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RFC 5405

Unicast UDP Usage Guidelines for Application Designers

Pages: 27
Obsoleted by:  8085

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Network Working Group                                          L. Eggert
Request for Comments: 5405                                         Nokia
BCP: 145                                                    G. Fairhurst
Category: Best Current Practice                   University of Aberdeen
                                                           November 2008

         Unicast UDP Usage Guidelines for Application Designers

Status of This Memo

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

Copyright Notice

   Copyright (c) 2008 IETF Trust and the persons identified as the
   document authors.  All rights reserved.

   This document is subject to BCP 78 and the IETF Trust's Legal
   Provisions Relating to IETF Documents (
   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.


The User Datagram Protocol (UDP) provides a minimal message-passing transport that has no inherent congestion control mechanisms. Because congestion control is critical to the stable operation of the Internet, applications and upper-layer protocols that choose to use UDP as an Internet transport must employ mechanisms to prevent congestion collapse and to establish some degree of fairness with concurrent traffic. This document provides guidelines on the use of UDP for the designers of unicast applications and upper-layer protocols. Congestion control guidelines are a primary focus, but the document also provides guidance on other topics, including message sizes, reliability, checksums, and middlebox traversal.
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Table of Contents

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

The User Datagram Protocol (UDP) [RFC0768] provides a minimal, unreliable, best-effort, message-passing transport to applications and upper-layer protocols (both simply called "applications" in the remainder of this document). Compared to other transport protocols, UDP and its UDP-Lite variant [RFC3828] are unique in that they do not establish end-to-end connections between communicating end systems. UDP communication consequently does not incur connection establishment and teardown overheads, and there is minimal associated end system state. Because of these characteristics, UDP can offer a very efficient communication transport to some applications. A second unique characteristic of UDP is that it provides no inherent congestion control mechanisms. On many platforms, applications can send UDP datagrams at the line rate of the link interface, which is often much greater than the available path capacity, and doing so contributes to congestion along the path. [RFC2914] describes the best current practice for congestion control in the Internet. It identifies two major reasons why congestion control mechanisms are critical for the stable operation of the Internet: 1. The prevention of congestion collapse, i.e., a state where an increase in network load results in a decrease in useful work done by the network. 2. The establishment of a degree of fairness, i.e., allowing multiple flows to share the capacity of a path reasonably equitably. Because UDP itself provides no congestion control mechanisms, it is up to the applications that use UDP for Internet communication to employ suitable mechanisms to prevent congestion collapse and establish a degree of fairness. [RFC2309] discusses the dangers of congestion-unresponsive flows and states that "all UDP-based streaming applications should incorporate effective congestion avoidance mechanisms". This is an important requirement, even for applications that do not use UDP for streaming. In addition, congestion-controlled transmission is of benefit to an application itself, because it can reduce self-induced packet loss, minimize retransmissions, and hence reduce delays. Congestion control is essential even at relatively slow transmission rates. For example, an application that generates five 1500-byte UDP datagrams in one second can already exceed the capacity of a 56 Kb/s path. For applications that can operate at higher, potentially unbounded data rates, congestion control becomes vital to prevent congestion collapse and establish some degree of fairness. Section 3 describes a number of simple guidelines for the designers of such applications.
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   A UDP datagram is carried in a single IP packet and is hence limited
   to a maximum payload of 65,507 bytes for IPv4 and 65,527 bytes for
   IPv6.  The transmission of large IP packets usually requires IP
   fragmentation.  Fragmentation decreases communication reliability and
   efficiency and should be avoided.  IPv6 allows the option of
   transmitting large packets ("jumbograms") without fragmentation when
   all link layers along the path support this [RFC2675].  Some of the
   guidelines in Section 3 describe how applications should determine
   appropriate message sizes.  Other sections of this document provide
   guidance on reliability, checksums, and middlebox traversal.

   This document provides guidelines and recommendations.  Although most
   unicast UDP applications are expected to follow these guidelines,
   there do exist valid reasons why a specific application may decide
   not to follow a given guideline.  In such cases, it is RECOMMENDED
   that the application designers document the rationale for their
   design choice in the technical specification of their application or

   This document provides guidelines to designers of applications that
   use UDP for unicast transmission, which is the most common case.
   Specialized classes of applications use UDP for IP multicast
   [RFC1112], broadcast [RFC0919], or anycast [RFC1546] transmissions.
   The design of such specialized applications requires expertise that
   goes beyond the simple, unicast-specific guidelines given in this
   document.  Multicast and broadcast senders may transmit to multiple
   receivers across potentially very heterogeneous paths at the same
   time, which significantly complicates congestion control, flow
   control, and reliability mechanisms.  The IETF has defined a reliable
   multicast framework [RFC3048] and several building blocks to aid the
   designers of multicast applications, such as [RFC3738] or [RFC4654].
   Anycast senders must be aware that successive messages sent to the
   same anycast IP address may be delivered to different anycast nodes,
   i.e., arrive at different locations in the topology.  It is not
   intended that the guidelines in this document apply to multicast,
   broadcast, or anycast applications that use UDP.

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

The key words "MUST", "MUST NOT", "REQUIRED", "SHALL", "SHALL NOT", "SHOULD", "SHOULD NOT", "RECOMMENDED", "MAY", and "OPTIONAL" in this document are to be interpreted as described in BCP 14, RFC 2119 [RFC2119].

