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Internet Engineering Task Force (IETF)                         L. EggertRequest for Comments: 8085                                        NetAppBCP: 145                                                    G. FairhurstObsoletes:5405                                   University of AberdeenCategory: Best Current Practice                              G. ShepherdISSN: 2070-1721                                            Cisco Systems                                                              March 2017UDP Usage GuidelinesAbstract   The User Datagram Protocol (UDP) provides a minimal message-passing   transport that has no inherent congestion control mechanisms.  This   document provides guidelines on the use of UDP for the designers of   applications, tunnels, and other protocols that use UDP.  Congestion   control guidelines are a primary focus, but the document also   provides guidance on other topics, including message sizes,   reliability, checksums, middlebox traversal, the use of Explicit   Congestion Notification (ECN), Differentiated Services Code Points   (DSCPs), and ports.   Because congestion control is critical to the stable operation of the   Internet, applications and other protocols that choose to use UDP as   an Internet transport must employ mechanisms to prevent congestion   collapse and to establish some degree of fairness with concurrent   traffic.  They may also need to implement additional mechanisms,   depending on how they use UDP.   Some guidance is also applicable to the design of other protocols   (e.g., protocols layered directly on IP or via IP-based tunnels),   especially when these protocols do not themselves provide congestion   control.   This document obsoletesRFC 5405 and adds guidelines for multicast   UDP usage.Eggert, et al.            Best Current Practice                 [Page 1]

RFC 8085                  UDP Usage Guidelines                March 2017Status of This Memo   This memo documents an Internet Best Current Practice.   This document is a product of the Internet Engineering Task Force   (IETF).  It represents the consensus of the IETF community.  It has   received public review and has been approved for publication by the   Internet Engineering Steering Group (IESG).  Further information on   BCPs is available inSection 2 of RFC 7841.   Information about the current status of this document, any errata,   and how to provide feedback on it may be obtained athttp://www.rfc-editor.org/info/rfc8085.Copyright Notice   Copyright (c) 2017 IETF Trust and the persons identified as the   document authors.  All rights reserved.   This document is subject toBCP 78 and the IETF Trust's Legal   Provisions Relating to IETF Documents   (http://trustee.ietf.org/license-info) in effect on the date of   publication of this document.  Please review these documents   carefully, as they describe your rights and restrictions with respect   to this document.  Code Components extracted from this document must   include Simplified BSD License text as described in Section 4.e of   the Trust Legal Provisions and are provided without warranty as   described in the Simplified BSD License.Eggert, et al.            Best Current Practice                 [Page 2]

RFC 8085                  UDP Usage Guidelines                March 2017Table of Contents1. Introduction ....................................................32. Terminology .....................................................53. UDP Usage Guidelines ............................................53.1. Congestion Control Guidelines ..............................63.2. Message Size Guidelines ...................................193.3. Reliability Guidelines ....................................213.4. Checksum Guidelines .......................................223.5. Middlebox Traversal Guidelines ............................253.6. Limited Applicability and Controlled Environments .........274. Multicast UDP Usage Guidelines .................................284.1. Multicast Congestion Control Guidelines ...................304.2. Message Size Guidelines for Multicast .....................325. Programming Guidelines .........................................325.1. Using UDP Ports ...........................................345.2. ICMP Guidelines ...........................................376. Security Considerations ........................................387. Summary ........................................................408. References .....................................................428.1. Normative References ......................................428.2. Informative References ....................................43Appendix A. .......................................................53   Acknowledgments ...................................................55   Authors' Addresses ................................................551.  Introduction   The User Datagram Protocol (UDP) [RFC768] provides a minimal,   unreliable, best-effort, message-passing transport to applications   and other protocols (such as tunnels) that wish to operate over IP.   Both are simply called "applications" in the remainder of this   document.   Compared to other transport protocols, UDP and its UDP-Lite variant   [RFC3828] are unique in that they do not establish end-to-end   connections between communicating end systems.  UDP communication   consequently does not incur connection establishment and teardown   overheads, and there is minimal associated end-system state.  Because   of these characteristics, UDP can offer a very efficient   communication transport to some applications.   A second unique characteristic of UDP is that it provides no inherent   congestion control mechanisms.  On many platforms, applications can   send UDP datagrams at the line rate of the platform's link interface,   which is often much greater than the available end-to-end path   capacity, and doing so contributes to congestion along the path.   [RFC2914] describes the best current practice for congestion controlEggert, et al.            Best Current Practice                 [Page 3]

RFC 8085                  UDP Usage Guidelines                March 2017   in the Internet.  It identifies two major reasons why congestion   control mechanisms are critical for the stable operation of the   Internet:   1.  The prevention of congestion collapse, i.e., a state where an       increase in network load results in a decrease in useful work       done by the network.   2.  The establishment of a degree of fairness, i.e., allowing       multiple flows to share the capacity of a path reasonably       equitably.   Because UDP itself provides no congestion control mechanisms, it is   up to the applications that use UDP for Internet communication to   employ suitable mechanisms to prevent congestion collapse and   establish a degree of fairness.  [RFC2309] discusses the dangers of   congestion-unresponsive flows and states that "all UDP-based   streaming applications should incorporate effective congestion   avoidance mechanisms."  [RFC7567] reaffirms this statement.  This is   an important requirement, even for applications that do not use UDP   for streaming.  In addition, congestion-controlled transmission is of   benefit to an application itself, because it can reduce self-induced   packet loss, minimize retransmissions, and hence reduce delays.   Congestion control is essential even at relatively slow transmission   rates.  For example, an application that generates five 1500-byte UDP   datagrams in one second can already exceed the capacity of a 56 Kb/s   path.  For applications that can operate at higher, potentially   unbounded data rates, congestion control becomes vital to prevent   congestion collapse and establish some degree of fairness.Section 3   describes a number of simple guidelines for the designers of such   applications.   A UDP datagram is carried in a single IP packet and is hence limited   to a maximum payload of 65,507 bytes for IPv4 and 65,527 bytes for   IPv6.  The transmission of large IP packets usually requires IP   fragmentation.  Fragmentation decreases communication reliability and   efficiency and should be avoided.  IPv6 allows the option of   transmitting large packets ("jumbograms") without fragmentation when   all link layers along the path support this [RFC2675].  Some of the   guidelines inSection 3 describe how applications should determine   appropriate message sizes.  Other sections of this document provide   guidance on reliability, checksums, middlebox traversal and use of   multicast.   This document provides guidelines and recommendations.  Although most   UDP applications are expected to follow these guidelines, there do   exist valid reasons why a specific application may decide not to   follow a given guideline.  In such cases, it is RECOMMENDED thatEggert, et al.            Best Current Practice                 [Page 4]

RFC 8085                  UDP Usage Guidelines                March 2017   application designers cite the respective section(s) of this document   in the technical specification of their application or protocol and   explain their rationale for their design choice.   [RFC5405] was scoped to provide guidelines for unicast applications   only, whereas this document also provides guidelines for UDP flows   that use IP anycast, multicast, broadcast, and applications that use   UDP tunnels to support IP flows.   Finally, although this document specifically refers to usage of UDP,   the spirit of some of its guidelines also applies to other message-   passing applications and protocols (specifically on the topics of   congestion control, message sizes, and reliability).  Examples   include signaling, tunnel or control applications that choose to run   directly over IP by registering their own IP protocol number with   IANA.  This document is expected to provide useful background reading   to the designers of such applications and protocols.2.  Terminology   The key words "MUST", "MUST NOT", "REQUIRED", "SHALL", "SHALL NOT",   "SHOULD", "SHOULD NOT", "RECOMMENDED", "NOT RECOMMENDED", "MAY", and   "OPTIONAL" in this document are to be interpreted as described in   [RFC2119].3.  UDP Usage Guidelines   Internet paths can have widely varying characteristics, including   transmission delays, available bandwidths, congestion levels,   reordering probabilities, supported message sizes, or loss rates.   Furthermore, the same Internet path can have very different   conditions over time.  Consequently, applications that may be used on   the Internet MUST NOT make assumptions about specific path   characteristics.  They MUST instead use mechanisms that let them   operate safely under very different path conditions.  Typically, this   requires conservatively probing the current conditions of the   Internet path they communicate over to establish a transmission   behavior that it can sustain and that is reasonably fair to other   traffic sharing the path.   These mechanisms are difficult to implement correctly.  For most   applications, the use of one of the existing IETF transport protocols   is the simplest method of acquiring the required mechanisms.  Doing   so also avoids issues that protocols using a new IP protocol number   face when being deployed over the Internet, where middleboxes that   only support TCP and UDP are sometimes present.  Consequently, the   RECOMMENDED alternative to the UDP usage described in the remainder   of this section is the use of an IETF transport protocol such as TCPEggert, et al.            Best Current Practice                 [Page 5]

RFC 8085                  UDP Usage Guidelines                March 2017   [RFC793], Stream Control Transmission Protocol (SCTP) [RFC4960], and   SCTP Partial Reliability Extension (SCTP-PR) [RFC3758], or Datagram   Congestion Control Protocol (DCCP) [RFC4340] with its different   congestion control types [RFC4341][RFC4342][RFC5622], or transport   protocols specified by the IETF in the future.  (UDP-encapsulated   SCTP [RFC6951] and DCCP [RFC6773] can offer support for traversing   firewalls and other middleboxes where the native protocols are not   supported.)   If used correctly, these more fully featured transport protocols are   not as "heavyweight" as often claimed.  For example, the TCP   algorithms have been continuously improved over decades, and they   have reached a level of efficiency and correctness that custom   application-layer mechanisms will struggle to easily duplicate.  In   addition, many TCP implementations allow connections to be tuned by   an application to its purposes.  For example, TCP's "Nagle" algorithm   [RFC1122] can be disabled, improving communication latency at the   expense of more frequent -- but still congestion controlled -- packet   transmissions.  Another example is the TCP SYN cookie mechanism   [RFC4987], which is available on many platforms.  TCP with SYN   cookies does not require a server to maintain per-connection state   until the connection is established.  TCP also requires the end that   closes a connection to maintain the TIME-WAIT state that prevents   delayed segments from one connection instance from interfering with a   later one.  Applications that are aware of and designed for this   behavior can shift maintenance of the TIME-WAIT state to conserve   resources by controlling which end closes a TCP connection [FABER].   Finally, TCP's built-in capacity-probing and awareness of the maximum   transmission unit supported by the path (PMTU) results in efficient   data transmission that quickly compensates for the initial connection   setup delay, in the case of transfers that exchange more than a few   segments.3.1.  Congestion Control Guidelines   If an application or protocol chooses not to use a congestion-   controlled transport protocol, it SHOULD control the rate at which it   sends UDP datagrams to a destination host, in order to fulfill the   requirements of [RFC2914].  It is important to stress that an   application SHOULD perform congestion control over all UDP traffic it   sends to a destination, independently from how it generates this   traffic.  For example, an application that forks multiple worker   processes or otherwise uses multiple sockets to generate UDP   datagrams SHOULD perform congestion control over the aggregate   traffic.Eggert, et al.            Best Current Practice                 [Page 6]

RFC 8085                  UDP Usage Guidelines                March 2017   Several approaches to perform congestion control are discussed in the   remainder of this section.  This section describes generic topics   with an intended emphasis on unicast and anycast [RFC1546] usage.   Not all approaches discussed below are appropriate for all UDP-   transmitting applications.Section 3.1.2 discusses congestion   control options for applications that perform bulk transfers over   UDP.  Such applications can employ schemes that sample the path over   several subsequent round-trips during which data is exchanged to   determine a sending rate that the path at its current load can   support.  Other applications only exchange a few UDP datagrams with a   destination.Section 3.1.3 discusses congestion control options for   such "low data-volume" applications.  Because they typically do not   transmit enough data to iteratively sample the path to determine a   safe sending rate, they need to employ different kinds of congestion   control mechanisms.Section 3.1.11 discusses congestion control   considerations when UDP is used as a tunneling protocol.Section 4   provides additional recommendations for broadcast and multicast   usage.   It is important to note that congestion control should not be viewed   as an add-on to a finished application.  Many of the mechanisms   discussed in the guidelines below require application support to   operate correctly.  Application designers need to consider congestion   control throughout the design of their application, similar to how   they consider security aspects throughout the design process.   In the past, the IETF has also investigated integrated congestion   control mechanisms that act on the traffic aggregate between two   hosts, i.e., a framework such as the Congestion Manager [RFC3124],   where active sessions may share current congestion information in a   way that is independent of the transport protocol.  Such mechanisms   have currently failed to see deployment, but would otherwise simplify   the design of congestion control mechanisms for UDP sessions, so that   they fulfill the requirements in [RFC2914].3.1.1.  Protocol Timer Guidelines   Understanding the latency between communicating endpoints is usually   a crucial part of effective congestion control implementations for   protocols and applications.  Latency estimation can be used in a   number of protocol functions, such as calculating a congestion-   controlled transmission rate, triggering retransmission, and   detecting packet loss.  Additional protocol functions, for example,   determining an interval for probing a path, determining an interval   between keep-alive messages, determining an interval for measuring   the quality of experience, or determining if a remote endpoint hasEggert, et al.            Best Current Practice                 [Page 7]

RFC 8085                  UDP Usage Guidelines                March 2017   responded to a request to perform an action, typically operate over   longer timescales than congestion control and therefore are not   covered in this section.   The general recommendation in this document is that applications   SHOULD leverage existing congestion control techniques and the   latency estimators specified therein (see next subsection).  The   following guidelines are provided for applications that need to   design their own latency estimation mechanisms.   The guidelines are framed in terms of "latency" and not "round-trip   time" because some situations require characterizing only the   network-based latency (e.g., TCP-Friendly Rate Control (TFRC)   [RFC5348]), while other cases necessitate inclusion of the time   required by the remote endpoint to provide feedback (e.g., developing   an understanding of when to retransmit a message).   The latency between endpoints is generally a dynamic property.   Therefore, estimates SHOULD represent some sort of averaging of   multiple recent measurement samples to account for variance.   Leveraging an Exponentially Weighted Moving Average (EWMA) has proven   useful for this purpose (e.g., in TCP [RFC6298] and TFRC [RFC5348]).   Independent latency estimates SHOULD be maintained for each   destination with which an endpoint communicates.   Latency samples MUST NOT be derived from ambiguous transactions.  The   canonical example is in a protocol that retransmits data, but   subsequently cannot determine which copy is being acknowledged.  This   ambiguity makes correct computation of the latency problematic.  See   the discussion of Karn's algorithm in [RFC6298].  This requirement   ensures a sender establishes a sound estimate of the latency without   relying on misleading measurements.   When a latency estimate is used to arm a timer that provides loss   detection -- with or without retransmission -- expiry of the timer   MUST be interpreted as an indication of congestion in the network,   causing the sending rate to be adapted to a safe conservative rate   (e.g., TCP collapses the congestion window to one segment [RFC5681]).   Some applications require an initial latency estimate before the   latency between endpoints can be empirically sampled.  For instance,   when arming a retransmission timer, an initial value is needed to   protect the messages sent before the endpoints sample the latency.   This initial latency estimate SHOULD generally be as conservative   (large) as possible for the given application.  For instance, in the   absence of any knowledge about the latency of a path, TCP requires   the initial Retransmission Timeout (RTO) to be set to no less than 1Eggert, et al.            Best Current Practice                 [Page 8]

