.ietf-tsvwg-circuit-breaker].  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.

   [I-D.ietf-tsvwg-circuit-breaker] provides guidance on the use of
   circuit breakers and examples of usage.  The use of a circuit breaker
   in RTP is specified in [I-D.ietf-avtcore-rtp-circuit-breakers].

   Applications used in the general Internet SHOULD implement a
   transport circuit breaker if they do not implement congestion control
   or operate a low volume data service.  All applications MAY implement
   a transport circuit breaker [I-D.ietf-tsvwg-circuit-breaker] 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.

3.1.7.  UDP Tunnels

   One increasingly popular use of UDP is as a tunneling protocol, where
   a tunnel endpoint encapsulates the packets of another protocol inside
   UDP datagrams and transmits them to another tunnel endpoint, which
   decapsulates the UDP datagrams and forwards the original packets
   contained in the payload.  Tunnels establish virtual links that
   appear to directly connect locations that are distant in the physical
   Internet topology and can be used to create virtual (private)
   networks.  Using UDP as a tunneling protocol is attractive when the

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   payload protocol is not supported by middleboxes that may exist along
   the path, because many middleboxes support transmission using UDP.

   Well-implemented tunnels are generally invisible to the endpoints
   that happen to transmit over a path that includes tunneled links.  On
   the other hand, to the routers along the path of a UDP tunnel, i.e.,
   the routers between the two tunnel endpoints, the traffic that a UDP
   tunnel generates is a regular UDP flow, and the encapsulator and
   decapsulator appear as regular UDP-sending and -receiving
   applications.  Because other flows can share the path with one or
   more UDP tunnels, congestion control needs to be considered.

   Two factors determine whether a UDP tunnel needs to employ specific
   congestion control mechanisms -- first, whether the payload traffic
   is IP-based; second, whether the tunneling scheme generates UDP
   traffic at a volume that corresponds to the volume of payload traffic
   carried within the tunnel.

   IP-based traffic is generally assumed to be congestion-controlled,
   i.e., it is assumed that the transport protocols generating IP-based
   traffic at the sender already employ mechanisms that are sufficient
   to address congestion on the path.  Consequently, a tunnel carrying
   IP-based traffic should already interact appropriately with other
   traffic sharing the path, and specific congestion control mechanisms
   for the tunnel are not necessary.

   However, if the IP traffic in the tunnel is known to not be
   congestion-controlled, additional measures are RECOMMENDED in order
   to limit the impact of the tunneled traffic on other traffic sharing
   the path.

   The following guidelines define these possible cases in more detail:

   1.  A tunnel generates UDP traffic at a volume that corresponds to
       the volume of payload traffic, and the payload traffic is IP-
       based and congestion-controlled.

       This is arguably the most common case for Internet tunnels.  In
       this case, the UDP tunnel SHOULD NOT employ its own congestion
       control mechanism, because congestion losses of tunneled traffic
       will already trigger an appropriate congestion response at the
       original senders of the tunneled traffic.  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

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       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 3.1.1 apply.

   3.  A tunnel generates UDP traffic at a volume that does not
       correspond to the volume of payload traffic, independent of
       whether the payload traffic is IP-based or congestion-controlled.

       Examples of this class include UDP tunnels that send at a
       constant rate, increase their transmission rates under loss, for
       example, due to increasing redundancy when Forward Error
       Correction is used, or are otherwise 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.

   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 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).

   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,
   e.g., handling fragmentation, treatment of DSCP values, generating
   and responding to ICMP messages, etc.  In particular, tunnels that
   carry or encapsulate using ECN code points MUST follow the
   requirements specified in [RFC6040].  At the same time, the tunneled

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   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, and instead choose to drop all
   fragments.  Finally, [RFC4963] documents other issues specific to
   IPv4 fragmentation.

   Due to these issues, an application SHOULD NOT send UDP datagrams
   that result in IP packets that exceed the MTU of the path to the
   destination.  Consequently, an application SHOULD either use the path
   MTU information provided by the IP layer or implement path MTU
   discovery itself [RFC1191][RFC1981][RFC4821] to determine whether the
   path to a destination will support its desired message size without
   fragmentation.