3. UDP Usage Guidelines

Internet paths can have widely varying characteristics, including transmission delays, available bandwidths, congestion levels, reordering probabilities, supported message sizes, or loss rates. Furthermore, the same Internet path can have very different conditions over time. Consequently, applications that may be used on the Internet MUST NOT make assumptions about specific path characteristics. They MUST instead use mechanisms that let them operate safely under very different path conditions. Typically, this requires conservatively probing the current conditions of the Internet path they communicate over to establish a transmission behavior that it can sustain and that is reasonably fair to other traffic sharing the path. These mechanisms are difficult to implement correctly. For most applications, the use of one of the existing IETF transport protocols is the simplest method of acquiring the required mechanisms. Consequently, the RECOMMENDED alternative to the UDP usage described in the remainder of this section is the use of an IETF transport protocol such as TCP [RFC0793], Stream Control Transmission Protocol (SCTP) [RFC4960], and SCTP Partial Reliability Extension (SCTP-PR) [RFC3758], or Datagram Congestion Control Protocol (DCCP) [RFC4340] with its different congestion control types [RFC4341][RFC4342][CCID4]. If used correctly, these more fully-featured transport protocols are not as "heavyweight" as often claimed. For example, the TCP algorithms have been continuously improved over decades, and have reached a level of efficiency and correctness that custom application-layer mechanisms will struggle to easily duplicate. In addition, many TCP implementations allow connections to be tuned by an application to its purposes. For example, TCP's "Nagle" algorithm [RFC0896] can be disabled, improving communication latency at the expense of more frequent -- but still congestion-controlled -- packet transmissions. Another example is the TCP SYN cookie mechanism [RFC4987], which is available on many platforms. TCP with SYN cookies does not require a server to maintain per-connection state until the connection is established. TCP also requires the end that closes a connection to maintain the TIME-WAIT state that prevents delayed segments from one connection instance from interfering with a
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   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

3.1. Congestion Control Guidelines

If an application or upper-layer protocol chooses not to use a congestion-controlled transport protocol, it SHOULD control the rate at which it sends UDP datagrams to a destination host, in order to fulfill the requirements of [RFC2914]. It is important to stress that an application SHOULD perform congestion control over all UDP traffic it sends to a destination, independently from how it generates this traffic. For example, an application that forks multiple worker processes or otherwise uses multiple sockets to generate UDP datagrams SHOULD perform congestion control over the aggregate traffic. Several approaches to perform congestion control are discussed in the remainder of this section. Not all approaches discussed below are appropriate for all UDP-transmitting applications. Section 3.1.1 discusses congestion control options for applications that perform bulk transfers over UDP. Such applications can employ schemes that sample the path over several subsequent RTTs during which data is exchanged, in order to determine a sending rate that the path at its current load can support. Other applications only exchange a few UDP datagrams with a destination. Section 3.1.2 discusses congestion control options for such "low data-volume" applications. Because they typically do not transmit enough data to iteratively sample the path to determine a safe sending rate, they need to employ different kinds of congestion control mechanisms. Section 3.1.3 discusses congestion control considerations when UDP is used as a tunneling protocol. It is important to note that congestion control should not be viewed as an add-on to a finished application. Many of the mechanisms discussed in the guidelines below require application support to operate correctly. Application designers need to consider congestion control throughout the design of their application, similar to how they consider security aspects throughout the design process. In the past, the IETF has also investigated integrated congestion control mechanisms that act on the traffic aggregate between two hosts, i.e., a framework such as the Congestion Manager [RFC3124],
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   where active sessions may share current congestion information in a
   way that is independent of the transport protocol.  Such mechanisms
   have currently failed to see deployment, but would otherwise simplify
   the design of congestion control mechanisms for UDP sessions, so that
   they fulfill the requirements in [RFC2914].

3.1.1. Bulk Transfer Applications

Applications that perform bulk transmission of data to a peer over UDP, i.e., applications that exchange more than a small number of UDP datagrams per RTT, SHOULD implement TCP-Friendly Rate Control (TFRC) [RFC5348], window-based, TCP-like congestion control, or otherwise ensure that the application complies with the congestion control principles. TFRC has been designed to provide both congestion control and fairness in a way that is compatible with the IETF's other transport protocols. If an application implements TFRC, it need not follow the remaining guidelines in Section 3.1.1, because TFRC already addresses them, but SHOULD still follow the remaining guidelines in the subsequent subsections of Section 3. Bulk transfer applications that choose not to implement TFRC or TCP- like windowing SHOULD implement a congestion control scheme that results in bandwidth use that competes fairly with TCP within an order of magnitude. Section 2 of [RFC3551] suggests that applications SHOULD monitor the packet loss rate to ensure that it is within acceptable parameters. Packet loss is considered acceptable if a TCP flow across the same network path under the same network conditions would achieve an average throughput, measured on a reasonable timescale, that is not less than that of the UDP flow. The comparison to TCP cannot be specified exactly, but is intended as an "order-of-magnitude" comparison in timescale and throughput. Finally, some bulk transfer applications may choose not to implement any congestion control mechanism and instead rely on transmitting across reserved path capacity. This might be an acceptable choice for a subset of restricted networking environments, but is by no means a safe practice for operation in the Internet. When the UDP traffic of such applications leaks out on unprovisioned Internet paths, it can significantly degrade the performance of other traffic sharing the path and even result in congestion collapse. Applications that support an uncontrolled or unadaptive transmission behavior SHOULD NOT do so by default and SHOULD instead require users to explicitly enable this mode of operation.
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3.1.2. Low Data-Volume Applications