RFC 8085                  UDP Usage Guidelines                March 2017   second [RFC6298].  UDP applications SHOULD similarly use an initial   latency estimate of 1 second.  Values shorter than 1 second can be   problematic (see the data analysis in the appendix of [RFC6298]).3.1.2.  Bulk-Transfer Applications   Applications that perform bulk transmission of data to a peer over   UDP, i.e., applications that exchange more than a few UDP datagrams   per RTT, SHOULD implement TFRC [RFC5348], window-based TCP-like   congestion control, or otherwise ensure that the application complies   with the congestion control principles.   TFRC has been designed to provide both congestion control and   fairness in a way that is compatible with the IETF's other transport   protocols.  If an application implements TFRC, it need not follow the   remaining guidelines inSection 3.1.2, because TFRC already addresses   them, but it SHOULD still follow the remaining guidelines in the   subsequent subsections ofSection 3.   Bulk-transfer applications that choose not to implement TFRC or TCP-   like windowing SHOULD implement a congestion control scheme that   results in bandwidth (capacity) use that competes fairly with TCP   within an order of magnitude.Section 2 of [RFC3551] suggests that applications SHOULD monitor the   packet-loss rate to ensure that it is within acceptable parameters.   Packet loss is considered acceptable if a TCP flow across the same   network path under the same network conditions would achieve an   average throughput, measured on a reasonable timescale, that is not   less than that of the UDP flow.  The comparison to TCP cannot be   specified exactly, but is intended as an "order-of-magnitude"   comparison in timescale and throughput.  The recommendations for   managing timers specified inSection 3.1.1 also apply.   Finally, some bulk-transfer applications may choose not to implement   any congestion control mechanism and instead rely on transmitting   across reserved path capacity (seeSection 3.1.9).  This might be an   acceptable choice for a subset of restricted networking environments,   but is by no means a safe practice for operation over the wider   Internet.  When the UDP traffic of such applications leaks out into   unprovisioned Internet paths, it can significantly degrade the   performance of other traffic sharing the path and even result in   congestion collapse.  Applications that support an uncontrolled or   unadaptive transmission behavior SHOULD NOT do so by default and   SHOULD instead require users to explicitly enable this mode of   operation, and they SHOULD verify that sufficient path capacity has   been reserved for them.Eggert, et al.            Best Current Practice                 [Page 9]

RFC 8085                  UDP Usage Guidelines                March 20173.1.3.  Low Data-Volume Applications   When applications that at any time exchange only a few UDP datagrams   with a destination implement TFRC or one of the other congestion   control schemes inSection 3.1.2, the network sees little benefit,   because those mechanisms perform congestion control in a way that is   only effective for longer transmissions.   Applications that at any time exchange only a few UDP datagrams with   a destination SHOULD still control their transmission behavior by not   sending on average more than one UDP datagram per RTT to a   destination.  Similar to the recommendation in [RFC1536], an   application SHOULD maintain an estimate of the RTT for any   destination with which it communicates using the methods specified inSection 3.1.1.   Some applications cannot maintain a reliable RTT estimate for a   destination.  These applications do not need to or are unable to use   protocol timers to measure the RTT (Section 3.1.1).  Two cases can be   identified:   1.  The first case is that of applications that exchange too few UDP       datagrams with a peer to establish a statistically accurate RTT       estimate but that can monitor the reliability of transmission       (Section 3.3).  Such applications MAY use a predetermined       transmission interval that is exponentially backed off when       packets are deemed lost.  TCP specifies an initial value of 1       second [RFC6298], which is also RECOMMENDED as an initial value       for UDP applications.  Some low data-volume applications, e.g.,       SIP [RFC3261] and General Internet Signaling Transport (GIST)       [RFC5971] use an interval of 500 ms, and shorter values are       likely problematic in many cases.  As in the previous case, note       that the initial timeout is not the maximum possible timeout, seeSection 3.1.1.   2.  A second case of applications cannot maintain an RTT estimate for       a destination, because the destination does not send return       traffic.  Such applications SHOULD NOT send more than one UDP       datagram every 3 seconds and SHOULD use an even less aggressive       rate when possible.  Shorter values are likely problematic in       many cases.  Note that the sending rate in this case must be more       conservative than in the previous cases, because the lack of       return traffic prevents the detection of packet loss, i.e.,       congestion, and the application therefore cannot perform       exponential back off to reduce load.Eggert, et al.            Best Current Practice                [Page 10]

RFC 8085                  UDP Usage Guidelines                March 20173.1.4.  Applications Supporting Bidirectional Communications   Applications that communicate bidirectionally SHOULD employ   congestion control for both directions of the communication.  For   example, for a client-server, request-response-style application,   clients SHOULD congestion-control their request transmission to a   server, and the server SHOULD congestion-control its responses to the   clients.  Congestion in the forward and reverse directions is   uncorrelated, and an application SHOULD either independently detect   and respond to congestion along both directions or limit new and   retransmitted requests based on acknowledged responses across the   entire round-trip path.3.1.5.  Implications of RTT and Loss Measurements on Congestion Control   Transports such as TCP, SCTP, and DCCP provide timely detection of   congestion that results in an immediate reduction of their maximum   sending rate when congestion is experienced.  This reaction is   typically completed 1-2 RTTs after loss/congestion is encountered.   Applications using UDP SHOULD implement a congestion control scheme   that provides a prompt reaction to signals indicating congestion   (e.g., by reducing the rate within the next RTT following a   congestion signal).   The operation of a UDP congestion control algorithm can be very   different from the way TCP operates.  This includes congestion   controls that respond on timescales that fit applications that cannot   usefully work within the "change rate every RTT" model of TCP.   Applications that experience a low or varying RTT are particularly   vulnerable to sampling errors (e.g., due to measurement noise or   timer accuracy).  This suggests the need to average loss/congestion   and RTT measurements over a longer interval; however, this also can   contribute additional delay in detecting congestion.  Some   applications may not react by reducing their sending rate immediately   for various reasons, including the following: RTT and loss   measurements are only made periodically (e.g., using RTCP),   additional time is required to filter information, or the application   is only able to change its sending rate at predetermined interval   (e.g., some video codecs).   When designing a congestion control algorithm, the designer therefore   needs to consider the total time taken to reduce the load following a   lack of feedback or a congestion event.  An application where the   most recent RTT measurement is smaller than the actual RTT or the   measured loss rate is smaller than the current rate, can result in   over estimating the available capacity.  Such over-estimation canEggert, et al.            Best Current Practice                [Page 11]

RFC 8085                  UDP Usage Guidelines                March 2017   result in a sending rate that creates congestion to the application   or other flows sharing the path capacity, and can contribute to   congestion collapse -- both of these need to be avoided.   A congestion control designed for UDP SHOULD respond as quickly as   possible when it experiences congestion, and it SHOULD take into   account both the loss rate and the response time when choosing a new   rate.  The implemented congestion control scheme SHOULD result in   bandwidth (capacity) use that is comparable to that of TCP within an   order of magnitude, so that it does not starve other flows sharing a   common bottleneck.3.1.6.  Burst Mitigation and Pacing   UDP applications SHOULD provide mechanisms to regulate the bursts of   transmission that the application may send to the network.  Many TCP   and SCTP implementations provide mechanisms that prevent a sender   from generating long bursts at line-rate, since these are known to   induce early loss to applications sharing a common network   bottleneck.  The use of pacing with TCP [ALLMAN] has also been shown   to improve the coexistence of TCP flows with other flows.  The need   to avoid excessive transmission bursts is also noted in   specifications for applications (e.g., [RFC7143]).   Even low data-volume UDP flows may benefit from packet pacing, e.g.,   an application that sends three copies of a packet to improve   robustness to loss is RECOMMENDED to pace out those three packets   over several RTTs, to reduce the probability that all three packets   will be lost due to the same congestion event (or other event, such   as burst corruption).3.1.7.  Explicit Congestion Notification   Internet applications can use Explicit Congestion Notification (ECN)   [RFC3168] to gain benefits for the services they support [RFC8087].   Internet transports, such as TCP, provide a set of mechanisms that   are needed to utilize ECN.  ECN operates by setting an ECN-capable   codepoint (ECT(0) or ECT(1)) in the IP header of packets that are   sent.  This indicates to ECN-capable network devices (routers and   other devices) that they may mark (set the congestion experienced,   Congestion Experience (CE) codepoint) rather than drop the IP packet   as a signal of incipient congestion.   UDP applications can also benefit from enabling ECN, providing that   the API supports ECN and that they implement the required protocol   mechanisms to support ECN.Eggert, et al.            Best Current Practice                [Page 12]

RFC 8085                  UDP Usage Guidelines                March 2017   The set of mechanisms required for an application to use ECN over UDP   are:   o  A sender MUST provide a method to determine (e.g., negotiate) that      the corresponding application is able to provide ECN feedback      using a compatible ECN method.   o  A receiver that enables the use of ECN for a UDP port MUST check      the ECN field at the receiver for each UDP datagram that it      receives on this port.   o  The receiving application needs to provide feedback of congestion      information to the sending application.  This MUST report the      presence of datagrams received with a CE-mark by providing a      mechanism to feed this congestion information back to the sending      application.  The feedback MAY also report the presence of ECT(1)      and ECT(0)/Not-ECT packets [RFC7560].  ([RFC3168] and [RFC7560]      specify methods for TCP.)   o  An application sending ECN-capable datagrams MUST provide an      appropriate congestion reaction when it receives feedback      indicating that congestion has been experienced.  This ought to      result in reduction of the sending rate by the UDP congestion      control method (seeSection 3.1) that is not less than the      reaction of TCP under equivalent conditions.   o  A sender SHOULD detect network paths that do not support the ECN      field correctly.  When detected, they need to either      conservatively react to congestion or even fall back to not using      ECN [RFC8087].  This method needs to be robust to changes within      the network path that may occur over the lifetime of a session.   o  A sender is encouraged to provide a mechanism to detect and react      appropriately to misbehaving receivers that fail to report      CE-marked packets [RFC8087].   [RFC6679] provides guidance and an example of this support, by   describing a method to allow ECN to be used for UDP-based   applications using the Real-Time Protocol (RTP).  Applications that   cannot provide this set of mechanisms, but wish to gain the benefits   of using ECN, are encouraged to use a transport protocol that already   supports ECN (such as TCP).3.1.8.  Differentiated Services Model   An application using UDP can use the differentiated services   (DiffServ) Quality of Service (QoS) framework.  To enable   differentiated services processing, a UDP sender sets theEggert, et al.            Best Current Practice                [Page 13]

RFC 8085                  UDP Usage Guidelines                March 2017   Differentiated Services Code Point (DSCP) field [RFC2475] in packets   sent to the network.  Normally, a UDP source/destination port pair   will set a single DSCP value for all packets belonging to a flow, but   multiple DSCPs can be used as described later in this section.  A   DSCP may be chosen from a small set of fixed values (the class   selector code points), or from a set of recommended values defined in   the Per Hop Behavior (PHB) specifications, or from values that have   purely local meanings to a specific network that supports DiffServ.   In general, packets may be forwarded across multiple networks between   source and destination.   In setting a non-default DSCP value, an application must be aware   that DSCP markings may be changed or removed between the traffic   source and destination.  This has implications on the design of   applications that use DSCPs.  Specifically, applications SHOULD be   designed not to rely on implementation of a specific network   treatment; they need instead to implement congestion control methods   to determine if their current sending rate is inducing congestion in   the network.   [RFC7657] describes the implications of using DSCPs and provides   recommendations on using multiple DSCPs within a single network five-   tuple (source and destination addresses, source and destination   ports, and the transport protocol used, in this case, UDP or   UDP-Lite), and particularly the expected impact on transport protocol   interactions, with congestion control or reliability functionality   (e.g., retransmission, reordering).  Use of multiple DSCPs can result   in reordering by increasing the set of network forwarding resources   used by a sender.  It can also increase exposure to resource   depletion or failure.3.1.9.  QoS, Pre-Provisioned, or Reserved Capacity   The IETF usually specifies protocols for use within the Best Effort   General Internet.  Sometimes it is relevant to specify protocols with   a different applicability.  An application using UDP can use the   integrated services QoS framework.  This framework is usually made   available within controlled environments (e.g., within a single   administrative domain or bilaterally agreed connection between   domains).  Applications intended for the Internet SHOULD NOT assume   that QoS mechanisms are supported by the networks they use, and   therefore need to provide congestion control, error recovery, etc.,   in case the actual network path does not provide provisioned service.   Some UDP applications are only expected to be deployed over network   paths that use pre-provisioned capacity or capacity reserved using   dynamic provisioning, e.g., through the Resource Reservation Protocol   (RSVP).  Multicast applications are also used with pre-provisionedEggert, et al.            Best Current Practice                [Page 14]

RFC 8085                  UDP Usage Guidelines                March 2017   capacity (e.g., IPTV deployments within access networks).  These   applications MAY choose not to implement any congestion control   mechanism and instead rely on transmitting only on paths where the   capacity is provisioned and reserved for this use.  This might be an   acceptable choice for a subset of restricted networking environments,   but is by no means a safe practice for operation over the wider   Internet.  Applications that choose this option SHOULD carefully and   in detail describe the provisioning and management procedures that   result in the desired containment.   Applications that support an uncontrolled or unadaptive transmission   behavior SHOULD NOT do so by default and SHOULD instead require users   to explicitly enable this mode of operation.   Applications designed for use within a controlled environment (seeSection 3.6) may be able to exploit network management functions to   detect whether they are causing congestion, and react accordingly.   If the traffic of such applications leaks out into unprovisioned   Internet paths, it can significantly degrade the performance of other   traffic sharing the path and even result in congestion collapse.   Protocols designed for such networks SHOULD provide mechanisms at the   network edge to prevent leakage of traffic into unprovisioned   Internet paths (e.g., [RFC7510]).  To protect other applications   sharing the same path, applications SHOULD also deploy an appropriate   circuit breaker, as described inSection 3.1.10.   An IETF specification targeting a controlled environment is expected   to provide an applicability statement that restricts the application   to the controlled environment (seeSection 3.6).3.1.10.  Circuit Breaker Mechanisms   A transport circuit breaker is an automatic mechanism that is used to   estimate the congestion caused by a flow, and to terminate (or   significantly reduce the rate of) the flow when excessive congestion   is detected [RFC8084].  This is a safety measure to prevent   congestion collapse (starvation of resources available to other   flows), essential for an Internet that is heterogeneous and for   traffic that is hard to predict in advance.   A circuit breaker is intended as a protection mechanism of last   resort.  Under normal circumstances, a circuit breaker should not be   triggered; it is designed to protect things when there is severe   overload.  The goal is usually to limit the maximum transmission rate   that reflects the available capacity of a network path.  Circuit   breakers can operate on individual UDP flows or traffic aggregates,   e.g., traffic sent using a network tunnel.Eggert, et al.            Best Current Practice                [Page 15]