   Applications that do not follow this recommendation to do PMTU
   discovery SHOULD still avoid sending UDP datagrams that would result
   in IP packets that exceed the path MTU.  Because the actual path MTU
   is unknown, such applications SHOULD fall back to sending messages
   that are shorter than the default effective MTU for sending (EMTU_S
   in [RFC1122]).  For IPv4, EMTU_S is the smaller of 576 bytes and the
   first-hop MTU [RFC1122].  For IPv6, EMTU_S is 1280 bytes [RFC2460].
   The effective PMTU for a directly connected destination (with no
   routers on the path) is the configured interface MTU, which could be
   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

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   header (8 bytes) from the PMTU size.  This size, known as the 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, 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.

   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.  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 updates not only the PMTU algorithm, it also impacts
   loss recovery, congestion control, etc.  These updated mechanisms can
   be implemented within a connection-oriented transport (e.g., TCP,
   SCTP, DCCP), but are not a part of UDP.  PLPMTUD therefore places
   additional design requirements on a UDP application that wishes to
   use this method.

3.3.  Reliability Guidelines

   Application designers are generally aware that UDP does not provide
   any reliability, e.g., it does not retransmit any lost packets.
   Often, this is a main reason to consider UDP as a transport.
   Applications that do require reliable message delivery MUST implement
   an appropriate mechanism themselves.

   UDP also does not protect against datagram duplication, i.e., an
   application may receive multiple copies of the same UDP datagram,
   with some duplicates arriving potentially much later than the first.
   Application designers SHOULD verify that their application handles
   such datagram duplication gracefully, and 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,

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   where UDP datagrams arrive at the receiver in an order different from
   the transmission order.

   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.
   [RFC0793] 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.

   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) included with the data to verify the integrity of an
   entire object/file sent over the UDP service.

   The UDP checksum provides a statistical guarantee that the payload
   was not corrupted in transit.  It also allows the receiver to verify
   that it was the intended destination of the packet, because it covers
   the IP addresses, port numbers, and protocol number, and it verifies
   that the packet is not truncated or padded, because it covers the
   size field.  It therefore protects an application against receiving
   corrupted payload data in place of, or in addition to, the data that
   was sent.  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.  For IPv4, [RFC0768]
   permits an option to disable their use.

   When UDP is used over IPv6, the UDP checksum is relied upon to
   protect both the IPv6 and UDP headers from corruption, and MUST be
   used as specified in [RFC2460], unless the requirements in [RFC6935]
   and [RFC6936] for use of UDP zero-checksum mode with a tunnel
   protocol are satisfied.  The application MUST implement mechanisms

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   and/or usage restrictions for this mode.  These additional design
   requirements for using a zero IPv6 UDP checksum [RFC6936] are not
   present for IPv4, since the IPv4 header validates information that is
   not protected in an IPv6 packet.  Key requirements apply to
   implementation and use of UDP zero-checksum mode for IPv6:

   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 MUST comply with all implementation requirements
      specified in Section 4 of [RFC6936] and with 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.

   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
   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.

3.4.1.  UDP-Lite

   A special class of applications can derive benefit from having
   partially-damaged payloads delivered, rather than discarded, when
   using paths that include error-prone links.  Such applications can
   tolerate payload corruption and MAY choose to use the Lightweight
   User Datagram Protocol (UDP-Lite) [RFC3828] variant of UDP instead of
   basic UDP.  Applications that choose to use UDP-Lite instead of UDP
   should still follow the congestion control and other guidelines
   described for use with UDP in Section 3.

   UDP-Lite changes the semantics of the UDP "payload length" field to
   that of a "checksum coverage length" field.  Otherwise, UDP-Lite is

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   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.  Applications using UDP-Lite therefore MUST NOT make
   assumptions regarding the correctness of the data received in the
   insensitive part of the UDP-Lite payload.

   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 may be negotiated by
   an application.  UDP-Lite does not provide mechanisms to negotiate
   the checksum coverage between the sender and receiver.  This
   therefore needs to be performed by the application.

   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).

   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.

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3.5.  Middlebox Traversal Guidelines

   Network address translators (NATs) and firewalls are examples of
   intermediary devices ("middleboxes") that can exist along an end-to-
   end path.  A middlebox typically performs a function that requires it
   to maintain per-flow state.  For connection-oriented protocols, such
   as TCP, middleboxes snoop and parse the connection-management
   information, and create and destroy per-flow state accordingly.  For
   a connectionless protocol such as UDP, this approach is not possible.
   Consequently, middleboxes may create per-flow state when they see a
   packet that -- according to some local criteria -- indicates a new
   flow, and destroy the state after some period of time during which no
   packets belonging to the same flow have arrived.