When applications that at any time exchange only a small number of UDP datagrams with a destination implement TFRC or one of the other congestion control schemes in Section 3.1.1, the network sees little benefit, because those mechanisms perform congestion control in a way that is only effective for longer transmissions. Applications that at any time exchange only a small number of UDP datagrams with a destination SHOULD still control their transmission behavior by not sending on average more than one UDP datagram per round-trip time (RTT) to a destination. Similar to the recommendation in [RFC1536], an application SHOULD maintain an estimate of the RTT for any destination with which it communicates. Applications SHOULD implement the algorithm specified in [RFC2988] to compute a smoothed RTT (SRTT) estimate. They SHOULD also detect packet loss and exponentially back-off their retransmission timer when a loss event occurs. When implementing this scheme, applications need to choose a sensible initial value for the RTT. This value SHOULD generally be as conservative as possible for the given application. TCP uses an initial value of 3 seconds [RFC2988], which is also RECOMMENDED as an initial value for UDP applications. SIP [RFC3261] and GIST [GIST] use an initial value of 500 ms, and initial timeouts that are shorter than this are likely problematic in many cases. It is also important to note that the initial timeout is not the maximum possible timeout -- the RECOMMENDED algorithm in [RFC2988] yields timeout values after a series of losses that are much longer than the initial value. Some applications cannot maintain a reliable RTT estimate for a destination. The first case is that of applications that exchange too few UDP datagrams with a peer to establish a statistically accurate RTT estimate. Such applications MAY use a predetermined transmission interval that is exponentially backed-off when packets are lost. TCP uses an initial value of 3 seconds [RFC2988], which is also RECOMMENDED as an initial value for UDP applications. SIP [RFC3261] and GIST [GIST] use an interval of 500 ms, and shorter values are likely problematic in many cases. As in the previous case, note that the initial timeout is not the maximum possible timeout. A second class of applications cannot maintain an RTT estimate for a destination, because the destination does not send return traffic. Such applications SHOULD NOT send more than one UDP datagram every 3 seconds, and SHOULD use an even less aggressive rate when possible. The 3-second interval was chosen based on TCP's retransmission timeout when the RTT is unknown [RFC2988], and shorter values are likely problematic in many cases. Note that the sending rate in this
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   case must be more conservative than in the two previous cases,
   because the lack of return traffic prevents the detection of packet
   loss, i.e., congestion events, and the application therefore cannot
   perform exponential back-off to reduce load.

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

3.1.3. UDP Tunnels

One increasingly popular use of UDP is as a tunneling protocol, where a tunnel endpoint encapsulates the packets of another protocol inside UDP datagrams and transmits them to another tunnel endpoint, which decapsulates the UDP datagrams and forwards the original packets contained in the payload. Tunnels establish virtual links that appear to directly connect locations that are distant in the physical Internet topology and can be used to create virtual (private) networks. Using UDP as a tunneling protocol is attractive when the payload protocol is not supported by middleboxes that may exist along the path, because many middleboxes support transmission using UDP. Well-implemented tunnels are generally invisible to the endpoints that happen to transmit over a path that includes tunneled links. On the other hand, to the routers along the path of a UDP tunnel, i.e., the routers between the two tunnel endpoints, the traffic that a UDP tunnel generates is a regular UDP flow, and the encapsulator and decapsulator appear as regular UDP-sending and -receiving applications. Because other flows can share the path with one or more UDP tunnels, congestion control needs to be considered. Two factors determine whether a UDP tunnel needs to employ specific congestion control mechanisms -- first, whether the payload traffic is IP-based; second, whether the tunneling scheme generates UDP traffic at a volume that corresponds to the volume of payload traffic carried within the tunnel. IP-based traffic is generally assumed to be congestion-controlled, i.e., it is assumed that the transport protocols generating IP-based traffic at the sender already employ mechanisms that are sufficient to address congestion on the path. Consequently, a tunnel carrying
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   IP-based traffic should already interact appropriately with other
   traffic sharing the path, and specific congestion control mechanisms
   for the tunnel are not necessary.

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

   The following guidelines define these possible cases in more detail:

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

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

       Note that this guideline is built on the assumption that most IP-
       based communication is congestion-controlled.  If a UDP tunnel is
       used for IP-based traffic that is known to not be congestion-
       controlled, the next set of guidelines applies.

   2.  A tunnel generates UDP traffic at a volume that corresponds to
       the volume of payload traffic, and the payload traffic is not
       known to be IP-based, or is known to be IP-based but not

       This can be the case, for example, when some link-layer protocols
       are encapsulated within UDP (but not all link-layer protocols;
       some are congestion-controlled).  Because it is not known that
       congestion losses of tunneled non-IP traffic will trigger an
       appropriate congestion response at the senders, the UDP tunnel
       SHOULD employ an appropriate congestion control mechanism.
       Because tunnels are usually bulk-transfer applications as far as
       the intermediate routers are concerned, the guidelines in
       Section 3.1.1 apply.

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

       Examples of this class include UDP tunnels that send at a
       constant rate, increase their transmission rates under loss, for
       example, due to increasing redundancy when Forward Error
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       Correction is used, or are otherwise constrained in their
       transmission behavior.  These specialized uses of UDP for
       tunneling go beyond the scope of the general guidelines given in
       this document.  The implementer of such specialized tunnels
       SHOULD carefully consider congestion control in the design of
       their tunneling mechanism.