RFC 8085                  UDP Usage Guidelines                March 2017   [RFC8084] provides guidance and examples on the use of circuit   breakers.  The use of a circuit breaker in RTP is specified in   [RFC8083].   Applications used in the general Internet SHOULD implement a   transport circuit breaker if they do not implement congestion control   or operate a low data-volume service (seeSection 3.6).  All   applications MAY implement a transport circuit breaker [RFC8084] and   are encouraged to consider implementing at least a slow-acting   transport circuit breaker to provide a protection of last resort for   their network traffic.3.1.11.  UDP Tunnels   One increasingly popular use of UDP is as a tunneling protocol   [INT-TUNNELS], where a tunnel endpoint encapsulates the packets of   another protocol inside UDP datagrams and transmits them to another   tunnel endpoint, which decapsulates the UDP datagrams and forwards   the original packets contained in the payload.  One example of such a   protocol is Teredo [RFC4380].  Tunnels establish virtual links that   appear to directly connect locations that are distant in the physical   Internet topology and can be used to create virtual (private)   networks.  Using UDP as a tunneling protocol is attractive when the   payload protocol is not supported by middleboxes that may exist along   the path, because many middleboxes support transmission using UDP.   Well-implemented tunnels are generally invisible to the endpoints   that happen to transmit over a path that includes tunneled links.  On   the other hand, to the routers along the path of a UDP tunnel, i.e.,   the routers between the two tunnel endpoints, the traffic that a UDP   tunnel generates is a regular UDP flow, and the encapsulator and   decapsulator appear as regular UDP-sending and UDP-receiving   applications.  Because other flows can share the path with one or   more UDP tunnels, congestion control needs to be considered.   Two factors determine whether a UDP tunnel needs to employ specific   congestion control mechanisms: first, whether the payload traffic is   IP-based; and second, whether the tunneling scheme generates UDP   traffic at a volume that corresponds to the volume of payload traffic   carried within the tunnel.   IP-based unicast traffic is generally assumed to be congestion   controlled, i.e., it is assumed that the transport protocols   generating IP-based unicast traffic at the sender already employ   mechanisms that are sufficient to address congestion on the path.   Consequently, a tunnel carrying IP-based unicast traffic shouldEggert, et al.            Best Current Practice                [Page 16]

RFC 8085                  UDP Usage Guidelines                March 2017   already interact appropriately with other traffic sharing the path,   and specific congestion control mechanisms for the tunnel are not   necessary.   However, if the IP traffic in the tunnel is known not to be   congestion controlled, additional measures are RECOMMENDED to limit   the impact of the tunneled traffic on other traffic sharing the path.   For the specific case of a tunnel that carries IP multicast traffic,   seeSection 4.1.   The following guidelines define these possible cases in more detail:   1.  A tunnel generates UDP traffic at a volume that corresponds to       the volume of payload traffic, and the payload traffic is IP       based and congestion controlled.       This is arguably the most common case for Internet tunnels.  In       this case, the UDP tunnel SHOULD NOT employ its own congestion       control mechanism, because congestion losses of tunneled traffic       will already trigger an appropriate congestion response at the       original senders of the tunneled traffic.  A circuit breaker       mechanism may provide benefit by controlling the envelope of the       aggregated traffic.       Note that this guideline is built on the assumption that most       IP-based communication is congestion controlled.  If a UDP tunnel       is used for IP-based traffic that is known to not be congestion       controlled, the next set of guidelines applies.   2.  A tunnel generates UDP traffic at a volume that corresponds to       the volume of payload traffic, and the payload traffic is not       known to be IP based, or is known to be IP based but not       congestion controlled.       This can be the case, for example, when some link-layer protocols       are encapsulated within UDP (but not all link-layer protocols;       some are congestion controlled).  Because it is not known that       congestion losses of tunneled non-IP traffic will trigger an       appropriate congestion response at the senders, the UDP tunnel       SHOULD employ an appropriate congestion control mechanism or       circuit breaker mechanism designed for the traffic it carries.       Because tunnels are usually bulk-transfer applications as far as       the intermediate routers are concerned, the guidelines inSection 3.1.2 apply.   3.  A tunnel generates UDP traffic at a volume that does not       correspond to the volume of payload traffic, independent of       whether the payload traffic is IP based or congestion controlled.Eggert, et al.            Best Current Practice                [Page 17]

RFC 8085                  UDP Usage Guidelines                March 2017       Examples of this class include UDP tunnels that send at a       constant rate, increase their transmission rates under loss, for       example, due to increasing redundancy when Forward Error       Correction is used, or are otherwise unconstrained in their       transmission behavior.  These specialized uses of UDP for       tunneling go beyond the scope of the general guidelines given in       this document.  The implementer of such specialized tunnels       SHOULD carefully consider congestion control in the design of       their tunneling mechanism and SHOULD consider use of a circuit       breaker mechanism.   The type of encapsulated payload might be identified by a UDP port;   identified by an Ethernet Type or IP protocol number.  A tunnel   SHOULD provide mechanisms to restrict the types of flows that may be   carried by the tunnel.  For instance, a UDP tunnel designed to carry   IP needs to filter out non-IP traffic at the ingress.  This is   particularly important when a generic tunnel encapsulation is used   (e.g., one that encapsulates using an EtherType value).  Such tunnels   SHOULD provide a mechanism to restrict the types of traffic that are   allowed to be encapsulated for a given deployment (see   [INT-TUNNELS]).   Designing a tunneling mechanism requires significantly more expertise   than needed for many other UDP applications, because tunnels are   usually intended to be transparent to the endpoints transmitting over   them, so they need to correctly emulate the behavior of an IP link   [INT-TUNNELS], for example:   o  Requirements for tunnels that carry or encapsulate using ECN code      points [RFC6040].   o  Usage of the IP DSCP field by tunnel endpoints [RFC2983].   o  Encapsulation considerations in the design of tunnels [ENCAP].   o  Usage of ICMP messages [INT-TUNNELS].   o  Handling of fragmentation and packet size for tunnels      [INT-TUNNELS].   o  Source port usage for tunnels designed to support equal cost      multipath (ECMP) routing (seeSection 5.1.1).   o  Guidance on the need to protect headers [INT-TUNNELS] and the use      of checksums for IPv6 tunnels (seeSection 3.4.1).   o  Support for operations and maintenance [INT-TUNNELS].Eggert, et al.            Best Current Practice                [Page 18]

RFC 8085                  UDP Usage Guidelines                March 2017   At the same time, the tunneled traffic is application traffic like   any other from the perspective of the networks the tunnel transmits   over.  This document only touches upon the congestion control   considerations for implementing UDP tunnels; a discussion of other   required tunneling behavior is out of scope.3.2.  Message Size Guidelines   IP fragmentation lowers the efficiency and reliability of Internet   communication.  The loss of a single fragment results in the loss of   an entire fragmented packet, because even if all other fragments are   received correctly, the original packet cannot be reassembled and   delivered.  This fundamental issue with fragmentation exists for both   IPv4 and IPv6.   In addition, some network address translators (NATs) and firewalls   drop IP fragments.  The network address translation performed by a   NAT only operates on complete IP packets, and some firewall policies   also require inspection of complete IP packets.  Even with these   being the case, some NATs and firewalls simply do not implement the   necessary reassembly functionality; instead, they choose to drop all   fragments.  Finally, [RFC4963] documents other issues specific to   IPv4 fragmentation.   Due to these issues, an application SHOULD NOT send UDP datagrams   that result in IP packets that exceed the Maximum Transmission Unit   (MTU) along the path to the destination.  Consequently, an   application SHOULD either use the path MTU information provided by   the IP layer or implement Path MTU Discovery (PMTUD) itself [RFC1191]   [RFC1981] [RFC4821] to determine whether the path to a destination   will support its desired message size without fragmentation.   However, the ICMP messages that enable path MTU discovery are being   increasingly filtered by middleboxes (including Firewalls) [RFC4890].   When the path includes a tunnel, some devices acting as a tunnel   ingress discard ICMP messages that originate from network devices   over which the tunnel passes, preventing these from reaching the UDP   endpoint.   Packetization Layer Path MTU Discovery (PLPMTUD) [RFC4821] does not   rely upon network support for ICMP messages and is therefore   considered more robust than standard PMTUD.  It is not susceptible to   "black holing" of ICMP messages.  To operate, PLPMTUD requires   changes to the way the transport is used: both to transmit probe   packets and to account for the loss or success of these probes.  This   not only updates the PMTU algorithm, it also impacts loss recovery,   congestion control, etc.  These updated mechanisms can be implementedEggert, et al.            Best Current Practice                [Page 19]

RFC 8085                  UDP Usage Guidelines                March 2017   within a connection-oriented transport (e.g., TCP, SCTP, DCCP), but   they are not a part of UDP; this type of feedback is not typically   present for unidirectional applications.   Therefore, PLPMTUD places additional design requirements on a UDP   application that wishes to use this method.  This is especially true   for UDP tunnels, because the overhead of sending probe packets needs   to be accounted for and may require adding a congestion control   mechanism to the tunnel (seeSection 3.1.11) as well as complicating   the data path at a tunnel decapsulator.   Applications that do not follow the recommendation to do PMTU/PLPMTUD   discovery SHOULD still avoid sending UDP datagrams that would result   in IP packets that exceed the path MTU.  Because the actual path MTU   is unknown, such applications SHOULD fall back to sending messages   that are shorter than the default effective MTU for sending (EMTU_S   in [RFC1122]).  For IPv4, EMTU_S is the smaller of 576 bytes and the   first-hop MTU [RFC1122].  For IPv6, EMTU_S is 1280 bytes [RFC2460].   The effective PMTU for a directly connected destination (with no   routers on the path) is the configured interface MTU, which could be   less than the maximum link payload size.  Transmission of minimum-   sized UDP datagrams is inefficient over paths that support a larger   PMTU, which is a second reason to implement PMTU discovery.   To determine an appropriate UDP payload size, applications MUST   subtract the size of the IP header (which includes any IPv4 optional   headers or IPv6 extension headers) as well as the length of the UDP   header (8 bytes) from the PMTU size.  This size, known as the Maximum   Segment Size (MSS), can be obtained from the TCP/IP stack [RFC1122].   Applications that do not send messages that exceed the effective PMTU   of IPv4 or IPv6 need not implement any of the above mechanisms.  Note   that the presence of tunnels can cause an additional reduction of the   effective PMTU [INT-TUNNELS], so implementing PMTU discovery may be   beneficial.   Applications that fragment an application-layer message into multiple   UDP datagrams SHOULD perform this fragmentation so that each datagram   can be received independently, and be independently retransmitted in   the case where an application implements its own reliability   mechanisms.Eggert, et al.            Best Current Practice                [Page 20]

RFC 8085                  UDP Usage Guidelines                March 20173.3.  Reliability Guidelines   Application designers are generally aware that UDP does not provide   any reliability, e.g., it does not retransmit any lost packets.   Often, this is a main reason to consider UDP as a transport protocol.   Applications that do require reliable message delivery MUST implement   an appropriate mechanism themselves.   UDP also does not protect against datagram duplication, i.e., an   application may receive multiple copies of the same UDP datagram,   with some duplicates arriving potentially much later than the first.   Application designers SHOULD handle such datagram duplication   gracefully, and they may consequently need to implement mechanisms to   detect duplicates.  Even if UDP datagram reception triggers only   idempotent operations, applications may want to suppress duplicate   datagrams to reduce load.   Applications that require ordered delivery MUST reestablish datagram   ordering themselves.  The Internet can significantly delay some   packets with respect to others, e.g., due to routing transients,   intermittent connectivity, or mobility.  This can cause reordering,   where UDP datagrams arrive at the receiver in an order different from   the transmission order.   Applications that use multiple transport ports need to be robust to   reordering between sessions.  Load-balancing techniques within the   network, such as Equal Cost Multipath (ECMP) forwarding can also   result in a lack of ordering between different transport sessions,   even between the same two network endpoints.   It is important to note that the time by which packets are reordered   or after which duplicates can still arrive can be very large.  Even   more importantly, there is no well-defined upper boundary here.   [RFC793] defines the maximum delay a TCP segment should experience --   the Maximum Segment Lifetime (MSL) -- as 2 minutes.  No other RFC   defines an MSL for other transport protocols or IP itself.  The MSL   value defined for TCP is conservative enough that it SHOULD be used   by other protocols, including UDP.  Therefore, applications SHOULD be   robust to the reception of delayed or duplicate packets that are   received within this 2-minute interval.   Retransmission of lost packets or messages is a common reliability   mechanism.  Such retransmissions can increase network load in   response to congestion, worsening that congestion.  Any application   that uses retransmission is responsible for congestion control of its   retransmissions (as well as the application's original traffic);   hence, it is subject to the Congestion Control guidelines inEggert, et al.            Best Current Practice                [Page 21]

