Internet DRAFT - draft-eggert-tsvwg-rfc5405bis
draft-eggert-tsvwg-rfc5405bis
Transport Area Working Group L. Eggert
Internet-Draft NetApp
Obsoletes: 5405 (if approved) G. Fairhurst
Intended status: Best Current Practice University of Aberdeen
Expires: December 19, 2014 G. Shepherd
Cisco Systems
June 17, 2014
UDP Usage Guidelines
draft-eggert-tsvwg-rfc5405bis-01
Abstract
The User Datagram Protocol (UDP) provides a minimal message-passing
transport that has no inherent congestion control mechanisms.
Because congestion control is critical to the stable operation of the
Internet, applications and other protocols that choose to use UDP as
an Internet transport must employ mechanisms to prevent congestion
collapse and to establish some degree of fairness with concurrent
traffic. They may also need to implement additional mechanisms,
depending on how they use UDP.
This document provides guidelines on the use of UDP for the designers
of applications, tunnels and other protocols that use UDP.
Congestion control guidelines are a primary focus, but the document
also provides guidance on other topics, including message sizes,
reliability, checksums, and middlebox traversal.
If published as an RFC, this document will obsolete RFC5405.
Status of This Memo
This Internet-Draft is submitted in full conformance with the
provisions of BCP 78 and BCP 79.
Internet-Drafts are working documents of the Internet Engineering
Task Force (IETF). Note that other groups may also distribute
working documents as Internet-Drafts. The list of current Internet-
Drafts is at http://datatracker.ietf.org/drafts/current/.
Internet-Drafts are draft documents valid for a maximum of six months
and may be updated, replaced, or obsoleted by other documents at any
time. It is inappropriate to use Internet-Drafts as reference
material or to cite them other than as "work in progress."
This Internet-Draft will expire on December 19, 2014.
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Copyright Notice
Copyright (c) 2014 IETF Trust and the persons identified as the
document authors. All rights reserved.
This document is subject to BCP 78 and the IETF Trust's Legal
Provisions Relating to IETF Documents
(http://trustee.ietf.org/license-info) in effect on the date of
publication of this document. Please review these documents
carefully, as they describe your rights and restrictions with respect
to this document. Code Components extracted from this document must
include Simplified BSD License text as described in Section 4.e of
the Trust Legal Provisions and are provided without warranty as
described in the Simplified BSD License.
This document may not be modified, and derivative works of it may not
be created, and it may not be published except as an Internet-Draft.
Table of Contents
1. Introduction . . . . . . . . . . . . . . . . . . . . . . . . 2
2. Terminology . . . . . . . . . . . . . . . . . . . . . . . . . 4
3. UDP Usage Guidelines . . . . . . . . . . . . . . . . . . . . 4
3.1. Congestion Control Guidelines . . . . . . . . . . . . . . 5
3.2. Message Size Guidelines . . . . . . . . . . . . . . . . . 12
3.3. Reliability Guidelines . . . . . . . . . . . . . . . . . 14
3.4. Checksum Guidelines . . . . . . . . . . . . . . . . . . . 15
3.5. Middlebox Traversal Guidelines . . . . . . . . . . . . . 17
4. Multicast UDP Usage Guidelines . . . . . . . . . . . . . . . 19
4.1. Multicast Congestion Control Guidelines . . . . . . . . . 20
4.2. Message Size Guidelines for Multicast . . . . . . . . . . 22
5. Programming Guidelines . . . . . . . . . . . . . . . . . . . 22
5.1. Using UDP Ports . . . . . . . . . . . . . . . . . . . . . 23
5.2. ICMP Guidelines . . . . . . . . . . . . . . . . . . . . . 25
6. Security Considerations . . . . . . . . . . . . . . . . . . . 26
7. Summary . . . . . . . . . . . . . . . . . . . . . . . . . . . 27
8. IANA Considerations . . . . . . . . . . . . . . . . . . . . . 29
9. Acknowledgments . . . . . . . . . . . . . . . . . . . . . . . 29
10. References . . . . . . . . . . . . . . . . . . . . . . . . . 29
10.1. Normative References . . . . . . . . . . . . . . . . . . 29
10.2. Informative References . . . . . . . . . . . . . . . . . 30
Appendix A. Revision Notes . . . . . . . . . . . . . . . . . . . 36
1. Introduction
The User Datagram Protocol (UDP) [RFC0768] provides a minimal,
unreliable, best-effort, message-passing transport to applications
and other protocols (such as tunnels) that desire to operate over UDP
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(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 tear-down 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.
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
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fragmentation. Fragmentation decreases communication reliability and
efficiency and should be avoided. IPv6 allows the option of
transmitting large packets ("jumbograms") without fragmentation when
all link layers along the path support this [RFC2675]. Some of the
guidelines in Section 3 describe how applications should determine
appropriate message sizes. Other sections of this document provide
guidance on reliability, checksums, and middlebox traversal.
This document provides guidelines and recommendations. Although most
UDP applications are expected to follow these guidelines, there do
exist valid reasons why a specific application may decide not to
follow a given guideline. In such cases, it is RECOMMENDED that
application designers cite the respective section(s) of this document
in the technical specification of their application or protocol and
explain their rationale for their design choice.
[RFC5405] was scoped to provide guidelines for unicast applications
only, whereas this document also provides guidelines for UDP flows
that use IP anycast, multicast and broadcast, and applications that
use UDP tunnels to support IP flows.
Finally, although this document specifically refers to applications
that use UDP, the spirit of some of its guidelines also applies to
other message-passing applications and protocols (specifically on the
topics of congestion control, message sizes, and reliability).
Examples include signaling or control applications that choose to run
directly over IP by registering their own IP protocol number with
IANA. This document may provide useful background reading to the
designers of such applications and protocols.
2. Terminology
The key words "MUST", "MUST NOT", "REQUIRED", "SHALL", "SHALL NOT",
"SHOULD", "SHOULD NOT", "RECOMMENDED", "NOT RECOMMENDED", "MAY", and
"OPTIONAL" in this document are to be interpreted as described in
[RFC2119].
3. UDP Usage Guidelines
Internet paths can have widely varying characteristics, including
transmission delays, available bandwidths, congestion levels,
reordering probabilities, supported message sizes, or loss rates.
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
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Internet path they communicate over to establish a transmission
behavior that it can sustain and that is reasonably fair to other
traffic sharing the path.
These mechanisms are difficult to implement correctly. For most
applications, the use of one of the existing IETF transport protocols
is the simplest method of acquiring the required mechanisms.
Consequently, the RECOMMENDED alternative to the UDP usage described
in the remainder of this section is the use of an IETF transport
protocol such as TCP [RFC0793], Stream Control Transmission Protocol
(SCTP) [RFC4960], and SCTP Partial Reliability Extension (SCTP-PR)
[RFC3758], or Datagram Congestion Control Protocol (DCCP) [RFC4340]
with its different congestion control types
[RFC4341][RFC4342][RFC5622].
If used correctly, these more fully-featured transport protocols are
not as "heavyweight" as often claimed. For example, the TCP
algorithms have been continuously improved over decades, and have
reached a level of efficiency and correctness that custom
application-layer mechanisms will struggle to easily duplicate. In
addition, many TCP implementations allow connections to be tuned by
an application to its purposes. For example, TCP's "Nagle" algorithm
[RFC0896] can be disabled, improving communication latency at the
expense of more frequent -- but still congestion-controlled -- packet
transmissions. Another example is the TCP SYN cookie mechanism
[RFC4987], which is available on many platforms. TCP with SYN
cookies does not require a server to maintain per-connection state
until the connection is established. TCP also requires the end that
closes a connection to maintain the TIME-WAIT state that prevents
delayed segments from one connection instance from interfering with a
later one. Applications that are aware of and designed for this
behavior can shift maintenance of the TIME-WAIT state to conserve
resources by controlling which end closes a TCP connection [FABER].