   Designing a tunneling mechanism requires significantly more expertise
   than needed for many other UDP applications, because tunnels
   virtualize lower-layer components of the Internet, and the
   virtualized components need to correctly interact with the
   infrastructure at that layer.  This document only touches upon the
   congestion control considerations for implementing UDP tunnels; a
   discussion of other required tunneling behavior is out of scope.

3.2. Message Size Guidelines

IP fragmentation lowers the efficiency and reliability of Internet communication. The loss of a single fragment results in the loss of an entire fragmented packet, because even if all other fragments are received correctly, the original packet cannot be reassembled and delivered. This fundamental issue with fragmentation exists for both IPv4 and IPv6. In addition, some network address translators (NATs) and firewalls drop IP fragments. The network address translation performed by a NAT only operates on complete IP packets, and some firewall policies also require inspection of complete IP packets. Even with these being the case, some NATs and firewalls simply do not implement the necessary reassembly functionality, and instead choose to drop all fragments. Finally, [RFC4963] documents other issues specific to IPv4 fragmentation. Due to these issues, an application SHOULD NOT send UDP datagrams that result in IP packets that exceed the MTU of the path to the destination. Consequently, an application SHOULD either use the path MTU information provided by the IP layer or implement path MTU discovery itself [RFC1191][RFC1981][RFC4821] to determine whether the path to a destination will support its desired message size without fragmentation. Applications that do not follow this recommendation to do PMTU discovery SHOULD still avoid sending UDP datagrams that would result in IP packets that exceed the path MTU. Because the actual path MTU is unknown, such applications SHOULD fall back to sending messages that are shorter than the default effective MTU for sending (EMTU_S in [RFC1122]). For IPv4, EMTU_S is the smaller of 576 bytes and the first-hop MTU [RFC1122]. For IPv6, EMTU_S is 1280 bytes [RFC2460]. The effective PMTU for a directly connected destination (with no routers on the path) is the configured interface MTU, which could be
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   less than the maximum link payload size.  Transmission of minimum-
   sized UDP datagrams is inefficient over paths that support a larger
   PMTU, which is a second reason to implement PMTU discovery.

   To determine an appropriate UDP payload size, applications MUST
   subtract the size of the IP header (which includes any IPv4 optional
   headers or IPv6 extension headers) as well as the length of the UDP
   header (8 bytes) from the PMTU size.  This size, known as the MMS_S,
   can be obtained from the TCP/IP stack [RFC1122].

   Applications that do not send messages that exceed the effective PMTU
   of IPv4 or IPv6 need not implement any of the above mechanisms.  Note
   that the presence of tunnels can cause an additional reduction of the
   effective PMTU, so implementing PMTU discovery may be beneficial.

   Applications that fragment an application-layer message into multiple
   UDP datagrams SHOULD perform this fragmentation so that each datagram
   can be received independently, and be independently retransmitted in
   the case where an application implements its own reliability

3.3. Reliability Guidelines

Application designers are generally aware that UDP does not provide any reliability, e.g., it does not retransmit any lost packets. Often, this is a main reason to consider UDP as a transport. Applications that do require reliable message delivery MUST implement an appropriate mechanism themselves. UDP also does not protect against datagram duplication, i.e., an application may receive multiple copies of the same UDP datagram. Application designers SHOULD verify that their application handles datagram duplication gracefully, and may consequently need to implement mechanisms to detect duplicates. Even if UDP datagram reception triggers idempotent operations, applications may want to suppress duplicate datagrams to reduce load. In addition, the Internet can significantly delay some packets with respect to others, e.g., due to routing transients, intermittent connectivity, or mobility. This can cause reordering, where UDP datagrams arrive at the receiver in an order different from the transmission order. Applications that require ordered delivery MUST reestablish datagram ordering themselves. Finally, it is important to note that delay spikes can be very large. This can cause reordered packets to arrive many seconds after they were sent. [RFC0793] defines the maximum delay a TCP segment should experience -- the Maximum Segment Lifetime (MSL) -- as 2 minutes. No
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   other RFC defines an MSL for other transport protocols or IP itself.
   This document clarifies that the MSL value to be used for UDP SHOULD
   be the same 2 minutes as for TCP.  Applications SHOULD be robust to
   the reception of delayed or duplicate packets that are received
   within this 2-minute interval.

   An application that requires reliable and ordered message delivery
   SHOULD choose an IETF standard transport protocol that provides these
   features.  If this is not possible, it will need to implement a set
   of appropriate mechanisms itself.

3.4. Checksum Guidelines

The UDP header includes an optional, 16-bit one's complement checksum that provides an integrity check. This results in a relatively weak protection in terms of coding theory [RFC3819], and application developers SHOULD implement additional checks where data integrity is important, e.g., through a Cyclic Redundancy Check (CRC) included with the data to verify the integrity of an entire object/file sent over the UDP service. The UDP checksum provides a statistical guarantee that the payload was not corrupted in transit. It also allows the receiver to verify that it was the intended destination of the packet, because it covers the IP addresses, port numbers, and protocol number, and it verifies that the packet is not truncated or padded, because it covers the size field. It therefore protects an application against receiving corrupted payload data in place of, or in addition to, the data that was sent. This check is not strong from a coding or cryptographic perspective, and is not designed to detect physical-layer errors or malicious modification of the datagram [RFC3819]. Applications SHOULD enable UDP checksums, although [RFC0768] permits the option to disable their use. Applications that choose to disable UDP checksums when transmitting over IPv4 therefore MUST NOT make assumptions regarding the correctness of received data and MUST behave correctly when a UDP datagram is received that was originally sent to a different destination or is otherwise corrupted. The use of the UDP checksum is REQUIRED when applications transmit UDP over IPv6 [RFC2460].