RFC 8085                  UDP Usage Guidelines                March 2017Section 3.1.  Guidance on the appropriate measurement of RTT inSection 3.1.1 also applies for timers used for retransmission packet-   loss detection.   Instead of implementing these relatively complex reliability   mechanisms by itself, an application that requires reliable and   ordered message delivery SHOULD whenever possible choose an IETF   standard transport protocol that provides these features.3.4.  Checksum Guidelines   The UDP header includes an optional, 16-bit one's complement checksum   that provides an integrity check.  These checks are not strong from a   coding or cryptographic perspective and are not designed to detect   physical-layer errors or malicious modification of the datagram   [RFC3819].  Application developers SHOULD implement additional checks   where data integrity is important, e.g., through a Cyclic Redundancy   Check (CRC) or keyed or non-keyed cryptographic hash included with   the data to verify the integrity of an entire object/file sent over   the UDP service.   The UDP checksum provides a statistical guarantee that the payload   was not corrupted in transit.  It also allows the receiver to verify   that it was the intended destination of the packet, because it covers   the IP addresses, port numbers, and protocol number, and it verifies   that the packet is not truncated or padded, because it covers the   size field.  Therefore, it protects an application against receiving   corrupted payload data in place of, or in addition to, the data that   was sent.  More description of the set of checks performed using the   checksum field is provided inSection 3.1 of [RFC6396].   Applications SHOULD enable UDP checksums [RFC1122].  For IPv4,   [RFC768] permits an option to disable their use, by setting a zero   checksum value.  An application is permitted to optionally discard   UDP datagrams with a zero checksum [RFC1122].   When UDP is used over IPv6, the UDP checksum is relied upon to   protect both the IPv6 and UDP headers from corruption (because IPv6   lacks a checksum) and MUST be used as specified in [RFC2460].  Under   specific conditions, a UDP application is allowed to use a zero UDP   zero-checksum mode with a tunnel protocol (seeSection 3.4.1).   Applications that choose to disable UDP checksums MUST NOT make   assumptions regarding the correctness of received data and MUST   behave correctly when a UDP datagram is received that was originally   sent to a different destination or is otherwise corrupted.Eggert, et al.            Best Current Practice                [Page 22]

RFC 8085                  UDP Usage Guidelines                March 20173.4.1.  IPv6 Zero UDP Checksum   [RFC6935] defines a method that enables use of a zero UDP zero-   checksum mode with a tunnel protocol, providing that the method   satisfies the requirements in [RFC6936].  The application MUST   implement mechanisms and/or usage restrictions when enabling this   mode.  This includes defining the scope for usage and measures to   prevent leakage of traffic to other UDP applications (seeAppendix A   andSection 3.6).  These additional design requirements for using a   zero IPv6 UDP checksum are not present for IPv4, since the IPv4   header validates information that is not protected in an IPv6 packet.   Key requirements are:   o  Use of the UDP checksum with IPv6 MUST be the default      configuration for all implementations [RFC6935].  The receiving      endpoint MUST only allow the use of UDP zero-checksum mode for      IPv6 on a UDP destination port that is specifically enabled.   o  An application that supports a checksum different than that in      [RFC2460] MUST comply with all implementation requirements      specified inSection 4 of [RFC6936] and with the usage      requirements specified inSection 5 of [RFC6936].   o  A UDP application MUST check that the source and destination IPv6      addresses are valid for any packets with a UDP zero-checksum and      MUST discard any packet for which this check fails.  To protect      from misdelivery, new encapsulation designs SHOULD include an      integrity check at the transport layer that includes at least the      IPv6 header, the UDP header and the shim header for the      encapsulation, if any [RFC6936].   o  One way to help satisfy the requirements of [RFC6936] may be to      limit the usage of such tunnels, e.g., to constrain traffic to an      operator network, as discussed inSection 3.6.  The encapsulation      defined for MPLS in UDP [RFC7510] chooses this approach.   As in IPv4, IPv6 applications that choose to disable UDP checksums   MUST NOT make assumptions regarding the correctness of received data   and MUST behave correctly when a UDP datagram is received that was   originally sent to a different destination or is otherwise corrupted.   IPv6 datagrams with a zero UDP checksum will not be passed by any   middlebox that validates the checksum based on [RFC2460] or that   updates the UDP checksum field, such as NATs or firewalls.  Changing   this behavior would require such middleboxes to be updated to   correctly handle datagrams with zero UDP checksums.  To ensure end-   to-end robustness, applications that may be deployed in the general   Internet MUST provide a mechanism to safely fall back to using aEggert, et al.            Best Current Practice                [Page 23]

RFC 8085                  UDP Usage Guidelines                March 2017   checksum when a path change occurs that redirects a zero UDP checksum   flow over a path that includes a middlebox that discards IPv6   datagrams with a zero UDP checksum.3.4.2.  UDP-Lite   A special class of applications can derive benefit from having   partially damaged payloads delivered, rather than discarded, when   using paths that include error-prone links.  Such applications can   tolerate payload corruption and MAY choose to use the Lightweight   User Datagram Protocol (UDP-Lite) [RFC3828] variant of UDP instead of   basic UDP.  Applications that choose to use UDP-Lite instead of UDP   should still follow the congestion control and other guidelines   described for use with UDP inSection 3.   UDP-Lite changes the semantics of the UDP "payload length" field to   that of a "checksum coverage length" field.  Otherwise, UDP-Lite is   semantically identical to UDP.  The interface of UDP-Lite differs   from that of UDP by the addition of a single (socket) option that   communicates the checksum coverage length: at the sender, this   specifies the intended checksum coverage, with the remaining   unprotected part of the payload called the "error-insensitive part".   By default, the UDP-Lite checksum coverage extends across the entire   datagram.  If required, an application may dynamically modify this   length value, e.g., to offer greater protection to some messages.   UDP-Lite always verifies that a packet was delivered to the intended   destination, i.e., always verifies the header fields.  Errors in the   insensitive part will not cause a UDP datagram to be discarded by the   destination.  Therefore, applications using UDP-Lite MUST NOT make   assumptions regarding the correctness of the data received in the   insensitive part of the UDP-Lite payload.   A UDP-Lite sender SHOULD select the minimum checksum coverage to   include all sensitive payload information.  For example, applications   that use the Real-Time Protocol (RTP) [RFC3550] will likely want to   protect the RTP header against corruption.  Applications, where   appropriate, MUST also introduce their own appropriate validity   checks for protocol information carried in the insensitive part of   the UDP-Lite payload (e.g., internal CRCs).   A UDP-Lite receiver MUST set a minimum coverage threshold for   incoming packets that is not smaller than the smallest coverage used   by the sender [RFC3828].  The receiver SHOULD select a threshold that   is sufficiently large to block packets with an inappropriately short   coverage field.  This may be a fixed value, or it may be negotiated   by an application.  UDP-Lite does not provide mechanisms to negotiate   the checksum coverage between the sender and receiver.  Therefore,   this needs to be performed by the application.Eggert, et al.            Best Current Practice                [Page 24]

RFC 8085                  UDP Usage Guidelines                March 2017   Applications can still experience packet loss when using UDP-Lite.   The enhancements offered by UDP-Lite rely upon a link being able to   intercept the UDP-Lite header to correctly identify the partial   coverage required.  When tunnels and/or encryption are used, this can   result in UDP-Lite datagrams being treated the same as UDP datagrams,   i.e., result in packet loss.  Use of IP fragmentation can also   prevent special treatment for UDP-Lite datagrams, and this is another   reason why applications SHOULD avoid IP fragmentation (Section 3.2).   UDP-Lite is supported in some endpoint protocol stacks.  Current   support for middlebox traversal using UDP-Lite is poor, because UDP-   Lite uses a different IPv4 protocol number or IPv6 "next header"   value than that used for UDP; therefore, few middleboxes are   currently able to interpret UDP-Lite and take appropriate actions   when forwarding the packet.  This makes UDP-Lite less suited for   applications needing general Internet support, until such time as   UDP-Lite has achieved better support in middleboxes.3.5.  Middlebox Traversal Guidelines   NATs and firewalls are examples of intermediary devices   ("middleboxes") that can exist along an end-to-end path.  A middlebox   typically performs a function that requires it to maintain per-flow   state.  For connection-oriented protocols, such as TCP, middleboxes   snoop and parse the connection-management information, and create and   destroy per-flow state accordingly.  For a connectionless protocol   such as UDP, this approach is not possible.  Consequently,   middleboxes can create per-flow state when they see a packet that --   according to some local criteria -- indicates a new flow, and destroy   the state after some time during which no packets belonging to the   same flow have arrived.   Depending on the specific function that the middlebox performs, this   behavior can introduce a time-dependency that restricts the kinds of   UDP traffic exchanges that will be successful across the middlebox.   For example, NATs and firewalls typically define the partial path on   one side of them to be interior to the domain they serve, whereas the   partial path on their other side is defined to be exterior to that   domain.  Per-flow state is typically created when the first packet   crosses from the interior to the exterior, and while the state is   present, NATs and firewalls will forward return traffic.  Return   traffic that arrives after the per-flow state has timed out is   dropped, as is other traffic that arrives from the exterior.Eggert, et al.            Best Current Practice                [Page 25]

RFC 8085                  UDP Usage Guidelines                March 2017   Many applications that use UDP for communication operate across   middleboxes without needing to employ additional mechanisms.  One   example is the Domain Name System (DNS), which has a strict request-   response communication pattern that typically completes within   seconds.   Other applications may experience communication failures when   middleboxes destroy the per-flow state associated with an application   session during periods when the application does not exchange any UDP   traffic.  Applications SHOULD be able to gracefully handle such   communication failures and implement mechanisms to re-establish   application-layer sessions and state.   For some applications, such as media transmissions, this   re-synchronization is highly undesirable, because it can cause user-   perceivable playback artifacts.  Such specialized applications MAY   send periodic keep-alive messages to attempt to refresh middlebox   state (e.g., [RFC7675]).  It is important to note that keep-alive   messages are not recommended for general use -- they are unnecessary   for many applications and can consume significant amounts of system   and network resources.   An application that needs to employ keep-alive messages to deliver   useful service over UDP in the presence of middleboxes SHOULD NOT   transmit them more frequently than once every 15 seconds and SHOULD   use longer intervals when possible.  No common timeout has been   specified for per-flow UDP state for arbitrary middleboxes.  NATs   require a state timeout of 2 minutes or longer [RFC4787].  However,   empirical evidence suggests that a significant fraction of currently   deployed middleboxes unfortunately use shorter timeouts.  The timeout   of 15 seconds originates with the Interactive Connectivity   Establishment (ICE) protocol [RFC5245].  When an application is   deployed in a controlled environment, the deployer SHOULD investigate   whether the target environment allows applications to use longer   intervals, or whether it offers mechanisms to explicitly control   middlebox state timeout durations, for example, using the Port   Control Protocol (PCP) [RFC6887], Middlebox Communications (MIDCOM)   [RFC3303], Next Steps in Signaling (NSIS) [RFC5973], or Universal   Plug and Play (UPnP) [UPnP].  It is RECOMMENDED that applications   apply slight random variations ("jitter") to the timing of keep-alive   transmissions, to reduce the potential for persistent synchronization   between keep-alive transmissions from different hosts [RFC7675].Eggert, et al.            Best Current Practice                [Page 26]

RFC 8085                  UDP Usage Guidelines                March 2017   Sending keep-alive messages is not a substitute for implementing a   mechanism to recover from broken sessions.  Like all UDP datagrams,   keep-alive messages can be delayed or dropped, causing middlebox   state to time out.  In addition, the congestion control guidelines inSection 3.1 cover all UDP transmissions by an application, including   the transmission of middlebox keep-alive messages.  Congestion   control may thus lead to delays or temporary suspension of keep-alive   transmission.   Keep-alive messages are NOT RECOMMENDED for general use.  They are   unnecessary for many applications and may consume significant   resources.  For example, on battery-powered devices, if an   application needs to maintain connectivity for long periods with   little traffic, the frequency at which keep-alive messages are sent   can become the determining factor that governs power consumption,   depending on the underlying network technology.   Because many middleboxes are designed to require keep-alive messages   for TCP connections at a frequency that is much lower than that   needed for UDP, this difference alone can often be sufficient to   prefer TCP over UDP for these deployments.  On the other hand, there   is anecdotal evidence that suggests that direct communication through   middleboxes, e.g., by using ICE [RFC5245], does succeed less often   with TCP than with UDP.  The trade-offs between different transport   protocols -- especially when it comes to middlebox traversal --   deserve careful analysis.   UDP applications that could be deployed in the Internet need to be   designed understanding that there are many variants of middlebox   behavior, and although UDP is connectionless, middleboxes often   maintain state for each UDP flow.  Using multiple UDP flows can   consume available state space and also can lead to changes in the way   the middlebox handles subsequent packets (either to protect its   internal resources, or to prevent perceived misuse).  The probability   of path failure can increase when applications use multiple UDP flows   in parallel (seeSection 5.1.2 for recommendations on usage of   multiple ports).3.6.  Limited Applicability and Controlled Environments   Two different types of applicability have been identified for the   specification of IETF applications that utilize UDP:   General Internet.  By default, IETF specifications target deployment      on the general Internet.  Experience has shown that successful      protocols developed in one specific context or for a particular      application tend to become used in a wider range of contexts.  For      example, a protocol with an initial deployment within a local areaEggert, et al.            Best Current Practice                [Page 27]

RFC 8085                  UDP Usage Guidelines                March 2017      network may subsequently be used over a virtual network that      traverses the Internet, or in the Internet in general.      Applications designed for general Internet use may experience a      range of network device behaviors and, in particular, should      consider whether applications need to operate over paths that may      include middleboxes.   Controlled Environment.  A protocol/encapsulation/tunnel could be      designed to be used only within a controlled environment.  For      example, an application designed for use by a network operator      might only be deployed within the network of that single network      operator or on networks of an adjacent set of cooperating network      operators.  The application traffic may then be managed to avoid      congestion, rather than relying on built-in mechanisms, which are      required when operating over the general Internet.  Applications      that target a limited applicability use case may be able to take      advantage of specific hardware (e.g., carrier-grade equipment) or      underlying protocol features of the subnetwork over which they are      used.   Specifications addressing a limited applicability use case or a   controlled environment SHOULD identify how, in their restricted   deployment, a level of safety is provided that is equivalent to that   of a protocol designed for operation over the general Internet (e.g.,   a design based on extensive experience with deployments of particular   methods that provide features that cannot be expected in general   Internet equipment and the robustness of the design of MPLS to   corruption of headers both helped justify use of an alternate UDP   integrity check [RFC7510]).   An IETF specification targeting a controlled environment is expected   to provide an applicability statement that restricts the application   traffic to the controlled environment, and it would be expected to   describe how methods can be provided to discourage or prevent escape   of corrupted packets from the environment (for example,Section 5 of   [RFC7510]).4.  Multicast UDP Usage Guidelines   This section complementsSection 3 by providing additional guidelines   that are applicable to multicast and broadcast usage of UDP.   Multicast and broadcast transmission [RFC1112] usually employ the UDP   transport protocol, although they may be used with other transport   protocols (e.g., UDP-Lite).Eggert, et al.            Best Current Practice                [Page 28]