   Depending on the specific function that the middlebox performs, this
   behavior can introduce a time-dependency that restricts the kinds of
   UDP traffic exchanges that will be successful across the middlebox.
   For example, NATs and firewalls typically define the partial path on
   one side of them to be interior to the domain they serve, whereas the
   partial path on their other side is defined to be exterior to that
   domain.  Per-flow state is typically created when the first packet
   crosses from the interior to the exterior, and while the state is
   present, NATs and firewalls will forward return traffic.  Return
   traffic that arrives after the per-flow state has timed out is
   dropped, as is other traffic that arrives from the exterior.

   Many applications that use UDP for communication operate across
   middleboxes without needing to employ additional mechanisms.  One
   example is the Domain Name System (DNS), which has a strict request-
   response communication pattern that typically completes within
   seconds.

   Other applications may experience communication failures when
   middleboxes destroy the per-flow state associated with an application
   session during periods when the application does not exchange any UDP
   traffic.  Applications SHOULD be able to gracefully handle such
   communication failures and implement mechanisms to re-establish
   application-layer sessions and state.

   For some applications, such as media transmissions, this re-
   synchronization is highly undesirable, because it can cause user-
   perceivable playback artifacts.  Such specialized applications MAY
   send periodic keep-alive messages to attempt to refresh middlebox
   state.  It is important to note that keep-alive messages are NOT
   RECOMMENDED for general use -- they are unnecessary for many
   applications and can consume significant amounts of system and
   network resources.

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   An application that needs to employ keep-alives to deliver useful
   service over UDP in the presence of middleboxes SHOULD NOT transmit
   them more frequently than once every 15 seconds and SHOULD use longer
   intervals when possible.  No common timeout has been specified for
   per-flow UDP state for arbitrary middleboxes.  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 network 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 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.

   Sending keep-alives is not a substitute for implementing a mechanism
   to recover from broken sessions.  Like all UDP datagrams, keep-alives
   can be delayed or dropped, causing middlebox state to time out.  In
   addition, the congestion control guidelines in Section 3.1 cover all
   UDP transmissions by an application, including the transmission of
   middlebox keep-alives.  Congestion control may thus lead to delays or
   temporary suspension of keep-alive transmission.

   Keep-alive messages are NOT RECOMMENDED for general use.  They are
   unnecessary for many applications and may consume significant
   resources.  For example, on battery-powered devices, if an
   application needs to maintain connectivity for long periods with
   little traffic, the frequency at which keep-alives are sent can
   become the determining factor that governs power consumption,
   depending on the underlying network technology.  Because many
   middleboxes are designed to require keep-alives for TCP connections
   at a frequency that is much lower than that needed for UDP, this
   difference alone can often be sufficient to prefer TCP over UDP for
   these deployments.  On the other hand, there is anecdotal evidence
   that suggests that direct communication through middleboxes, e.g., by
   using ICE [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 may 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

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   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 5.1.1).

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).

   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 [RFC0919].  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].  Anycast senders must
   be aware that successive messages sent to the same anycast IP address
   may be delivered to different anycast nodes, i.e., arrive at
   different locations in the topology.

   Most UDP tunnels that carry IP multicast traffic use a tunnel
   encapsulation with a unicast destination address.  These MUST follow
   the same requirements as a tunnel carrying unicast data (see

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   Section 3.1.7).  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
   bi-laterally 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).

4.1.  Multicast Congestion Control Guidelines

   Unicast congestion-controlled transport mechanism are often not
   applicable to multicast distribution services, or simply do not scale
   to large multicast trees, since they require bi-directional
   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.

   Congestion control mechanisms for multicast may operate on longer
   timescales than for unicast (e.g., due to the higher group RTT of a
   heterogeneous group); appropriate methods are particularly for any
   multicast session were all or part of the multicast distribution tree
   spans an access network (e.g., a home gateway).

   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.

   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 there
   is no way at the transport layer to prevent a join message
   propagating to the next-hop router.  A multicast congestion control
   method MAY therefore decide not to reduce the rate of the entire
   multicast group in response to a report received by a single
   receiver; instead it can decide to expel each congested receiver from
   the multicast group and to then distribute content to these congested

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   receivers at a lower rate using unicast congestion control.  Care
   needs to be taken when this action results in many flows being
   simultaneously transitioned, so that this does not result in
   excessive traffic exasperating congestion and potentially
   contributing to congestion collapse.

   Some classes of multicast applications support real-time
   transmissions in which the quality of the transfer may be monitored
   at the receiver.  Applications that can detect when there is a
   significant reduction in user quality SHOULD regard this as a
   congestion signal (e.g., to leave a group using layered multicast
   encoding) or SHOULD employ a circuit breaker to control the traffic.