Finally, TCP's built-in capacity-probing and awareness of the maximum
transmission unit supported by the path (PMTU) results in efficient
data transmission that quickly compensates for the initial connection
setup delay, in the case of transfers that exchange more than a few
segments.
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
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processes or otherwise uses multiple sockets to generate UDP
datagrams SHOULD perform congestion control over the aggregate
traffic.
Several approaches to perform congestion control are discussed in the
remainder of this section. The section describes generic topics with
an intended emphasis on unicast and anycast [RFC1546] usage. Not all
approaches discussed below are appropriate for all UDP-transmitting
applications. Section 3.1.1 discusses congestion control options for
applications that perform bulk transfers over UDP. Such applications
can employ schemes that sample the path over several subsequent RTTs
during which data is exchanged, in order to determine a sending rate
that the path at its current load can support. Other applications
only exchange a few UDP datagrams with a destination. Section 3.1.2
discusses congestion control options for such "low data-volume"
applications. Because they typically do not transmit enough data to
iteratively sample the path to determine a safe sending rate, they
need to employ different kinds of congestion control mechanisms.
Section 3.1.6 discusses congestion control considerations when UDP is
used as a tunneling protocol. Section 4 provides additional
recommendations for broadcast and multicast usage.
UDP applications may take advantage of Explicit Congestion
Notification (ECN), providing that the application programming
interface can support ECN and the congestion control can
appropriately react to ECN-marked packets. [RFC6679] provides
guidance on how to use ECN for UDP-based applications using the Real-
Time Protocol (RTP).
It is important to note that congestion control should not be viewed
as an add-on to a finished application. Many of the mechanisms
discussed in the guidelines below require application support to
operate correctly. Application designers need to consider congestion
control throughout the design of their application, similar to how
they consider security aspects throughout the design process.
In the past, the IETF has also investigated integrated congestion
control mechanisms that act on the traffic aggregate between two
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].
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3.1.1. Bulk Transfer Applications
Applications that perform bulk transmission of data to a peer over
UDP, i.e., applications that exchange more than a few UDP datagrams
per RTT, SHOULD implement TCP-Friendly Rate Control (TFRC) [RFC5348],
window-based TCP-like congestion control, or otherwise ensure that
the application complies with the congestion control principles.
TFRC has been designed to provide both congestion control and
fairness in a way that is compatible with the IETF's other transport
protocols. If an application implements TFRC, it need not follow the
remaining guidelines in Section 3.1.1, because TFRC already addresses
them, but SHOULD still follow the remaining guidelines in the
subsequent subsections of Section 3.
Bulk transfer applications that choose not to implement TFRC or TCP-
like windowing SHOULD implement a congestion control scheme that
results in bandwidth use that competes fairly with TCP within an
order of magnitude. Section 2 of [RFC3551] suggests that
applications SHOULD monitor the packet loss rate to ensure that it is
within acceptable parameters. Packet loss is considered acceptable
if a TCP flow across the same network path under the same network
conditions would achieve an average throughput, measured on a
reasonable timescale, that is not less than that of the UDP flow.
The comparison to TCP cannot be specified exactly, but is intended as
an "order-of-magnitude" comparison in timescale and throughput.
Finally, some bulk transfer applications may choose not to implement
any congestion control mechanism and instead rely on transmitting
across reserved path capacity. This might be an acceptable choice
for a subset of restricted networking environments, but is by no
means a safe practice for operation over the wider Internet. When
the UDP traffic of such applications leaks out into unprovisioned
Internet paths, it can significantly degrade the performance of other
traffic sharing the path and even result in congestion collapse.
Applications that support an uncontrolled or unadaptive transmission
behavior SHOULD NOT do so by default and SHOULD instead require users
to explicitly enable this mode of operation.
3.1.2. Low Data-Volume Applications
When applications that at any time exchange only a few UDP datagrams
with a destination implement TFRC or one of the other congestion
control schemes in Section 3.1.1, the network sees little benefit,
because those mechanisms perform congestion control in a way that is
only effective for longer transmissions.
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Applications that at any time exchange only a few UDP datagrams with
a destination SHOULD still control their transmission behavior by not
sending on average more than one UDP datagram per round-trip time
(RTT) to a destination. Similar to the recommendation in [RFC1536],
an application SHOULD maintain an estimate of the RTT for any
destination with which it communicates. Applications SHOULD
implement the algorithm specified in [RFC6298] to compute a smoothed
RTT (SRTT) estimate. They SHOULD also detect packet loss and
exponentially back their retransmission timer off when a loss event
occurs. When implementing this scheme, applications need to choose a
sensible initial value for the RTT. This value SHOULD generally be
as conservative as possible for the given application. TCP uses an
initial value of 3 seconds [RFC6298], which is also RECOMMENDED as an
initial value for UDP applications. SIP [RFC3261] and GIST [RFC5971]
use an initial value of 500 ms, and initial timeouts that are shorter
than this are likely problematic in many cases. It is also important
to note that the initial timeout is not the maximum possible timeout
-- the RECOMMENDED algorithm in [RFC6298] yields timeout values after
a series of losses that are much longer than the initial value.
Some applications cannot maintain a reliable RTT estimate for a
destination. The first case is that of applications that exchange
too few UDP datagrams with a peer to establish a statistically
accurate RTT estimate. Such applications MAY use a predetermined
transmission interval that is exponentially backed-off when packets
are lost. TCP uses an initial value of 3 seconds [RFC6298], which is
also RECOMMENDED as an initial value for UDP applications. SIP
[RFC3261] and GIST [RFC5971] use an interval of 500 ms, and shorter
values are likely problematic in many cases. As in the previous
case, note that the initial timeout is not the maximum possible
timeout.
A second class of applications cannot maintain an RTT estimate for a
destination, because the destination does not send return traffic.
Such applications SHOULD NOT send more than one UDP datagram every 3
seconds, and SHOULD use an even less aggressive rate when possible.
The 3-second interval was chosen based on TCP's retransmission
timeout when the RTT is unknown [RFC6298], and shorter values are
likely problematic in many cases. Note that the sending rate in this
case must be more conservative than in the two previous cases,
because the lack of return traffic prevents the detection of packet
loss, i.e., congestion, and the application therefore cannot perform
exponential back-off to reduce load.
Applications that communicate bidirectionally SHOULD employ
congestion control for both directions of the communication. For
example, for a client-server, request-response-style application,
clients SHOULD congestion-control their request transmission to a
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server, and the server SHOULD congestion-control its responses to the
clients. Congestion in the forward and reverse direction is
uncorrelated, and an application SHOULD either independently detect
and respond to congestion along both directions, or limit new and
retransmitted requests based on acknowledged responses across the
entire round-trip path.
3.1.3. Burst Mitigation and Pacing
UDP applications SHOULD provide mechanisms to regulate the bursts of
transmission that the application may send to the network. Many TCP
and SCTP implementations provide mechanisms that prevent a sender
from generating long bursts at line-rate, since these are known to
induce early loss to applications sharing a common network
bottleneck. The use of pacing with TCP has also been shown to
improve the coexistence of TCP flows with other flows.
Even low data-volume UDP flows may benefit from rate control, e.g.,
an application that sends three copies of a packet to improve
robustness to loss is RECOMMENDED to pace out those three packets
over several RTTs, to reduce the probability that all three packets
will be lost due to the same congestion event.
3.1.4. QoS, Pre-Provisioned or Reserved Capacity
An application using UDP can use the differentiated services and
integrated services QoS frameworks. These are usually available
within controlled environments (e.g., within a single administrative
domain or bilaterally agreed connection between domains).
Applications intended for the Internet should not assume that QoS
mechanisms are supported by the networks they use, and therefore need
to provide congestion control, error recovery, etc. in case the
actual network path does not provide provisioned service.
