Transport Area Working Group                                   L. Eggert
Internet-Draft                                                     Nokia
Intended status: Best Current                               G. Fairhurst
Practice                                          University of Aberdeen
Expires: November 29, 2007                                  May 28, 2007

             UDP Usage Guidelines for Application Designers

Status of this Memo

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Copyright Notice

   Copyright (C) The IETF Trust (2007).


   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 establish some degree of fairness with
   concurrent traffic.  This document provides guidelines on the use of

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   UDP for the designers of such applications and upper-layer protocols
   that cover congestion-control and other topics, including message
   sizes and reliability.

Table of Contents

   1.  Introduction . . . . . . . . . . . . . . . . . . . . . . . . .  3
   2.  Terminology  . . . . . . . . . . . . . . . . . . . . . . . . .  4
   3.  UDP Usage Guidelines . . . . . . . . . . . . . . . . . . . . .  4
     3.1.  Congestion Control Guidelines  . . . . . . . . . . . . . .  5
     3.2.  Message Size Guidelines  . . . . . . . . . . . . . . . . .  7
     3.3.  Reliability Guidelines . . . . . . . . . . . . . . . . . .  7
     3.4.  Checksum Guidelines  . . . . . . . . . . . . . . . . . . .  8
   4.  Security Considerations  . . . . . . . . . . . . . . . . . . .  9
   5.  IANA Considerations  . . . . . . . . . . . . . . . . . . . . . 10
   6.  Acknowledgments  . . . . . . . . . . . . . . . . . . . . . . . 10
   7.  References . . . . . . . . . . . . . . . . . . . . . . . . . . 10
     7.1.  Normative References . . . . . . . . . . . . . . . . . . . 10
     7.2.  Informative References . . . . . . . . . . . . . . . . . . 11
   Authors' Addresses . . . . . . . . . . . . . . . . . . . . . . . . 12
   Intellectual Property and Copyright Statements . . . . . . . . . . 14

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1.  Introduction

   The User Datagram Protocol (UDP) [RFC0768] provides a minimal,
   unreliable, 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 no 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.  [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

   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.  For example, an
   application that generates five 1500-byte UDP packets 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.  Section 3 describes a number of
   simple guidelines for the designers of such applications.

   A UDP message is carried in a single IP packet and is hence limited
   to a maximum payload of 65,487 bytes.  The transmission of large IP
   packets frequently requires IP fragmentation, which decreases
   communication reliability and efficiency and should be avoided
   [I-D.heffner-frag-harmful].  Some of the guidelines in Section 3
   describe how applications should determine appropriate message sizes.

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   This document provides guidelines to designers of applications that
   use UDP for unicast transmission.  A special class of applications
   uses UDP for IP multicast transmissions.  Congestion control, flow
   control or reliability for multicast transmissions is more difficult
   to establish than for unicast transmissions, because a single sender
   may transmit to multiple receivers across potentially very
   heterogeneous paths at the same time.  Designing multicast
   applications requires expertise that goes beyond the simple
   guidelines given in this document.  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].

2.  Terminology

   The key words "MUST", "MUST NOT", "REQUIRED", "SHALL", "SHALL NOT",
   document are to be interpreted as described in BCP 14, RFC 2119

3.  UDP Usage Guidelines

   The RECOMMENDED alternative to the UDP usage guidelines described in
   this section is the use of a transport protocol that is congestion-
   controlled, such as TCP [RFC0793], SCTP [RFC2960] or DCCP [RFC4340]
   with its different congestion control types
   [RFC4341][RFC4342][I-D.floyd-dccp-ccid4].  Congestion control
   mechanisms are difficult to implement correctly, and for most
   applications, the use of one of the existing, congestion-controlled
   protocols is the simplest method of satisfying [RFC2914].  The same
   is true for message size determination and reliability mechanisms.

   If used correctly, congestion-controlled transport protocols are not
   as "heavyweight" as often claimed.  For example, TCP with SYN cookies
   [I-D.ietf-tcpm-syn-flood], which are available on many platforms,
   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 to interfere with a later one.
   Applications that are aware of this behavior can shift maintenance of
   the TIME-WAIT state to conserve resources.  Finally, TCP's built-in
   capacity-probing and PMTU awareness results in efficient data
   transmission that quickly compensates for the initial connection
   setup delay, for transfers that exchange more than a few packets.