RFC 8085                  UDP Usage Guidelines                March 2017   There are currently two models of multicast delivery: the Any-Source   Multicast (ASM) model as defined in [RFC1112] and the Source-Specific   Multicast (SSM) model as defined in [RFC4607].  ASM group members   will receive all data sent to the group by any source, while SSM   constrains the distribution tree to only one single source.   Specialized classes of applications also use UDP for IP multicast or   broadcast [RFC919].  The design of such specialized applications   requires expertise that goes beyond simple, unicast-specific   guidelines, since these senders may transmit to potentially very many   receivers across potentially very heterogeneous paths at the same   time, which significantly complicates congestion control, flow   control, and reliability mechanisms.   This section provides guidance on multicast and broadcast UDP usage.   Use of broadcast by an application is normally constrained by routers   to the local subnetwork.  However, use of tunneling techniques and   proxies can and does result in some broadcast traffic traversing   Internet paths.  These guidelines therefore also apply to broadcast   traffic.   The IETF has defined a reliable multicast framework [RFC3048] and   several building blocks to aid the designers of multicast   applications, such as [RFC3738] or [RFC4654].   Senders to anycast destinations must be aware that successive   messages sent to the same anycast IP address may be delivered to   different anycast nodes, i.e., arrive at different locations in the   topology.   Most UDP tunnels that carry IP multicast traffic use a tunnel   encapsulation with a unicast destination address, such as Automatic   Multicast Tunneling [RFC7450].  These MUST follow the same   requirements as a tunnel carrying unicast data (seeSection 3.1.11).   There are deployment cases and solutions where the outer header of a   UDP tunnel contains a multicast destination address, such as   [RFC6513].  These cases are primarily deployed in controlled   environments over reserved capacity, often operating within a single   administrative domain, or between two domains over a bilaterally   agreed upon path with reserved capacity, and so congestion control is   OPTIONAL, but circuit breaker techniques are still RECOMMENDED in   order to restore some degree of service should the offered load   exceed the reserved capacity (e.g., due to misconfiguration).Eggert, et al.            Best Current Practice                [Page 29]

RFC 8085                  UDP Usage Guidelines                March 20174.1.  Multicast Congestion Control Guidelines   Unicast congestion-controlled transport mechanisms are often not   applicable to multicast distribution services, or simply do not scale   to large multicast trees, since they require bidirectional   communication and adapt the sending rate to accommodate the network   conditions to a single receiver.  In contrast, multicast distribution   trees may fan out to massive numbers of receivers, which limits the   scalability of an in-band return channel to control the sending rate,   and the one-to-many nature of multicast distribution trees prevents   adapting the rate to the requirements of an individual receiver.  For   this reason, generating TCP-compatible aggregate flow rates for   Internet multicast data, either native or tunneled, is the   responsibility of the application implementing the congestion   control.   Applications using multicast SHOULD provide appropriate congestion   control.  Multicast congestion control needs to be designed using   mechanisms that are robust to the potential heterogeneity of both the   multicast distribution tree and the receivers belonging to a group.   Heterogeneity may manifest itself in some receivers experiencing more   loss that others, higher delay, and/or less ability to respond to   network conditions.  Congestion control is particularly important for   any multicast session where all or part of the multicast distribution   tree spans an access network (e.g., a home gateway).  Two styles of   congestion control have been defined in the RFC Series:   o  Feedback-based congestion control, in which the sender receives      multicast or unicast UDP messages from the receivers allowing it      to assess the level of congestion and then adjust the sender      rate(s) (e.g., [RFC5740],[RFC4654]).  Multicast methods may      operate on longer timescales than for unicast (e.g., due to the      higher group RTT of a heterogeneous group).  A control method      could decide not to reduce the rate of the entire multicast group      in response to a control message received from a single receiver      (e.g., a sender could set a minimum rate and decide to request a      congested receiver to leave the multicast group and could also      decide to distribute content to these congested receivers at a      lower rate using unicast congestion control).   o  Receiver-driven congestion control, which does not require a      receiver to send explicit UDP control messages for congestion      control (e.g., [RFC3738], [RFC5775]).  Instead, the sender      distributes the data across multiple IP multicast groups (e.g.,      using a set of {S,G} channels).  Each receiver determines its own      level of congestion and controls its reception rate using only      multicast join/leave messages sent in the network control plane.      This method scales to arbitrary large groups of receivers.Eggert, et al.            Best Current Practice                [Page 30]

RFC 8085                  UDP Usage Guidelines                March 2017   Any multicast-enabled receiver may attempt to join and receive   traffic from any group.  This may imply the need for rate limits on   individual receivers or the aggregate multicast service.  Note, at   the transport layer, there is no way to prevent a join message   propagating to the next-hop router.   Some classes of multicast applications support applications that can   monitor the user-level quality of the transfer at the receiver.   Applications that can detect a significant reduction in user quality   SHOULD regard this as a congestion signal (e.g., to leave a group   using layered multicast encoding); if not, they SHOULD use this   signal to provide a circuit breaker to terminate the flow by leaving   the multicast group.4.1.1.  Bulk-Transfer Multicast Applications   Applications that perform bulk transmission of data over a multicast   distribution tree, i.e., applications that exchange more than a few   UDP datagrams per RTT, SHOULD implement a method for congestion   control.  The currently RECOMMENDED IETF methods are as follows:   Asynchronous Layered Coding (ALC) [RFC5775], TCP-Friendly Multicast   Congestion Control (TFMCC) [RFC4654], Wave and Equation Based Rate   Control (WEBRC) [RFC3738], NACK-Oriented Reliable Multicast (NORM)   transport protocol [RFC5740], File Delivery over Unidirectional   Transport (FLUTE) [RFC6726], Real Time Protocol/Control Protocol   (RTP/RTCP) [RFC3550].   An application can alternatively implement another congestion control   scheme following the guidelines of [RFC2887] and utilizing the   framework of [RFC3048].  Bulk-transfer applications that choose not   to implement [RFC4654], [RFC5775], [RFC3738], [RFC5740], [RFC6726],   or [RFC3550] SHOULD implement a congestion control scheme that   results in bandwidth use that competes fairly with TCP within an   order of magnitude.Section 2 of [RFC3551] states that multimedia applications SHOULD   monitor the packet-loss rate to ensure that it is within acceptable   parameters.  Packet loss is considered acceptable if a TCP flow   across the same network path under the same network conditions would   achieve an average throughput, measured on a reasonable timescale,   that is not less than that of the UDP flow.  The comparison to TCP   cannot be specified exactly, but is intended as an "order-of-   magnitude" comparison in timescale and throughput.4.1.2.  Low Data-Volume Multicast Applications   All the recommendations inSection 3.1.3 are also applicable to low   data-volume multicast applications.Eggert, et al.            Best Current Practice                [Page 31]

RFC 8085                  UDP Usage Guidelines                March 20174.2.  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 inSection 3 of   [RFC6807].  Consequently, an application SHOULD either use the   effective MTU information provided by the "Population Count   Extensions to Protocol Independent Multicast (PIM)" [RFC6807] or   implement path MTU discovery itself (seeSection 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 aEggert, et al.            Best Current Practice                [Page 32]

RFC 8085                  UDP Usage Guidelines                March 2017   local UDP socket is bound.  Binding a socket also allows configuring   options that affect the UDP or IP layers, for example, use of the UDP   checksum or the IP Timestamp option.  On some stacks, a bound socket   also allows an application to be notified when ICMP error messages   are received for its transmissions [RFC1122].   If a client/server application executes on a host with more than one   IP interface, the application SHOULD send any UDP responses with an   IP source address that matches the IP destination address of the UDP   datagram that carried the request (see[RFC1122], Section 4.1.3.5).   Many middleboxes expect this transmission behavior and drop replies   that are sent from a different IP address, as explained inSection 3.5.   A UDP receiver can receive a valid UDP datagram with a zero-length   payload.  Note that this is different from a return value of zero   from a read() socket call, which for TCP indicates the end of the   connection.   UDP provides no flow-control, i.e., the sender at any given time does   not know whether the receiver is able to handle incoming   transmissions.  This is another reason why UDP-based applications   need to be robust in the presence of packet loss.  This loss can also   occur within the sending host, when an application sends data faster   than the line rate of the outbound network interface.  It can also   occur at the destination, where receive calls fail to return all the   data that was sent when the application issues them too infrequently   (i.e., such that the receive buffer overflows).  Robust flow control   mechanisms are difficult to implement, which is why applications that   need this functionality SHOULD consider using a full-featured   transport protocol such as TCP.   When an application closes a TCP, SCTP, or DCCP socket, the transport   protocol on the receiving host is required to maintain TIME-WAIT   state.  This prevents delayed packets from the closed connection   instance from being mistakenly associated with a later connection   instance that happens to reuse the same IP address and port pairs.   The UDP protocol does not implement such a mechanism.  Therefore,   UDP-based applications need to be robust to reordering and delay.   One application may close a socket or terminate, followed in time by   another application receiving on the same port.  This later   application may then receive packets intended for the first   application that were delayed in the network.Eggert, et al.            Best Current Practice                [Page 33]

RFC 8085                  UDP Usage Guidelines                March 20175.1.  Using UDP Ports   The rules and procedures for the management of the "Service Name and   Transport Protocol Port Number Registry" are specified in [RFC6335].   Recommendations for use of UDP ports are provided in [RFC7605].   A UDP sender SHOULD NOT use a source port value of zero.  A source   port number that cannot be easily determined from the address or   payload type provides protection at the receiver from data injection   attacks by off-path devices.  A UDP receiver SHOULD NOT bind to port   zero.   Applications SHOULD implement receiver port and address checks at the   application layer or explicitly request that the operating system   filter the received packets to prevent receiving packets with an   arbitrary port.  This measure is designed to provide additional   protection from data injection attacks from an off-path source (where   the port values may not be known).   Applications SHOULD provide a check that protects from off-path data   injection, avoiding an application receiving packets that were   created by an unauthorized third party.  TCP stacks commonly use a   randomized source port to provide this protection [RFC6056]; UDP   applications should follow the same technique.  Middleboxes and end   systems often make assumptions about the system ports or user ports;   hence, it is recommended to use randomized ports in the Dynamic and/   or Private Port range.  Setting a "randomized" source port also   provides greater assurance that reported ICMP errors originate from   network systems on the path used by a particular flow.  Some UDP   applications choose to use a predetermined value for the source port   (including some multicast applications), these applications need to   therefore employ a different technique.  Protection from off-path   data attacks can also be provided by randomizing the initial value of   another protocol field within the datagram payload, and checking the   validity of this field at the receiver (e.g., RTP has random initial   sequence number and random media timestamp offsets [RFC3550]).   When using multicast, IP routers perform a reverse-path forwarding   (RPF) check for each multicast packet.  This provides protection from   off-path data injection, restricting opportunities to forge a   packet's source address.  When a receiver joins a multicast group and   filters based on the source address the filter verifies the sender's   IP address.  This is always the case when using an SSM {S,G} channel.Eggert, et al.            Best Current Practice                [Page 34]

RFC 8085                  UDP Usage Guidelines                March 20175.1.1.  Usage of UDP for Source Port Entropy and the IPv6 Flow Label   Some applications use the UDP datagram header as a source of entropy   for network devices that implement ECMP [RFC6438].  A UDP tunnel   application targeting this usage encapsulates an inner packet using   UDP, where the UDP source port value forms a part of the entropy that   can be used to balance forwarding of network traffic by the devices   that use ECMP.  A sending tunnel endpoint selects a source port value   in the UDP datagram header that is computed from the inner flow   information (e.g., the encapsulated packet headers).  To provide   sufficient entropy, the sending tunnel endpoint maps the encapsulated   traffic to one of a range of UDP source values.  The value SHOULD be   within the ephemeral port range, i.e., 49152 to 65535, where the high   order two bits of the port are set to one.  The available source port   entropy of 14 bits (using the ephemeral port range) plus the outer IP   addresses seems sufficient for entropy for most ECMP applications   [ENCAP].   To avoid reordering within an IP flow, the same UDP source port value   SHOULD be used for all packets assigned to an encapsulated flow   (e.g., using a hash of the relevant headers).  The entropy mapping   for a flow MAY change over the lifetime of the encapsulated flow   [ENCAP].  For instance, this could be changed as a Denial of Service   (DOS) mitigation, or as a means to effect routing through the ECMP   network.  However, the source port selected for a flow SHOULD NOT   change more than once in every thirty seconds (e.g., as in   [RFC8086]).   The use of the source port field for entropy has several side effects   that need to be considered, including:   o  It can increase the probability of misdelivery of corrupted      packets, which increases the need for checksum computation or an      equivalent mechanism to protect other UDP applications from      misdelivery errorsSection 3.4.   o  It is expected to reduce the probability of successful middlebox      traversalSection 3.5.  This use of the source port field will      often not be suitable for applications targeting deployment in the      general Internet.   o  It can prevent the field being usable to protect from off-path      attacks (described inSection 5.1).  Designers therefore need to      consider other mechanisms to provide equivalent protection (e.g.,      to restrict use to a controlled environment[RFC7510]      Section 3.6).Eggert, et al.            Best Current Practice                [Page 35]

RFC 8085                  UDP Usage Guidelines                March 2017   The UDP source port number field has also been leveraged to produce   entropy with IPv6.  However, in the case of IPv6, the "flow label"   [RFC6437] may also alternatively be used to provide entropy for load   balancing [RFC6438].  This use of the flow label for load balancing   is consistent with the definition of the field, although further   clarity was needed to ensure the field can be consistently used for   this purpose.  Therefore, an updated IPv6 flow label [RFC6437] and   ECMP routing [RFC6438] usage was specified.   To ensure future opportunities to use the flow label, UDP   applications SHOULD set the flow label field, even when an entropy   value is also set in the source port field (e.g., An IPv6 tunnel   endpoint could copy the source port flow entropy value to the IPv6   flow label field [RFC8086]).  Router vendors are encouraged to start   using the IPv6 flow label as a part of the flow hash, providing   support for IP-level ECMP without requiring use of UDP.  The end-to-   end use of flow labels for load balancing is a long-term solution.   Even if the usage of the flow label has been clarified, there will be   a transition time before a significant proportion of endpoints start   to assign a good quality flow label to the flows that they originate.   The use of load balancing using the transport header fields will   likely continue until widespread deployment is finally achieved.5.1.2.  Applications Using Multiple UDP Ports   A single application may exchange several types of data.  In some   cases, this may require multiple UDP flows (e.g., multiple sets of   flows, identified by different five-tuples).  [RFC6335] recommends   application developers not to apply to IANA to be assigned multiple   well-known ports (user or system).  It does not discuss the   implications of using multiple flows with the same well-known port or   pairs of dynamic ports (e.g., identified by a service name or   signaling protocol).   Use of multiple flows can affect the network in several ways:   o  Starting a series of successive connections can increase the      number of state bindings in middleboxes (e.g., NAPT or Firewall)      along the network path.  UDP-based middlebox traversal usually      relies on timeouts to remove old state, since middleboxes are      unaware when a particular flow ceases to be used by an      application.   o  Using several flows at the same time may result in seeing      different network characteristics for each flow.  It cannot be      assumed both follow the same path (e.g., when ECMP is used,      traffic is intentionally hashed onto different parallel paths      based on the port numbers).Eggert, et al.            Best Current Practice                [Page 36]