Some UDP applications are only expected to be deployed over network
paths that use pre-provisioned capacity or capacity reserved using
dynamic provisioning, e.g., through the Resource Reservation Protocol
(RSVP). Multicast applications are also used with pre-provisioned
capacity (e.g., IPTV deployments within access networks). These
applications MAY choose not to implement any congestion control
mechanism and instead rely on transmitting only on paths where the
capacity is provisioned and reserved for this use. This might be an
acceptable choice for a subset of restricted networking environments,
but is by no means a safe practice for operation over the wider
Internet.
If the traffic of such applications leaks out into unprovisioned
Internet paths, it can significantly degrade the performance of other
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traffic sharing the path and even result in congestion collapse. For
this reason, and to protect other applications sharing the same path,
applications SHOULD deploy an appropriate circuit breaker, as
described in Section 3.1.5. 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 used in networks within a controlled environment may be
able to exploit network management functions to detect whether they
are causing congestion, and react accordingly.
3.1.5. Circuit Breaker Mechanisms
A transport circuit breaker is an automatic mechanism that is used to
estimate the congestion caused by a flow, and to terminate (or
significantly reduce the rate of) the flow when excessive congestion
is detected [I-D.fairhurst-tsvwg-circuit-breaker]. This is a safety
measure to prevent congestion collapse (starvation of resources
available to other flows), essential for an Internet that is
heterogeneous and for traffic that is hard to predict in advance.
A circuit breaker is intended as a protection mechanism of last
resort. Under normal circumstances, a circuit breaker should not be
triggered; it is designed to protect things when there is severe
overload. The goal is usually to limit the maximum transmission rate
that reflects the available capacity of a network path. circuit
breakers can operate on individual UDP flows or traffic aggregates,
e.g., traffic sent using a network tunnel. Later sections provide
examples of cases where circuit breakers may or may not be desirable.
[I-D.fairhurst-tsvwg-circuit-breaker] provides guidance on the use of
circuit breakers and examples of usage. The use of a circuit breaker
in RTP is specified in [I-D.ietf-avtcore-rtp-circuit-breakers].
3.1.6. 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.
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Well-implemented tunnels are generally invisible to the endpoints
that happen to transmit over a path that includes tunneled links. On
the other hand, to the routers along the path of a UDP tunnel, i.e.,
the routers between the two tunnel endpoints, the traffic that a UDP
tunnel generates is a regular UDP flow, and the encapsulator and
decapsulator appear as regular UDP-sending and -receiving
applications. Because other flows can share the path with one or
more UDP tunnels, congestion control needs to be considered.
Two factors determine whether a UDP tunnel needs to employ specific
congestion control mechanisms -- first, whether the payload traffic
is IP-based; second, whether the tunneling scheme generates UDP
traffic at a volume that corresponds to the volume of payload traffic
carried within the tunnel.
IP-based traffic is generally assumed to be congestion-controlled,
i.e., it is assumed that the transport protocols generating IP-based
traffic at the sender already employ mechanisms that are sufficient
to address congestion on the path. Consequently, a tunnel carrying
IP-based traffic should already interact appropriately with other
traffic sharing the path, and specific congestion control mechanisms
for the tunnel are not necessary.
However, if the IP traffic in the tunnel is known to not be
congestion-controlled, additional measures are RECOMMENDED in order
to limit the impact of the tunneled traffic on other traffic sharing
the path.
The following guidelines define these possible cases in more detail:
1. A tunnel generates UDP traffic at a volume that corresponds to
the volume of payload traffic, and the payload traffic is IP-
based and congestion-controlled.
This is arguably the most common case for Internet tunnels. In
this case, the UDP tunnel SHOULD NOT employ its own congestion
control mechanism, because congestion losses of tunneled traffic
will already trigger an appropriate congestion response at the
original senders of the tunneled traffic.
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
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known to be IP-based, or is known to be IP-based but not
congestion-controlled.
This can be the case, for example, when some link-layer protocols
are encapsulated within UDP (but not all link-layer protocols;
some are congestion-controlled). Because it is not known that
congestion losses of tunneled non-IP traffic will trigger an
appropriate congestion response at the senders, the UDP tunnel
SHOULD employ an appropriate congestion control mechanism.
Because tunnels are usually bulk-transfer applications as far as
the intermediate routers are concerned, the guidelines in
Section 3.1.1 apply.
3. A tunnel generates UDP traffic at a volume that does not
correspond to the volume of payload traffic, independent of
whether the payload traffic is IP-based or congestion-controlled.
Examples of this class include UDP tunnels that send at a
constant rate, increase their transmission rates under loss, for
example, due to increasing redundancy when Forward Error
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.
Designing a tunneling mechanism requires significantly more expertise
than needed for many other UDP applications, because tunnels are
usually intended to be transparent to the endpoints transmitting over
them, so they need to correctly emulate the behavior of an IP link,
e.g., handling fragmentation, generating and responding to ICMP
messages, etc. 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
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performed by a NAT only operates on complete IP packets, and some
firewall policies also require inspection of complete IP packets.
Even with these being the case, some NATs and firewalls simply do not
implement the necessary reassembly functionality, and instead choose
to drop all fragments. Finally, [RFC4963] documents other issues
specific to IPv4 fragmentation.
Due to these issues, an application SHOULD NOT send UDP datagrams
that result in IP packets that exceed the MTU of the path to the
destination. Consequently, an application SHOULD either use the path
MTU information provided by the IP layer or implement path MTU
discovery itself [RFC1191][RFC1981][RFC4821] to determine whether the
path to a destination will support its desired message size without
fragmentation.
Applications that do not follow this recommendation to do PMTU
discovery SHOULD still avoid sending UDP datagrams that would result
in IP packets that exceed the path MTU. Because the actual path MTU
is unknown, such applications SHOULD fall back to sending messages
that are shorter than the default effective MTU for sending (EMTU_S
in [RFC1122]). For IPv4, EMTU_S is the smaller of 576 bytes and the
first-hop MTU [RFC1122]. For IPv6, EMTU_S is 1280 bytes [RFC2460].
The effective PMTU for a directly connected destination (with no
routers on the path) is the configured interface MTU, which could be
less than the maximum link payload size. Transmission of minimum-
sized UDP datagrams is inefficient over paths that support a larger
PMTU, which is a second reason to implement PMTU discovery.
To determine an appropriate UDP payload size, applications MUST
subtract the size of the IP header (which includes any IPv4 optional
headers or IPv6 extension headers) as well as the length of the UDP
header (8 bytes) from the PMTU size. This size, known as the MSS,
can be obtained from the TCP/IP stack [RFC1122].
Applications that do not send messages that exceed the effective PMTU
of IPv4 or IPv6 need not implement any of the above mechanisms. Note
that the presence of tunnels can cause an additional reduction of the
effective PMTU, so implementing PMTU discovery may be beneficial.
Applications that fragment an application-layer message into multiple
UDP datagrams SHOULD perform this fragmentation so that each datagram
can be received independently, and be independently retransmitted in
the case where an application implements its own reliability
mechanisms.
Packetization Layer Path MTU Discovery (PLPMTUD) [RFC4821] does not
rely upon network support for ICMP messages and is therefore
considered more robust than standard PMTUD. To operate, PLPMTUD
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requires changes to the way the transport is used, both to transmit
probe packets, and to account for the loss or success of these
probes. This updates not only the PMTU algorithm, it also impacts
loss recovery, congestion control, etc. These updated mechanisms can
be implemented within a connection-oriented transport (e.g., TCP,
SCTP, DCCP), but are not a part of UDP. PLPMTUD therefore places
additional design requirements on a UDP application that wishes to
use this method.
3.3. Reliability Guidelines
Application designers are generally aware that UDP does not provide
any reliability, e.g., it does not retransmit any lost packets.
Often, this is a main reason to consider UDP as a transport.
Applications that do require reliable message delivery MUST implement
an appropriate mechanism themselves.