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: 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
   schemes 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 3.1.2 are also applicable to low
   data-volume multicast applications.

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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 [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

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   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.

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5.1.  Using UDP Ports

   The rules 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
   [I-D.ietf-tsvwg-port-use].

   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.

   TCP commonly uses source port randomization for this reason
   [RFC6056].  Setting a "randomized" source port also helps provide
   greater assurance that reported ICMP errors originate from network
   systems on the path used by a particular flow.

   Protection from off-path data attacks can also be provided by
   randomizing the initial value of a protocol field within the datagram
   payload, and checking the validity of this field at the receiver.
   TCP also uses a random initial sequence number for the same reason,
   and similar techniques can be used by UDP applications (e.g., RTP has
   random initial sequence number and random media timestamp offsets
   [RFC3550]).

   A UDP receiver SHOULD NOT bind to port zero.  Applications SHOULD
   implement corresponding receiver 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).  Although the source port value is often not directly
   used in multicast applications, this should still be set to a random
   or predetermined value.

   The UDP source port number field has been used as a basis to design
   load-balancing solutions for IPv4.  This approach has also been
   leveraged for IPv6 [RFC6438], but for IPv6 the "flow label" [RFC6437]
   may also be used as entropy for load balancing.  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 were specified.
   Router vendors are encouraged to start using the 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

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   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.

5.1.1.  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).  This 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 can not 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).

   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 congestion control to react to congestion events, unless the
      congestion state is shared between all flows in a session.

   Therefore, applications MUST NOT assume consistent behavior of
   middleboxes when multiple UDP flows are used; many devices respond
   differently as the number of ports used 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 3.1.5.

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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, e.g., transient ICMP "unreachable"
   messages should not normally cause a communication abort.

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.  Applications that respond to short requests
   with potentially large responses are vulnerable to amplification
   attacks, and SHOULD authenticate the sender before responding.  The
   source IP address of a request is not a useful authenticator, because
   it can easily be spoofed.

   One option of securing UDP communications is with IPsec [RFC4301],
   which can provide authentication for flows of IP packets through the
   Authentication Header (AH) [RFC4302] and encryption and/or
   authentication through the Encapsulating Security Payload (ESP)
   [RFC4303].  Applications use the Internet Key Exchange (IKE)
   [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.

   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

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   (DTLS) [RFC6347].  DTLS provides communication privacy by encrypting
   UDP payloads.  It does not protect the UDP headers.  Applications can
   implement DTLS without relying on support from the operating system.

   Many other options for authenticating or encrypting UDP payloads
   exist.  For example, the GSS-API security framework [RFC2743] or
   Cryptographic Message Syntax (CMS) [RFC5652] could be used to protect
   UDP payloads.  The IETF standard for securing RTP [RFC3550]
   communication sessions over UDP is the Secure Real-time Transport
   Protocol (SRTP) [RFC3711].  In some applications, a better solution
   is to protect larger stand-alone objects, such as files or messages,
   instead of individual UDP payloads.  In these situations, CMS
   [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 (especially 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.
   [I-D.ietf-tsvwg-circuit-breaker] 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.

7.  Summary

   This section summarizes the guidelines made in Sections 3 and 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 (TCP, SCTP, DCCP)  |         |
   |                                                         |         |
   | SHOULD control rate of transmission                     | 3.1     |
   | SHOULD perform congestion control over all traffic      |         |