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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 messages to a destination host.  It is
   important to stress that an application SHOULD perform congestion
   control over all UDP traffic it sends to a destination, independent
   of how it generates this traffic.  For example, an application that
   forks multiple worker processes or otherwise uses multiple sockets to
   generate UDP messages SHOULD perform congestion control over the
   aggregate traffic.  The remainder of this section discusses several
   approaches for this purpose.

   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.

3.1.1.  Bulk Transfer Applications

   Applications that perform bulk transmission of data to a peer over
   UDP SHOULD consider implementing TCP-Friendly Rate Control (TFRC)
   [RFC3448], window-based, TCP-like congestion control, or otherwise
   ensure that the application complies with the congestion control

   TFRC has been designed to provide both congestion control and
   fairness in a way that is compatible with the IETF's other transport
   protocols.  TFRC is currently being updated
   [I-D.ietf-dccp-rfc3448bis], and application designers SHOULD always
   evaluate whether the latest published specification fits their needs.
   If an application implements TFRC, it need not follow the remaining
   guidelines in Section 3.1, but SHOULD still follow the guidelines on
   message sizes in Section 3.2 and reliability in Section 3.2.

   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.  [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.

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   Finally, some bulk transfer applications chose 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 paths, results are

3.1.2.  Low Data-Volume Applications

   Applications that exchange only a small number of messages with a
   destination at any time may not benefit from implementing TFRC or one
   of the other congestion control schemes in Section 3.1.1.  Such
   applications SHOULD still control their transmission behavior by not
   sending more than one UDP message 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 it communicates with.  Applications SHOULD implement the
   algorithm specified in [RFC2988] to compute a smoothed RTT (SRTT)
   estimate.  A lost response from the peer SHOULD be treated as a very
   large RTT sample, instead of being ignored, in order to cause a
   sufficiently large (exponential) back-off.  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 [I-D.ietf-nsis-ntlp] 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 applications that exchange too few
   messages with a peer to establish a statistically accurate RTT
   estimate.  Such applications MAY use a fixed transmission interval
   that is exponentially backed-off during loss.  TCP uses an initial
   value of 3 seconds [RFC2988], which is also RECOMMENDED as an initial
   value for UDP applications.  SIP [RFC3261] and GIST
   [I-D.ietf-nsis-ntlp] 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 message every 3
   seconds, and SHOULD consider if they can use an even less aggressive

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   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 initial
   timeout interval 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 independently detect and
   respond to congestion along both directions.

3.2.  Message Size Guidelines

   Because IP fragmentation lowers the efficiency and reliability of
   Internet communication [I-D.heffner-frag-harmful], an application
   SHOULD NOT send UDP messages 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 choose not to do so SHOULD NOT send UDP messages
   that exceed the minimum PMTU.  The minimum PMTU depends on the IP
   version used for transmission, and is the lesser of 576 bytes and the
   first-hop MTU for IPv4 [RFC1122] and 1280 bytes for IPv6 [RFC2460].
   To determine an appropriate UDP payload size, applications must
   subtract IP header and option lengths as well as the length of the
   UDP header from the PMTU size.  Transmission of minimum-sized
   messages is inefficient over paths that support a larger PMTU, which
   is a second reason to implement PMTU discovery.

   Applications that do not send messages that exceed the minimum PMTU
   of IPv4 or IPv6 need not implement any of the above mechanisms.

3.3.  Reliability Guidelines

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

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   UDP also does not protect against message duplication, i.e., an
   application may receive multiple copies of the same message.
   Application designers SHOULD consider whether their application
   handles message duplication gracefully, and may need to implement
   mechanisms to detect duplicates.  Even if message reception triggers
   idempotent operations, applications may want to suppress duplicate
   messages to reduce load.

   Finally, UDP messages may be reordered in the network and arrive at
   the receiver in an order different from the send order.  Applications
   that require ordered delivery SHOULD reestablish message ordering

3.4.  Checksum Guidelines

   The UDP header includes a 16-bit ones' complement checksum that
   provides an integrity check.  The UDP checksum provides assurance
   that the payload was not corrupted in transit.  It also verifies that
   the datagram was delivered to the intended end point, because it
   covers the IP addresses, port numbers and protocol number, and it
   verifies that the datagram 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.