3.4.1. UDP-Lite

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

   UDP-Lite changes the semantics of the UDP "payload length" field to
   that of a "checksum coverage length" field.  Otherwise, UDP-Lite is
   semantically identical to UDP.  The interface of UDP-Lite differs
   from that of UDP by the addition of a single (socket) option that
   communicates a checksum coverage length value: at the sender, this
   specifies the intended checksum coverage, with the remaining
   unprotected part of the payload called the "error-insensitive part".
   By default, the UDP-Lite checksum coverage extends across the entire
   datagram.  If required, an application may dynamically modify this
   length value, e.g., to offer greater protection to some messages.
   UDP-Lite always verifies that a packet was delivered to the intended
   destination, i.e., always verifies the header fields.  Errors in the
   insensitive part will not cause a UDP datagram to be discarded by the
   destination.  Applications using UDP-Lite therefore MUST NOT make
   assumptions regarding the correctness of the data received in the
   insensitive part of the UDP-Lite payload.

   The sending application SHOULD select the minimum checksum coverage
   to include all sensitive protocol headers.  For example, applications
   that use the Real-Time Protocol (RTP) [RFC3550] will likely want to
   protect the RTP header against corruption.  Applications, where
   appropriate, MUST also introduce their own appropriate validity
   checks for protocol information carried in the insensitive part of
   the UDP-Lite payload (e.g., internal CRCs).

   The receiver must set a minimum coverage threshold for incoming
   packets that is not smaller than the smallest coverage used by the
   sender [RFC3828].  The receiver SHOULD select a threshold that is
   sufficiently large to block packets with an inappropriately short
   coverage field.  This may be a fixed value, or may be negotiated by
   an application.  UDP-Lite does not provide mechanisms to negotiate
   the checksum coverage between the sender and receiver.

   Applications may still experience packet loss, rather than
   corruption, when using UDP-Lite.  The enhancements offered by UDP-
   Lite rely upon a link being able to intercept the UDP-Lite header to
   correctly identify the partial coverage required.  When tunnels
   and/or encryption are used, this can result in UDP-Lite datagrams
   being treated the same as UDP datagrams, i.e., result in packet loss.
   Use of IP fragmentation can also prevent special treatment for UDP-
   Lite datagrams, and this is another reason why applications SHOULD
   avoid IP fragmentation (Section 3.2).
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3.5. Middlebox Traversal Guidelines

Network address translators (NATs) and firewalls are examples of intermediary devices ("middleboxes") that can exist along an end-to- end path. A middlebox typically performs a function that requires it to maintain per-flow state. For connection-oriented protocols, such as TCP, middleboxes snoop and parse the connection-management traffic and create and destroy per-flow state accordingly. For a connectionless protocol such as UDP, this approach is not possible. Consequently, middleboxes may create per-flow state when they see a packet that indicates a new flow, and destroy the state after some period of time during which no packets belonging to the same flow have arrived. Depending on the specific function that the middlebox performs, this behavior can introduce a time-dependency that restricts the kinds of UDP traffic exchanges that will be successful across the middlebox. For example, NATs and firewalls typically define the partial path on one side of them to be interior to the domain they serve, whereas the partial path on their other side is defined to be exterior to that domain. Per-flow state is typically created when the first packet crosses from the interior to the exterior, and while the state is present, NATs and firewalls will forward return traffic. Return traffic that arrives after the per-flow state has timed out is dropped, as is other traffic that arrives from the exterior. Many applications that use UDP for communication operate across middleboxes without needing to employ additional mechanisms. One example is the Domain Name System (DNS), which has a strict request- response communication pattern that typically completes within seconds. Other applications may experience communication failures when middleboxes destroy the per-flow state associated with an application session during periods when the application does not exchange any UDP traffic. Applications SHOULD be able to gracefully handle such communication failures and implement mechanisms to re-establish application-layer sessions and state. For some applications, such as media transmissions, this re- synchronization is highly undesirable, because it can cause user- perceivable playback artifacts. Such specialized applications MAY send periodic keep-alive messages to attempt to refresh middlebox state. It is important to note that keep-alive messages are NOT RECOMMENDED for general use -- they are unnecessary for many applications and can consume significant amounts of system and network resources.
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   An application that needs to employ keep-alives to deliver useful
   service over UDP in the presence of middleboxes SHOULD NOT transmit
   them more frequently than once every 15 seconds and SHOULD use longer
   intervals when possible.  No common timeout has been specified for
   per-flow UDP state for arbitrary middleboxes.  NATs require a state
   timeout of 2 minutes or longer [RFC4787].  However, empirical
   evidence suggests that a significant fraction of currently deployed
   middleboxes unfortunately use shorter timeouts.  The timeout of 15
   seconds originates with the Interactive Connectivity Establishment
   (ICE) protocol [ICE].  When applications are deployed in more
   controlled network environments, the deployers SHOULD investigate
   whether the target environment allows applications to use longer
   intervals, or whether it offers mechanisms to explicitly control
   middlebox state timeout durations, for example, using Middlebox
   Communications (MIDCOM) [RFC3303], Next Steps in Signaling (NSIS)
   [NSLP], or Universal Plug and Play (UPnP) [UPnP].  It is RECOMMENDED
   that applications apply slight random variations ("jitter") to the
   timing of keep-alive transmissions, to reduce the potential for
   persistent synchronization between keep-alive transmissions from
   different hosts.