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   |                                                         |         |
   | for bulk transfers,                                     | 3.1.1   |
   | SHOULD consider implementing TFRC                       |         |
   | else, SHOULD in other ways use bandwidth similar to TCP |         |
   |                                                         |         |
   | for non-bulk transfers,                                 | 3.1.2   |
   | SHOULD measure RTT and transmit max. 1 datagram/RTT     |         |
   | else, SHOULD send at most 1 datagram every 3 seconds    |         |
   | SHOULD back-off retransmission timers following loss    |         |
   |                                                         |         |
   | SHOULD provide mechanisms to regulate the bursts of     | 3.1.3   |
   | transmission                                            |         |
   |                                                         |         |
   | for DiffServ, SHOULD NOT rely on implementation of PHBs | 3.1.4   |
   |                                                         |         |
   | for QoS-enabled paths, MAY choose not to use CC         | 3.1.5   |
   |                                                         |         |
   | for neither CC-controlled or pre-provisioned capacity,  | 3.1.6   |
   | SHOULD implement a transport circuit breaker            |         |
   | MAY implement a circuit breaker for other applications  |         |
   |                                                         |         |
   | for tunnels carrying IP Traffic,                        | 3.1.7   |
   | SHOULD NOT perform congestion control                   |         |
   |                                                         |         |
   | for non-IP tunnels or rate not determined by traffic,   |         |
   | SHOULD perform CC or use circuit breaker                |         |
   |                                                         |         |
   | 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   |         |
   | mechanisms are REQUIRED if a zero IPv6 UDP checksum is  |         |
   | used.                                                   |         |
   | else, MAY use UDP-Lite with suitable checksum coverage  | 3.4.1   |
   |                                                         |         |
   | SHOULD NOT always send middlebox keep-alives            | 3.5     |
   | MAY use keep-alives when needed (min. interval 15 sec)  |         |
   |                                                         |         |
   | Bulk multicast apps SHOULD implement congestion control | 4.1.1   |
   |                                                         |         |
   | Low volume multicast apps SHOULD implement congestion   | 4.1.2   |
   | control                                                 |         |

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   |                                                         |         |
   | Multicast apps SHOULD use a safe PMTU                   | 4.2     |
   |                                                         |         |
   | SHOULD avoid using multiple ports                       | 5       |
   | MUST check received IP source address                   |         |
   | and, for client/server applications,                    |         |
   | SHOULD send responses from src address matching request |         |
   |                                                         |         |
   | SHOULD use standard IETF security protocols when needed | 6       |
   +---------------------------------------------------------+---------+

                    Table 1: Summary of recommendations

8.  IANA Considerations

   Note to RFC-Editor: please remove this entire section prior to
   publication.

   This document raises no IANA considerations.

9.  Acknowledgments

   The middlebox traversal guidelines in Section 3.5 incorporate ideas
   from Section 5 of [I-D.ford-behave-app] by Bryan Ford, Pyda
   Srisuresh, and Dan Kegel.  G.  Fairhurst acknowledges support
   provided by the EU H2020 NEAT project.  Lars Eggert has received
   funding from the European Union's Horizon 2020 research and
   innovation program 2014-2018 under grant agreement No. 644866.  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.

10.  References

10.1.  Normative References

   [RFC0768]  Postel, J., "User Datagram Protocol", STD 6, RFC 768,
              August 1980.

   [RFC0793]  Postel, J., "Transmission Control Protocol", STD 7, RFC
              793, September 1981.

   [RFC1122]  Braden, R., "Requirements for Internet Hosts -
              Communication Layers", STD 3, RFC 1122, October 1989.

   [RFC1191]  Mogul, J. and S. Deering, "Path MTU discovery", RFC 1191,
              November 1990.

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   [RFC1981]  McCann, J., Deering, S., and J. Mogul, "Path MTU Discovery
              for IP version 6", RFC 1981, August 1996.

   [RFC2119]  Bradner, S., "Key words for use in RFCs to Indicate
              Requirement Levels", BCP 14, RFC 2119, March 1997.

   [RFC2460]  Deering, S. and R. Hinden, "Internet Protocol, Version 6
              (IPv6) Specification", RFC 2460, December 1998.

   [RFC2914]  Floyd, S., "Congestion Control Principles", BCP 41, RFC
              2914, September 2000.

   [RFC3828]  Larzon, L-A., Degermark, M., Pink, S., Jonsson, L-E., and
              G. Fairhurst, "The Lightweight User Datagram Protocol
              (UDP-Lite)", RFC 3828, July 2004.

   [RFC4787]  Audet, F. and C. Jennings, "Network Address Translation
              (NAT) Behavioral Requirements for Unicast UDP", BCP 127,
              RFC 4787, January 2007.

   [RFC4821]  Mathis, M. and J. Heffner, "Packetization Layer Path MTU
              Discovery", RFC 4821, March 2007.

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

   [RFC5405]  Eggert, L. and G. Fairhurst, "Unicast UDP Usage Guidelines
              for Application Designers", BCP 145, RFC 5405, November
              2008.

   [RFC6040]  Briscoe, Bob., "Tunnelling of Explicit Congestion
              Notification", November 2010.

   [RFC6298]  Paxson, V., Allman, M., Chu, J., and M. Sargent,
              "Computing TCP's Retransmission Timer", RFC 6298, June
              2011.

10.2.  Informative References

   [ALLMAN]   Allman, M. and E. Blanton, "Notes on burst mitigation for
              transport protocols", March 2005.

   [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.

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

   [I-D.ietf-aqm-ecn-benefits]
              Fairhurst, G. and M. Welzl, "The Benefits of using
              Explicit Congestion Notification (ECN)", draft-ietf-aqm-
              ecn-benefits-03 (work in progress), April 2015.

   [I-D.ietf-avtcore-rtp-circuit-breakers]
              Perkins, C. and V. Singh, "Multimedia Congestion Control:
              Circuit Breakers for Unicast RTP Sessions", draft-ietf-
              avtcore-rtp-circuit-breakers-10 (work in progress), March
              2015.

   [I-D.ietf-dart-dscp-rtp]
              Black, D. and P. Jones, "Differentiated Services
              (DiffServ) and Real-time Communication", draft-ietf-dart-
              dscp-rtp-10 (work in progress), November 2014.

   [I-D.ietf-tsvwg-circuit-breaker]
              Fairhurst, G., "Network Transport Circuit Breakers",
              draft-ietf-tsvwg-circuit-breaker-01 (work in progress),
              March 2015.

   [I-D.ietf-tsvwg-port-use]
              Touch, J., "Recommendations on Using Assigned Transport
              Port Numbers", draft-ietf-tsvwg-port-use-10 (work in
              progress), March 2015.

   [POSIX]    IEEE Std. 1003.1-2001, , "Standard for Information
              Technology - Portable Operating System Interface (POSIX)",
              Open Group Technical Standard: Base Specifications Issue
              6, ISO/IEC 9945:2002, December 2001.

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

   [RFC0919]  Mogul, J., "Broadcasting Internet Datagrams", STD 5, RFC
              919, October 1984.

   [RFC1112]  Deering, S., "Host extensions for IP multicasting", STD 5,
              RFC 1112, August 1989.

   [RFC1536]  Kumar, A., Postel, J., Neuman, C., Danzig, P., and S.
              Miller, "Common DNS Implementation Errors and Suggested
              Fixes", RFC 1536, October 1993.

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   [RFC1546]  Partridge, C., Mendez, T., and W. Milliken, "Host
              Anycasting Service", RFC 1546, November 1993.

   [RFC2309]  Braden, B., Clark, D., Crowcroft, J., Davie, B., Deering,
              S., Estrin, D., Floyd, S., Jacobson, V., Minshall, G.,
              Partridge, C., Peterson, L., Ramakrishnan, K., Shenker,
              S., Wroclawski, J., and L. Zhang, "Recommendations on
              Queue Management and Congestion Avoidance in the
              Internet", RFC 2309, April 1998.

   [RFC2475]  Blake, S., Black, D., Carlson, M., Davies, E., Wang, Z.,
              and W. Weiss, "An Architecture for Differentiated
              Services", RFC 2475, December 1998.

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

   [RFC2743]  Linn, J., "Generic Security Service Application Program
              Interface Version 2, Update 1", RFC 2743, January 2000.

   [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, August 2000.

   [RFC3048]  Whetten, B., Vicisano, L., Kermode, R., Handley, M.,
              Floyd, S., and M. Luby, "Reliable Multicast Transport
              Building Blocks for One-to-Many Bulk-Data Transfer", RFC
              3048, January 2001.

   [RFC3124]  Balakrishnan, H. and S. Seshan, "The Congestion Manager",
              RFC 3124, June 2001.

   [RFC3261]  Rosenberg, J., Schulzrinne, H., Camarillo, G., Johnston,
              A., Peterson, J., Sparks, R., Handley, M., and E.
              Schooler, "SIP: Session Initiation Protocol", RFC 3261,
              June 2002.

   [RFC3303]  Srisuresh, P., Kuthan, J., Rosenberg, J., Molitor, A., and
              A. Rayhan, "Middlebox communication architecture and
              framework", RFC 3303, August 2002.

   [RFC3493]  Gilligan, R., Thomson, S., Bound, J., McCann, J., and W.
              Stevens, "Basic Socket Interface Extensions for IPv6", RFC
              3493, February 2003.

   [RFC3550]  Schulzrinne, H., Casner, S., Frederick, R., and V.
              Jacobson, "RTP: A Transport Protocol for Real-Time
              Applications", STD 64, RFC 3550, July 2003.

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   [RFC3551]  Schulzrinne, H. and S. Casner, "RTP Profile for Audio and
              Video Conferences with Minimal Control", STD 65, RFC 3551,
              July 2003.