UDP also does not protect against datagram duplication, i.e., an
application may receive multiple copies of the same UDP datagram,
with some duplicates arriving potentially much later than the first.
Application designers SHOULD verify that their application handles
such datagram duplication gracefully, and may consequently need to
implement mechanisms to detect duplicates. Even if UDP datagram
reception triggers only idempotent operations, applications may want
to suppress duplicate datagrams to reduce load.
Applications that require ordered delivery MUST reestablish datagram
ordering themselves. The Internet can significantly delay some
packets with respect to others, e.g., due to routing transients,
intermittent connectivity, or mobility. This can cause reordering,
where UDP datagrams arrive at the receiver in an order different from
the transmission order.
It is important to note that the time by which packets are reordered
or after which duplicates can still arrive can be very large. Even
more importantly, there is no well-defined upper boundary here.
[RFC0793] defines the maximum delay a TCP segment should experience
-- the Maximum Segment Lifetime (MSL) -- as 2 minutes. No other RFC
defines an MSL for other transport protocols or IP itself. The MSL
value defined for TCP is conservative enough that it SHOULD be used
by other protocols, including UDP. Therefore, applications SHOULD be
robust to the reception of delayed or duplicate packets that are
received within this 2-minute interval.
Instead of implementing these relatively complex reliability
mechanisms by itself, an application that requires reliable and
ordered message delivery SHOULD whenever possible choose an IETF
standard transport protocol that provides these features.
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3.4. Checksum Guidelines
The UDP header includes an optional, 16-bit one's complement checksum
that provides an integrity check. These checks are not strong from a
coding or cryptographic perspective, and are not designed to detect
physical-layer errors or malicious modification of the datagram
[RFC3819]. Application developers SHOULD implement additional checks
where data integrity is important, e.g., through a Cyclic Redundancy
Check (CRC) included with the data to verify the integrity of an
entire object/file sent over the UDP service.
The UDP checksum provides a statistical guarantee that the payload
was not corrupted in transit. It also allows the receiver to verify
that it was the intended destination of the packet, because it covers
the IP addresses, port numbers, and protocol number, and it verifies
that the packet is not truncated or padded, because it covers the
size field. It therefore protects an application against receiving
corrupted payload data in place of, or in addition to, the data that
was sent. More description of the set of checks performed using the
checksum field are provided in Section 3.1 of [RFC6396].
Applications SHOULD enable UDP checksums. For IPv4, [RFC0768]
permits the option to disable their use. The use of the UDP checksum
was required when applications transmit UDP over IPv6 [RFC2460].
This requirement was updated in [RFC6395], but only for specific
protocols and applications, and the implementation of the set of
functions defined in [RFC6396] is then REQUIRED. These additional
design requirements for using a zero IPv6 UDP checksum [RFC6396] are
not present for IPv4, since the network-layer header validates
information that is not protected for an IPv6 packet.
Applications that choose to disable UDP checksums when transmitting
over IPv4 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.
3.4.1. UDP-Lite
A special class of applications can derive benefit from having
partially-damaged payloads delivered, rather than discarded, when
using paths that include error-prone links. Such applications can
tolerate payload corruption and MAY choose to use the Lightweight
User Datagram Protocol (UDP-Lite) [RFC3828] variant of UDP instead of
basic UDP. Applications that choose to use UDP-Lite instead of UDP
should still follow the congestion control and other guidelines
described for use with UDP in Section 3.
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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.
A UDP-Lite sender SHOULD select the minimum checksum coverage to
include all sensitive payload information. For example, applications
that use the Real-Time Protocol (RTP) [RFC3550] will likely want to
protect the RTP header against corruption. Applications, where
appropriate, MUST also introduce their own appropriate validity
checks for protocol information carried in the insensitive part of
the UDP-Lite payload (e.g., internal CRCs).
A UDP-Lite receiver MUST set a minimum coverage threshold for
incoming packets that is not smaller than the smallest coverage used
by the sender [RFC3828]. The receiver SHOULD select a threshold that
is sufficiently large to block packets with an inappropriately short
coverage field. This may be a fixed value, or may be negotiated by
an application. UDP-Lite does not provide mechanisms to negotiate
the checksum coverage between the sender and receiver.
Applications can still experience packet loss when using UDP-Lite.
The enhancements offered by UDP-Lite rely upon a link being able to
intercept the UDP-Lite header to correctly identify the partial
coverage required. When tunnels and/or encryption are used, this can
result in UDP-Lite datagrams being treated the same as UDP datagrams,
i.e., result in packet loss. Use of IP fragmentation can also
prevent special treatment for UDP-Lite datagrams, and this is another
reason why applications SHOULD avoid IP fragmentation (Section 3.2).
Current support for middlebox traversal using UDP-Lite is poor,
because UDP-Lite uses a different IPv4 protocol number or IPv6 "next
header" value than that used for UDP; therefore, few middleboxes are
currently able to interpret UDP-Lite and take appropriate actions
when forwarding the packet. This makes UDP-Lite less suited for
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applications needing general Internet support, until such time as
UDP-Lite has achieved better support in middleboxes and endpoints.
3.5. Middlebox Traversal Guidelines
Network address translators (NATs) and firewalls are examples of
intermediary devices ("middleboxes") that can exist along an end-to-
end path. A middlebox typically performs a function that requires it
to maintain per-flow state. For connection-oriented protocols, such
as TCP, middleboxes snoop and parse the connection-management
information and create and destroy per-flow state accordingly. For a
connectionless protocol such as UDP, this approach is not possible.
Consequently, middleboxes may create per-flow state when they see a
packet that -- according to some local criteria -- indicates a new
flow, and destroy the state after some period of time during which no
packets belonging to the same flow have arrived.
Depending on the specific function that the middlebox performs, this
behavior can introduce a time-dependency that restricts the kinds of
UDP traffic exchanges that will be successful across the middlebox.
For example, NATs and firewalls typically define the partial path on
one side of them to be interior to the domain they serve, whereas the
partial path on their other side is defined to be exterior to that
domain. Per-flow state is typically created when the first packet
crosses from the interior to the exterior, and while the state is
present, NATs and firewalls will forward return traffic. Return
traffic that arrives after the per-flow state has timed out is
dropped, as is other traffic that arrives from the exterior.
Many applications that use UDP for communication operate across
middleboxes without needing to employ additional mechanisms. One
example is the Domain Name System (DNS), which has a strict request-
response communication pattern that typically completes within
seconds.
Other applications may experience communication failures when
middleboxes destroy the per-flow state associated with an application
session during periods when the application does not exchange any UDP
traffic. Applications SHOULD be able to gracefully handle such
communication failures and implement mechanisms to re-establish
application-layer sessions and state.
For some applications, such as media transmissions, this re-
synchronization is highly undesirable, because it can cause user-
perceivable playback artifacts. Such specialized applications MAY
send periodic keep-alive messages to attempt to refresh middlebox
state. It is important to note that keep-alive messages are NOT
RECOMMENDED for general use -- they are unnecessary for many
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applications and can consume significant amounts of system and
network resources.
An application that needs to employ keep-alives to deliver useful
service over UDP in the presence of middleboxes SHOULD NOT transmit
them more frequently than once every 15 seconds and SHOULD use longer
intervals when possible. No common timeout has been specified for
per-flow UDP state for arbitrary middleboxes. NATs require a state
timeout of 2 minutes or longer [RFC4787]. However, empirical
evidence suggests that a significant fraction of currently deployed
middleboxes unfortunately use shorter timeouts. The timeout of 15
seconds originates with the Interactive Connectivity Establishment
(ICE) protocol [RFC5245]. When an application is deployed in a
controlled network environment, the deployer SHOULD investigate
whether the target environment allows applications to use longer
intervals, or whether it offers mechanisms to explicitly control
middlebox state timeout durations, for example, using Middlebox
Communications (MIDCOM) [RFC3303], Next Steps in Signaling (NSIS)
[RFC5973], or Universal Plug and Play (UPnP) [UPnP]. It is
RECOMMENDED that applications apply slight random variations
("jitter") to the timing of keep-alive transmissions, to reduce the
potential for persistent synchronization between keep-alive
transmissions from different hosts.
Sending keep-alives is not a substitute for implementing a mechanism
to recover from broken sessions. Like all UDP datagrams, keep-alives
can be delayed or dropped, causing middlebox state to time out. In
addition, the congestion control guidelines in Section 3.1 cover all
UDP transmissions by an application, including the transmission of
middlebox keep-alives. Congestion control may thus lead to delays or
temporary suspension of keep-alive transmission.
Keep-alive messages are NOT RECOMMENDED for general use. They are
unnecessary for many applications and may consume significant
resources. For example, on battery-powered devices, if an
application needs to maintain connectivity for long periods with
little traffic, the frequency at which keep-alives are sent can
become the determining factor that governs power consumption,
depending on the underlying network technology. Because many
middleboxes are designed to require keep-alives for TCP connections
at a frequency that is much lower than that needed for UDP, this
difference alone can often be sufficient to prefer TCP over UDP for
these deployments. On the other hand, there is anecdotal evidence
that suggests that direct communication through middleboxes, e.g., by
using ICE [RFC5245], does succeed less often with TCP than with UDP.
The trade-offs between different transport protocols -- especially
when it comes to middlebox traversal -- deserve careful analysis.
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UDP applications need to be designed understanding that there are
many variants of middlebox behavior, and although UDP is connection-
less, middleboxes often maintain state for each UDP flow. Using
multiple 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).
This has implications on applications that use multiple UDP flows in
parallel, even on multiple ports Section 5.1.1.
4. Multicast UDP Usage Guidelines
This section complements Section 3 by providing additional guidelines
that are applicable to multicast and broacast usage of UDP.
Multicast and broadcast transmission [RFC1112] usually employ the UDP
transport protocol, although they may be used with other transport
protocols (e.g., UDP-Lite).
There are currently two models of multicast delivery: the Any-Source
Multicast (ASM) model as defined in [RFC1112] and the Source-Specific
Multicast (SSM) model as defined in [RFC4607]. ASM group members
will receive all data sent to the group by any source, while SSM
constrains the distribution tree to only one single source.
Specialized classes of applications also use UDP for IP multicast or
broadcast [RFC0919]. The design of such specialized applications
requires expertise that goes beyond simple, unicast-specific
guidelines, since these senders may transmit to potentially very many
receivers across potentially very heterogeneous paths at the same
time, which significantly complicates congestion control, flow
control, and reliability mechanisms. This section provides guidance
on multicast UDP usage.
Use of broadcast by an application is normally constrained by routers
to the local subnetwork. However, use of tunneling techniques and
proxies can and does result in some broadcast traffic traversing
Internet paths. These guidelines therefore also apply to broadcast
traffic.
The IETF has defined a reliable multicast framework [RFC3048] and
several building blocks to aid the designers of multicast
applications, such as [RFC3738] or [RFC4654]. Anycast senders must
be aware that successive messages sent to the same anycast IP address
may be delivered to different anycast nodes, i.e., arrive at
different locations in the topology.
Most UDP tunnels that carry IP multicast traffic use a tunnel
encapsulation with a unicast destination address. These MUST follow
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the same requirements as a tunnel carrying unicast data (see
Section 3.1.6). There are deployment cases and solutions where the
outer header of a UDP tunnel contains a multicast destination
address, such as [RFC6513]. These cases are primarily deployed in
controlled environments over reserved capacity, often operating
within a single administrative domain, or between two domains over a
bi-laterally agreed upon path with reserved bandwidth, and so
congestion control is OPTIONAL, but circuit breaker techniques are
still RECOMMENDED in order to restore some degree of service should
the offered load exceed the reserved capacity (e.g., due to
misconfiguration).
4.1. Multicast Congestion Control Guidelines
Unicast congestion-controlled transport mechanism are often not
applicable to multicast distribution services, or simply do not scale
to large multicast trees, since they require bi-directional
communication and adapt the sending rate to accommodate the network
conditions to a single receiver. In contrast, multicast distribution
trees may fan out to massive numbers of receivers, which limits the
scalability of an in-band return channel to control the sending rate,
and the one-to-many nature of multicast distribution trees prevents
adapting the rate to the requirements of an individual receiver. For
this reason, generating TCP-compatible aggregate flow rates for
Internet multicast data, either native or tunneled, is the
responsibility of the application.
Congestion control mechanisms for multicast may operate on longer
timescales than for unicast (e.g., due to the higher group RTT of a
heterogeneous group); appropriate methods are particularly for any
multicast session were all or part of the multicast distribution tree
spans an access network (e.g., a home gateway).
Multicast congestion control needs to consider the potential
heterogeneity of both the multicast distribution tree and the
receivers belonging to a group. Heterogeneity may manifest itself in
some receivers experiencing more loss that others, higher delay, and/
or less ability to respond to network conditions. Any multicast-
enabled receiver may attempt to join and receive traffic from any
group. This may imply the need for rate limits on individual
receivers or the aggregate multicast service. Note there is no way
at the transport layer to prevent a join message propagating to the
next-hop router. A multicast congestion control method MAY therefore
decide not to reduce the rate of the entire multicast group in
response to a report received by a single receiver; instead it can
decide to expel each congested receiver from the multicast group and
to then distribute content to these congested receivers at a lower-
rate using unicast congestion-control. Care needs to be taken when
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this action results in many flows being simultaneously transitioned,
so that this does not result in excessive traffic exasperating
congestion and potentially contributing to congestion collapse.
Some classes of multicast applications support real-time
transmissions in which the quality of the transfer may be monitored
at the receiver. Applications that detect a significant reduction in
user quality SHOULD regard this as a congestion signal (e.g., to
leave a group using layered multicast encoding).
4.1.1. Bulk Transfer Multicast Applications
Applications that perform bulk transmission of data over a multicast
distribution tree, i.e., applications that exchange more than a few
UDP datagrams per RTT, SHOULD implement a method for congestion
control. The currently RECOMMENDED IETF methods are: Asynchronous
Layered Coding (ALC) [RFC5775], TCP-Friendly Multicast Congestion
Control (TFMCC) [RFC4654], Wave and Equation Based Rate Control
(WEBRC) [RFC3738], NACK-Oriented Reliable Multicast (NORM) transport
protocol [RFC5740], File Delivery over Unidirectional Transport
(FLUTE) [RFC6726], Real Time Protocol/Control Protocol (RTP/RTCP),
[RFC3550].
An application can alternatively implement another congestion control
schemes following the guidelines of [RFC2887] and utilizing the
framework of [RFC3048]. Bulk transfer applications that choose not
to implement , [RFC4654][RFC5775], [RFC3738], [RFC5740], [RFC6726],
or [RFC3550] SHOULD implement a congestion control scheme that
results in bandwidth use that competes fairly with TCP within an
order of magnitude.
Section 2 of [RFC3551] states that multimedia applications SHOULD
monitor the packet loss rate to ensure that it is within acceptable
parameters. Packet loss is considered acceptable if a TCP flow
across the same network path under the same network conditions would
achieve an average throughput, measured on a reasonable timescale,
that is not less than that of the UDP flow. The comparison to TCP
cannot be specified exactly, but is intended as an "order-of-
magnitude" comparison in timescale and throughput.
4.1.2. Low Data-Volume Multicast Applications
All the recommendations in Section 3.1.2 are also applicable to such
multicast applications.
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4.2. Message Size Guidelines for Multicast
A multicast application SHOULD NOT send UDP datagrams that result in
IP packets that exceed the effective MTU as described in section 3 of
[RFC6807]. Consequently, an application SHOULD either use the
effective MTU information provided by the Population Count Extensions
to Protocol Independent Multicast [RFC6807] or implement path MTU
discovery itself (see Section 3.2) to determine whether the path to
each destination will support its desired message size without
fragmentation.
5. Programming Guidelines
The de facto standard application programming interface (API) for
TCP/IP applications is the "sockets" interface [POSIX]. Some
platforms also offer applications the ability to directly assemble
and transmit IP packets through "raw sockets" or similar facilities.
This is a second, more cumbersome method of using UDP. The
guidelines in this document cover all such methods through which an
application may use UDP. Because the sockets API is by far the most
common method, the remainder of this section discusses it in more
detail.
Although the sockets API was developed for UNIX in the early 1980s, a
wide variety of non-UNIX operating systems also implement it. The
sockets API supports both IPv4 and IPv6 [RFC3493]. The UDP sockets
API differs from that for TCP in several key ways. Because
application programmers are typically more familiar with the TCP
sockets API, this section discusses these differences. [STEVENS]
provides usage examples of the UDP sockets API.
UDP datagrams may be directly sent and received, without any
connection setup. Using the sockets API, applications can receive
packets from more than one IP source address on a single UDP socket.
Some servers use this to exchange data with more than one remote host
through a single UDP socket at the same time. Many applications need
to ensure that they receive packets from a particular source address;
these applications MUST implement corresponding checks at the
application layer or explicitly request that the operating system
filter the received packets.
If a client/server application executes on a host with more than one
IP interface, the application SHOULD send any UDP responses with an
IP source address that matches the IP destination address of the UDP
datagram that carried the request (see [RFC1122], Section 4.1.3.5).
Many middleboxes expect this transmission behavior and drop replies
that are sent from a different IP address, as explained in
Section 3.5.
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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
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, 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 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 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 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.
5.1. Using UDP Ports
The rules procedures for the management of the Service Name and
Transport Protocol Port Number Registry are specified in [RFC6335].
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Recommendations for use of UDP ports are provided in
[I-D.ietf-tsvwg-port-use].
A UDP sender SHOULD NOT use a zero source port value, and a UDP
receiver should not bind to port zero. Applications SHOULD implement
corresponding receiver checks at the application layer or explicitly
request that the operating system filter the received packets to
prevent receiving packets with an arbitrary port. This measure is
designed to provide additional protection from data injection attacks
from an off-path source (where the port values may not be known).
Although the source port value is often not directly used in
multicast applications, this should still be set to a random or pre-
determined value.
The UDP port number fields have been used as a basis to design load-
balancing solutions for IPv4. This approach has also been leveraged
for IPv6 [RFC6438], but the IPv6 "flow label" [RFC6437]may also be
used as a basis for entropy for load balancing. This use of the flow
label for load balancing is consistent with the intended use,
although further clarity was needed to ensure the field can be
consistently used for this purpose. Therefore, an updated IPv6 flow
label [RFC6437] and ECMP routing [RFC6438] usage were specified.
Router vendors are encouraged to start using the flow label as a part
of the flow hash, providing support for IP-level ECMP without
requiring use of UDP. The end-to-end use of flow labels for load
balancing is a long-term solution. Even if the usage of the flow
label has been clarified, there will be a transition time before a
significant proportion of endpoints start to assign a good quality
flow label to the flows that they originate. The use of load
balancing using the transport header fields will likely continue
until widespread deployment is finally achieved.
5.1.1. Applications using Multiple UDP Ports
A single application may exchange several types of data. In some
cases, this may require multiple UDP flows (e.g., multiple sets of
flows, identified by different 5-tuples). [RFC6335] recommends
applications developers not to apply to IANA to be assigned multiple
well-known ports (user or system). This does not discuss the
implications of using multiple flows with the same well-known port or
pairs of dynamic ports (e.g., identified by a service name or
signaling protocol).
Use of multiple flows can impact 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
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relies on timeouts to remove old state, since middleboxes are
unaware when a particular flow ceases to be used by an
application.
o Using several flows at the same time may result in seeing
different network characteristics for each flow. It can not be
assumed both follow the same path (e.g., when ECMP is used,
traffic is intentionally hashed onto different parallel paths
based on the port numbers).
o Using several flows can also increase the occupancy of a binding
or lookup table in a middlebox (e.g., NAPT or Firewall) which may
cause the device to change the way it manages the flow state.
o Further, using excessive numbers of flows can degrade the ability
of congestion control to react to congestion events, unless the
congestion state is shared between all flows in a session.
Therefore, applications MUST NOT assume consistent behavior of
middleboxes when multiple UDP flows are used; many devices respond
differently as the number of ports used increases. Using multiple
flows with different QoS requirements requires applications to verify
that the expected performance is achieved using each individual flow
(five-tuple), see Section 3.1.4.
5.2. ICMP Guidelines
Applications can utilize information about ICMP error messages that
the UDP layer passes up for a variety of purposes [RFC1122].
Applications SHOULD appropriately validate the payload of ICMP
messages to ensure these are received in response to transmitted
traffic (i.e., a reported error condition that corresponds to a UDP
datagram actually sent by the application). This requires context,
such as local state about communication instances to each
destination, that although readily available in connection-oriented
transport protocols is not always maintained by UDP-based
applications. Note that not all platforms have the necessary APIs to
support this validation, and some platforms already perform this
validation internally before passing ICMP information to the
application.
Any application response to ICMP error messages SHOULD be robust to
temporary routing failures, e.g., transient ICMP "unreachable"
messages should not normally cause a communication abort.
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6. Security Considerations
UDP does not provide communications security. Applications that need
to protect their communications against eavesdropping, tampering, or
message forgery SHOULD employ end-to-end security services provided
by other IETF protocols. Applications that respond to short requests
with potentially large responses are vulnerable to amplification
attacks, and SHOULD authenticate the sender before responding. The
source IP address of a request is not a useful authenticator, because
it can easily be spoofed.
One option of securing UDP communications is with IPsec [RFC4301],
which can provide authentication for flows of IP packets through the
Authentication Header (AH) [RFC4302] and encryption and/or
authentication through the Encapsulating Security Payload (ESP)
[RFC4303]. Applications use the Internet Key Exchange (IKE)
[RFC5996] 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)
[RFC6347]. DTLS provides communication privacy by encrypting UDP
payloads. It does not protect the UDP headers. Applications can
implement DTLS without relying on support from the operating system.
Many other options for authenticating or encrypting UDP payloads
exist. For example, the GSS-API security framework [RFC2743] or
Cryptographic Message Syntax (CMS) [RFC5652] could be used to protect
UDP payloads. The IETF standard for securing RTP [RFC3550]
communication sessions over UDP is the Secure Real-time Transport
Protocol (SRTP) [RFC3711]. In some applications, a better solution
is to protect larger stand-alone objects, such as files or messages,
instead of individual UDP payloads. In these situations, CMS
[RFC5652], S/MIME [RFC5751] or OpenPGP [RFC4880] could be used. In
addition, there are many non-IETF protocols in this area.
Like congestion control mechanisms, security mechanisms are difficult
to design and implement correctly. It is hence RECOMMENDED that
applications employ well-known standard security mechanisms such as
DTLS or IPsec, rather than inventing their own.
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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.
[I-D.fairhurst-tsvwg-circuit-breaker] describes methods that can be
used to set a performance envelope that can assist in preventing
congestion collapse in the absence of congestion control or when the
congestion control fails to react to congestion events. This
document provides guidelines to designers of UDP-based applications
to congestion-control their transmissions, and does not raise any
additional security concerns.
7. Summary
This section summarizes the guidelines made in Sections 3 and 6 in a
tabular format (Table 1) for easy referencing.
+---------------------------------------------------------+---------+
| Recommendation | Section |
+---------------------------------------------------------+---------+
| MUST tolerate a wide range of Internet path conditions | 3 |
| | |
| SHOULD use a full-featured transport (TCP, SCTP, DCCP) | |
| | |
| | |
| | |
| SHOULD control rate of transmission | 3.1 |
| | |
| SHOULD perform congestion control over all traffic | |
| | |
| | |
| | |
| 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 | |
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| | |
| SHOULD back-off retransmission timers following loss | |
| | |
| | |
| | |
| for tunnels carrying IP Traffic, | 3.1.6 |
| | |
| SHOULD NOT perform congestion control | |
| | |
| | |
| | |
| for non-IP tunnels or rate not determined by traffic, | 3.1.6 |
| | |
| SHOULD perform congestion control | |
| | |
| | |
| | |
| SHOULD NOT send datagrams that exceed the PMTU, i.e., | 3.2 |
| | |
| SHOULD discover PMTU or send datagrams < minimum PMTU; | |
| Specific application mechanisms are REQUIRED if PLPMTUD | |
| is used. | |
| | |
| | |
| | |
| SHOULD handle datagram loss, duplication, reordering | 3.3 |
| | |
| SHOULD be robust to delivery delays up to 2 minutes | |
| | |
| | |
| | |
| SHOULD enable IPv4 UDP checksum | 3.4 |
| | |
| SHOULD enable IPv6 UDP checksum; Specific application | |
| mechanisms are REQUIRED if a zero IPv6 UDP checksum is | |
| used. | |
| | |
| else, MAY use UDP-Lite with suitable checksum coverage | 3.4.1 |
| | |
| | |
| | |
| SHOULD NOT always send middlebox keep-alives | 3.5 |
| | |
| MAY use keep-alives when needed (min. interval 15 sec) | |
| | |
| | |
| | |
| MUST check IP source address | 5 |
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| | |
| and, for client/server applications | |
| | |
| SHOULD send responses from src address matching request | |
| | |
| | |
| | |
| SHOULD use standard IETF security protocols when needed | 6 |
+---------------------------------------------------------+---------+
Table 1: Summary of recommendations
8. IANA Considerations
Note to RFC-Editor: please remove this entire section prior to
publication.
This document raises no IANA considerations.
9. Acknowledgments
The middlebox traversal guidelines in Section 3.5 incorporate ideas
from Section 5 of [I-D.ford-behave-app] by Bryan Ford, Pyda
Srisuresh, and Dan Kegel.
10. References
10.1. Normative References
[RFC0768] Postel, J., "User Datagram Protocol", STD 6, RFC 768,
August 1980.
[RFC0793] Postel, J., "Transmission Control Protocol", STD 7, RFC
793, September 1981.
[RFC1122] Braden, R., "Requirements for Internet Hosts -
Communication Layers", STD 3, RFC 1122, October 1989.
[RFC1191] Mogul, J. and S. Deering, "Path MTU discovery", RFC 1191,
November 1990.
[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.
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[RFC2460] Deering, S. and R. Hinden, "Internet Protocol, Version 6
(IPv6) Specification", RFC 2460, December 1998.
[RFC2914] Floyd, S., "Congestion Control Principles", BCP 41, RFC
2914, September 2000.
[RFC3828] Larzon, L-A., Degermark, M., Pink, S., Jonsson, L-E., and
G. Fairhurst, "The Lightweight User Datagram Protocol
(UDP-Lite)", RFC 3828, July 2004.
[RFC4787] Audet, F. and C. Jennings, "Network Address Translation
(NAT) Behavioral Requirements for Unicast UDP", BCP 127,
RFC 4787, January 2007.
[RFC4821] Mathis, M. and J. Heffner, "Packetization Layer Path MTU
Discovery", RFC 4821, March 2007.
[RFC5348] Floyd, S., Handley, M., Padhye, J., and J. Widmer, "TCP
Friendly Rate Control (TFRC): Protocol Specification", RFC
5348, September 2008.
[RFC5405] Eggert, L. and G. Fairhurst, "Unicast UDP Usage Guidelines
for Application Designers", BCP 145, RFC 5405, November
2008.
[RFC6298] Paxson, V., Allman, M., Chu, J., and M. Sargent,
"Computing TCP's Retransmission Timer", RFC 6298, June
2011.
10.2. Informative References
[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.
[I-D.fairhurst-tsvwg-circuit-breaker]
Fairhurst, G., "Network Transport Circuit Breakers",
draft-fairhurst-tsvwg-circuit-breaker-01 (work in
progress), May 2014.
[I-D.ford-behave-app]
Ford, B., "Application Design Guidelines for Traversal
through Network Address Translators", draft-ford-behave-
app-05 (work in progress), March 2007.
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[I-D.ietf-avtcore-rtp-circuit-breakers]
Perkins, C. and V. Singh, "Multimedia Congestion Control:
Circuit Breakers for Unicast RTP Sessions", draft-ietf-
avtcore-rtp-circuit-breakers-05 (work in progress),
February 2014.
[I-D.ietf-tsvwg-port-use]
Touch, J., "Recommendations for Transport Port Uses",
draft-ietf-tsvwg-port-use-04 (work in progress), May 2014.
[POSIX] IEEE Std. 1003.1-2001, , "Standard for Information
Technology - Portable Operating System Interface (POSIX)",
Open Group Technical Standard: Base Specifications Issue
6, ISO/IEC 9945:2002, December 2001.
[RFC0896] Nagle, J., "Congestion control in IP/TCP internetworks",
RFC 896, January 1984.
[RFC0919] Mogul, J., "Broadcasting Internet Datagrams", STD 5, RFC
919, October 1984.
[RFC1112] Deering, S., "Host extensions for IP multicasting", STD 5,
RFC 1112, August 1989.
[RFC1536] Kumar, A., Postel, J., Neuman, C., Danzig, P., and S.
Miller, "Common DNS Implementation Errors and Suggested
Fixes", RFC 1536, October 1993.
[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.
[RFC2675] Borman, D., Deering, S., and R. Hinden, "IPv6 Jumbograms",
RFC 2675, August 1999.
[RFC2743] Linn, J., "Generic Security Service Application Program
Interface Version 2, Update 1", RFC 2743, January 2000.
[RFC2887] Handley, M., Floyd, S., Whetten, B., Kermode, R.,
Vicisano, L., and M. Luby, "The Reliable Multicast Design
Space for Bulk Data Transfer", RFC 2887, August 2000.
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[RFC3048] Whetten, B., Vicisano, L., Kermode, R., Handley, M.,
Floyd, S., and M. Luby, "Reliable Multicast Transport
Building Blocks for One-to-Many Bulk-Data Transfer", RFC
3048, January 2001.
[RFC3124] Balakrishnan, H. and S. Seshan, "The Congestion Manager",
RFC 3124, June 2001.
[RFC3261] Rosenberg, J., Schulzrinne, H., Camarillo, G., Johnston,
A., Peterson, J., Sparks, R., Handley, M., and E.
Schooler, "SIP: Session Initiation Protocol", RFC 3261,
June 2002.
[RFC3303] Srisuresh, P., Kuthan, J., Rosenberg, J., Molitor, A., and
A. Rayhan, "Middlebox communication architecture and
framework", RFC 3303, August 2002.
[RFC3493] Gilligan, R., Thomson, S., Bound, J., McCann, J., and W.
Stevens, "Basic Socket Interface Extensions for IPv6", RFC
3493, February 2003.
[RFC3550] Schulzrinne, H., Casner, S., Frederick, R., and V.
Jacobson, "RTP: A Transport Protocol for Real-Time
Applications", STD 64, RFC 3550, July 2003.
[RFC3551] Schulzrinne, H. and S. Casner, "RTP Profile for Audio and
Video Conferences with Minimal Control", STD 65, RFC 3551,
July 2003.
[RFC3711] Baugher, M., McGrew, D., Naslund, M., Carrara, E., and K.
Norrman, "The Secure Real-time Transport Protocol (SRTP)",
RFC 3711, March 2004.
[RFC3738] Luby, M. and V. Goyal, "Wave and Equation Based Rate
Control (WEBRC) Building Block", RFC 3738, April 2004.
[RFC3758] Stewart, R., Ramalho, M., Xie, Q., Tuexen, M., and P.
Conrad, "Stream Control Transmission Protocol (SCTP)
Partial Reliability Extension", RFC 3758, May 2004.
[RFC3819] Karn, P., Bormann, C., Fairhurst, G., Grossman, D.,
Ludwig, R., Mahdavi, J., Montenegro, G., Touch, J., and L.
Wood, "Advice for Internet Subnetwork Designers", BCP 89,
RFC 3819, July 2004.
[RFC4301] Kent, S. and K. Seo, "Security Architecture for the
Internet Protocol", RFC 4301, December 2005.
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[RFC4302] Kent, S., "IP Authentication Header", RFC 4302, December
2005.
[RFC4303] Kent, S., "IP Encapsulating Security Payload (ESP)", RFC
4303, December 2005.
[RFC4340] Kohler, E., Handley, M., and S. Floyd, "Datagram
Congestion Control Protocol (DCCP)", RFC 4340, March 2006.
[RFC4341] Floyd, S. and E. Kohler, "Profile for Datagram Congestion
Control Protocol (DCCP) Congestion Control ID 2: TCP-like
Congestion Control", RFC 4341, March 2006.
[RFC4342] Floyd, S., Kohler, E., and J. Padhye, "Profile for
Datagram Congestion Control Protocol (DCCP) Congestion
Control ID 3: TCP-Friendly Rate Control (TFRC)", RFC 4342,
March 2006.
[RFC4607] Holbrook, H. and B. Cain, "Source-Specific Multicast for
IP", RFC 4607, August 2006.
[RFC4654] Widmer, J. and M. Handley, "TCP-Friendly Multicast
Congestion Control (TFMCC): Protocol Specification", RFC
4654, August 2006.
[RFC4880] Callas, J., Donnerhacke, L., Finney, H., Shaw, D., and R.
Thayer, "OpenPGP Message Format", RFC 4880, November 2007.
[RFC4960] Stewart, R., "Stream Control Transmission Protocol", RFC
4960, September 2007.
[RFC4963] Heffner, J., Mathis, M., and B. Chandler, "IPv4 Reassembly
Errors at High Data Rates", RFC 4963, July 2007.
[RFC4987] Eddy, W., "TCP SYN Flooding Attacks and Common
Mitigations", RFC 4987, August 2007.
[RFC5082] Gill, V., Heasley, J., Meyer, D., Savola, P., and C.
Pignataro, "The Generalized TTL Security Mechanism
(GTSM)", RFC 5082, October 2007.
[RFC5245] Rosenberg, J., "Interactive Connectivity Establishment
(ICE): A Protocol for Network Address Translator (NAT)
Traversal for Offer/Answer Protocols", RFC 5245, April
2010.
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[RFC5622] Floyd, S. and E. Kohler, "Profile for Datagram Congestion
Control Protocol (DCCP) Congestion ID 4: TCP-Friendly Rate
Control for Small Packets (TFRC-SP)", RFC 5622, August
2009.
[RFC5652] Housley, R., "Cryptographic Message Syntax (CMS)", STD 70,
RFC 5652, September 2009.
[RFC5740] Adamson, B., Bormann, C., Handley, M., and J. Macker,
"NACK-Oriented Reliable Multicast (NORM) Transport
Protocol", RFC 5740, November 2009.
[RFC5751] Ramsdell, B. and S. Turner, "Secure/Multipurpose Internet
Mail Extensions (S/MIME) Version 3.2 Message
Specification", RFC 5751, January 2010.
[RFC5775] Luby, M., Watson, M., and L. Vicisano, "Asynchronous
Layered Coding (ALC) Protocol Instantiation", RFC 5775,
April 2010.
[RFC5971] Schulzrinne, H. and R. Hancock, "GIST: General Internet
Signalling Transport", RFC 5971, October 2010.
[RFC5973] Stiemerling, M., Tschofenig, H., Aoun, C., and E. Davies,
"NAT/Firewall NSIS Signaling Layer Protocol (NSLP)", RFC
5973, October 2010.
[RFC5996] Kaufman, C., Hoffman, P., Nir, Y., and P. Eronen,
"Internet Key Exchange Protocol Version 2 (IKEv2)", RFC
5996, September 2010.
[RFC6335] Cotton, M., Eggert, L., Touch, J., Westerlund, M., and S.
Cheshire, "Internet Assigned Numbers Authority (IANA)
Procedures for the Management of the Service Name and
Transport Protocol Port Number Registry", BCP 165, RFC
6335, August 2011.
[RFC6347] Rescorla, E. and N. Modadugu, "Datagram Transport Layer
Security Version 1.2", RFC 6347, January 2012.
[RFC6395] Gulrajani, S. and S. Venaas, "An Interface Identifier (ID)
Hello Option for PIM", RFC 6395, October 2011.
[RFC6396] Blunk, L., Karir, M., and C. Labovitz, "Multi-Threaded
Routing Toolkit (MRT) Routing Information Export Format",
RFC 6396, October 2011.
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[RFC6437] Amante, S., Carpenter, B., Jiang, S., and J. Rajahalme,
"IPv6 Flow Label Specification", RFC 6437, November 2011.
[RFC6438] Carpenter, B. and S. Amante, "Using the IPv6 Flow Label
for Equal Cost Multipath Routing and Link Aggregation in
Tunnels", RFC 6438, November 2011.
[RFC6513] Rosen, E. and R. Aggarwal, "Multicast in MPLS/BGP IP
VPNs", RFC 6513, February 2012.
[RFC6679] Westerlund, M., Johansson, I., Perkins, C., O'Hanlon, P.,
and K. Carlberg, "Explicit Congestion Notification (ECN)
for RTP over UDP", RFC 6679, August 2012.
[RFC6726] Paila, T., Walsh, R., Luby, M., Roca, V., and R. Lehtonen,
"FLUTE - File Delivery over Unidirectional Transport", RFC
6726, November 2012.
[RFC6807] Farinacci, D., Shepherd, G., Venaas, S., and Y. Cai,
"Population Count Extensions to Protocol Independent
Multicast (PIM)", RFC 6807, December 2012.
[STEVENS] Stevens, W., Fenner, B., and A. Rudoff, "UNIX Network
Programming, The sockets Networking API", Addison-Wesley,
2004.
[UPnP] UPnP Forum, , "Internet Gateway Device (IGD) Standardized
Device Control Protocol V 1.0", November 2001.
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Appendix A. Revision Notes
Note to RFC-Editor: please remove this entire section prior to
publication.
Changes in draft-eggert-tsvwg-rfc5405bis-01:
o Added Greg Shepherd as a co-author, based on the multicast
guidelines that originated with him.
Changes in draft-eggert-tsvwg-rfc5405bis-00 (relative to RFC5405):
o The words "application designers" were removed from the draft
title and the wording of the abstract was clarified abstract.
o New text to clarify various issues and set new recommendations not
previously included in RFC 5405. These include new
recommendations for multicast, the use of checksums with IPv6,
ECMP, recommendations on port usage, use of ECN, use of DiffServ,
circuit breakers (initial text), etc.
Authors' Addresses
Lars Eggert
NetApp
Sonnenallee 1
Kirchheim 85551
Germany
Phone: +49 151 120 55791
EMail: lars@netapp.com
URI: https://eggert.org/
Godred Fairhurst
University of Aberdeen
Department of Engineering
Fraser Noble Building
Aberdeen AB24 3UE
Scotland
EMail: gorry@erg.abdn.ac.uk
URI: http://www.erg.abdn.ac.uk/
Eggert, et al. Expires December 19, 2014 [Page 36]
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Greg Shepherd
Cisco Systems
Tasman Drive
San Jose
USA
EMail: gjshep@gmail.com
Eggert, et al. Expires December 19, 2014 [Page 37]