RFC 8085                  UDP Usage Guidelines                March 2017   o  Using several flows can also increase the occupancy of a binding      or lookup table in a middlebox (e.g., NAPT or Firewall), which may      cause the device to change the way it manages the flow state.   o  Further, using excessive numbers of flows can degrade the ability      of a unicast congestion control to react to congestion events,      unless the congestion state is shared between all flows in a      session.  A receiver-driven multicast congestion control requires      the sending application to distribute its data over a set of IP      multicast groups, each receiver is therefore expected to receive      data from a modest number of simultaneously active UDP ports.   Therefore, applications MUST NOT assume consistent behavior of   middleboxes when multiple UDP flows are used; many devices respond   differently as the number of used ports increases.  Using multiple   flows with different QoS requirements requires applications to verify   that the expected performance is achieved using each individual flow   (five-tuple), seeSection 3.1.9.5.2.  ICMP Guidelines   Applications can utilize information about ICMP error messages that   the UDP layer passes up for a variety of purposes [RFC1122].   Applications SHOULD appropriately validate the payload of ICMP   messages to ensure these are received in response to transmitted   traffic (i.e., a reported error condition that corresponds to a UDP   datagram actually sent by the application).  This requires context,   such as local state about communication instances to each   destination, that although readily available in connection-oriented   transport protocols is not always maintained by UDP-based   applications.  Note that not all platforms have the necessary APIs to   support this validation, and some platforms already perform this   validation internally before passing ICMP information to the   application.   Any application response to ICMP error messages SHOULD be robust to   temporary routing failures (sometimes called "soft errors"), e.g.,   transient ICMP "unreachable" messages ought to not normally cause a   communication abort.   ICMP messages are being increasingly filtered by middleboxes.  A UDP   application therefore SHOULD NOT rely on their delivery for correct   and safe operation.Eggert, et al.            Best Current Practice                [Page 37]

RFC 8085                  UDP Usage Guidelines                March 20176.  Security Considerations   UDP does not provide communications security.  Applications that need   to protect their communications against eavesdropping, tampering, or   message forgery SHOULD employ end-to-end security services provided   by other IETF protocols.   UDP applications SHOULD provide protection from off-path data   injection attacks using a randomized source port or equivalent   technique (seeSection 5.1).   Applications that respond to short requests with potentially large   responses are a potential vector for amplification attacks, and   SHOULD take steps to minimize their potential for being abused as   part of a DoS attack.  That could mean authenticating the sender   before responding; noting that the source IP address of a request is   not a useful authenticator, because it can easily be spoofed.  Or it   may mean otherwise limiting the cases where short unauthenticated   requests produce large responses.  Applications MAY also want to   offer ways to limit the number of requests they respond to in a time   interval, in order to cap the bandwidth they consume.   One option for securing UDP communications is with IPsec [RFC4301],   which can provide authentication for flows of IP packets through the   Authentication Header (AH) [RFC4302] and encryption and/or   authentication through the Encapsulating Security Payload (ESP)   [RFC4303].  Applications use the Internet Key Exchange (IKE)   [RFC7296] to configure IPsec for their sessions.  Depending on how   IPsec is configured for a flow, it can authenticate or encrypt the   UDP headers as well as UDP payloads.  If an application only requires   authentication, ESP with no encryption but with authentication is   often a better option than AH, because ESP can operate across   middleboxes.  An application that uses IPsec requires the support of   an operating system that implements the IPsec protocol suite, and the   network path must permit IKE and IPsec traffic.  This may become more   common with IPv6 deployments [RFC6092].   Although it is possible to use IPsec to secure UDP communications,   not all operating systems support IPsec or allow applications to   easily configure it for their flows.  A second option for securing   UDP communications is through Datagram Transport Layer Security   (DTLS) [RFC6347][RFC7525].  DTLS provides communication privacy by   encrypting UDP payloads.  It does not protect the UDP headers.   Applications can implement DTLS without relying on support from the   operating system.Eggert, et al.            Best Current Practice                [Page 38]

RFC 8085                  UDP Usage Guidelines                March 2017   Many other options for authenticating or encrypting UDP payloads   exist.  For example, the GSS-API security framework [RFC2743] or   Cryptographic Message Syntax (CMS) [RFC5652] could be used to protect   UDP payloads.  There exist a number of security options for RTP   [RFC3550] over UDP, especially to accomplish key-management, see   [RFC7201].  These options covers many usages, including point-to-   point, centralized group communication as well as multicast.  In some   applications, a better solution is to protect larger stand-alone   objects, such as files or messages, instead of individual UDP   payloads.  In these situations, CMS [RFC5652], S/MIME [RFC5751] or   OpenPGP [RFC4880] could be used.  In addition, there are many   non-IETF protocols in this area.   Like congestion control mechanisms, security mechanisms are difficult   to design and implement correctly.  It is hence RECOMMENDED that   applications employ well-known standard security mechanisms such as   DTLS or IPsec, rather than inventing their own.   The Generalized TTL Security Mechanism (GTSM) [RFC5082] may be used   with UDP applications when the intended endpoint is on the same link   as the sender.  This lightweight mechanism allows a receiver to   filter unwanted packets.   In terms of congestion control, [RFC2309] and [RFC2914] discuss the   dangers of congestion-unresponsive flows to the Internet.  [RFC8084]   describes methods that can be used to set a performance envelope that   can assist in preventing congestion collapse in the absence of   congestion control or when the congestion control fails to react to   congestion events.  This document provides guidelines to designers of   UDP-based applications to congestion-control their transmissions, and   does not raise any additional security concerns.   Some network operators have experienced surges of UDP attack traffic   that are multiple orders of magnitude above the baseline traffic rate   for UDP.  This can motivate operators to limit the data rate or   packet rate of UDP traffic.  This may in turn limit the throughput   that an application can achieve using UDP and could also result in   higher packet loss for UDP traffic that would not be experienced if   other transport protocols had been used.   A UDP application with a long-lived association between the sender   and receiver, ought to be designed so that the sender periodically   checks that the receiver still wants ("consents") to receive traffic   and need to be designed to stop if there is no explicit confirmation   of this [RFC7675].  Applications that require communications in two   directions to implement protocol functions (such as reliability orEggert, et al.            Best Current Practice                [Page 39]

RFC 8085                  UDP Usage Guidelines                March 2017   congestion control) will need to independently check both directions   of communication, and may have to exchange keep-alive messages to   traverse middleboxes (seeSection 3.5).7.  Summary   This section summarizes the key guidelines made in Sections3 -6 in   a tabular format (Table 1) for easy referencing.   +---------------------------------------------------------+---------+   | Recommendation                                          | Section |   +---------------------------------------------------------+---------+   | MUST tolerate a wide range of Internet path conditions  | 3       |   | SHOULD use a full-featured transport (e.g., TCP)        |         |   |                                                         |         |   | SHOULD control rate of transmission                     | 3.1     |   | SHOULD perform congestion control over all traffic      |         |   |                                                         |         |   | for bulk transfers,                                     | 3.1.2   |   | SHOULD consider implementing TFRC                       |         |   | else, SHOULD in other ways use bandwidth similar to TCP |         |   |                                                         |         |   | for non-bulk transfers,                                 | 3.1.3   |   | SHOULD measure RTT and transmit max. 1 datagram/RTT     | 3.1.1   |   | else, SHOULD send at most 1 datagram every 3 seconds    |         |   | SHOULD back-off retransmission timers following loss    |         |   |                                                         |         |   | SHOULD provide mechanisms to regulate the bursts of     | 3.1.6   |   | transmission                                            |         |   |                                                         |         |   | MAY implement ECN; a specific set of application        | 3.1.7   |   | mechanisms are REQUIRED if ECN is used.                 |         |   |                                                         |         |   | for DiffServ, SHOULD NOT rely on implementation of PHBs | 3.1.8   |   |                                                         |         |   | for QoS-enabled paths, MAY choose not to use CC         | 3.1.9   |   |                                                         |         |   | SHOULD NOT rely solely on QoS for their capacity        | 3.1.10  |   | non-CC controlled flows SHOULD implement a transport    |         |   | circuit breaker                                         |         |   | MAY implement a circuit breaker for other applications  |         |   |                                                         |         |   | for tunnels carrying IP traffic,                        | 3.1.11  |   | SHOULD NOT perform congestion control                   |         |   | MUST correctly process the IP ECN field                 |         |   |                                                         |         |Eggert, et al.            Best Current Practice                [Page 40]

RFC 8085                  UDP Usage Guidelines                March 2017   | for non-IP tunnels or rate not determined by traffic,   |         |   | SHOULD perform CC or use circuit breaker                | 3.1.11  |   | SHOULD restrict types of traffic transported by the     |         |   | tunnel                                                  |         |   |                                                         |         |   | SHOULD NOT send datagrams that exceed the PMTU, i.e.,   | 3.2     |   | SHOULD discover PMTU or send datagrams < minimum PMTU;  |         |   | Specific application mechanisms are REQUIRED if PLPMTUD |         |   | is used.                                                |         |   |                                                         |         |   | SHOULD handle datagram loss, duplication, reordering    | 3.3     |   | SHOULD be robust to delivery delays up to 2 minutes     |         |   |                                                         |         |   | SHOULD enable IPv4 UDP checksum                         | 3.4     |   | SHOULD enable IPv6 UDP checksum; Specific application   | 3.4.1   |   | mechanisms are REQUIRED if a zero IPv6 UDP checksum is  |         |   | used.                                                   |         |   |                                                         |         |   | SHOULD provide protection from off-path attacks         | 5.1     |   | else, MAY use UDP-Lite with suitable checksum coverage  | 3.4.2   |   |                                                         |         |   | SHOULD NOT always send middlebox keep-alive messages    | 3.5     |   | MAY use keep-alives when needed (min. interval 15 sec)  |         |   |                                                         |         |   | Applications specified for use in limited use (or       | 3.6     |   | controlled environments) SHOULD identify equivalent     |         |   | mechanisms and describe their use case.                 |         |   |                                                         |         |   | Bulk-multicast apps SHOULD implement congestion control | 4.1.1   |   |                                                         |         |   | Low volume multicast apps SHOULD implement congestion   | 4.1.2   |   | control                                                 |         |   |                                                         |         |   | Multicast apps SHOULD use a safe PMTU                   | 4.2     |   |                                                         |         |   | SHOULD avoid using multiple ports                       | 5.1.2   |   | MUST check received IP source address                   |         |   |                                                         |         |   | SHOULD validate payload in ICMP messages                | 5.2     |   |                                                         |         |   | SHOULD use a randomized source port or equivalent       | 6       |   | technique, and, for client/server applications, SHOULD  |         |   | send responses from source address matching request     |         |   | 5.1                                                     |         |   | SHOULD use standard IETF security protocols when needed | 6       |   +---------------------------------------------------------+---------+                    Table 1: Summary of RecommendationsEggert, et al.            Best Current Practice                [Page 41]

RFC 8085                  UDP Usage Guidelines                March 20178.  References8.1.  Normative References   [RFC768]   Postel, J., "User Datagram Protocol", STD 6,RFC 768,              DOI 10.17487/RFC0768, August 1980,              <http://www.rfc-editor.org/info/rfc768>.   [RFC793]   Postel, J., "Transmission Control Protocol", STD 7,RFC 793, DOI 10.17487/RFC0793, September 1981,              <http://www.rfc-editor.org/info/rfc793>.   [RFC1122]  Braden, R., Ed., "Requirements for Internet Hosts -              Communication Layers", STD 3,RFC 1122,              DOI 10.17487/RFC1122, October 1989,              <http://www.rfc-editor.org/info/rfc1122>.   [RFC1191]  Mogul, J. and S. Deering, "Path MTU discovery",RFC 1191,              DOI 10.17487/RFC1191, November 1990,              <http://www.rfc-editor.org/info/rfc1191>.   [RFC1981]  McCann, J., Deering, S., and J. Mogul, "Path MTU Discovery              for IP version 6",RFC 1981, DOI 10.17487/RFC1981, August              1996, <http://www.rfc-editor.org/info/rfc1981>.   [RFC2119]  Bradner, S., "Key words for use in RFCs to Indicate              Requirement Levels",BCP 14,RFC 2119,              DOI 10.17487/RFC2119, March 1997,              <http://www.rfc-editor.org/info/rfc2119>.   [RFC2460]  Deering, S. and R. Hinden, "Internet Protocol, Version 6              (IPv6) Specification",RFC 2460, DOI 10.17487/RFC2460,              December 1998, <http://www.rfc-editor.org/info/rfc2460>.   [RFC2914]  Floyd, S., "Congestion Control Principles",BCP 41,RFC 2914, DOI 10.17487/RFC2914, September 2000,              <http://www.rfc-editor.org/info/rfc2914>.   [RFC3828]  Larzon, L-A., Degermark, M., Pink, S., Jonsson, L-E., Ed.,              and G. Fairhurst, Ed., "The Lightweight User Datagram              Protocol (UDP-Lite)",RFC 3828, DOI 10.17487/RFC3828, July              2004, <http://www.rfc-editor.org/info/rfc3828>.   [RFC4787]  Audet, F., Ed. and C. Jennings, "Network Address              Translation (NAT) Behavioral Requirements for Unicast              UDP",BCP 127,RFC 4787, DOI 10.17487/RFC4787, January              2007, <http://www.rfc-editor.org/info/rfc4787>.Eggert, et al.            Best Current Practice                [Page 42]

RFC 8085                  UDP Usage Guidelines                March 2017   [RFC4821]  Mathis, M. and J. Heffner, "Packetization Layer Path MTU              Discovery",RFC 4821, DOI 10.17487/RFC4821, March 2007,              <http://www.rfc-editor.org/info/rfc4821>.   [RFC5348]  Floyd, S., Handley, M., Padhye, J., and J. Widmer, "TCP              Friendly Rate Control (TFRC): Protocol Specification",RFC 5348, DOI 10.17487/RFC5348, September 2008,              <http://www.rfc-editor.org/info/rfc5348>.   [RFC5405]  Eggert, L. and G. Fairhurst, "Unicast UDP Usage Guidelines              for Application Designers",BCP 145,RFC 5405,              DOI 10.17487/RFC5405, November 2008,              <http://www.rfc-editor.org/info/rfc5405>.   [RFC6040]  Briscoe, B., "Tunnelling of Explicit Congestion              Notification",RFC 6040, DOI 10.17487/RFC6040, November              2010, <http://www.rfc-editor.org/info/rfc6040>.   [RFC6298]  Paxson, V., Allman, M., Chu, J., and M. Sargent,              "Computing TCP's Retransmission Timer",RFC 6298,              DOI 10.17487/RFC6298, June 2011,              <http://www.rfc-editor.org/info/rfc6298>.   [RFC8084]  Fairhurst, G., "Network Transport Circuit Breakers",BCP 208,RFC 8084, DOI 10.17487/RFC8084, March 2017,              <http://www.rfc-editor.org/info/rfc8084>.8.2.  Informative References   [ALLMAN]   Allman, M. and E. Blanton, "Notes on burst mitigation for              transport protocols", March 2005.   [BEHAVE-APP]              Ford, B., "Application Design Guidelines for Traversal              through Network Address Translators", Work in Progress,draft-ford-behave-app-05, March 2007.   [ENCAP]    Nordmark, E., Ed., Tian, A., Gross, J., Hudson, J.,              Kreeger, L., Garg, P., Thaler, P., and T. Herbert,              "Encapsulation Considerations", Work in Progress,draft-ietf-rtgwg-dt-encap-02, October 2016.   [FABER]    Faber, T., Touch, J., and W. Yue, "The TIME-WAIT State in              TCP and Its Effect on Busy Servers", Proc. IEEE Infocom,              March 1999.Eggert, et al.            Best Current Practice                [Page 43]

RFC 8085                  UDP Usage Guidelines                March 2017   [INT-TUNNELS]              Touch, J. and W. Townsley, "IP Tunnels in the Internet              Architecture", Work in Progress,draft-ietf-intarea-tunnels-03, July 2016.   [POSIX]    IEEE Std. 1003.1-2001, , "Standard for Information              Technology - Portable Operating System Interface (POSIX)",              Open Group Technical Standard: Base Specifications Issue              6, ISO/IEC 9945:2002, December 2001.   [RFC919]   Mogul, J., "Broadcasting Internet Datagrams", STD 5,RFC 919, DOI 10.17487/RFC0919, October 1984,              <http://www.rfc-editor.org/info/rfc919>.   [RFC1112]  Deering, S., "Host extensions for IP multicasting", STD 5,RFC 1112, DOI 10.17487/RFC1112, August 1989,              <http://www.rfc-editor.org/info/rfc1112>.   [RFC1536]  Kumar, A., Postel, J., Neuman, C., Danzig, P., and S.              Miller, "Common DNS Implementation Errors and Suggested              Fixes",RFC 1536, DOI 10.17487/RFC1536, October 1993,              <http://www.rfc-editor.org/info/rfc1536>.   [RFC1546]  Partridge, C., Mendez, T., and W. Milliken, "Host              Anycasting Service",RFC 1546, DOI 10.17487/RFC1546,              November 1993, <http://www.rfc-editor.org/info/rfc1546>.   [RFC2309]  Braden, B., Clark, D., Crowcroft, J., Davie, B., Deering,              S., Estrin, D., Floyd, S., Jacobson, V., Minshall, G.,              Partridge, C., Peterson, L., Ramakrishnan, K., Shenker,              S., Wroclawski, J., and L. Zhang, "Recommendations on              Queue Management and Congestion Avoidance in the              Internet",RFC 2309, DOI 10.17487/RFC2309, April 1998,              <http://www.rfc-editor.org/info/rfc2309>.   [RFC2475]  Blake, S., Black, D., Carlson, M., Davies, E., Wang, Z.,              and W. Weiss, "An Architecture for Differentiated              Services",RFC 2475, DOI 10.17487/RFC2475, December 1998,              <http://www.rfc-editor.org/info/rfc2475>.   [RFC2675]  Borman, D., Deering, S., and R. Hinden, "IPv6 Jumbograms",RFC 2675, DOI 10.17487/RFC2675, August 1999,              <http://www.rfc-editor.org/info/rfc2675>.   [RFC2743]  Linn, J., "Generic Security Service Application Program              Interface Version 2, Update 1",RFC 2743,              DOI 10.17487/RFC2743, January 2000,              <http://www.rfc-editor.org/info/rfc2743>.Eggert, et al.            Best Current Practice                [Page 44]

RFC 8085                  UDP Usage Guidelines                March 2017   [RFC2887]  Handley, M., Floyd, S., Whetten, B., Kermode, R.,              Vicisano, L., and M. Luby, "The Reliable Multicast Design              Space for Bulk Data Transfer",RFC 2887,              DOI 10.17487/RFC2887, August 2000,              <http://www.rfc-editor.org/info/rfc2887>.   [RFC2983]  Black, D., "Differentiated Services and Tunnels",RFC 2983, DOI 10.17487/RFC2983, October 2000,              <http://www.rfc-editor.org/info/rfc2983>.   [RFC3048]  Whetten, B., Vicisano, L., Kermode, R., Handley, M.,              Floyd, S., and M. Luby, "Reliable Multicast Transport              Building Blocks for One-to-Many Bulk-Data Transfer",RFC 3048, DOI 10.17487/RFC3048, January 2001,              <http://www.rfc-editor.org/info/rfc3048>.   [RFC3124]  Balakrishnan, H. and S. Seshan, "The Congestion Manager",RFC 3124, DOI 10.17487/RFC3124, June 2001,              <http://www.rfc-editor.org/info/rfc3124>.   [RFC3168]  Ramakrishnan, K., Floyd, S., and D. Black, "The Addition              of Explicit Congestion Notification (ECN) to IP",RFC 3168, DOI 10.17487/RFC3168, September 2001,              <http://www.rfc-editor.org/info/rfc3168>.   [RFC3261]  Rosenberg, J., Schulzrinne, H., Camarillo, G., Johnston,              A., Peterson, J., Sparks, R., Handley, M., and E.              Schooler, "SIP: Session Initiation Protocol",RFC 3261,              DOI 10.17487/RFC3261, June 2002,              <http://www.rfc-editor.org/info/rfc3261>.   [RFC3303]  Srisuresh, P., Kuthan, J., Rosenberg, J., Molitor, A., and              A. Rayhan, "Middlebox communication architecture and              framework",RFC 3303, DOI 10.17487/RFC3303, August 2002,              <http://www.rfc-editor.org/info/rfc3303>.   [RFC3493]  Gilligan, R., Thomson, S., Bound, J., McCann, J., and W.              Stevens, "Basic Socket Interface Extensions for IPv6",RFC 3493, DOI 10.17487/RFC3493, February 2003,              <http://www.rfc-editor.org/info/rfc3493>.   [RFC3550]  Schulzrinne, H., Casner, S., Frederick, R., and V.              Jacobson, "RTP: A Transport Protocol for Real-Time              Applications", STD 64,RFC 3550, DOI 10.17487/RFC3550,              July 2003, <http://www.rfc-editor.org/info/rfc3550>.Eggert, et al.            Best Current Practice                [Page 45]

RFC 8085                  UDP Usage Guidelines                March 2017   [RFC3551]  Schulzrinne, H. and S. Casner, "RTP Profile for Audio and              Video Conferences with Minimal Control", STD 65,RFC 3551,              DOI 10.17487/RFC3551, July 2003,              <http://www.rfc-editor.org/info/rfc3551>.   [RFC3738]  Luby, M. and V. Goyal, "Wave and Equation Based Rate              Control (WEBRC) Building Block",RFC 3738,              DOI 10.17487/RFC3738, April 2004,              <http://www.rfc-editor.org/info/rfc3738>.   [RFC3758]  Stewart, R., Ramalho, M., Xie, Q., Tuexen, M., and P.              Conrad, "Stream Control Transmission Protocol (SCTP)              Partial Reliability Extension",RFC 3758,              DOI 10.17487/RFC3758, May 2004,              <http://www.rfc-editor.org/info/rfc3758>.   [RFC3819]  Karn, P., Ed., Bormann, C., Fairhurst, G., Grossman, D.,              Ludwig, R., Mahdavi, J., Montenegro, G., Touch, J., and L.              Wood, "Advice for Internet Subnetwork Designers",BCP 89,RFC 3819, DOI 10.17487/RFC3819, July 2004,              <http://www.rfc-editor.org/info/rfc3819>.   [RFC4301]  Kent, S. and K. Seo, "Security Architecture for the              Internet Protocol",RFC 4301, DOI 10.17487/RFC4301,              December 2005, <http://www.rfc-editor.org/info/rfc4301>.   [RFC4302]  Kent, S., "IP Authentication Header",RFC 4302,              DOI 10.17487/RFC4302, December 2005,              <http://www.rfc-editor.org/info/rfc4302>.   [RFC4303]  Kent, S., "IP Encapsulating Security Payload (ESP)",RFC 4303, DOI 10.17487/RFC4303, December 2005,              <http://www.rfc-editor.org/info/rfc4303>.   [RFC4340]  Kohler, E., Handley, M., and S. Floyd, "Datagram              Congestion Control Protocol (DCCP)",RFC 4340,              DOI 10.17487/RFC4340, March 2006,              <http://www.rfc-editor.org/info/rfc4340>.   [RFC4341]  Floyd, S. and E. Kohler, "Profile for Datagram Congestion              Control Protocol (DCCP) Congestion Control ID 2: TCP-like              Congestion Control",RFC 4341, DOI 10.17487/RFC4341, March              2006, <http://www.rfc-editor.org/info/rfc4341>.Eggert, et al.            Best Current Practice                [Page 46]

RFC 8085                  UDP Usage Guidelines                March 2017   [RFC4342]  Floyd, S., Kohler, E., and J. Padhye, "Profile for              Datagram Congestion Control Protocol (DCCP) Congestion              Control ID 3: TCP-Friendly Rate Control (TFRC)",RFC 4342,              DOI 10.17487/RFC4342, March 2006,              <http://www.rfc-editor.org/info/rfc4342>.   [RFC4380]  Huitema, C., "Teredo: Tunneling IPv6 over UDP through              Network Address Translations (NATs)",RFC 4380,              DOI 10.17487/RFC4380, February 2006,              <http://www.rfc-editor.org/info/rfc4380>.   [RFC4607]  Holbrook, H. and B. Cain, "Source-Specific Multicast for              IP",RFC 4607, DOI 10.17487/RFC4607, August 2006,              <http://www.rfc-editor.org/info/rfc4607>.   [RFC4654]  Widmer, J. and M. Handley, "TCP-Friendly Multicast              Congestion Control (TFMCC): Protocol Specification",RFC 4654, DOI 10.17487/RFC4654, August 2006,              <http://www.rfc-editor.org/info/rfc4654>.   [RFC4880]  Callas, J., Donnerhacke, L., Finney, H., Shaw, D., and R.              Thayer, "OpenPGP Message Format",RFC 4880,              DOI 10.17487/RFC4880, November 2007,              <http://www.rfc-editor.org/info/rfc4880>.   [RFC4890]  Davies, E. and J. Mohacsi, "Recommendations for Filtering              ICMPv6 Messages in Firewalls",RFC 4890,              DOI 10.17487/RFC4890, May 2007,              <http://www.rfc-editor.org/info/rfc4890>.   [RFC4960]  Stewart, R., Ed., "Stream Control Transmission Protocol",RFC 4960, DOI 10.17487/RFC4960, September 2007,              <http://www.rfc-editor.org/info/rfc4960>.   [RFC4963]  Heffner, J., Mathis, M., and B. Chandler, "IPv4 Reassembly              Errors at High Data Rates",RFC 4963,              DOI 10.17487/RFC4963, July 2007,              <http://www.rfc-editor.org/info/rfc4963>.   [RFC4987]  Eddy, W., "TCP SYN Flooding Attacks and Common              Mitigations",RFC 4987, DOI 10.17487/RFC4987, August 2007,              <http://www.rfc-editor.org/info/rfc4987>.   [RFC5082]  Gill, V., Heasley, J., Meyer, D., Savola, P., Ed., and C.              Pignataro, "The Generalized TTL Security Mechanism              (GTSM)",RFC 5082, DOI 10.17487/RFC5082, October 2007,              <http://www.rfc-editor.org/info/rfc5082>.Eggert, et al.            Best Current Practice                [Page 47]

RFC 8085                  UDP Usage Guidelines                March 2017   [RFC5245]  Rosenberg, J., "Interactive Connectivity Establishment              (ICE): A Protocol for Network Address Translator (NAT)              Traversal for Offer/Answer Protocols",RFC 5245,              DOI 10.17487/RFC5245, April 2010,              <http://www.rfc-editor.org/info/rfc5245>.   [RFC5622]  Floyd, S. and E. Kohler, "Profile for Datagram Congestion              Control Protocol (DCCP) Congestion ID 4: TCP-Friendly Rate              Control for Small Packets (TFRC-SP)",RFC 5622,              DOI 10.17487/RFC5622, August 2009,              <http://www.rfc-editor.org/info/rfc5622>.   [RFC5652]  Housley, R., "Cryptographic Message Syntax (CMS)", STD 70,RFC 5652, DOI 10.17487/RFC5652, September 2009,              <http://www.rfc-editor.org/info/rfc5652>.   [RFC5681]  Allman, M., Paxson, V., and E. Blanton, "TCP Congestion              Control",RFC 5681, DOI 10.17487/RFC5681, September 2009,              <http://www.rfc-editor.org/info/rfc5681>.   [RFC5740]  Adamson, B., Bormann, C., Handley, M., and J. Macker,              "NACK-Oriented Reliable Multicast (NORM) Transport              Protocol",RFC 5740, DOI 10.17487/RFC5740, November 2009,              <http://www.rfc-editor.org/info/rfc5740>.   [RFC5751]  Ramsdell, B. and S. Turner, "Secure/Multipurpose Internet              Mail Extensions (S/MIME) Version 3.2 Message              Specification",RFC 5751, DOI 10.17487/RFC5751, January              2010, <http://www.rfc-editor.org/info/rfc5751>.   [RFC5775]  Luby, M., Watson, M., and L. Vicisano, "Asynchronous              Layered Coding (ALC) Protocol Instantiation",RFC 5775,              DOI 10.17487/RFC5775, April 2010,              <http://www.rfc-editor.org/info/rfc5775>.   [RFC5971]  Schulzrinne, H. and R. Hancock, "GIST: General Internet              Signalling Transport",RFC 5971, DOI 10.17487/RFC5971,              October 2010, <http://www.rfc-editor.org/info/rfc5971>.   [RFC5973]  Stiemerling, M., Tschofenig, H., Aoun, C., and E. Davies,              "NAT/Firewall NSIS Signaling Layer Protocol (NSLP)",RFC 5973, DOI 10.17487/RFC5973, October 2010,              <http://www.rfc-editor.org/info/rfc5973>.   [RFC6056]  Larsen, M. and F. Gont, "Recommendations for Transport-              Protocol Port Randomization",BCP 156,RFC 6056,              DOI 10.17487/RFC6056, January 2011,              <http://www.rfc-editor.org/info/rfc6056>.Eggert, et al.            Best Current Practice                [Page 48]

RFC 8085                  UDP Usage Guidelines                March 2017   [RFC6092]  Woodyatt, J., Ed., "Recommended Simple Security              Capabilities in Customer Premises Equipment (CPE) for              Providing Residential IPv6 Internet Service",RFC 6092,              DOI 10.17487/RFC6092, January 2011,              <http://www.rfc-editor.org/info/rfc6092>.   [RFC6335]  Cotton, M., Eggert, L., Touch, J., Westerlund, M., and S.              Cheshire, "Internet Assigned Numbers Authority (IANA)              Procedures for the Management of the Service Name and              Transport Protocol Port Number Registry",BCP 165,RFC 6335, DOI 10.17487/RFC6335, August 2011,              <http://www.rfc-editor.org/info/rfc6335>.   [RFC6347]  Rescorla, E. and N. Modadugu, "Datagram Transport Layer              Security Version 1.2",RFC 6347, DOI 10.17487/RFC6347,              January 2012, <http://www.rfc-editor.org/info/rfc6347>.   [RFC6396]  Blunk, L., Karir, M., and C. Labovitz, "Multi-Threaded              Routing Toolkit (MRT) Routing Information Export Format",RFC 6396, DOI 10.17487/RFC6396, October 2011,              <http://www.rfc-editor.org/info/rfc6396>.   [RFC6437]  Amante, S., Carpenter, B., Jiang, S., and J. Rajahalme,              "IPv6 Flow Label Specification",RFC 6437,              DOI 10.17487/RFC6437, November 2011,              <http://www.rfc-editor.org/info/rfc6437>.   [RFC6438]  Carpenter, B. and S. Amante, "Using the IPv6 Flow Label              for Equal Cost Multipath Routing and Link Aggregation in              Tunnels",RFC 6438, DOI 10.17487/RFC6438, November 2011,              <http://www.rfc-editor.org/info/rfc6438>.   [RFC6513]  Rosen, E., Ed. and R. Aggarwal, Ed., "Multicast in MPLS/              BGP IP VPNs",RFC 6513, DOI 10.17487/RFC6513, February              2012, <http://www.rfc-editor.org/info/rfc6513>.   [RFC6679]  Westerlund, M., Johansson, I., Perkins, C., O'Hanlon, P.,              and K. Carlberg, "Explicit Congestion Notification (ECN)              for RTP over UDP",RFC 6679, DOI 10.17487/RFC6679, August              2012, <http://www.rfc-editor.org/info/rfc6679>.   [RFC6726]  Paila, T., Walsh, R., Luby, M., Roca, V., and R. Lehtonen,              "FLUTE - File Delivery over Unidirectional Transport",RFC 6726, DOI 10.17487/RFC6726, November 2012,              <http://www.rfc-editor.org/info/rfc6726>.Eggert, et al.            Best Current Practice                [Page 49]

RFC 8085                  UDP Usage Guidelines                March 2017   [RFC6773]  Phelan, T., Fairhurst, G., and C. Perkins, "DCCP-UDP: A              Datagram Congestion Control Protocol UDP Encapsulation for              NAT Traversal",RFC 6773, DOI 10.17487/RFC6773, November              2012, <http://www.rfc-editor.org/info/rfc6773>.   [RFC6807]  Farinacci, D., Shepherd, G., Venaas, S., and Y. Cai,              "Population Count Extensions to Protocol Independent              Multicast (PIM)",RFC 6807, DOI 10.17487/RFC6807, December              2012, <http://www.rfc-editor.org/info/rfc6807>.   [RFC6887]  Wing, D., Ed., Cheshire, S., Boucadair, M., Penno, R., and              P. Selkirk, "Port Control Protocol (PCP)",RFC 6887,              DOI 10.17487/RFC6887, April 2013,              <http://www.rfc-editor.org/info/rfc6887>.   [RFC6935]  Eubanks, M., Chimento, P., and M. Westerlund, "IPv6 and              UDP Checksums for Tunneled Packets",RFC 6935,              DOI 10.17487/RFC6935, April 2013,              <http://www.rfc-editor.org/info/rfc6935>.   [RFC6936]  Fairhurst, G. and M. Westerlund, "Applicability Statement              for the Use of IPv6 UDP Datagrams with Zero Checksums",RFC 6936, DOI 10.17487/RFC6936, April 2013,              <http://www.rfc-editor.org/info/rfc6936>.   [RFC6951]  Tuexen, M. and R. Stewart, "UDP Encapsulation of Stream              Control Transmission Protocol (SCTP) Packets for End-Host              to End-Host Communication",RFC 6951,              DOI 10.17487/RFC6951, May 2013,              <http://www.rfc-editor.org/info/rfc6951>.   [RFC7143]  Chadalapaka, M., Satran, J., Meth, K., and D. Black,              "Internet Small Computer System Interface (iSCSI) Protocol              (Consolidated)",RFC 7143, DOI 10.17487/RFC7143, April              2014, <http://www.rfc-editor.org/info/rfc7143>.   [RFC7201]  Westerlund, M. and C. Perkins, "Options for Securing RTP              Sessions",RFC 7201, DOI 10.17487/RFC7201, April 2014,              <http://www.rfc-editor.org/info/rfc7201>.   [RFC7296]  Kaufman, C., Hoffman, P., Nir, Y., Eronen, P., and T.              Kivinen, "Internet Key Exchange Protocol Version 2              (IKEv2)", STD 79,RFC 7296, DOI 10.17487/RFC7296, October              2014, <http://www.rfc-editor.org/info/rfc7296>.   [RFC7450]  Bumgardner, G., "Automatic Multicast Tunneling",RFC 7450,              DOI 10.17487/RFC7450, February 2015,              <http://www.rfc-editor.org/info/rfc7450>.Eggert, et al.            Best Current Practice                [Page 50]

RFC 8085                  UDP Usage Guidelines                March 2017   [RFC7510]  Xu, X., Sheth, N., Yong, L., Callon, R., and D. Black,              "Encapsulating MPLS in UDP",RFC 7510,              DOI 10.17487/RFC7510, April 2015,              <http://www.rfc-editor.org/info/rfc7510>.   [RFC7525]  Sheffer, Y., Holz, R., and P. Saint-Andre,              "Recommendations for Secure Use of Transport Layer              Security (TLS) and Datagram Transport Layer Security              (DTLS)",BCP 195,RFC 7525, DOI 10.17487/RFC7525, May              2015, <http://www.rfc-editor.org/info/rfc7525>.   [RFC7560]  Kuehlewind, M., Ed., Scheffenegger, R., and B. Briscoe,              "Problem Statement and Requirements for Increased Accuracy              in Explicit Congestion Notification (ECN) Feedback",RFC 7560, DOI 10.17487/RFC7560, August 2015,              <http://www.rfc-editor.org/info/rfc7560>.   [RFC7567]  Baker, F., Ed. and G. Fairhurst, Ed., "IETF              Recommendations Regarding Active Queue Management",BCP 197,RFC 7567, DOI 10.17487/RFC7567, July 2015,              <http://www.rfc-editor.org/info/rfc7567>.   [RFC7605]  Touch, J., "Recommendations on Using Assigned Transport              Port Numbers",BCP 165,RFC 7605, DOI 10.17487/RFC7605,              August 2015, <http://www.rfc-editor.org/info/rfc7605>.   [RFC7657]  Black, D., Ed. and P. Jones, "Differentiated Services              (Diffserv) and Real-Time Communication",RFC 7657,              DOI 10.17487/RFC7657, November 2015,              <http://www.rfc-editor.org/info/rfc7657>.   [RFC7675]  Perumal, M., Wing, D., Ravindranath, R., Reddy, T., and M.              Thomson, "Session Traversal Utilities for NAT (STUN) Usage              for Consent Freshness",RFC 7675, DOI 10.17487/RFC7675,              October 2015, <http://www.rfc-editor.org/info/rfc7675>.   [RFC8083]  Perkins, C. and V. Singh, "Multimedia Congestion Control:              Circuit Breakers for Unicast RTP Sessions",RFC 8083,              DOI 10.17487/RFC8083, March 2017,              <http://www.rfc-editor.org/info/rfc8083>.   [RFC8086]  Yong, L., Ed., Crabbe, E., Xu, X., and T. Herbert, "GRE-              in-UDP Encapsulation",RFC 8086, DOI 10.17487/RFC8086,              March 2017, <http://www.rfc-editor.org/info/rfc8086>.Eggert, et al.            Best Current Practice                [Page 51]

RFC 8085                  UDP Usage Guidelines                March 2017   [RFC8087]  Fairhurst, G. and M. Welzl, "The Benefits of Using              Explicit Congestion Notification (ECN)",RFC 8087,              DOI 10.17487/RFC8087, March 2017,              <http://www.rfc-editor.org/info/rfc8087>.   [STEVENS]  Stevens, W., Fenner, B., and A. Rudoff, "UNIX Network              Programming, The sockets Networking API", Addison-Wesley,              2004.   [UPnP]     UPnP Forum, , "Internet Gateway Device (IGD) Standardized              Device Control Protocol V 1.0", November 2001.Eggert, et al.            Best Current Practice                [Page 52]

RFC 8085                  UDP Usage Guidelines                March 2017Appendix A.  Case Study of the Use of IPv6 UDP Zero-Checksum Mode   This appendix provides a brief review of MPLS-in-UDP as an example of   a UDP Tunnel Encapsulation that defines a UDP encapsulation.  The   purpose of the appendix is to provide a concrete example of which   mechanisms were required in order to safely use UDP zero-checksum   mode for MPLS-in-UDP tunnels over IPv6.  By default, UDP requires a   checksum for use with IPv6.  An option has been specified that   permits a zero IPv6 UDP checksum when used in specific environments,   specified in [RFC7510], and defines a set of operational constraints   for use of this mode.  These are summarized below:   A UDP tunnel or encapsulation using a zero-checksum mode with IPv6   must only be deployed within a single network (with a single network   operator) or networks of an adjacent set of cooperating network   operators where traffic is managed to avoid congestion, rather than   over the Internet where congestion control is required.  MPLS-in-UDP   has been specified for networks under single administrative control   (such as within a single operator's network) where it is known   (perhaps through knowledge of equipment types and lower-layer checks)   that packet corruption is exceptionally unlikely and where the   operator is willing to take the risk of undetected packet corruption.   The tunnel encapsulator SHOULD use different IPv6 addresses for each   UDP tunnel that uses the UDP zero-checksum mode, regardless of the   decapsulator, to strengthen the decapsulator's check of the IPv6   source address (i.e., the same IPv6 source address SHOULD NOT be used   with more than one IPv6 destination address, independent of whether   that destination address is a unicast or multicast address).  Use of   MPLS-in-UDP may be extended to networks within a set of closely   cooperating network administrations (such as network operators who   have agreed to work together to jointly provide specific services)   [RFC7510].   The requirement for MPLS-in-UDP endpoints to check the source IPv6   address in addition to the destination IPv6 address, plus the strong   recommendation against reuse of source IPv6 addresses among MPLS-in-   UDP tunnels collectively provide some mitigation for the absence of   UDP checksum coverage of the IPv6 header.  In addition, the MPLS data   plane only forwards packets with valid labels (i.e., labels that have   been distributed by the tunnel egress Label Switched Router, LSR),   providing some additional opportunity to detect MPLS-in-UDP packet   misdelivery when the misdelivered packet contains a label that is not   valid for forwarding at the receiving LSR.  The expected result for   IPv6 UDP zero-checksum mode for MPLS-in-UDP is that corruption of the   destination IPv6 address will usually cause packet discard, as   offsetting corruptions to the source IPv6 and/or MPLS top label are   unlikely.Eggert, et al.            Best Current Practice                [Page 53]

RFC 8085                  UDP Usage Guidelines                March 2017   Additional assurance is provided by the restrictions in the above   exceptions that limit usage of IPv6 UDP zero-checksum mode to well-   managed networks for which MPLS packet corruption has not been a   problem in practice.  Hence, MPLS-in-UDP is suitable for transmission   over lower layers in well-managed networks that are allowed by the   exceptions stated above and the rate of corruption of the inner IP   packet on such networks is not expected to increase by comparison to   MPLS traffic that is not encapsulated in UDP.  For these reasons,   MPLS-in-UDP does not provide an additional integrity check when UDP   zero-checksum mode is used with IPv6, and this design is in   accordance with requirements 2, 3, and 5 specified inSection 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 inSection 5   of [RFC6936].  MPLS does not accumulate incorrect state as a   consequence of label-stack corruption.  A corrupt MPLS label results   in either packet discard or forwarding (and forgetting) of the packet   without accumulation of MPLS protocol state.  Active monitoring of   MPLS-in-UDP traffic for errors is REQUIRED because the occurrence of   errors will result in some accumulation of error information outside   the MPLS protocol for operational and management purposes.  This   design is in accordance with requirement 4 specified inSection 5 of   [RFC6936].  In addition, IPv6 traffic with a zero UDP checksum MUST   be actively monitored for errors by the network operator.   Operators SHOULD also deploy packet filters to prevent IPv6 packets   with a zero UDP checksum from escaping from the network due to   misconfiguration or packet errors.  In addition, IPv6 traffic with a   zero UDP checksum MUST be actively monitored for errors by the   network operator.Eggert, et al.            Best Current Practice                [Page 54]

RFC 8085                  UDP Usage Guidelines                March 2017Acknowledgments   The middlebox traversal guidelines inSection 3.5 incorporate ideas   from Section 5 of [BEHAVE-APP] by Bryan Ford, Pyda Srisuresh, and Dan   Kegel.  The protocol timer guidelines inSection 3.1.1 were largely   contributed by Mark Allman.   G.  Fairhurst received funding from the European Union's Horizon 2020   research and innovation program 2014-2018 under grant agreement No.   644334 (NEAT).  Lars Eggert has received funding from the European   Union's Horizon 2020 research and innovation program 2014-2018 under   grant agreement No. 644866 (SSICLOPS).  This document reflects only   the authors' views and the European Commission is not responsible for   any use that may be made of the information it contains.Authors' Addresses   Lars Eggert   NetApp   Sonnenallee 1   Kirchheim  85551   Germany   Phone: +49 151 120 55791   Email: lars@netapp.com   URI:https://eggert.org/   Godred Fairhurst   University of Aberdeen   Department of Engineering   Fraser Noble Building   Aberdeen  AB24 3UE   Scotland   Email: gorry@erg.abdn.ac.uk   URI:http://www.erg.abdn.ac.uk/   Greg Shepherd   Cisco Systems   Tasman Drive   San Jose   United States of America   Email: gjshep@gmail.comEggert, et al.            Best Current Practice                [Page 55]