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

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

The de facto standard application programming interface (API) for TCP/IP applications is the "sockets" interface [POSIX]. Some platforms also offer applications the ability to directly assemble and transmit IP packets through "raw sockets" or similar facilities. This is a second, more cumbersome method of using UDP. The guidelines in this document cover all such methods through which an application may use UDP. Because the sockets API is by far the most common method, the remainder of this section discusses it in more detail. Although the sockets API was developed for UNIX in the early 1980s, a wide variety of non-UNIX operating systems also implement this. The sockets API supports both IPv4 and IPv6 [RFC3493]. The UDP sockets API differs from that for TCP in several key ways. Because application programmers are typically more familiar with the TCP sockets API, the remainder of this section discusses these differences. [STEVENS] provides usage examples of the UDP sockets API. UDP datagrams may be directly sent and received, without any connection setup. Using the sockets API, applications can receive packets from more than one IP source address on a single UDP socket. Some servers use this to exchange data with more than one remote host through a single UDP socket at the same time. Many applications need to ensure that they receive packets from a particular source address; these applications MUST implement corresponding checks at the application layer or explicitly request that the operating system filter the received packets. If a client/server application executes on a host with more than one IP interface, the application SHOULD send any UDP responses with an IP source address that matches the IP destination address of the UDP datagram that carried the request (see [RFC1122], Section Many middleboxes expect this transmission behavior and drop replies that are sent from a different IP address, as explained in Section 3.5. A UDP receiver can receive a valid UDP datagram with a zero-length payload. Note that this is different from a return value of zero from a read() socket call, which for TCP indicates the end of the connection. Many operating systems also allow a UDP socket to be connected, i.e., to bind a UDP socket to a specific pair of addresses and ports. This is similar to the corresponding TCP sockets API functionality. However, for UDP, this is only a local operation that serves to
ToP   noToC   RFC5405 - Page 18
   simplify the local send/receive functions and to filter the traffic
   for the specified addresses and ports.  Binding a UDP socket does not
   establish a connection -- UDP does not notify the remote end when a
   local UDP socket is bound.  Binding a socket also allows configuring
   options that affect the UDP or IP layers, for example, use of the UDP
   checksum or the IP Timestamp option.  On some stacks, a bound socket
   also allows an application to be notified when ICMP error messages
   are received for its transmissions [RFC1122].

   UDP provides no flow-control.  This is another reason why UDP-based
   applications need to be robust in the presence of packet loss.  This
   loss can also occur within the sending host, when an application
   sends data faster than the line rate of the outbound network
   interface.  It can also occur on the destination, where receive calls
   fail to return all the data that was sent when the application issues
   them too infrequently (i.e., such that the receive buffer overflows).
   Robust flow control mechanisms are difficult to implement, which is
   why applications that need this functionality SHOULD consider using a
   full-featured transport protocol.

   When an application closes a TCP, SCTP or DCCP socket, the transport
   protocol on the receiving host is required to maintain TIME-WAIT
   state.  This prevents delayed packets from the closed connection
   instance from being mistakenly associated with a later connection
   instance that happens to reuse the same IP address and port pairs.
   The UDP protocol does not implement such a mechanism.  Therefore,
   UDP-based applications need to be robust in this case.  One
   application may close a socket or terminate, followed in time by
   another application receiving on the same port.  This later
   application may then receive packets intended for the first
   application that were delayed in the network.

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

3.7. ICMP Guidelines

Applications can utilize information about ICMP error messages that the UDP layer passes up for a variety of purposes [RFC1122]. Applications SHOULD validate that the information in the ICMP message payload, e.g., a reported error condition, corresponds to a UDP datagram that the application actually sent. Note that not all APIs
ToP   noToC   RFC5405 - Page 19
   have the necessary functions to support this validation, and some
   APIs already perform this validation internally before passing ICMP
   information to the application.

   Any application response to ICMP error messages SHOULD be robust to
   temporary routing failures, i.e., transient ICMP "unreachable"
   messages should not normally cause a communication abort.
   Applications SHOULD appropriately process ICMP messages generated in
   response to transmitted traffic.  A correct response often requires
   context, such as local state about communication instances to each
   destination, that although readily available in connection-oriented
   transport protocols is not always maintained by UDP-based