   Applications SHOULD enable UDP checksums, although [RFC0793] 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 packet is received that was originally sent
   to a different end point or is otherwise corrupted.  The use of the
   UDP checksum is MANDATORY when applications transmit UDP over IPv6
   [RFC2460] and applications consequently MUST NOT disable their use.
   (The IPv6 header does not have a separate checksum field to protect
   the IP addressing information.)

   The UDP checksum provides relatively weak protection from a coding
   point of view [RFC3819] and, where appropriate, applications
   developers SHOULD provide additional checks, e.g., through a CRC
   included with the data to verify the integrity of an entire object/
   file sent over UDP service.

3.4.1.  UDP-Lite

   A special class of applications 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

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   Datagram Protocol (UDP-Lite) [RFC3828] variant of UDP instead of
   basic UDP.  Applications that choose to use UDP-Lite instead of UDP
   MUST still follow the congestion control and other guidelines
   described for use with UDP in Section 3.1.

   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 datagram checksum coverage, with the remaining
   unprotected part of the payload called the "error insensitive part".
   If required, an application may dynamically modify this length value,
   e.g., to offer greater protection to some packets.  UDP-Lite always
   verifies that a datagram was delivered to the intended end point,
   i.e., always verifies the header fields.  Errors in the insensitive
   part will not cause a packet to be discarded by the receiving end
   host.  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 (e.g., the RTP header),
   and, 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
   datagrams that is not smaller than the smallest coverage used by the
   sender.  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 packets being
   treated the same as UDP packets, i.e., result in packet loss.  Use of
   IP fragmentation can also prevent special treatment for UDP-Lite
   packets, and is another reason why applications SHOULD avoid IP
   fragmentation Section 3.2.

4.  Security Considerations

   [RFC2309] and [RFC2914] discuss the dangers of congestion-
   unresponsive flows to the Internet.  This document provides

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   guidelines to designers of UDP-based applications to congestion-
   control their transmissions.  As such, it does not raise any
   additional security concerns.

5.  IANA Considerations

   This document raises no IANA considerations.

6.  Acknowledgments

   Thanks to Mark Allman, Sally Floyd, Joerg Ott, Colin Perkins, Pasi
   Sarolahti and Magnus Westerlund for their comments on this document.

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.

   [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.

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

   [RFC2960]  Stewart, R., Xie, Q., Morneault, K., Sharp, C.,
              Schwarzbauer, H., Taylor, T., Rytina, I., Kalla, M.,
              Zhang, L., and V. Paxson, "Stream Control Transmission
              Protocol", RFC 2960, October 2000.

   [RFC2988]  Paxson, V. and M. Allman, "Computing TCP's Retransmission
              Timer", RFC 2988, November 2000.

   [RFC3448]  Handley, M., Floyd, S., Padhye, J., and J. Widmer, "TCP
              Friendly Rate Control (TFRC): Protocol Specification",

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              RFC 3448, January 2003.

   [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.

   [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.

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

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

7.2.  Informative References

              Floyd, S. and E. Kohler, "Profile for Datagram Congestion
              Control Protocol (DCCP) Congestion ID 4:  TCP-Friendly
              Rate Control for Small Packets (TFRC-SP)",
              draft-floyd-dccp-ccid4-00 (work in progress),
              November 2006.

              Heffner, J., "IPv4 Reassembly Errors at High Data Rates",
              draft-heffner-frag-harmful-05 (work in progress),
              May 2007.

              Handley, M., "TCP Friendly Rate Control (TFRC): Protocol
              Specification", draft-ietf-dccp-rfc3448bis-01 (work in
              progress), March 2007.

              Schulzrinne, H. and R. Hancock, "GIST: General Internet
              Signalling Transport", draft-ietf-nsis-ntlp-13 (work in
              progress), April 2007.

              Eddy, W., "TCP SYN Flooding Attacks and Common
              Mitigations", draft-ietf-tcpm-syn-flood-04 (work in
              progress), May 2007.

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

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

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

   [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.

   [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.

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

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

   [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.

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

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Authors' Addresses

   Lars Eggert
   Nokia Research Center
   P.O. Box 407
   Nokia Group  00045

   Phone: +358 50 48 24461

   Godred Fairhurst
   University of Aberdeen
   Department of Engineering
   Fraser Noble Building
   Aberdeen  AB24 3UE


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