   [RFC3711]  Baugher, M., McGrew, D., Naslund, M., Carrara, E., and K.
              Norrman, "The Secure Real-time Transport Protocol (SRTP)",
              RFC 3711, March 2004.

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

   [RFC3758]  Stewart, R., Ramalho, M., Xie, Q., Tuexen, M., and P.
              Conrad, "Stream Control Transmission Protocol (SCTP)
              Partial Reliability Extension", RFC 3758, May 2004.

   [RFC3819]  Karn, P., Bormann, C., Fairhurst, G., Grossman, D.,
              Ludwig, R., Mahdavi, J., Montenegro, G., Touch, J., and L.
              Wood, "Advice for Internet Subnetwork Designers", BCP 89,
              RFC 3819, July 2004.

   [RFC4301]  Kent, S. and K. Seo, "Security Architecture for the
              Internet Protocol", RFC 4301, December 2005.

   [RFC4302]  Kent, S., "IP Authentication Header", RFC 4302, December
              2005.

   [RFC4303]  Kent, S., "IP Encapsulating Security Payload (ESP)", RFC
              4303, December 2005.

   [RFC4340]  Kohler, E., Handley, M., and S. Floyd, "Datagram
              Congestion Control Protocol (DCCP)", RFC 4340, March 2006.

   [RFC4341]  Floyd, S. and E. Kohler, "Profile for Datagram Congestion
              Control Protocol (DCCP) Congestion Control ID 2: TCP-like
              Congestion Control", RFC 4341, March 2006.

   [RFC4342]  Floyd, S., Kohler, E., and J. Padhye, "Profile for
              Datagram Congestion Control Protocol (DCCP) Congestion
              Control ID 3: TCP-Friendly Rate Control (TFRC)", RFC 4342,
              March 2006.

   [RFC4607]  Holbrook, H. and B. Cain, "Source-Specific Multicast for
              IP", RFC 4607, August 2006.

   [RFC4654]  Widmer, J. and M. Handley, "TCP-Friendly Multicast
              Congestion Control (TFMCC): Protocol Specification", RFC
              4654, August 2006.

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   [RFC4880]  Callas, J., Donnerhacke, L., Finney, H., Shaw, D., and R.
              Thayer, "OpenPGP Message Format", RFC 4880, November 2007.

   [RFC4960]  Stewart, R., "Stream Control Transmission Protocol", RFC
              4960, September 2007.

   [RFC4963]  Heffner, J., Mathis, M., and B. Chandler, "IPv4 Reassembly
              Errors at High Data Rates", RFC 4963, July 2007.

   [RFC4987]  Eddy, W., "TCP SYN Flooding Attacks and Common
              Mitigations", RFC 4987, August 2007.

   [RFC5082]  Gill, V., Heasley, J., Meyer, D., Savola, P., and C.
              Pignataro, "The Generalized TTL Security Mechanism
              (GTSM)", RFC 5082, October 2007.

   [RFC5245]  Rosenberg, J., "Interactive Connectivity Establishment
              (ICE): A Protocol for Network Address Translator (NAT)
              Traversal for Offer/Answer Protocols", RFC 5245, April
              2010.

   [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, August
              2009.

   [RFC5652]  Housley, R., "Cryptographic Message Syntax (CMS)", STD 70,
              RFC 5652, September 2009.

   [RFC5740]  Adamson, B., Bormann, C., Handley, M., and J. Macker,
              "NACK-Oriented Reliable Multicast (NORM) Transport
              Protocol", RFC 5740, November 2009.

   [RFC5751]  Ramsdell, B. and S. Turner, "Secure/Multipurpose Internet
              Mail Extensions (S/MIME) Version 3.2 Message
              Specification", RFC 5751, January 2010.

   [RFC5775]  Luby, M., Watson, M., and L. Vicisano, "Asynchronous
              Layered Coding (ALC) Protocol Instantiation", RFC 5775,
              April 2010.

   [RFC5971]  Schulzrinne, H. and R. Hancock, "GIST: General Internet
              Signalling Transport", RFC 5971, October 2010.

   [RFC5973]  Stiemerling, M., Tschofenig, H., Aoun, C., and E. Davies,
              "NAT/Firewall NSIS Signaling Layer Protocol (NSLP)", RFC
              5973, October 2010.

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   [RFC6056]  Larsen, M. and F. Gont, "Recommendations for Transport-
              Protocol Port Randomization", BCP 156, RFC 6056, January
              2011.

   [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, August 2011.

   [RFC6347]  Rescorla, E. and N. Modadugu, "Datagram Transport Layer
              Security Version 1.2", RFC 6347, January 2012.

   [RFC6396]  Blunk, L., Karir, M., and C. Labovitz, "Multi-Threaded
              Routing Toolkit (MRT) Routing Information Export Format",
              RFC 6396, October 2011.

   [RFC6437]  Amante, S., Carpenter, B., Jiang, S., and J. Rajahalme,
              "IPv6 Flow Label Specification", RFC 6437, November 2011.

   [RFC6438]  Carpenter, B. and S. Amante, "Using the IPv6 Flow Label
              for Equal Cost Multipath Routing and Link Aggregation in
              Tunnels", RFC 6438, November 2011.

   [RFC6513]  Rosen, E. and R. Aggarwal, "Multicast in MPLS/BGP IP
              VPNs", RFC 6513, February 2012.

   [RFC6679]  Westerlund, M., Johansson, I., Perkins, C., O'Hanlon, P.,
              and K. Carlberg, "Explicit Congestion Notification (ECN)
              for RTP over UDP", RFC 6679, August 2012.

   [RFC6726]  Paila, T., Walsh, R., Luby, M., Roca, V., and R. Lehtonen,
              "FLUTE - File Delivery over Unidirectional Transport", RFC
              6726, November 2012.

   [RFC6807]  Farinacci, D., Shepherd, G., Venaas, S., and Y. Cai,
              "Population Count Extensions to Protocol Independent
              Multicast (PIM)", RFC 6807, December 2012.

   [RFC6935]  Eubanks, M., Chimento, P., and M. Westerlund, "IPv6 and
              UDP Checksums for Tunneled Packets", RFC 6935, April 2013.

   [RFC6936]  Fairhurst, G. and M. Westerlund, "Applicability Statement
              for the Use of IPv6 UDP Datagrams with Zero Checksums",
              RFC 6936, April 2013.

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   [RFC7296]  Kaufman, C., Hoffman, P., Nir, Y., Eronen, P., and T.
              Kivinen, "Internet Key Exchange Protocol Version 2
              (IKEv2)", STD 79, RFC 7296, October 2014.

   [RFC7510]  Xu, X., Sheth, N., Yong, L., Callon, R., and D. Black,
              "Encapsulating MPLS in UDP", RFC 7510, April 2015.

   [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.

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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 co-operating 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].

   MPLS-in-UDP endpoints must 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

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   offsetting corruptions to the source IPv6 and/or MPLS top label are
   unlikely.

   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 as 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.

Appendix B.  Revision Notes

   Note to RFC-Editor: please remove this entire section prior to
   publication.

   Changes in draft-ietf-tsvwg-rfc5405bis-02:

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   o  Added note that guidance may be applicable beyond UDP to abstract
      (from Erik Nordmark).

   o  Small editorial changes to fix English nits.

   o  Added a circuit may provide benefit to CC tunnels by controlling
      envelope.

   o  Added tunnels should ingress-filter by packet type (from Erik
      Nordmark).

   o  Added tunnels should perform IETF ECN processing when supporting
      ECN.

   o  Multicast apps may employ CC or a circuit breaker.

   o  Added programming guidance on off-path attacks (with C.  Perkins).

   o  Added reference to ECN benefits.

   Changes in draft-ietf-tsvwg-rfc5405bis-01:

   o  Added text on DSCP-usage.

   o  More guidance on use of the checksum, including an example of how
      MPLS/UDP allowed support of a zero IPv6 UDP Checksum in some
      cases.

   o  Added description of diffuse usage.

   o  Clarified usage of the source port field.

   draft-eggert-tsvwg-rfc5405bis-01 was adopted by the TSVWG and
   resubmitted as draft-ietf-tsvwg-rfc5405bis-00.  There were no
   technical changes.

   Changes in draft-eggert-tsvwg-rfc5405bis-01:

   o  Added Greg Shepherd as a co-author, based on the multicast
      guidelines that originated with him.

   Changes in draft-eggert-tsvwg-rfc5405bis-00 (relative to RFC5405):

   o  The words "application designers" were removed from the draft
      title and the wording of the abstract was clarified abstract.

   o  New text to clarify various issues and set new recommendations not
      previously included in RFC 5405.  These include new

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      recommendations for multicast, the use of checksums with IPv6,
      ECMP, recommendations on port usage, use of ECN, use of DiffServ,
      circuit breakers (initial text), etc.

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
   USA

   EMail: gjshep@gmail.com

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