4. Security Considerations

UDP does not provide communications security. Applications that need to protect their communications against eavesdropping, tampering, or message forgery SHOULD employ end-to-end security services provided by other IETF protocols. Applications that respond to short requests with potentially large responses are vulnerable to amplification attacks, and SHOULD authenticate the sender before responding. The source IP address of a request is not a useful authenticator, because it can be spoofed. One option of securing UDP communications is with IPsec [RFC4301], which can provide authentication for flows of IP packets through the Authentication Header (AH) [RFC4302] and encryption and/or authentication through the Encapsulating Security Payload (ESP) [RFC4303]. Applications use the Internet Key Exchange (IKE) [RFC4306] to configure IPsec for their sessions. Depending on how IPsec is configured for a flow, it can authenticate or encrypt the UDP headers as well as UDP payloads. If an application only requires authentication, ESP with no encryption but with authentication is often a better option than AH, because ESP can operate across middleboxes. An application that uses IPsec requires the support of an operating system that implements the IPsec protocol suite. Although it is possible to use IPsec to secure UDP communications, not all operating systems support IPsec or allow applications to easily configure it for their flows. A second option of securing UDP communications is through Datagram Transport Layer Security (DTLS) [RFC4347]. DTLS provides communication privacy by encrypting UDP payloads. It does not protect the UDP headers. Applications can implement DTLS without relying on support from the operating system.
ToP   noToC   RFC5405 - Page 20
   Many other options for authenticating or encrypting UDP payloads
   exist.  For example, the GSS-API security framework [RFC2743] or
   Cryptographic Message Syntax (CMS) [RFC3852] could be used to protect
   UDP payloads.  The IETF standard for securing RTP [RFC3550]
   communication sessions over UDP is the Secure Real-time Transport
   Protocol (SRTP) [RFC3711].  In some applications, a better solution
   is to protect larger stand-alone objects, such as files or messages,
   instead of individual UDP payloads.  In these situations, CMS
   [RFC3852], S/MIME [RFC3851] or OpenPGP [RFC4880] could be used.  In
   addition, there are many non-IETF protocols in this area.

   Like congestion control mechanisms, security mechanisms are difficult
   to design and implement correctly.  It is hence RECOMMENDED that
   applications employ well-known standard security mechanisms such as
   DTLS or IPsec, rather than inventing their own.

   The Generalized TTL Security Mechanism (GTSM) [RFC5082] may be used
   with UDP applications (especially when the intended endpoint is on
   the same link as the sender).  This is a lightweight mechanism that
   allows a receiver to filter unwanted packets.

   In terms of congestion control, [RFC2309] and [RFC2914] discuss the
   dangers of congestion-unresponsive flows to the Internet.  This
   document provides guidelines to designers of UDP-based applications
   to congestion-control their transmissions, and does not raise any
   additional security concerns.

5. Summary

This section summarizes the guidelines made in Sections 3 and 4 in a tabular format (Table 1) for easy referencing.
ToP   noToC   RFC5405 - Page 21
   | Recommendation                                          | Section |
   | MUST tolerate a wide range of Internet path conditions  | 3       |
   | SHOULD use a full-featured transport (TCP, SCTP, DCCP)  |         |
   |                                                         |         |
   | SHOULD control rate of transmission                     | 3.1     |
   | SHOULD perform congestion control over all traffic      |         |
   |                                                         |         |
   | for bulk transfers,                                     | 3.1.1   |
   | SHOULD consider implementing TFRC                       |         |
   | else, SHOULD in other ways use bandwidth similar to TCP |         |
   |                                                         |         |
   | for non-bulk transfers,                                 | 3.1.2   |
   | SHOULD measure RTT and transmit max. 1 datagram/RTT     |         |
   | else, SHOULD send at most 1 datagram every 3 seconds    |         |
   | SHOULD back-off retransmission timers following loss    |         |
   |                                                         |         |
   | for tunnels carrying IP Traffic,                        | 3.1.3   |
   | SHOULD NOT perform congestion control                   |         |
   |                                                         |         |
   | for non-IP tunnels or rate not determined by traffic,   | 3.1.3   |
   | SHOULD perform congestion control                       |         |
   |                                                         |         |
   | SHOULD NOT send datagrams that exceed the PMTU, i.e.,   | 3.2     |
   | SHOULD discover PMTU or send datagrams < minimum PMTU   |         |
   |                                                         |         |
   | SHOULD handle datagram loss, duplication, reordering    | 3.3     |
   | SHOULD be robust to delivery delays up to 2 minutes     |         |
   |                                                         |         |
   | SHOULD enable IPv4 UDP checksum                         | 3.4     |
   | MUST enable IPv6 UDP checksum                           |         |
   | else, MAY use UDP-Lite with suitable checksum coverage  | 3.4.1   |
   |                                                         |         |
   | SHOULD NOT always send middlebox keep-alives            | 3.5     |
   | MAY use keep-alives when needed (min. interval 15 sec)  |         |
   |                                                         |         |
   | MUST check IP source address                            | 3.6     |
   | and, for client/server applications                     |         |
   | SHOULD send responses from src address matching request |         |
   |                                                         |         |
   | SHOULD use standard IETF security protocols when needed | 4       |

                    Table 1: Summary of recommendations
ToP   noToC   RFC5405 - Page 22

6. Acknowledgments

Thanks to Paul Aitken, Mark Allman, Francois Audet, Iljitsch van Beijnum, Stewart Bryant, Remi Denis-Courmont, Lisa Dusseault, Wesley Eddy, Pasi Eronen, Sally Floyd, Robert Hancock, Jeffrey Hutzelman, Cullen Jennings, Tero Kivinen, Peter Koch, Jukka Manner, Philip Matthews, Joerg Ott, Colin Perkins, Tom Petch, Carlos Pignataro, Pasi Sarolahti, Pascal Thubert, Joe Touch, Dave Ward, and Magnus Westerlund for their comments on this document. The middlebox traversal guidelines in Section 3.5 incorporate ideas from Section 5 of [BEHAVE-APP] by Bryan Ford, Pyda Srisuresh, and Dan Kegel. Lars Eggert is partly funded by [TRILOGY], a research project supported by the European Commission under its Seventh Framework Program. Gorry Fairhurst was partly funded by the EC SatNEx project.

7. References

7.1. Normative References

[RFC0768] Postel, J., "User Datagram Protocol", STD 6, RFC 768, August 1980. [RFC0793] Postel, J., "Transmission Control Protocol", STD 7, RFC 793, September 1981. [RFC1122] Braden, R., "Requirements for Internet Hosts - Communication Layers", STD 3, RFC 1122, October 1989. [RFC1191] Mogul, J. and S. Deering, "Path MTU discovery", RFC 1191, November 1990. [RFC1981] McCann, J., Deering, S., and J. Mogul, "Path MTU Discovery for IP version 6", RFC 1981, August 1996. [RFC2119] Bradner, S., "Key words for use in RFCs to Indicate Requirement Levels", BCP 14, RFC 2119, March 1997. [RFC2460] Deering, S. and R. Hinden, "Internet Protocol, Version 6 (IPv6) Specification", RFC 2460, December 1998. [RFC2914] Floyd, S., "Congestion Control Principles", BCP 41, RFC 2914, September 2000. [RFC2988] Paxson, V. and M. Allman, "Computing TCP's Retransmission Timer", RFC 2988, November 2000.
ToP   noToC   RFC5405 - Page 23
   [RFC3828]     Larzon, L-A., Degermark, M., Pink, S., Jonsson, L-E.,
                 and G. Fairhurst, "The Lightweight User Datagram
                 Protocol (UDP-Lite)", RFC 3828, July 2004.

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

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

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

7.2. Informative References

[BEHAVE-APP] Ford, B., "Application Design Guidelines for Traversal through Network Address Translators", Work in Progress, March 2007. [CCID4] Floyd, S. and E. Kohler, "Profile for Datagram Congestion Control Protocol (DCCP) Congestion ID 4: TCP-Friendly Rate Control for Small Packets (TFRC-SP)", Work in Progress, February 2008. [FABER] Faber, T., Touch, J., and W. Yue, "The TIME-WAIT State in TCP and Its Effect on Busy Servers", Proc. IEEE Infocom, March 1999. [GIST] Schulzrinne, H. and R. Hancock, "GIST: General Internet Signalling Transport", Work in Progress, July 2008. [ICE] Rosenberg, J., "Interactive Connectivity Establishment (ICE): A Protocol for Network Address Translator (NAT) Traversal for Offer/Answer Protocols", Work in Progress, October 2007. [NSLP] Stiemerling, M., Tschofenig, H., Aoun, C., and E. Davies, "NAT/Firewall NSIS Signaling Layer Protocol (NSLP)", Work in Progress, September 2008. [POSIX] IEEE Std. 1003.1-2001, "Standard for Information Technology - Portable Operating System Interface (POSIX)", Open Group Technical Standard: Base Specifications Issue 6, ISO/IEC 9945:2002, December 2001.
ToP   noToC   RFC5405 - Page 24
   [RFC0896]     Nagle, J., "Congestion control in IP/TCP
                 internetworks", RFC 896, January 1984.

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

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

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

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

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

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

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

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

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

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

   [RFC3261]     Rosenberg, J., Schulzrinne, H., Camarillo, G.,
                 Johnston, A., Peterson, J., Sparks, R., Handley, M.,
                 and E. Schooler, "SIP: Session Initiation Protocol",
                 RFC 3261, June 2002.
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   [RFC3303]     Srisuresh, P., Kuthan, J., Rosenberg, J., Molitor, A.,
                 and A. Rayhan, "Middlebox communication architecture
                 and framework", RFC 3303, August 2002.

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

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

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

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

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

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

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

   [RFC3851]     Ramsdell, B., "Secure/Multipurpose Internet Mail
                 Extensions (S/MIME) Version 3.1 Message Specification",
                 RFC 3851, July 2004.

   [RFC3852]     Housley, R., "Cryptographic Message Syntax (CMS)",
                 RFC 3852, July 2004.

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

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

   [RFC4303]     Kent, S., "IP Encapsulating Security Payload (ESP)",
                 RFC 4303, December 2005.
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   [RFC4306]     Kaufman, C., "Internet Key Exchange (IKEv2) Protocol",
                 RFC 4306, December 2005.

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

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

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

   [RFC4347]     Rescorla, E. and N. Modadugu, "Datagram Transport Layer
                 Security", RFC 4347, April 2006.

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

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

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

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

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

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

   [STEVENS]     Stevens, W., Fenner, B., and A. Rudoff, "UNIX Network
                 Programming, The sockets Networking API", Addison-
                 Wesley, 2004.

   [TRILOGY]     "Trilogy Project", <>.
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   [UPnP]        UPnP Forum, "Internet Gateway Device (IGD) Standardized
                 Device Control Protocol V 1.0", November 2001.

Authors' Addresses

Lars Eggert Nokia Research Center P.O. Box 407 Nokia Group 00045 Finland Phone: +358 50 48 24461 EMail: URI: Godred Fairhurst University of Aberdeen Department of Engineering Fraser Noble Building Aberdeen AB24 3UE Scotland EMail: URI: