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Optimized Link State Routing Protocol (OLSR)
RFC 3626

Document Type RFC - Experimental (October 2003) Errata
Authors Thomas H. Clausen , Philippe Jacquet
Last updated 2017-08-22
RFC stream Internet Engineering Task Force (IETF)
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IESG Responsible AD Alex D. Zinin
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RFC 3626
Transport Area Working Group                                   L. Eggert
Internet-Draft                                                    NetApp
Obsoletes: 5405 (if approved)                               G. Fairhurst
Intended status: Best Current Practice            University of Aberdeen
Expires: May 5, 2016                                         G. Shepherd
                                                           Cisco Systems
                                                        November 2, 2015

                          UDP Usage Guidelines
                     draft-ietf-tsvwg-rfc5405bis-07

Abstract

   The User Datagram Protocol (UDP) provides a minimal message-passing
   transport that has no inherent congestion control mechanisms.  This
   document provides guidelines on the use of UDP for the designers of
   applications, tunnels and other protocols that use UDP.  Congestion
   control guidelines are a primary focus, but the document also
   provides guidance on other topics, including message sizes,
   reliability, checksums, middlebox traversal, the use of ECN, DSCPs,
   and ports.

   Because congestion control is critical to the stable operation of the
   Internet, applications and other protocols that choose to use UDP as
   an Internet transport must employ mechanisms to prevent congestion
   collapse and to establish some degree of fairness with concurrent
   traffic.  They may also need to implement additional mechanisms,
   depending on how they use UDP.

   Some guidance is also applicable to the design of other protocols
   (e.g., protocols layered directly on IP or via IP-based tunnels),
   especially when these protocols do not themselves provide congestion
   control.

   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/.

Eggert, et al.             Expires May 5, 2016                  [Page 1]
Internet-Draft            UDP Usage Guidelines             November 2015

   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 May 5, 2016.

Copyright Notice

   Copyright (c) 2015 IETF Trust and the persons identified as the
   document authors.  All rights reserved.

   This document is subject to BCP 78 and the IETF Trust's Legal
   Provisions Relating to IETF Documents
   (http://trustee.ietf.org/license-info) in effect on the date of
   publication of this document.  Please review these documents
   carefully, as they describe your rights and restrictions with respect
   to this document.  Code Components extracted from this document must
   include Simplified BSD License text as described in Section 4.e of
   the Trust Legal Provisions and are provided without warranty as
   described in the Simplified BSD License.

Table of Contents

   1.  Introduction  . . . . . . . . . . . . . . . . . . . . . . . .   3
   2.  Terminology . . . . . . . . . . . . . . . . . . . . . . . . .   4
   3.  UDP Usage Guidelines  . . . . . . . . . . . . . . . . . . . .   5
     3.1.  Congestion Control Guidelines . . . . . . . . . . . . . .   6
     3.2.  Message Size Guidelines . . . . . . . . . . . . . . . . .  16
     3.3.  Reliability Guidelines  . . . . . . . . . . . . . . . . .  18
     3.4.  Checksum Guidelines . . . . . . . . . . . . . . . . . . .  19
     3.5.  Middlebox Traversal Guidelines  . . . . . . . . . . . . .  22
     3.6.  Limited Applicability and Controlled Environments . . . .  24
   4.  Multicast UDP Usage Guidelines  . . . . . . . . . . . . . . .  25
     4.1.  Multicast Congestion Control Guidelines . . . . . . . . .  26
     4.2.  Message Size Guidelines for Multicast . . . . . . . . . .  28
   5.  Programming Guidelines  . . . . . . . . . . . . . . . . . . .  28
     5.1.  Using UDP Ports . . . . . . . . . . . . . . . . . . . . .  30
     5.2.  ICMP Guidelines . . . . . . . . . . . . . . . . . . . . .  32
   6.  Security Considerations . . . . . . . . . . . . . . . . . . .  33
   7.  Summary . . . . . . . . . . . . . . . . . . . . . . . . . . .  35
   8.  IANA Considerations . . . . . . . . . . . . . . . . . . . . .  37
   9.  Acknowledgments . . . . . . . . . . . . . . . . . . . . . . .  37
   10. References  . . . . . . . . . . . . . . . . . . . . . . . . .  37
     10.1.  Normative References . . . . . . . . . . . . . . . . . .  37
     10.2.  Informative References . . . . . . . . . . . . . . . . .  39
   Appendix A.  Case Study of the Use of IPv6 UDP Zero-Checksum Mode  47
   Appendix B.  Revision Notes . . . . . . . . . . . . . . . . . . .  48

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3.5.  Message Emission and Jitter

   As a basic implementation requirement, synchronization of control
   messages SHOULD be avoided.  As a consequence, OLSR control messages
   SHOULD be emitted such that they avoid synchronization.

   Emission of control traffic from neighboring nodes may, for various
   reasons (mainly timer interactions with packet processing), become
   synchronized such that several neighbor nodes attempt to transmit
   control traffic simultaneously.  Depending on the nature of the
   underlying link-layer, this may or may not lead to collisions and
   hence message loss - possibly loss of several subsequent messages of
   the same type.

   To avoid such synchronizations, the following simple strategy for
   emitting control messages is proposed.  A node SHOULD add an amount
   of jitter to the interval at which messages are generated.  The
   jitter must be a random value for each message generated.  Thus, for
   a node utilizing jitter:

        Actual message interval = MESSAGE_INTERVAL - jitter

   Where jitter is a value, randomly selected from the interval
   [0,MAXJITTER] and MESSAGE_INTERVAL is the value of the message
   interval specified for the message being emitted (e.g.,
   HELLO_INTERVAL for HELLO messages, TC_INTERVAL for TC-messages etc.).

   Jitter SHOULD also be introduced when forwarding messages.  The
   following simple strategy may be adopted: when a message is to be
   forwarded by a node, it should be kept in the node during a short
   period of time :

           Keep message period = jitter

   Where jitter is a random value in [0,MAXJITTER].

   Notice that when the node sends a control message, the opportunity to
   piggyback other messages (before their keeping period is expired) may
   be taken to reduce the number of packet transmissions.

   Notice, that a minimal rate of control messages is imposed.  A node
   MAY send control messages at a higher rate, if beneficial for a
   specific deployment.

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4.  Information Repositories

   Through the exchange of OLSR control messages, each node accumulates
   information about the network.  This information is stored according
   to the descriptions in this section.

4.1.  Multiple Interface Association Information Base

   For each destination in the network, "Interface Association Tuples"
   (I_iface_addr, I_main_addr, I_time) are recorded.  I_iface_addr is an
   interface address of a node, I_main_addr is the main address of this
   node.  I_time specifies the time at which this tuple expires and
   *MUST* be removed.

   In a node, the set of Interface Association Tuples is denoted the
   "Interface Association Set".

4.2.  Link Sensing: Local Link Information Base

   The local link information base stores information about links to
   neighbors.

4.2.1.  Link Set

   A node records a set of "Link Tuples" (L_local_iface_addr,
   L_neighbor_iface_addr, L_SYM_time, L_ASYM_time, L_time).
   L_local_iface_addr is the interface address of the local node (i.e.,
   one endpoint of the link), L_neighbor_iface_addr is the interface
   address of the neighbor node (i.e., the other endpoint of the link),
   L_SYM_time is the time until which the link is considered symmetric,
   L_ASYM_time is the time until which the neighbor interface is
   considered heard, and L_time specifies the time at which this record
   expires and *MUST* be removed.  When L_SYM_time and L_ASYM_time are
   expired, the link is considered lost.

   This information is used when declaring the neighbor interfaces in
   the HELLO messages.

   L_SYM_time is used to decide the Link Type declared for the neighbor
   interface.  If L_SYM_time is not expired, the link MUST be declared
   symmetric.  If L_SYM_time is expired, the link MUST be declared
   asymmetric.  If both L_SYM_time and L_ASYM_time are expired, the link
   MUST be declared lost.

   In a node, the set of Link Tuples are denoted the "Link Set".

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4.3.  Neighbor Detection: Neighborhood Information Base

   The neighborhood information base stores information about neighbors,
   2-hop neighbors, MPRs and MPR selectors.

4.3.1.  Neighbor Set

   A node records a set of "neighbor tuples" (N_neighbor_main_addr,
   N_status, N_willingness), describing neighbors.  N_neighbor_main_addr
   is the main address of a neighbor, N_status specifies if the node is
   NOT_SYM or SYM.  N_willingness in an integer between 0 and 7, and
   specifies the node's willingness to carry traffic on behalf of other
   nodes.

4.3.2.  2-hop Neighbor Set

   A node records a set of "2-hop tuples" (N_neighbor_main_addr,
   N_2hop_addr, N_time), describing symmetric (and, since MPR links by
   definition are also symmetric, thereby also MPR) links between its
   neighbors and the symmetric 2-hop neighborhood.  N_neighbor_main_addr
   is the main address of a neighbor, N_2hop_addr is the main address of
   a 2-hop neighbor with a symmetric link to N_neighbor_main_addr, and
   N_time specifies the time at which the tuple expires and *MUST* be
   removed.

   In a node, the set of 2-hop tuples are denoted the "2-hop Neighbor
   Set".

4.3.3.  MPR Set

   A node maintains a set of neighbors which are selected as MPR.  Their
   main addresses are listed in the MPR Set.

4.3.4.  MPR Selector Set

   A node records a set of MPR-selector tuples (MS_main_addr, MS_time),
   describing the neighbors which have selected this node as a MPR.
   MS_main_addr is the main address of a node, which has selected this
   node as MPR.  MS_time specifies the time at which the tuple expires
   and *MUST* be removed.

   In a node, the set of MPR-selector tuples are denoted the "MPR
   Selector Set".

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4.4.  Topology Information Base

   Each node in the network maintains topology information about the
   network.  This information is acquired from TC-messages and is used
   for routing table calculations.

   Thus, for each destination in the network, at least one "Topology
   Tuple" (T_dest_addr, T_last_addr, T_seq, T_time) is recorded.
   T_dest_addr is the main address of a node, which may be reached in
   one hop from the node with the main address T_last_addr.  Typically,
   T_last_addr is a MPR of T_dest_addr.  T_seq is a sequence number, and
   T_time specifies the time at which this tuple expires and *MUST* be
   removed.

   In a node, the set of Topology Tuples are denoted the "Topology Set".

5.  Main Addresses and Multiple Interfaces

   For single OLSR interface nodes, the relationship between an OLSR
   interface address and the corresponding main address is trivial: the
   main address is the OLSR interface address.  For multiple OLSR
   interface nodes, the relationship between OLSR interface addresses
   and main addresses is defined through the exchange of Multiple
   Interface Declaration (MID) messages.  This section describes how MID
   messages are exchanged and processed.

   Each node with multiple interfaces MUST announce, periodically,
   information describing its interface configuration to other nodes in
   the network.  This is accomplished through flooding a Multiple
   Interface Declaration message to all nodes in the network through the
   MPR flooding mechanism.

   Each node in the network maintains interface information about the
   other nodes in the network.  This information acquired from MID
   messages, emitted by nodes with multiple interfaces participating in
   the MANET, and is used for routing table calculations.

   Specifically, multiple interface declaration associates multiple
   interfaces to a node (and to a main address) through populating the
   multiple interface association base in each node.

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5.1.  MID Message Format

   The proposed format of a MID message is as follows:

       0                   1                   2                   3
       0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1
      +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
      |                    OLSR Interface Address                     |
      +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
      |                    OLSR Interface Address                     |
      +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
      |                              ...                              |
      +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+

   This is sent as the data-portion of the general packet format
   described in section 3.4, with the "Message Type" set to MID_MESSAGE.
   The time to live SHOULD be set to 255 (maximum value) to diffuse the
   message into the entire network and Vtime set accordingly to the
   value of MID_HOLD_TIME, as specified in section 18.3.

     OLSR Interface Address

          This field contains the address of an OLSR interface of the
          node, excluding the nodes main address (which already
          indicated in the originator address).

   All interface addresses other than the main address of the originator
   node are put in the MID message.  If the maximum allowed message size
   (as imposed by the network) is reached while there are still
   interface addresses which have not been inserted into the MIDmessage,
   more MID messages are generated until the entire interface addresses
   set has been sent.

5.2.  MID Message Generation

   A MID message is sent by a node in the network to declare its
   multiple interfaces (if any).  I.e., the MID message contains the
   list of interface addresses which are associated to its main address.
   The list of addresses can be partial in each MID message (e.g., due
   to message size limitations, imposed by the network), but parsing of
   all MID messages describing the interface set from a node MUST be
   complete within a certain refreshing period (MID_INTERVAL).  The
   information diffused in the network by these MID messages will help
   each node to calculate its routing table.  A node which has only a
   single interface address participating in the MANET (i.e., running
   OLSR), MUST NOT generate any MID message.

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   A node with several interfaces, where only one is participating in
   the MANET and running OLSR (e.g., a node is connected to a wired
   network as well as to a MANET) MUST NOT generate any MID messages.

   A node with several interfaces, where more than one is participating
   in the MANET and running OLSR MUST generate MID messages as
   specified.

5.3.  MID Message Forwarding

   MID messages are broadcast and retransmitted by the MPRs in order to
   diffuse the messages in the entire network.  The "default forwarding
   algorithm" (described in section 3.4) MUST be used for forwarding of
   MID messages.

5.4.  MID Message Processing

   The tuples in the multiple interface association set are recorded
   with the information that is exchanged through MID messages.

   Upon receiving a MID message, the "validity time" MUST be computed
   from the Vtime field of the message header (as described in section
   3.3.2).  The Multiple Interface Association Information Base SHOULD
   then be updated as follows:

     1    If the sender interface (NB: not originator) of this message
          is not in the symmetric 1-hop neighborhood of this node, the
          message MUST be discarded.

     2    For each interface address listed in the MID message:

          2.1  If there exist some tuple in the interface association
               set where:

                    I_iface_addr == interface address, AND

                    I_main_addr  == originator address,

               then the holding time of that tuple is set to:

                    I_time       = current time + validity time.

          2.2  Otherwise, a new tuple is recorded in the interface
               association set where:

                    I_iface_addr = interface address,

                    I_main_addr  = originator address,

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                    I_time       = current time + validity time.

5.5.  Resolving a Main Address from an Interface Address

   In general, the only part of OLSR requiring use of "interface
   addresses" is link sensing.  The remaining parts of OLSR operate on
   nodes, uniquely identified by their "main addresses" (effectively,
   the main address of a node is its "node id" - which for convenience
   corresponds to the address of one of its interfaces).  In a network
   with only single interface nodes, the main address of a node will, by
   definition, be equal to the interface address of the node.  In
   networks with multiple interface nodes operating within a common OLSR
   area, it is required to be able to map any interface address to the
   corresponding main address.

   The exchange of MID messages provides a way in which interface
   information is acquired by nodes in the network.  This permits
   identification of a node's "main address", given one of its interface
   addresses.

   Given an interface address:

     1    if there exists some tuple in the interface association set
          where:

               I_iface_addr == interface address

          then the result of the main address search is the originator
          address I_main_addr of the tuple.

     2    Otherwise, the result of the main address search is the
          interface address itself.

6.  HELLO Message Format and Generation

   A common mechanism is employed for populating the local link
   information base and the neighborhood information base, namely
   periodic exchange of HELLO messages.  Thus this section describes the
   general HELLO message mechanism, followed by a description of link
   sensing and topology detection, respectively.

6.1.  HELLO Message Format

   To accommodate for link sensing, neighborhood detection and MPR
   selection signalling, as well as to accommodate for future
   extensions, an approach similar to the overall packet format is
   taken.  Thus the proposed format of a HELLO message is as follows:

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       0                   1                   2                   3
       0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1

      +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
      |          Reserved             |     Htime     |  Willingness  |
      +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
      |   Link Code   |   Reserved    |       Link Message Size       |
      +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
      |                  Neighbor Interface Address                   |
      +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
      |                  Neighbor Interface Address                   |
      +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
      :                             .  .  .                           :
      :                                                               :
      +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
      |   Link Code   |   Reserved    |       Link Message Size       |
      +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
      |                  Neighbor Interface Address                   |
      +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
      |                  Neighbor Interface Address                   |
      +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
      :                                                               :
      :                                       :
   (etc.)

   This is sent as the data-portion of the general packet format
   described in section 3.4, with the "Message Type" set to
   HELLO_MESSAGE, the TTL field set to 1 (one) and Vtime set accordingly
   to the value of NEIGHB_HOLD_TIME, specified in section 18.3.

      Reserved

         This field must be set to "0000000000000" to be in compliance
         with this specification.

      HTime

         This field specifies the HELLO emission interval used by the
         node on this particular interface, i.e., the time before the
         transmission of the next HELLO (this information may be used in
         advanced link sensing, see section 14).  The HELLO emission
         interval is represented by its mantissa (four highest bits of
         Htime field) and by its exponent (four lowest bits of Htime
         field).  In other words:

              HELLO emission interval=C*(1+a/16)*2^b  [in seconds]

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         where a is the integer represented by the four highest bits of
         Htime field and b the integer represented by the four lowest
         bits of Htime field.  The proposed value of the scaling factor
         C is specified in section 18.

      Willingness

         This field specifies the willingness of a node to carry and
         forward traffic for other nodes.

         A node with willingness WILL_NEVER (see section 18.8, for
         willingness constants) MUST never be selected as MPR by any
         node.  A node with willingness WILL_ALWAYS MUST always be
         selected as MPR.  By default, a node SHOULD advertise a
         willingness of WILL_DEFAULT.

      Link Code

         This field specifies information about the link between the
         interface of the sender and the following list of neighbor
         interfaces.  It also specifies information about the status of
         the neighbor.

         Link codes, not known by a node, are silently discarded.

      Link Message Size

         The size of the link message, counted in bytes and measured
         from the beginning of the "Link Code" field and until the next
         "Link Code" field (or - if there are no more link types - the
         end of the message).

      Neighbor Interface Address

         The address of an interface of a neighbor node.

6.1.1.  Link Code as Link Type and Neighbor Type

   This document only specifies processing of Link Codes < 16.

   If the Link Code value is less than or equal to 15, then it MUST be
   interpreted as holding two different fields, of two bits each:

          7       6       5       4       3       2       1       0
      +-------+-------+-------+-------+-------+-------+-------+-------+
      |   0   |   0   |   0   |   0   | Neighbor Type |   Link Type   |
      +-------+-------+-------+-------+-------+-------+-------+-------+

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   The following four "Link Types" are REQUIRED by OLSR:

     -    UNSPEC_LINK - indicating that no specific information about
          the links is given.

     -    ASYM_LINK - indicating that the links are asymmetric (i.e.,
          the neighbor interface is "heard").

     -    SYM_LINK - indicating that the links are symmetric with the
          interface.

     -    LOST_LINK - indicating that the links have been lost.

   The following three "Neighbor Types&Internet-Draft            UDP Usage Guidelines             November 2015

   Authors' Addresses  . . . . . . . . . . . . . . . . . . . . . . .  51

1.  Introduction

   The User Datagram Protocol (UDP) [RFC0768] provides a minimal,
   unreliable, best-effort, message-passing transport to applications
   and other protocols (such as tunnels) that desire to operate over
   UDP.  Both simply called "applications" in the remainder of this
   document.

   Compared to other transport protocols, UDP and its UDP-Lite variant
   [RFC3828] are unique in that they do not establish end-to-end
   connections between communicating end systems.  UDP communication
   consequently does not incur connection establishment and teardown
   overheads, and there is minimal associated end system state.  Because
   of these characteristics, UDP can offer a very efficient
   communication transport to some applications.

   A second unique characteristic of UDP is that it provides no inherent
   congestion control mechanisms.  On many platforms, applications can
   send UDP datagrams at the line rate of the platform's link interface,
   which is often much greater than the available end-to-end path
   capacity, and doing so contributes to congestion along the path.
   [RFC2914] describes the best current practice for congestion 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,

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   an application that generates five 1500-byte UDP datagrams in one
   second can already exceed the capacity of a 56 Kb/s path.  For
   applications that can operate at higher, potentially unbounded data
   rates, congestion control becomes vital to prevent congestion
   collapse and establish some degree of fairness.  Section 3 describes
   a number of simple guidelines for the designers of such applications.

   A UDP datagram is carried in a single IP packet and is hence limited
   to a maximum payload of 65,507 bytes for IPv4 and 65,527 bytes for
   IPv6.  The transmission of large IP packets usually requires IP
   fragmentation.  Fragmentation decreases communication reliability and
   efficiency and should be avoided.  IPv6 allows the option of
   transmitting large packets ("jumbograms") without fragmentation when
   all link layers along the path support this [RFC2675].  Some of the
   guidelines in Section 3 describe how applications should determine
   appropriate message sizes.  Other sections of this document provide
   guidance on reliability, checksums, middlebox traversal and use of
   multicast.

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

   [RFC5405] was scoped to provide guidelines for unicast applications
   only, whereas this document also provides guidelines for UDP flows
   that use IP anycast, multicast, broadcast, and applications that use
   UDP tunnels to support IP flows.

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

2.  Terminology

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

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3.  UDP Usage Guidelines

   Internet paths can have widely varying characteristics, including
   transmission delays, available bandwidths, congestion levels,
   reordering probabilities, supported message sizes, or loss rates.
   Furthermore, the same Internet path can have very different
   conditions over time.  Consequently, applications that may be used on
   the Internet MUST NOT make assumptions about specific path
   characteristics.  They MUST instead use mechanisms that let them
   operate safely under very different path conditions.  Typically, this
   requires conservatively probing the current conditions of the
   Internet path they communicate over to establish a transmission
   behavior that it can sustain and that is reasonably fair to other
   traffic sharing the path.

   These mechanisms are difficult to implement correctly.  For most
   applications, the use of one of the existing IETF transport protocols
   is the simplest method of acquiring the required mechanisms.  Doing
   so also avoids issues that protocols using a new IP protocol number
   face when being deployed over the Internet, where middleboxes that
   only support TCP and UDP are not rare.  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

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   data transmission that quickly compensates for the initial connection
   setup delay, in the case of transfers that exchange more than a few
   segments.

3.1.  Congestion Control Guidelines

   If an application or protocol chooses not to use a congestion-
   controlled transport protocol, it SHOULD control the rate at which it
   sends UDP datagrams to a destination host, in order to fulfill the
   requirements of [RFC2914].  It is important to stress that an
   application SHOULD perform congestion control over all UDP traffic it
   sends to a destination, independently from how it generates this
   traffic.  For example, an application that forks multiple worker
   processes or otherwise uses multiple sockets to generate UDP
   datagrams SHOULD perform congestion control over the aggregate
   traffic.

   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 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 3.1.9 discusses congestion control considerations when UDP is
   used as a tunneling protocol.  Section 4 provides additional
   recommendations for broadcast and multicast usage.

   It is important to note that congestion control should not be viewed
   as an add-on to a finished application.  Many of the mechanisms
   discussed in the guidelines below require application support to
   operate correctly.  Application designers need to consider congestion
   control throughout the design of their application, similar to how
   they consider security aspects throughout the design process.

   In the past, the IETF has also investigated integrated congestion
   control mechanisms that act on the traffic aggregate between two
   hosts, i.e., a framework such as the Congestion Manager [RFC3124],
   where active sessions may share current congestion information in a
   way that is independent of the transport protocol.  Such mechanisms
   have currently failed to see deployment, but would otherwise simplify

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   the design of congestion control mechanisms for UDP sessions, so that
   they fulfill the requirements in [RFC2914].

3.1.1.  Bulk Transfer Applications

   Applications that perform bulk transmission of data to a peer over
   UDP, i.e., applications that exchange more than a few UDP datagrams
   per RTT, SHOULD implement TCP-Friendly Rate Control (TFRC) [RFC5348],
   window-based TCP-like congestion control, or otherwise ensure that
   the application complies with the congestion control principles.

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

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

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

   Finally, some bulk transfer applications may choose not to implement
   any congestion control mechanism and instead rely on transmitting
   across reserved path capacity (see Section 3.1.7).  This might be an
   acceptable choice for a subset of restricted networking environments,
   but is by no means a safe practice for operation over the wider
   Internet.  When the UDP traffic of such applications leaks out into
   unprovisioned Internet paths, it can significantly degrade the
   performance of other traffic sharing the path and even result in
   congestion collapse.  Applications that support an uncontrolled or
   unadaptive transmission behavior SHOULD NOT do so by default and
   SHOULD instead require users to explicitly enable this mode of
   operation, and they SHOULD verify that sufficient path capacity has
   been reserved for them.

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quot; are REQUIRED by OLSR:

     -    SYM_NEIGH - indicating that the neighbors have at least one
          symmetrical link with this node.

     -    MPR_NEIGH - indicating that the neighbors have at least one
          symmetrical link AND have been selected as MPR by the sender.

     -    NOT_NEIGH - indicating that the nodes are either no longer or
          have not yet become symmetric neighbors.

   Note that an implementation should be careful in confusing neither
   Link Type with Neighbor Type nor the constants (confusing SYM_NEIGH
   with SYM_LINK for instance).

   A link code advertising:

          Link Type     == SYM_LINK AND

          Neighbor Type == NOT_NEIGH

   is invalid, and any links advertised as such MUST be silently
   discarded without any processing.

   Likewise a Neighbor Type field advertising a numerical value which is
   not one of the constants SYM_NEIGH, MPR_NEIGH, NOT_NEIGH, is invalid,
   and any links advertised as such MUST be silently discarded without
   any processing.

6.2.  HELLO Message Generation

   This involves transmitting the Link Set, the Neighbor Set and the MPR
   Set.  In principle, a HELLO message serves three independent tasks:

     -    link sensing

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     -    neighbor detection

     -    MPR selection signaling

   Three tasks are all are based on periodic information exchange within
   a nodes neighborhood, and serve the common purpose of "local topology
   discovery".  A HELLO message is therefore generated based on the
   information stored in the Local Link Set, the Neighbor Set and the
   MPR Set from the local link information base.

   A node must perform link sensing on each interface, in order to
   detect links between the interface and neighbor interfaces.
   Furthermore, a node must advertise its entire symmetric 1-hop
   neighborhood on each interface in order to perform neighbor
   detection.  Hence, for a given interface, a HELLO message will
   contain a list of links on that interface (with associated link
   types), as well as a list of the entire neighborhood (with an
   associated neighbor types).

   The Vtime field is set such that it corresponds to the value of the
   node's NEIGHB_HOLD_TIME parameter.  The Htime field is set such that
   it corresponds to the value of the node's HELLO_INTERVAL parameter
   (see section 18.3).

   The Willingness field is set such that it corresponds to the node's
   willingness to forward traffic on behalf of other nodes (see section
   18.8).  A node MUST advertise the same willingness on all interfaces.

   The lists of addresses declared in a HELLO message is a list of
   neighbor interface addresses computed as follows:

   For each tuple in the Link Set, where L_local_iface_addr is the
   interface where the HELLO is to be transmitted, and where L_time >=
   current time (i.e., not expired), L_neighbor_iface_addr is advertised
   with:

     1    The Link Type set according to the following:

          1.1  if L_SYM_time >= current time (not expired)

                    Link Type = SYM_LINK

          1.2  Otherwise, if L_ASYM_time >= current time (not expired)
               AND

                             L_SYM_time  <  current time (expired)

                    Link Type = ASYM_LINK

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          1.3  Otherwise, if L_ASYM_time < current time (expired) AND

                             L_SYM_time  < current time (expired)

                    Link Type = LOST_LINK

     2    The Neighbor Type is set according to the following:

          2.1  If the main address, corresponding to
               L_neighbor_iface_addr, is included in the MPR set:

                    Neighbor Type = MPR_NEIGH

          2.2  Otherwise, if the main address, corresponding to
               L_neighbor_iface_addr, is included in the neighbor set:

               2.2.1
                    if N_status == SYM

                         Neighbor Type = SYM_NEIGH

               2.2.2
                    Otherwise, if N_status == NOT_SYM
                         Neighbor Type = NOT_NEIGH

   For each tuple in the Neighbor Set, for which no
   L_neighbor_iface_addr from an associated link tuple has been
   advertised by the previous algorithm,  N_neighbor_main_addr is
   advertised with:

     - Link Type = UNSPEC_LINK,

     - Neighbor Type set as described in step 2 above

   For a node with a single OLSR interface, the main address is simply
   the address of the OLSR interface, i.e., for a node with a single
   OLSR interface the main address, corresponding to
   L_neighbor_iface_addr is simply L_neighbor_iface_addr.

   A HELLO message can be partial (e.g., due to message size
   limitations, imposed by the network), the rule being the following,
   on each interface: each link and each neighbor node MUST be cited at
   least once within a predetermined refreshing period,
   REFRESH_INTERVAL.  To keep track of fast connectivity changes, a
   HELLO message must be sent at least every HELLO_INTERVAL period,
   smaller than or equal to REFRESH_INTERVAL.

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   Notice that for limiting the impact from loss of control messages, it
   is desirable that a message (plus the generic packet header) can fit
   into a single MAC frame.

6.3.  HELLO Message Forwarding

   Each HELLO message generated is broadcast by the node on one
   interface to its neighbors (i.e. the interface for which the HELLO
   was generated).  HELLO messages MUST never be forwarded.

6.4.  HELLO Message Processing

   A node processes incoming HELLO messages for the purpose of
   conducting link sensing (detailed in section 7), neighbor detection
   and MPR selector set population (detailed in section 8)

7.  Link Sensing

   Link sensing populates the local link information base.  Link sensing
   is exclusively concerned with OLSR interface addresses and the
   ability to exchange packets between such OLSR interfaces.

   The mechanism for link sensing is the periodic exchange of HELLO
   messages.

7.1.  Populating the Link Set

   The Link Set is populated with information on links to neighbor
   nodes.  The process of populating this set is denoted "link sensing"
   and is performed using HELLO message exchange, updating a local link
   information base in each node.

   Each node should detect the links between itself and neighbor nodes.
   Uncertainties over radio propagation may make some links
   unidirectional.  Consequently, all links MUST be checked in both
   directions in order to be considered valid.

   A "link" is described by a pair of interfaces: a local and a remote
   interface.

   For the purpose of link sensing, each neighbor node (more
   specifically, the link to each neighbor) has an associated status of
   either "symmetric" or "asymmetric".  "Symmetric" indicates, that the
   link to that neighbor node has been verified to be bi-directional,
   i.e., it is possible to transmit data in both directions.
   "Asymmetric" indicates that HELLO messages from the node have been

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   heard (i.e., communication from the neighbor node is possible),
   however it is not confirmed that this node is also able to receive
   messages (i.e., communication to the neighbor node is not confirmed).

   The information, acquired through and used by the link sensing, is
   accumulated in the link set.

7.1.1.  HELLO Message Processing

   The "Originator Address" of a HELLO message is the main address of
   the node, which has emitted the message.

   Upon receiving a HELLO message, a node SHOULD update its Link Set.
   Notice, that a HELLO message MUST neither be forwarded nor be
   recorded in the duplicate set.

   Upon receiving a HELLO message, the "validity time" MUST be computed
   from the Vtime field of the message header (see section 3.3.2).
   Then, the Link Set SHOULD be updated as follows:

     1    Upon receiving a HELLO message, if there exists no link tuple
          with

               L_neighbor_iface_addr == Source Address

          a new tuple is created with

               L_neighbor_iface_addr = Source Address

               L_local_iface_addr    = Address of the interface
                                       which received the
                                       HELLO message

               L_SYM_time            = current time - 1 (expired)

               L_time                = current time + validity time

     2    The tuple (existing or new) with:

               L_neighbor_iface_addr == Source Address

          is then modified as follows:

          2.1  L_ASYM_time = current time + validity time;

          2.2  if the node finds the address of the interface which
               received the HELLO message among the addresses listed in
               the link message then the tuple is modified as follows:

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               2.2.1
                    if Link Type is equal to LOST_LINK then

                         L_SYM_time = current time - 1 (i.e., expired)

               2.2.2
                    else if Link Type is equal to SYM_LINK or ASYM_LINK
                    then

                         L_SYM_time = current time + validity time,

                         L_time     = L_SYM_time + NEIGHB_HOLD_TIME

          2.3  L_time = max(L_time, L_ASYM_time)

   The above rule for setting L_time is the following: a link losing its
   symmetry SHOULD still be advertised during at least the duration of
   the "validity time" advertised in the generated HELLO.  This allows
   neighbors to detect the link breakage.

8.  Neighbor Detection

   Neighbor detection populates the neighborhood information base and
   concerns itself with nodes and node main addresses.  The relationship
   between OLSR interface addresses and main addresses is described in
   section 5.

   The mechanism for neighbor detection is the periodic exchange of
   HELLO messages.

8.1.  Populating the Neighbor Set

   A node maintains a set of neighbor tuples, based on the link tuples.
   This information is updated according to changes in the Link Set.

   The Link Set keeps the information about the links, while the
   Neighbor Set keeps the information about the neighbors.  There is a
   clear association between those two sets, since a node is a neighbor
   of another node if and only if there is at least one link between the
   two nodes.

   In any case, the formal correspondence between links and neighbors is
   defined as follows:

          The "associated neighbor tuple" of a link tuple, is, if it
          exists, the neighbor tuple where:

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               N_neighbor_main_addr == main address of
                                       L_neighbor_iface_addr

          The "associated link tuples" of a neighbor tuple, are all the
          link tuples, where:

               N_neighbor_main_addr == main address of
                                       L_neighbor_iface_addr

   The Neighbor Set MUST be populated by maintaining the proper
   correspondence between link tuples and associated neighbor tuples, as
   follows:

     Creation

          Each time a link appears, that is, each time a link tuple is
          created, the associated neighbor tuple MUST be created, if it
          doesn't already exist, with the following values:

               N_neighbor_main_addr = main address of
                                      L_neighbor_iface_addr
                                      (from the link tuple)

          In any case, the N_status MUST then be computed as described
          in the next step

     Update

          Each time a link changes, that is, each time the information
          of a link tuple is modified, the node MUST ensure that the
          N_status of the associated neighbor tuple respects the
          property:

               If the neighbor has any associated link tuple which
               indicates a symmetric link (i.e., with L_SYM_time >=
               current time), then

                    N_status is set to SYM

               else N_status is set to NOT_SYM

     Removal

          Each time a link is deleted, that is, each time a link tuple
          is removed, the associated neighbor tuple MUST be removed if
          it has no longer any associated link tuples.

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   These rules ensure that there is exactly one associated neighbor
   tuple for a link tuple, and that every neighbor tuple has at least
   one associated link tuple.

8.1.1.  HELLO Message Processing

   The "Originator Address" of a HELLO message is the main address of
   the node, which has emitted the message.  Likewise, the "willingness"
   MUST be computed from the Willingness field of the HELLO message (see
   section 6.1).

   Upon receiving a HELLO message, a node SHOULD first update its Link
   Set as described before.  It SHOULD then update its Neighbor Set as
   follows:

     -    if the Originator Address is the N_neighbor_main_addr from a
          neighbor tuple included in the Neighbor Set:

               then, the neighbor tuple SHOULD be updated as follows:

               N_willingness = willingness from the HELLO message

8.2.  Populating the 2-hop Neighbor Set

   The 2-hop neighbor set describes the set of nodes which have a
   symmetric link to a symmetric neighbor.  This information set is
   maintained through periodic exchange of HELLO messages as described
   in this section.

8.2.1.  HELLO Message Processing

   The "Originator Address" of a HELLO message is the main address of
   the node, which has emitted the message.

   Upon receiving a HELLO message from a symmetric neighbor, a node
   SHOULD update its 2-hop Neighbor Set.  Notice, that a HELLO message
   MUST neither be forwarded nor be recorded in the duplicate set.

   Upon receiving a HELLO message, the "validity time" MUST be computed
   from the Vtime field of the message header (see section 3.3.2).

   If the Originator Address is the main address of a
   L_neighbor_iface_addr from a link tuple included in the Link Set with

          L_SYM_time >= current time (not expired)

   (in other words: if the Originator Address is a symmetric neighbor)
   then the 2-hop Neighbor Set SHOULD be updated as follows:

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     1    for each address (henceforth: 2-hop neighbor address), listed
          in the HELLO message with Neighbor Type equal to SYM_NEIGH or
          MPR_NEIGH:

          1.1  if the main address of the 2-hop neighbor address = main
               address of the receiving node:

                    silently discard the 2-hop neighbor address.

               (in other words: a node is not its own 2-hop neighbor).

          1.2  Otherwise, a 2-hop tuple is created with:

                    N_neighbor_main_addr =  Originator Address;

                    N_2hop_addr          =  main address of the
                                            2-hop neighbor;

                    N_time               =  current time
                                            + validity time.

               This tuple may replace an older similar tuple with same
               N_neighbor_main_addr and N_2hop_addr values.

     2    For each 2-hop node listed in the HELLO message with Neighbor
          Type equal to NOT_NEIGH, all 2-hop tuples where:

               N_neighbor_main_addr == Originator Address AND

               N_2hop_addr          == main address of the
                                       2-hop neighbor

          are deleted.

8.3.  Populating the MPR set

   MPRs are used to flood control messages from a node into the network
   while reducing the number of retransmissions that will occur in a
   region.  Thus, the concept of MPR is an optimization of a classical
   flooding mechanism.

   Each node in the network selects, independently, its own set of MPRs
   among its symmetric 1-hop neighborhood.  The symmetric links with
   MPRs are advertised with Link Type MPR_NEIGH instead of SYM_NEIGH in
   HELLO messages.

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   The MPR set MUST be calculated by a node in such a way that it,
   through the neighbors in the MPR-set, can reach all symmetric strict
   2-hop neighbors.  (Notice that a node, a, which is a direct neighbor
   of another node, b, is not also a strict 2-hop neighbor of node b).
   This means that the union of the symmetric 1-hop neighborhoods of the
   MPR nodes contains the symmetric strict 2-hop neighborhood.  MPR set
   recalculation should occur when changes are detected in the symmetric
   neighborhood or in the symmetric strict 2-hop neighborhood.

   MPRs are computed per interface, the union of the MPR sets of each
   interface make up the MPR set for the node.

   While it is not essential that the MPR set is minimal, it is
   essential that all strict 2-hop neighbors can be reached through the
   selected MPR nodes.  A node SHOULD select an MPR set such that any
   strict 2-hop neighbor is covered by at least one MPR node.  Keeping
   the MPR set small ensures that the overhead of the protocol is kept
   at a minimum.

   The MPR set can coincide with the entire symmetric neighbor set.
   This could be the case at network initialization (and will correspond
   to classic link-state routing).

8.3.1.  MPR Computation

   The following specifies a proposed heuristic for selection of MPRs.
   It constructs an MPR-set that enables a node to reach any node in the
   symmetrical strict 2-hop neighborhood through relaying by one MPR
   node with willingness different from WILL_NEVER.  The heuristic MUST
   be applied per interface, I.  The MPR set for a node is the union of
   the MPR sets found for each interface.  The following terminology
   will be used in describing the heuristics:

       neighbor of an interface

              a node is a "neighbor of an interface" if the interface
              (on the local node) has a link to any one interface of
              the neighbor node.

       2-hop neighbors reachable from an interface

              the list of 2-hop neighbors of the node that can be
              reached from neighbors of this interface.

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       3.1.2.  Low Data-Volume Applications

   When applications that at any time exchange only a few UDP datagrams
   with a destination implement TFRC or one of the other congestion
   control schemes in Section 3.1.1, the network sees little benefit,
   because those mechanisms perform congestion control in a way that is
   only effective for longer transmissions.

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

   Some applications cannot maintain a reliable RTT estimate for a
   destination.  The first case is that of applications that exchange
   too few UDP datagrams with a peer to establish a statistically
   accurate RTT estimate.  Such applications MAY use a predetermined
   transmission interval that is exponentially backed-off when packets
   are lost.  TCP uses an initial value of 3 seconds [RFC6298], which is
   also RECOMMENDED as an initial value for UDP applications.  SIP
   [RFC3261] and GIST [RFC5971] use an interval of 500 ms, and shorter
   values are likely problematic in many cases.  As in the previous
   case, note that the initial timeout is not the maximum possible
   timeout.

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

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

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

3.1.3.  Implications of RTT and Loss Measurements on Congestion Control

   Transports such as TCP, SCTP and DCCP provide timely detection of
   congestion that results in an immediate reduction of their maximum
   sending rate when congestion is experienced.  This reaction is
   typically completed 1-2 RTTs after loss/congestion is encountered.
   Applications using UDP SHOULD implement a congestion control scheme
   that provides a prompt reaction to signals indicating congestion
   (e.g., by reducing the rate within the next RTT following a
   congestion signal).

   The operation of a UDP Congestion Control algorithm can be very
   different to the way TCP operates.  This includes congestion controls
   that respond on timescales that fit applications that cannot usefully
   work within the "change rate every RTT" model of TCP.  Applications
   that experience a low or varying RTT are particularly vulnerable to
   sampling errors (e.g., due to measurement noise, or timer accuracy).
   This suggests the need to average loss/congestion and RTT
   measurements over a longer interval, however this also can contribute
   additional delay in detecting congestion.  Some applications may not
   react by reducing their sending rate immediately for various reasons,
   including: RTT and loss measurements are only made periodically
   (e.g., using RTCP), additional time is required to filter
   information, or the application is only able to change its sending
   rate at predetermined interval (e.g., some video codecs).

   When designing a congestion control algorithm, the designer therefore
   needs to consider the total time taken to reduce the load following a
   lack of feedback or a congestion event.  An application where the
   most recent RTT measurement is smaller than the actual RTT or the
   measured loss rate is smaller than the current rate, can result in
   over estimating the available capacity.  Such over estimation can

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   result in a sending rate that creates congestion to the application
   or other flows sharing the path capacity, and can contribute to
   congestion collapse - both of these need to be avoided.

   A congestion control designed for UDP SHOULD respond as quickly as
   possible when it experiences congestion, and SHOULD take into account
   both the loss rate and the response time when choosing a new rate.
   The implemented congestion control scheme SHOULD result in bandwidth
   (capacity) use that is comparable to that of TCP within an order of
   magnitude, so that it does not starve other flows sharing a common
   bottleneck.

3.1.4.  Burst Mitigation and Pacing

   UDP applications SHOULD provide mechanisms to regulate the bursts of
   transmission that the application may send to the network.  Many TCP
   and SCTP implementations provide mechanisms that prevent a sender
   from generating long bursts at line-rate, since these are known to
   induce early loss to applications sharing a common network
   bottleneck.  The use of pacing with TCP [ALLMAN] has also been shown
   to improve the coexistence of TCP flows with other flows.  The need
   to avoid excessive transmission bursts is also noted in
   specifications for applications (e.g., [RFC7143]).

   Even low data-volume UDP flows may benefit from packet pacing, e.g.,
   an application that sends three copies of a packet to improve
   robustness to loss is RECOMMENDED to pace out those three packets
   over several RTTs, to reduce the probability that all three packets
   will be lost due to the same congestion event (or other event, such
   as burst corruption).

3.1.5.  Explicit Congestion Notification

   Internet applications can use Explicit Congestion Notification (ECN)
   [RFC3168] to gain benefits for the services they support
   [I-D.ietf-aqm-ecn-benefits].

   Internet transports, such as TCP, provide a set of mechanisms that
   are needed to utilize ECN.  ECN operates by setting an ECN-capable
   codepoint (ECT(0) or ECT(1)) in the IP header of packets that are
   sent.  This indicates to ECN-capable network devices (routers, and
   other devices) that they may mark (set the congestion experienced, CE
   codepoint), rather than drop the IP packet as a signal of incipient
   congestion.

   UDP applications can also benefit from enabling ECN, providing that
   the API supports ECN and that they implement the required protocol
   mechanisms to support ECN.

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   The requirements for UDP-based tunnels to support ECN are described
   in Section 3.1.9.

   The set of mechanisms requires for an application to use ECN over UDP
   are:

   o  A sender MUST provide a method to determine (e.g., negotiate) that
      the corresponding application is able to provide ECN feedback
      using a compatible ECN method.

   o  A receiver that enables the use of ECN for a UDP port MUST check
      the ECN field at the receiver for each UDP datagram that it
      receives on this port.

   o  The receiving application needs to provide feedback of congestion
      information to the sending application.  This MUST report the
      presence of datagrams received with a CE-mark by providing a
      mechanism to feed this congestion information back to the sending
      application.  The feedback MAY also report the presence of ECT(1)
      and ECT(0)/Not-ECT packets [RFC7560].  ([RFC3168] and [RFC7560]
      specify methods for TCP.)

   o  An application sending ECN-capable datagrams MUST provide an
      appropriate congestion reaction when it receives feedback
      indicating that congestion has been experienced.  This must result
      in reduction of the sending rate by the UDP congestion control
      method Section 3.1 that is not less than the reaction of TCP under
      equivalent conditions.

   o  A sender SHOULD detect network paths that do not support the ECN
      field correctly.  When detected they need to either conservatively
      react to congestion or even fall back to not using ECN
      [I-D.ietf-aqm-ecn-benefits].  This method needs to be robust to
      changes within the network path that may occur over the lifetime
      of a session.

   o  A sender is encouraged to provide a mechanism to detect and react
      appropriately to misbehaving receivers that fail to report CE-
      marked packets [I-D.ietf-aqm-ecn-benefits].

   [RFC6679] provides guidance an example of this support, by describing
   a method to allow ECN to be used for UDP-based applications using the
   Real-Time Protocol (RTP).  Applications that cannot provide this set
   of mechanisms, but wish to gain the benefits of using ECN, are
   encouraged to use a transport protocol that already supports ECN
   (such as TCP).

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3.1.6.  Differentiated Services Model

   An application using UDP can use the differentiated services QoS
   framework.  To enable differentiated services processing, a UDP
   sender sets the Differentiated Services Code Point (DSCP) field
   [RFC2475] in packets sent to the network.  Normally, a UDP source/
   destination port pair will set a single DSCP value for all packets
   belonging to a flow, but multiple DSCPs can be used as described
   later in this section.  A DSCP may be chosen from a small set of
   fixed values (the class selector code points), or from a set of
   recommended values defined in the Per Hop Behavior (PHB)
   specifications, or from values that have purely local meanings to a
   specific network that supports DiffServ.  In general, packets may be
   forwarded across multiple networks the between source and
   destination.

   In setting a non-default DSCP value, an application must be aware
   that DSCP markings may be changed or removed between the traffic
   source and destination.  This has implications on the design of
   applications that use DSCPs.  Specifically, applications SHOULD be
   designed to not rely on implementation of a specific network
   treatment, they need instead to implement congestion control methods
   to determine if their current sending rate is inducing congestion in
   the network.

   [I-D.ietf-dart-dscp-rtp] describes the implications of using DSCPs
   and provides recommendations on using multiple DSCPs within a single
   network five-tuple (source and destination addresses, source and
   destination ports, and the transport protocol used, in this case, UDP
   or UDP-Lite), and particularly the expected impact on transport
   protocol interactions, with congestion control or reliability
   functionality (e.g., retransmission, reordering).  Use of multiple
   DSCPs can result in reordering by increasing the set of network
   forwarding resources used by a sender.  It can also increase exposure
   to resource depletion or failure.

3.1.7.  QoS, Preprovisioned or Reserved Capacity

   The IETF usually specifies protocols for use within the Best Effort
   General Internet.  Sometimes it is relevant to specify protocols with
   a different applicability.  An application using UDP can use the
   integrated services QoS framework.  This framework is usually made
   available within controlled environments (e.g., within a single
   administrative domain or bilaterally agreed connection between
   domains).  Applications intended for the Internet SHOULD NOT assume
   that QoS mechanisms are supported by the networks they use, and
   therefore need to provide congestion control, error recovery, etc. in
   case the actual network path does not provide provisioned service.

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   Some UDP applications are only expected to be deployed over network
   paths that use preprovisioned capacity or capacity reserved using
   dynamic provisioning, e.g., through the Resource Reservation Protocol
   (RSVP).  Multicast applications are also used with preprovisioned
   capacity (e.g., IPTV deployments within access networks).  These
   applications MAY choose not to implement any congestion control
   mechanism and instead rely on transmitting only on paths where the
   capacity is provisioned and reserved for this use.  This might be an
   acceptable choice for a subset of restricted networking environments,
   but is by no means a safe practice for operation over the wider
   Internet.  Applications that chose this option SHOULD carefully and
   in detail describe the provisioning and management procedures that
   result in the desired containment.

   Applications that support an uncontrolled or unadaptive transmission
   behavior SHOULD NOT do so by default and SHOULD instead require users
   to explicitly enable this mode of operation.

   Applications designed for use within a controlled network environment
   (see section Section 3.6 ) may be able to exploit network management
   functions to detect whether they are causing congestion, and react
   accordingly.  If the traffic of such applications leaks out into
   unprovisioned Internet paths, it can significantly degrade the
   performance of other traffic sharing the path and even result in
   congestion collapse.  Protocols designed for such networks SHOULD
   provide mechanisms at the network edge to prevent leakage of traffic
   into unprovisioned Internet paths (e.g., [RFC7510]) To protect other
   applications sharing the same path, applications SHOULD also deploy
   an appropriate circuit breaker, as described in Section 3.1.8.

3.1.8.  Circuit Breaker Mechanisms

   A transport circuit breaker is an automatic mechanism that is used to
   estimate the congestion caused by a flow, and to terminate (or
   significantly reduce the rate of) the flow when excessive congestion
   is detected [I-D.ietf-tsvwg-circuit-breaker].  This is a safety
   measure to prevent congestion collapse (starvation of resources
   available to other flows), essential for an Internet that is
   heterogeneous and for traffic that is hard to predict in advance.

   A circuit breaker is intended as a protection mechanism of last
   resort.  Under normal circumstances, a circuit breaker should not be
   triggered; it is designed to protect things when there is severe
   overload.  The goal is usually to limit the maximum transmission rate
   that reflects the available capacity of a network path.  Circuit
   breakers can operate on individual UDP flows or traffic aggregates,
   e.g., traffic sent using a network tunnel.

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

   Applications used in the general Internet SHOULD implement a
   transport circuit breaker if they do not implement congestion control
   or operate a low volume data service (see Section 3.6).  All
   applications MAY implement a transport circuit breaker
   [I-D.ietf-tsvwg-circuit-breaker] and are encouraged to consider
   implementing at least a slow-acting transport circuit breaker to
   provide a protection of last resort for their network traffic.

3.1.9.  UDP Tunnels

   One increasingly popular use of UDP is as a tunneling protocol
   [I-D.ietf-intarea-tunnels], where a tunnel endpoint encapsulates the
   packets of another protocol inside UDP datagrams and transmits them
   to another tunnel endpoint, which decapsulates the UDP datagrams and
   forwards the original packets contained in the payload.  Tunnels
   establish virtual links that appear to directly connect locations
   that are distant in the physical Internet topology and can be used to
   create virtual (private) networks.  Using UDP as a tunneling protocol
   is attractive when the payload protocol is not supported by
   middleboxes that may exist along the path, because many middleboxes
   support transmission using UDP.

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

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

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

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

   The following guidelines define these possible cases in more detail:

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

       This is arguably the most common case for Internet tunnels.  In
       this case, the UDP tunnel SHOULD NOT employ its own congestion
       control mechanism, because congestion losses of tunneled traffic
       will already trigger an appropriate congestion response at the
       original senders of the tunneled traffic.  A circuit breaker
       mechanism may provide benefit by controlling the envelope of the
       aggregated traffic.

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

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

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

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

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

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              a (sub)set of the neighbors of an interface with a
              willingness different from WILL_NEVER, selected such that
              through these selected nodes, all strict 2-hop neighbors
              reachable from that interface are reachable.

       N:
              N is the subset of neighbors of the node, which are
              neighbor of the interface I.

       N2:
              The set of 2-hop neighbors reachable from the interface
              I, excluding:

               (i)   the nodes only reachable by members of N with
                     willingness WILL_NEVER

               (ii)  the node performing the computation

               (iii) all the symmetric neighbors: the nodes for which
                     there exists a symmetric link to this node on some
                     interface.

    D(y):
              The degree of a 1-hop neighbor node y (where y is a
              member of N), is defined as the number of symmetric
              neighbors of node y, EXCLUDING all the members of N and
              EXCLUDING the node performing the computation.

   The proposed heuristic is as follows:

     1    Start with an MPR set made of all members of N with
          N_willingness equal to WILL_ALWAYS

     2    Calculate D(y), where y is a member of N, for all nodes in N.

     3    Add to the MPR set those nodes in N, which are the *only*
          nodes to provide reachability to a node in N2.  For example,
          if node b in N2 can be reached only through a symmetric link
          to node a in N, then add node a to the MPR set.  Remove the
          nodes from N2 which are now covered by a node in the MPR set.

     4    While there exist nodes in N2 which are not covered by at
          least one node in the MPR set:

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          4.1  For each node in N, calculate the reachability, i.e., the
               number of nodes in N2 which are not yet covered by at
               least one node in the MPR set, and which are reachable
               through this 1-hop neighbor;

          4.2  Select as a MPR the node with highest N_willingness among
               the nodes in N with non-zero reachability.  In case of
               multiple choice select the node which provides
               reachability to the maximum number of nodes in N2.  In
               case of multiple nodes providing the same amount of
               reachability, select the node as MPR whose D(y) is
               greater.  Remove the nodes from N2 which are now covered
               by a node in the MPR set.

     5    A node's MPR set is generated from the union of the MPR sets
          for each interface.  As an optimization, process each node, y,
          in the MPR set in increasing order of N_willingness.  If all
          nodes in N2 are still covered by at least one node in the MPR
          set excluding node y, and if N_willingness of node y is
          smaller than WILL_ALWAYS, then node y MAY be removed from the
          MPR set.

   Other algorithms, as well as improvements over this algorithm, are
   possible.  For example, assume that in a multiple-interface scenario
   there exists more than one link between nodes 'a' and 'b'.  If node
   'a' has selected node 'b' as MPR for one of its interfaces, then node
   'b' can be selected as MPR without additional performance loss by any
   other interfaces on node 'a'.

8.4.  Populating the MPR Selector Set

   The MPR selector set of a node, n, is populated by the main addresses
   of the nodes which have selected n as MPR.  MPR selection is signaled
   through HELLO messages.

8.4.1.  HELLO Message Processing

   Upon receiving a HELLO message, if a node finds one of its own
   interface addresses in the list with a Neighbor Type equal to
   MPR_NEIGH, information from the HELLO message must be recorded in the
   MPR Selector Set.

   The "validity time" MUST be computed from the Vtime field of the
   message header (see section 3.3.2).  The MPR Selector Set SHOULD then
   be updated as follows:

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     1    If there exists no MPR selector tuple with:

                    MS_main_addr   == Originator Address

               then a new tuple is created with:

                    MS_main_addr   =  Originator Address

     2    The tuple (new or otherwise) with

               MS_main_addr   == Originator Address

          is then modified as follows:

               MS_time        =  current time + validity time.

   Deletion of MPR selector tuples occurs in case of expiration of the
   timer or in case of link breakage as described in the "Neighborhood
   and 2-hop Neighborhood Changes".

8.5.  Neighborhood and 2-hop Neighborhood Changes

   A change in the neighborhood is detected when:

     -    The L_SYM_time field of a link tuple expires.  This is
          considered as a neighbor loss if the link described by the
          expired tuple was the last link with a neighbor node (on the
          contrary, a link with an interface may break while a link with
          another interface of the neighbor node remains without being
          observed as a neighborhood change).

     -    A new link tuple is inserted in the Link Set with a non
          expired L_SYM_time or a tuple with expired L_SYM_time is
          modified so that L_SYM_time becomes non-expired.  This is
          considered as a neighbor appearance if there was previously no
          link tuple describing a link with the corresponding neighbor
          node.

   A change in the 2-hop neighborhood is detected when a 2-hop neighbor
   tuple expires or is deleted according to section 8.2.

   The following processing occurs when changes in the neighborhood or
   the 2-hop neighborhood are detected:

     -    In case of neighbor loss, all 2-hop tuples with
          N_neighbor_main_addr == Main Address of the neighbor MUST be
          deleted.

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     -    In case of neighbor loss, all MPR selector tuples with
          MS_main_addr == Main Address of the neighbor MUST be deleted

     -    The MPR set MUST be re-calculated when a neighbor appearance
          or loss is detected, or when a change in the 2-hop
          neighborhood is detected.

     -    An additional HELLO message MAY be sent when the MPR set
          changes.

9.  Topology Discovery

   The link sensing and neighbor detection part of the protocol
   basically offers, to each node, a list of neighbors with which it can
   communicate directly and, in combination with the Packet Format and
   Forwarding part, an optimized flooding mechanism through MPRs.  Based
   on this, topology information is disseminated through the network.
   The present section describes which part of the information given by
   the link sensing and neighbor detection is disseminated to the entire
   network and how it is used to construct routes.

   Routes are constructed through advertised links and links with
   neighbors.  A node must at least disseminate links between itself and
   the nodes in its MPR-selector set, in order to provide sufficient
   information to enable routing.

9.1.  TC Message Format

   The proposed format of a TC message is as follows:

       0                   1                   2                   3
       0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1
      +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
      |              ANSN             |           Reserved            |
      +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
      |               Advertised Neighbor Main Address                |
      +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
      |               Advertised Neighbor Main Address                |
      +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
      |                              ...                              |
      +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+

   This is sent as the data-portion of the general message format with
   the "Message Type" set to TC_MESSAGE.  The time to live SHOULD be set
   to 255 (maximum value) to diffuse the message into the entire network
   and Vtime set accordingly to the value of TOP_HOLD_TIME, as specified
   in section 18.3.

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     Advertised Neighbor Sequence Number (ANSN)

          A sequence number is associated with the advertised neighbor
          set.  Every time a node detects a change in its advertised
          neighbor set, it increments this sequence number ("Wraparound"
          is handled as described in section 19).  This number is sent
          in this ANSN field of the TC message to keep track of the most
          recent information.  When a node receives a TC message, it can
          decide on the basis of this Advertised Neighbor Sequence
          Number, whether or not the received information about the
          advertised neighbors of the originator node is more recent
          than what it already has.

     Advertised Neighbor Main Address

          This field contains the main address of a neighbor node.  All
          main addresses of the advertised neighbors of the Originator
          node are put in the TC message.  If the maximum allowed
          message size (as imposed by the network) is reached while
          there are still advertised neighbor addresses which have not
          been inserted into the TC-message, more TC messages will be
          generated until the entire advertised neighbor set has been
          sent.  Extra main addresses of neighbor nodes may be included,
          if redundancy is desired.

     Reserved

          This field is reserved, and MUST be set to "0000000000000000"
          for compliance with this document.

9.2.  Advertised Neighbor Set

   A TC message is sent by a node in the network to declare a set of
   links, called advertised link set which MUST include at least the
   links to all nodes of its MPR Selector set, i.e., the neighbors which
   have selected the sender node as a MPR.

   If, for some reason, it is required to distribute redundant TC
   information, refer to section 15.

   The sequence number (ANSN) associated with the advertised neighbor
   set is also sent with the list.  The ANSN number MUST be incremented
   when links are removed from the advertised neighbor set; the ANSN
   number SHOULD be incremented when links are added to the advertised
   neighbor set.

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9.3.  TC Message Generation

   In order to build the topology information base, each node, which has
   been selected as MPR, broadcasts Topology Control (TC) messages.  TC
   messages are flooded to all nodes in the network and take advantage
   of MPRs.  MPRs enable a better scalability in the distribution of
   topology information [1].

   The list of addresses can be partial in each TC message (e.g., due to
   message size limitations, imposed by the network), but parsing of all
   TC messages describing the advertised link set of a node MUST be
   complete within a certain refreshing period (TC_INTERVAL).  The
   information diffused in the network by these TC messages will help
   each node calculate its routing table.

   When the advertised link set of a node becomes empty, this node
   SHOULD still send (empty) TC-messages during the a duration equal to
   the "validity time" (typically, this will be equal to TOP_HOLD_TIME)
   of its previously emitted TC-messages, in order to invalidate the
   previous TC-messages.  It SHOULD then stop sending TC-messages until
   some node is inserted in its advertised link set.

   A node MAY transmit additional TC-messages to increase its
   reactiveness to link failures.  When a change to the MPR selector set
   is detected and this change can be attributed to a link failure, a
   TC-message SHOULD be transmitted after an interval shorter than
   TC_INTERVAL.

9.4.  TC Message Forwarding

   TC messages are broadcast and retransmitted by the MPRs in order to
   diffuse the messages in the entire network.  TC messages MUST be
   forwarded according to the "default forwarding algorithm" (described
   in section 3.4).

9.5.  TC Message Processing

   Upon receiving a TC message, the "validity time" MUST be computed
   from the Vtime field of the message header (see section 3.3.2).  The
   topology set SHOULD then be updated as follows (using section 19 for
   comparison of ANSN):

     1    If the sender interface (NB: not originator) of this message
          is not in the symmetric 1-hop neighborhood of this node, the
          message MUST be discarded.

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     2    If there exist some tuple in the topology set where:

               T_last_addr == originator address AND

               T_seq       >  ANSN,

          then further processing of this TC message MUST NOT be
          performed and the message MUST be silently discarded (case:
          message received out of order).

     3    All tuples in the topology set where:

               T_last_addr == originator address AND

               T_seq       <  ANSN

          MUST be removed from the topology set.

     4    For each of the advertised neighbor main address received in
          the TC message:

          4.1  If there exist some tuple in the topology set where:

                    T_dest_addr == advertised neighbor main address, AND

                    T_last_addr == originator address,

               then the holding time of that tuple MUST be set to:

                    T_time      =  current time + validity time.

          4.2  Otherwise, a new tuple MUST be recorded in the topology
               set where:

                    T_dest_addr = advertised neighbor main address,

                    T_last_addr = originator address,

                    T_seq       = ANSN,

                    T_time      = current time + validity time.

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10.  Routing Table Calculation

   Each node maintains a routing table which allows it to route data,
   destined for the other nodes in the network.  The routing table is
   based on the information contained in the local link information base
   and the topology set.  Therefore, if any of these sets are changed,
   the routing table is recalculated to update the route information
   about each destination in the network.  The route entries are
   recorded in the routing table in the following format:

         1.  R_dest_addr    R_next_addr    R_dist   R_iface_addr
         2.  R_dest_addr    R_next_addr    R_dist   R_iface_addr
         3.      ,,             ,,           ,,          ,,

   Each entry in the table consists of R_dest_addr, R_next_addr, R_dist,
   and R_iface_addr.  Such entry specifies that the node identified by
   R_dest_addr is estimated to be R_dist hops away from the local node,
   that the symmetric neighbor node with interface address R_next_addr
   is the next hop node in the route to R_dest_addr, and that this
   symmetric neighbor node is reachable through the local interface with
   the address R_iface_addr.  Entries are recorded in the routing table
   for each destination in the network for which a route is known.  All
   the destinations, for which a route is broken or only partially
   known, are not recorded in the table.

   More precisely, the routing table is updated when a change is
   detected in either:

     -    the link set,

     -    the neighbor set,

     -    the 2-hop neighbor set,

     -    the topology set,

     -    the Multiple Interface Association Information Base,

   More precisely, the routing table is recalculated in case of neighbor
   appearance or loss, when a 2-hop tuple is created or removed, when a
   topology tuple is created or removed or when multiple interface
   association information changes.  The update of this routing
   information does not generate or trigger any messages to be
   transmitted, neither in the network, nor in the 1-hop neighborhood.

   To construct the routing table of node X, a shortest path algorithm
   is run on the directed graph containing the arcs X -> Y where Y is
   any symmetric neighbor of X (with Neighbor Type equal to SYM), the

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   arcs Y -> Z where Y is a neighbor node with willingness different of
   WILL_NEVER and there exists an entry in the 2-hop Neighbor set with Y
   as N_neighbor_main_addr and Z as N_2hop_addr, and the arcs U -> V,
   where there exists an entry in the topology set with V as T_dest_addr
   and U as T_last_addr.

   The following procedure is given as an example to calculate (or
   recalculate) the routing table:

     1    All the entries from the routing table are removed.

     2    The new routing entries are added starting with the
          symmetric neighbors (h=1) as the destination nodes. Thus, for
          each neighbor tuple in the neighbor set where:

               N_status   = SYM

          (there is a symmetric link to the neighbor), and for each
          associated link tuple of the neighbor node such that L_time >=
          current time, a new routing entry is recorded in the routing
          table with:

               R_dest_addr  = L_neighbor_iface_addr, of the
                              associated link tuple;

               R_next_addr  = L_neighbor_iface_addr, of the
                              associated link tuple;

               R_dist       = 1;

               R_iface_addr = L_local_iface_addr of the
                              associated link tuple.

          If in the above, no R_dest_addr is equal to the main address
          of the neighbor, then another new routing entry with MUST be
          added, with:

               R_dest_addr  = main address of the neighbor;

               R_next_addr  = L_neighbor_iface_addr of one of the
                              associated link tuple with L_time >=
               current time;

               R_dist       = 1;

               R_iface_addr = L_local_iface_addr of the
                              associated link tuple.

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       tunneling go beyond the scope of the general guidelines given in
       this document.  The implementer of such specialized tunnels
       SHOULD carefully consider congestion control in the design of
       their tunneling mechanism and SHOULD consider use of a circuit
       breaker mechanism.

   The type of encapsulated payload might be identified by a UDP port;
   identified by an Ethernet Type or IP protocol number.  A tunnel
   SHOULD provide mechanisms to restrict the types of flows that may be
   carried by the tunnel.  For instance, a UDP tunnel designed to carry
   IP needs to filter out non-IP traffic at the ingress.  This is
   particularly important when a generic tunnel encapsulation is used
   (e.g., one that encapsulates using an EtherType value).  Such tunnels
   SHOULD provide a mechanism to restrict the types of traffic that are
   allowed to be encapsulated for a given deployment (see
   [I-D.ietf-intarea-tunnels]).

   Designing a tunneling mechanism requires significantly more expertise
   than needed for many other UDP applications, because tunnels are
   usually intended to be transparent to the endpoints transmitting over
   them, so they need to correctly emulate the behavior of an IP link
   [I-D.ietf-intarea-tunnels], e.g., handling fragmentation and packet
   size, treatment of DSCP values, generating and responding to ICMP
   messages, indication of payload type and layering of tunnels,
   protection of headers, support for operations and maintenance, 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.

   o  Tunnels that carry or encapsulate using ECN code points MUST
      follow the requirements specified in [RFC6040].

   o  [I-D.ietf-rtgwg-dt-encap] describes encapsulation considerations
      in the design of tunnels.

   o  Fragmentation requirements for all types of tunnel and the
      requirements for MTU support are described in
      [I-D.ietf-intarea-tunnels].

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

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   delivered.  This fundamental issue with fragmentation exists for both
   IPv4 and IPv6.

   In addition, some network address translators (NATs) and firewalls
   drop IP fragments.  The network address translation performed by a
   NAT only operates on complete IP packets, and some firewall policies
   also require inspection of complete IP packets.  Even with these
   being the case, some NATs and firewalls simply do not implement the
   necessary reassembly functionality, and instead choose to drop all
   fragments.  Finally, [RFC4963] documents other issues specific to
   IPv4 fragmentation.

   Due to these issues, an application SHOULD NOT send UDP datagrams
   that result in IP packets that exceed the MTU of the path to the
   destination.  Consequently, an application SHOULD either use the path
   MTU information provided by the IP layer or implement path MTU
   discovery itself [RFC1191][RFC1981][RFC4821] to determine whether the
   path to a destination will support its desired message size without
   fragmentation.  However, the ICMP messages that enable path MTU
   discovery are being increasingly filtered by middleboxes (including
   Firewalls) fail to forward ICMP messages.  When the path includes a
   tunnel, some devices acting as a tunnel ingress discard ICMP messages
   that originate from network devices over which the tunnel passes,
   preventing these reaching the UDP endpoint.

   Packetization Layer Path MTU Discovery (PLPMTUD) [RFC4821] does not
   rely upon network support for ICMP messages and is therefore
   considered more robust than standard PMTUD.  To operate, PLPMTUD
   requires changes to the way the transport is used, both to transmit
   probe packets, and to account for the loss or success of these
   probes.  This updates not only the PMTU algorithm, it also impacts
   loss recovery, congestion control, etc.  These updated mechanisms can
   be implemented within a connection-oriented transport (e.g., TCP,
   SCTP, DCCP), but are not a part of UDP.  PLPMTUD therefore places
   additional design requirements on a UDP application that wishes to
   use this method.  This is especially true for UDP tunnels, because
   the overhead of sending probe packets needs to be accounted for and
   may require adding a congestion control mechanism to the tunnel (see
   Section 3.1.9) as well as complicating the data path at a tunnel
   decapsulator.

   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].

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   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 [I-D.ietf-intarea-tunnels], so implementing PMTU
   discovery may be beneficial.

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

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 protocol.
   Applications that do require reliable message delivery MUST implement
   an appropriate mechanism themselves.

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

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   Applications that use multiple transport ports need to be robust to
   reordering between sessions.  Load-balancing techniques within the
   network, such as Equal Cost Multipath (ECMP) forwarding can also
   result in a lack of ordering between different transport sessions,
   even between the same two network endpoints.

   It is important to note that the time by which packets are reordered
   or after which duplicates can still arrive can be very large.  Even
   more importantly, there is no well-defined upper boundary here.
   [RFC0793] defines the maximum delay a TCP segment should experience
   -- the Maximum Segment Lifetime (MSL) -- as 2 minutes.  No other RFC
   defines an MSL for other transport protocols or IP itself.  The MSL
   value defined for TCP is conservative enough that it SHOULD be used
   by other protocols, including UDP.  Therefore, applications SHOULD be
   robust to the reception of delayed or duplicate packets that are
   received within this 2-minute interval.

   Instead of implementing these relatively complex reliability
   mechanisms by itself, an application that requires reliable and
   ordered message delivery SHOULD whenever possible choose an IETF
   standard transport protocol that provides these features.

3.4.  Checksum Guidelines

   The UDP header includes an optional, 16-bit one's complement checksum
   that provides an integrity check.  These checks are not strong from a
   coding or cryptographic perspective, and are not designed to detect
   physical-layer errors or malicious modification of the datagram
   [RFC3819].  Application developers SHOULD implement additional checks
   where data integrity is important, e.g., through a Cyclic Redundancy
   Check (CRC) or keyed or non-keyed cryptographic hash included with
   the data to verify the integrity of an entire object/file sent over
   the UDP service.

   The UDP checksum provides a statistical guarantee that the payload
   was not corrupted in transit.  It also allows the receiver to verify
   that it was the intended destination of the packet, because it covers
   the IP addresses, port numbers, and protocol number, and it verifies
   that the packet is not truncated or padded, because it covers the
   size field.  It therefore protects an application against receiving
   corrupted payload data in place of, or in addition to, the data that
   was sent.  More description of the set of checks performed using the
   checksum field is provided in Section 3.1 of [RFC6396].

   Applications SHOULD enable UDP checksums.  For IPv4, [RFC0768]
   permits an option to disable their use.

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   When UDP is used over IPv6, the UDP checksum is relied upon to
   protect both the IPv6 and UDP headers from corruption, and MUST be
   used as specified in [RFC2460].  [RFC6935] defines a method that
   enables use of a zero UDP zero-checksum mode with a tunnel protocol,
   providing that the method satisfies the requirements in [RFC6936].
   The application MUST implement mechanisms and/or usage restrictions
   when enabling this mode.  This includes defining the scope for usage
   and measures to prevent leakage of traffic to other UDP applications
   (see Appendix A).  These additional design requirements for using a
   zero IPv6 UDP checksum are not present for IPv4, since the IPv4
   header validates information that is not protected in an IPv6 packet.
   Key requirements are:

   o  Use of the UDP checksum with IPv6 MUST be the default
      configuration for all implementations [RFC6935].  The receiving
      endpoint MUST only allow the use of UDP zero-checksum mode for
      IPv6 on a UDP destination port that is specifically enabled.

   o  An application that support a checksum different to that in
      [RFC2460] MUST comply with all implementation requirements
      specified in Section 4 of [RFC6936] and with the usage
      requirements specified in Section 5 of [RFC6936].

   o  A UDP application MUST check that the source and destination IPv6
      addresses are valid for any packets with a UDP zero-checksum and
      MUST discard any packet for which this check fails.  To protect
      from misdelivery, new encapsulation designs SHOULD include an
      integrity check at the transport layer that includes at least the
      IPv6 header, the UDP header and the shim header for the
      encapsulation, if any [RFC6936].

   o  One way to help satisfy the requirements of [RFC6936] may be to
      limit the usage (e.g., to constrain traffic to an operator network
      Section 3.6, as in [RFC7510]).

   Applications that choose to disable UDP checksums MUST NOT make
   assumptions regarding the correctness of received data and MUST
   behave correctly when a UDP datagram is received that was originally
   sent to a different destination or is otherwise corrupted.

   IPv6 datagrams with a zero UDP checksum will not be passed by any
   middlebox that validates the checksum based on [RFC2460] or that
   updates the UDP checksum field, such as NATs or firewalls.  Changing
   this behavior would require such middleboxes to be updated to
   correctly handle datagrams with zero UDP checksums To ensure end-to-
   end robustness, applications that may be deployed in the general
   Internet MUST provide a mechanism to safely fall back to using a
   checksum when a path change occurs that redirects a zero UDP checksum

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   flow over a path that includes a middlebox that discards IPv6
   datagrams with a zero UDP checksum.

3.4.1.  UDP-Lite

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

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

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

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

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   3    for each node in N2, i.e., a 2-hop neighbor which is not a
          neighbor node or the node itself, and such that there exist at
          least one entry in the 2-hop neighbor set where
          N_neighbor_main_addr correspond to a neighbor node with
          willingness different of WILL_NEVER, one selects one 2-hop
          tuple and creates one entry in the routing table with:

               R_dest_addr  =  the main address of the 2-hop neighbor;

               R_next_addr  = the R_next_addr of the entry in the
                              routing table with:

                                  R_dest_addr == N_neighbor_main_addr
                                                 of the 2-hop tuple;

               R_dist       = 2;

               R_iface_addr = the R_iface_addr of the entry in the
                              routing table with:

                                  R_dest_addr == N_neighbor_main_addr
                                                 of the 2-hop tuple;

     3    The new route entries for the destination nodes h+1 hops away
          are recorded in the routing table.  The following procedure
          MUST be executed for each value of h, starting with h=2 and
          incrementing it by 1 each time.  The execution will stop if no
          new entry is recorded in an iteration.

          3.1  For each topology entry in the topology table, if its
               T_dest_addr does not correspond to R_dest_addr of any
               route entry in the routing table AND its T_last_addr
               corresponds to R_dest_addr of a route entry whose R_dist
               is equal to h, then a new route entry MUST be recorded in
               the routing table (if it does not already exist) where:

                    R_dest_addr  = T_dest_addr;

                    R_next_addr  = R_next_addr of the recorded
                                   route entry where:

                                   R_dest_addr == T_last_addr

                    R_dist       = h+1; and

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                    R_iface_addr = R_iface_addr of the recorded
                                   route entry where:

                                      R_dest_addr == T_last_addr.

          3.2  Several topology entries may be used to select a next hop
               R_next_addr for reaching the node R_dest_addr.  When h=1,
               ties should be broken such that nodes with highest
               willingness and MPR selectors are preferred as next hop.

     4    For each entry in the multiple interface association base
          where there exists a routing entry such that:

               R_dest_addr  == I_main_addr  (of the multiple interface
                                            association entry)

          AND there is no routing entry such that:

               R_dest_addr  == I_iface_addr

          then a route entry is created in the routing table with:

               R_dest_addr  =  I_iface_addr (of the multiple interface
                                             association entry)

               R_next_addr  =  R_next_addr  (of the recorded
                                             route entry)

               R_dist       =  R_dist       (of the recorded
                                             route entry)

               R_iface_addr =  R_iface_addr (of the recorded
                                                route entry).

11.  Node Configuration

   This section outlines how a node should be configured, in order to
   operate in an OLSR MANET.

11.1.  Address Assignment

   The nodes in the MANET network SHOULD be assigned addresses within a
   defined address sequence, i.e., the nodes in the MANET SHOULD be
   addressable through a network address and a netmask.

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   Likewise, the nodes in each associated network SHOULD be assigned
   addresses from a defined address sequence, distinct from that being
   used in the MANET.

11.2.  Routing Configuration

   Any MANET node with associated networks or hosts SHOULD be configured
   such that it has routes set up to the interfaces with associated
   hosts or network.

11.3.  Data Packet Forwarding

   OLSR itself does not perform packet forwarding.  Rather, it maintains
   the routing table in the underlying operating system, which is
   assumed to be forwarding packets as specified in RFC1812.

12.  Non OLSR Interfaces

   A node MAY be equipped with multiple interfaces, some of which do not
   participate in the OLSR MANET.  These non OLSR interfaces may be
   point to point connections to other singular hosts or may connect to
   separate networks.

   In order to provide connectivity from the OLSR MANET interface(s) to
   these non OLSR interface(s), a node SHOULD be able to inject external
   route information to the OLSR MANET.

   Injecting routing information from the OLSR MANET to non OLSR
   interfaces is outside the scope of this specification.  It should be
   clear, however, that the routing information for the OLSR MANET can
   be extracted from the topology table (see section 4.4) or directly
   from the routing table of OLSR, and SHOULD be injected onto the non
   OLSR interfaces following whatever mechanism (routing protocol,
   static configuration etc.) is provided on these interfaces.

   An example of such a situation could be where a node is equipped with
   a fixed network (e.g., an Ethernet) connecting to a larger network as
   well as a wireless network interface running OLSR.

   Notice that this is a different case from that of "multiple
   interfaces", where all the interfaces are participating in the MANET
   through running the OLSR protocol.

   In order to provide this capability of injecting external routing
   information into an OLSR MANET, a node with such non-MANET interfaces
   periodically issues a Host and Network Association (HNA) message,
   containing sufficient information for the recipients to construct an
   appropriate routing table.

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12.1.  HNA Message Format

   The proposed format of an HNA-message is:

       0                   1                   2                   3
       0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1
      +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
      |                         Network Address                       |
      +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
      |                             Netmask                           |
      +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
      |                         Network Address                       |
      +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
      |                             Netmask                           |
      +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
      |                              ...                              |
      +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+

   This is sent as the data part of the general packet format with the
   "Message Type" set to HNA_MESSAGE, the TTL field set to 255 and Vtime
   set accordingly to the value of HNA_HOLD_TIME, as specified in
   section 18.3.

     Network Address

          The network address of the associated network

     Netmask

          The netmask, corresponding to the network address immediately
          above.

12.2.  Host and Network Association Information Base

   Each node maintains information concerning which nodes may act as
   "gateways" to associated hosts and networks by recording "association
   tuples" (A_gateway_addr, A_network_addr, A_netmask, A_time), where
   A_gateway_addr is the address of an OLSR interface of the gateway,
   A_network_addr and A_netmask specify the network address and netmask
   of a network, reachable through this gateway, and A_time specifies
   the time at which this tuple expires and hence *MUST* be removed.

   The set of all association tuples in a node is called the
   "association set".

   It should be noticed, that the HNA-message can be considered as a
   "generalized version" of the TC-message: the originator of both the
   HNA- and TC-messages announce "reachability" to some other host(s).

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   In the TC-message, no netmask is required, since all reachability is
   announced on a per-host basis.  In HNA-messages, announcing
   reachability to an address sequence through a network- and netmask
   address is typically preferred over announcing reachability to
   individual host addresses.

   An important difference between TC- and HNA-messages is, that a TC
   message may have a canceling effect on previous information (if the
   ANSN is incremented), whereas information in HNA-messages is removed
   only upon expiration.

12.3.  HNA Message Generation

   A node with associated hosts and/or networks SHOULD periodically
   generate a Host and Network Association (HNA) message, containing
   pairs of (network address, netmask) corresponding to the connected
   hosts and networks.  HNA-messages SHOULD be transmitted periodically
   every HNA_INTERVAL.  The Vtime is set accordingly to the value of
   HNA_HOLD_TIME, as specified in section 18.3.

   A node without any associated hosts and/or networks SHOULD NOT
   generate HNA-messages.

12.4.  HNA Message Forwarding

   Upon receiving a HNA message, and thus following the rules of section
   3, in this version of the specification, the message MUST be
   forwarded according to section 3.4.

12.5.  HNA Message Processing

   In this section, the term "originator address" is used to designate
   the main address on the OLSR MANET of the node which originally
   issued the HNA-message.

   Upon processing a HNA-message, the "validity time" MUST be computed
   from the Vtime field of the message header (see section 3.3.2).  The
   association base SHOULD then be updated as follows:

   1    If the sender interface (NB: not originator) of this message
        is not in the symmetric 1-hop neighborhood of this node, the
        message MUST be discarded.

   2    Otherwise, for each (network address, netmask) pair in the
        message:

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        2.1  if an entry in the association set already exists, where:

                  A_gateway_addr == originator address

                  A_network_addr == network address

                  A_netmask      == netmask

             then the holding time for that tuple MUST be set to:

                  A_time         =  current time + validity time

        2.2  otherwise, a new tuple MUST be recorded with:

                  A_gateway_addr =  originator address

                  A_network_addr =  network address

                  A_netmask      =  netmask

                  A_time         =  current time + validity time

12.6.  Routing Table Calculation

   In addition to the routing table computation as described in section
   10, the host and network association set MUST be added as follows:

   For each tuple in the association set,

     1    If there is no entry in the routing table with:

               R_dest_addr     == A_network_addr/A_netmask

          then a new routing entry is created.

     2    If a new routing entry was created at the previous step, or
          else if there existed one with:

               R_dest_addr     == A_network_addr/A_netmask

               R_dist          >  dist to A_gateway_addr of
                                  current association set tuple,

          then the routing entry is modified as follows:

               R_dest_addr     =  A_network_addr/A_netmask

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               R_next_addr     =  the next hop on the path
                                  from the node to A_gateway_addr

               R_dist          =  dist to A_gateway_addr

               R_next_addr and R_iface_addr MUST be set to the same
               values as the tuple from the routing set with R_dest_addr
               == A_gateway_addr.

12.7.  Interoperability Considerations

   Nodes, which do not implement support for non OLSR interfaces, can
   coexist in a network with nodes which do implement support for non
   OLSR interfaces: the generic packet format and message forwarding
   (section 3) ensures that HNA messages are correctly forwarded by all
   nodes.  Nodes which implement support for non OLSR interfaces may
   thus transmit and process HNA messages according to this section.

   Nodes, which do not implement support for non OLSR interfaces can not
   take advantage of the functionality specified in this section,
   however they will forward HNA messages correctly, as specified in
   section 3.

13.  Link Layer Notification

   OLSR is designed not to impose or expect any specific information
   from the link layer.  However, if information from the link-layer
   describing link breakage is available, a node MAY use this as
   described in this section.

   If link layer information describing connectivity to neighboring
   nodes is available (i.e., loss of connectivity such as through
   absence of a link layer acknowledgment), this information is used in
   addition to the information from the HELLO-messages to maintain the
   neighbor information base and the MPR selector set.

   Thus, upon receiving a link-layer notification that the link between
   a node and a neighbor interface is broken, the following actions are
   taken with respect to link sensing:

   Each link tuple in the local link set SHOULD, in addition to what is
   described in section 4.2, include a L_LOST_LINK_time field.
   L_LOST_LINK_time is a timer for declaring a link as lost when an
   established link becomes pending.  (Notice, that this is a subset of
   what is recommended in section 14, thus link hysteresis and link
   layer notifications can coexist).

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   HELLO message generation should consider those new fields as follows:

     1    if L_LOST_LINK_time is not expired, the link is advertised
          with a link type of LOST_LINK.  In addition, it is not
          considered as a symmetric link in the updates of the
          associated neighbor tuple (see section 8.1).

     2    if the link to a neighboring symmetric or asymmetric interface
          is broken, the corresponding link tuple is modified:
          L_LOST_LINK_time and L_time are set to current time +
          NEIGHB_HOLD_TIME.

     3    this is considered as a link loss and the appropriate
          processing described in section 8.5 should be
          performed.

13.1.  Interoperability Considerations

   Link layer notifications provide, for a node, an additional criterion
   by which a node may determine if a link to a neighbor node is lost.
   Once a link is detected as lost, it is advertised, in accordance with
   the provisions described in the previous sections of this
   specification.

14.  Link Hysteresis

   Established links should be as reliable as possible to avoid data
   packet loss.  This implies that link sensing should be robust against
   bursty loss or transient connectivity between nodes.  Hence, to
   enhance the robustness of the link sensing mechanism, the following
   implementation recommendations SHOULD be considered.

14.1.  Local Link Set

   Each link tuple in the local link set SHOULD, in addition to what is
   described in section 4.2, include a L_link_pending field, a
   L_link_quality field, and a L_LOST_LINK_time field.  L_link_pending
   is a boolean value specifying if the link is considered pending
   (i.e., the link is not considered established).  L_link_quality is a
   dimensionless number between 0 and 1 describing the quality of the
   link.  L_LOST_LINK_time is a timer for declaring a link as lost when
   an established link becomes pending.

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14.2.  Hello Message Generation

   HELLO message generation should consider those new fields as follows:

     1    if L_LOST_LINK_time is not expired, the link is advertised
          with a link type of LOST_LINK.

     2    otherwise, if L_LOST_LINK_time is expired and L_link_pending
          is set to "true", the link SHOULD NOT be advertised at all;

     3    otherwise, if L_LOST_LINK_time is expired and L_link_pending
          is set to "false", the link is advertised as described
          previously in section 6.

   A node considers that it has a symmetric link for each link tuple
   where:

     1    L_LOST_LINK_time is expired, AND

     2    L_link_pending is "false", AND

     3    L_SYM_time is not expired.

   This definition for "symmetric link" SHOULD be used in updating the
   associated neighbor tuple (see section 8.1) for computing the
   N_status of a neighbor node.  This definition SHOULD thereby also be
   used as basis for the symmetric neighborhood when computing the MPR
   set, as well as for "the symmetric neighbors" in the first steps of
   the routing table calculation.

   Apart from the above, what has been described previously does not
   interfere with the advanced link sensing fields in the link tuples.
   The L_link_quality, L_link_pending and L_LOST_LINK_time fields are
   exclusively updated according to the present section.  This section
   does not modify the function of any other fields in the link tuples.

14.3.  Hysteresis Strategy

   The link between a node and some of its neighbor interfaces might be
   "bad", i.e., from time to time let HELLOs pass through only to fade
   out immediately after.  In this case, the neighbor information base
   would contain a bad link for at least "validity time".  The following
   hysteresis strategy SHOULD be adopted to counter this situation.

   For each neighbor interface NI heard by interface I, the
   L_link_quality field of the corresponding Link Tuple determines the
   establishment of the link.  The value of L_link_quality is compared
   to two thresholds HYST_THRESHOLD_HIGH, HYST_THRESHOLD_LOW, fixed

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   between 0 and 1 and such that HYST_THRESHOLD_HIGH >=
   HYST_THRESHOLD_LOW.

   The L_link_pending field is set according to the following:

     1    if L_link_quality > HYST_THRESHOLD_HIGH:

               L_link_pending   = false

               L_LOST_LINK_time = current time - 1 (expired)

     2    otherwise, if L_link_quality < HYST_THRESHOLD_LOW:

               L_link_pending   = true

               L_LOST_LINK_time = min (L_time, current time +
               NEIGHB_HOLD_TIME)

               (the link is then considered as lost according to section
               8.5 and this may produce a neighbor loss).

     3    otherwise, if HYST_THRESHOLD_LOW <= L_link_quality
                                           <= HYST_THRESHOLD_HIGH:

               L_link_pending and L_LOST_LINK_time remain unchanged.

   The condition for considering a link established is thus stricter
   than the condition for dropping a link.  Notice thus, that a link can
   be dropped based on either timer expiration (as described in section
   7) or on L_link_quality dropping below HYST_THRESHOLD_LOW.

   Also notice, that even if a link is not considered as established by
   the link hysteresis, the link tuples are still updated for each
   received HELLO message (as described in section 7).  Specifically,
   this implies that, regardless of whether or not the link hysteresis
   considers a link as "established", tuples in the link set do not
   expire except as determined by the L_time field of the link tuples.

   As a basic implementation requirement, an estimation of the link
   quality must be maintained and stored in the L_link_quality field.
   If some measure of the signal/noise level on a received message is
   available (e.g., as a link layer notification), then it can be used
   as estimation after normalization.

   If no signal/noise information or other link quality information is
   available from the link layer, an algorithm such as the following can
   be utilized (it is an exponentially smoothed moving average of the
   transmission success rate).  The algorithm is parameterized by a

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   scaling parameter HYST_SCALING which is a number fixed between 0 and
   1.  For each neighbor interface NI heard by interface I, the first
   time NI is heard by I, L_link_quality is set to HYST_SCALING
   (L_link_pending is set to true and L_LOST_LINK_time to current time -
   1).

   A tuple is updated according to two rules.  Every time an OLSR packet
   emitted by NI is received by I, the stability rule is applied:

          L_link_quality = (1-HYST_SCALING)*L_link_quality
                           + HYST_SCALING.

     When an OLSR packet emitted by NI is lost by I, the instability
     rule is applied:

          L_link_quality = (1-HYST_SCALING)*L_link_quality.

   The loss of OLSR packet is detected by tracking the missing Packet
   Sequence Numbers on a per interface basis and by "long period of
   silence" from a node.  A "long period of silence may be detected
   thus: if no OLSR packet has been received on interface I from
   interface NI during HELLO emission interval of interface NI (computed
   from the Htime field in the last HELLO message received from NI), a
   loss of an OLSR packet is detected.

14.4.  Interoperability Considerations

   Link hysteresis determines, for a node, the criteria at which a link
   to a neighbor node is accepted or rejected.  Nodes in a network may
   have different criteria, according to the nature of the media over
   which they are communicating.  Once a link is accepted, it is
   advertised, in accordance with the provisions described in the
   previous sections of this specification.

15.  Redundant Topology Information

   In order to provide redundancy to topology information base, the
   advertised link set of a node MAY contain links to neighbor nodes
   which are not in MPR selector set of the node.  The advertised link
   set MAY contain links to the whole neighbor set of the node.  The
   minimal set of links that any node MUST advertise in its TC messages
   is the links to its MPR selectors.  The advertised link set can be
   built according to the following rule based on a local parameter
   called TC_REDUNDANCY parameter.

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15.1.  TC_REDUNDANCY Parameter

   The parameter TC_REDUNDANCY specifies, for the local node, the amount
   of information that MAY be included in the TC messages.  The
   parameter SHOULD be interpreted as follows:

     -    if the TC_REDUNDANCY parameter of the node is 0, then the
          advertised link set of the node is limited  to the MPR
          selector set (as described in section 8.3),

     -    if the TC_REDUNDANCY parameter of the node is 1, then the
          advertised link set of the node is the union of its MPR set
          and its MPR selector set,

     -    if the TC_REDUNDANCY parameter of the node is 2, then the
          advertised link set of the node is the full neighbor link set.

   A node with willingness equal to WILL_NEVER SHOULD have TC_REDUNDANCY
   also equal to zero.

15.2.  Interoperability Considerations

   A TC message is sent by a node in the network to declare a set of
   links, called advertised link set, which MUST include at least the
   links to all nodes of its MPR Selector set, i.e., the neighbors which
   have selected the sender node as a MPR.  This is sufficient
   information to ensure that routes can be computed in accordance with
   section 10.

   The provisions in this section specifies how additional information
   may be declared, as specified through a TC_REDUNDANCY parameter.
   TC_REDUNDANCY = 0 implies that the information declared corresponds
   exactly to the MPR Selector set, identical to section 9.  Other
   values of TC_REDUNDANCY specifies additional information to be
   declared, i.e., the contents of the MPR Selector set is always
   declared.  Thus, nodes with different values of TC_REDUNDANCY may
   coexist in a network: control messages are carried by all nodes in
   accordance with section 3, and all nodes will receive at least the
   link-state information required to construct routes as described in
   section 10.

16.  MPR Redundancy

   MPR redundancy specifies the ability for a node to select redundant
   MPRs.  Section 4.5 specifies that a node should select its MPR set to
   be as small as possible, in order to reduce protocol overhead.  The
   criteria for selecting MPRs is, that all strict 2-hop nodes must be
   reachable through, at least, one MPR node.  Redundancy of the MPR set

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   affects the overhead through affecting the amount of links being
   advertised, the amount of nodes advertising links and the efficiency
   of the MPR flooding mechanism.  On the other hand, redundancy in the
   MPR set ensures that reachability for a node is advertised by more
   nodes, thus additional links are diffused to the network.

   While, in general, a minimal MPR set provides the least overhead,
   there are situations in which overhead can be traded off for other
   benefits.  For example, a node may decide to increase its MPR
   coverage if it observes many changes in its neighbor information base
   caused by mobility, while otherwise keeping a low MPR coverage.

16.1.  MPR_COVERAGE Parameter

   The MPR coverage is defined by a single local parameter,
   MPR_COVERAGE, specifying by how many MPR nodes any strict 2-hop node
   should be covered.  MPR_COVERAGE=1 specifies that the overhead of the
   protocol is kept at a minimum and causes the MPR selection to operate
   as described in section 8.3.1.  MPR_COVERAGE=m ensures that, if
   possible, a node selects its MPR set such that all strict 2-hop nodes
   for an interface are reachable through at least m MPR nodes on that
   interface.  MPR_COVERAGE can assume any integer value > 0.  The
   heuristic MUST be applied per interface, I.  The MPR set for a node
   is the union of the MPR sets found for each interface.

   Notice that MPR_COVERAGE can be tuned locally without affecting the
   consistency of the protocol.  For example, nodes in a network may
   operate with different values of MPR_COVERAGE.

16.2.  MPR Computation

   Using MPR coverage, the MPR selection heuristics is extended from
   that described in the section 8.3.1 by one definition:

     Poorly covered node:

          A poorly covered node is a node in N2 which is covered by less
          than MPR_COVERAGE nodes in N.

   The proposed heuristic for selecting MPRs is then as follows:

     1    Start with an MPR set made of all members of N with
          willingness equal to WILL_ALWAYS

     2    Calculate D(y), where y is a member of N, for all nodes in N.

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     3    Select as MPRs those nodes in N which cover the poorly covered
          nodes in N2.  The nodes are then removed from N2 for the rest
          of the computation.

     4    While there exist nodes in N2 which are not covered by at
          least MPR_COVERAGE nodes in the MPR set:

          4.1  For each node in N, calculate the reachability, i.e.,
               the number of nodes in N2 which are not yet covered
               by at least MPR_COVERAGE nodes in the MPR set, and
               which are reachable through this 1-hop neighbor;

          4.2  Select as a MPR the node with highest willingness among
               the nodes in N with non-zero reachability.  In case of
               multiple choice select the node which provides
               reachability to the maximum number of nodes in N2.  In
               case of multiple nodes providing the same amount of
               reachability, select the node as MPR whose D(y) is
               greater.  Remove the nodes from N2 which are now covered
               by MPR_COVERAGE nodes in the MPR set.

     5    A node's MPR set is generated from the union of the MPR sets
          for each interface.  As an optimization, process each node, y,
          in the MPR set in increasing order of N_willingness.  If all
          nodes in N2 are still covered by at least MPR_COVERAGE nodes
          in the MPR set excluding node y, and if N_willingness of node
          y is smaller than WILL_ALWAYS, then node y MAY be removed from
          the MPR set.

   When the MPR set has been computed, all the corresponding main
   addresses are stored in the MPR Set.

16.3.  Interoperability Considerations

   The MPR set of a node MUST, according to section 8.3, be calculated
   by a node in such a way that it, through the neighbors in the MPR-
   set, can reach all symmetric strict 2-hop neighbors.  This is
   achieved by the heuristics in this section, for all values of
   MPR_COVERAGE > 0.  MPR_COVERAGE is a local parameter for each node.
   Setting this parameter affects only the amount of redundancy in part
   of the network.

   Notice that for MPR_COVERAGE=1, the heuristics in this section is
   identical to the heuristics specified in the section 8.3.1.

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   Nodes with different values of MPR_COVERAGE may coexist in a network:
   control messages are carried by all nodes in accordance with section
   3, and all nodes will receive at least the link-state information
   required to construct routes as described in sections 9 and 10.

17.  IPv6 Considerations

   All the operations and parameters described in this document used by
   OLSR for IP version 4 are the same as those used by OLSR for IP
   version 6.  To operate with IP version 6, the only required change is
   to replace the IPv4 addresses with IPv6 address.  The minimum packet
   and message sizes (under which there is rejection) should be adjusted
   accordingly, considering the greater size of IPv6 addresses.

18.  Proposed Values for Constants

   This section list the values for the constants used in the
   description of the protocol.

18.1.  Setting emission intervals and holding times

   The proposed constant for C is the following:

          C = 1/16 seconds (equal to 0.0625 seconds)

   C is a scaling factor for the "validity time" calculation ("Vtime"
   and "Htime" fields in message headers, see section 18.3).  The
   "validity time" advertisement is designed such that nodes in a
   network may have different and individually tuneable emission
   intervals, while still interoperate fully.  For protocol functioning
   and interoperability to work:

     -    the advertised holding time MUST always be greater than the
          refresh interval of the advertised information.  Moreover, it
          is recommended that the relation between the interval (from
          section 18.2), and the hold time is kept as specified
          in section 18.3, to allow for reasonable packet loss.

     -    the constant C SHOULD be set to the suggested value.  In order
          to achieve interoperability, C MUST be the same on all nodes.

     -    the emission intervals (section 18.2), along with the
          advertised holding times (subject to the above constraints)
          MAY be selected on a per node basis.

   Note that the timer resolution of a given implementation might not be
   sufficient to wake up the system on precise refresh times or on
   precise expire times: the implementation SHOULD round up the

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

   UDP-Lite is supported in some endpoint protocol stacks.  Current
   support for middlebox traversal using UDP-Lite is poor, because UDP-
   Lite uses a different IPv4 protocol number or IPv6 "next header"
   value than that used for UDP; therefore, few middleboxes are
   currently able to interpret UDP-Lite and take appropriate actions
   when forwarding the packet.  This makes UDP-Lite less suited for
   applications needing general Internet support, until such time as
   UDP-Lite has achieved better support in middleboxes.

3.5.  Middlebox Traversal Guidelines

   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 can create per-flow state when they see a
   packet that -- according to some local criteria -- indicates a new
   flow, and destroy the state after some time during which no packets
   belonging to the same flow have arrived.

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

   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-

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   response communication pattern that typically completes within
   seconds.

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

   For some applications, such as media transmissions, this re-
   synchronization is highly undesirable, because it can cause user-
   perceivable playback artifacts.  Such specialized applications MAY
   send periodic keep-alive messages to attempt to refresh middlebox
   state (e.g., [RFC7675]).  It is important to note that keep-alive
   messages are not recommended for general use -- they are unnecessary
   for many applications and can consume significant amounts of system
   and network resources.

   An application that needs to employ keep-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 the Port
   Control Protocol (PCP) [RFC6887], Middlebox Communications (MIDCOM)
   [RFC3303], Next Steps in Signaling (NSIS) [RFC5973], or Universal
   Plug and Play (UPnP) [UPnP].  It is RECOMMENDED that applications
   apply slight random variations ("jitter") to the timing of keep-alive
   transmissions, to reduce the potential for persistent synchronization
   between keep-alive transmissions from different hosts [RFC7675].

   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.

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   Keep-alive messages are NOT RECOMMENDED for general use.  They are
   unnecessary for many applications and may consume significant
   resources.  For example, on battery-powered devices, if an
   application needs to maintain connectivity for long periods with
   little traffic, the frequency at which keep-alives are sent can
   become the determining factor that governs power consumption,
   depending on the underlying network technology.

   Because many middleboxes are designed to require keep-alives for TCP
   connections at a frequency that is much lower than that needed for
   UDP, this difference alone can often be sufficient to prefer TCP over
   UDP for these deployments.  On the other hand, there is anecdotal
   evidence that suggests that direct communication through middleboxes,
   e.g., by using ICE [RFC5245], does succeed less often with TCP than
   with UDP.  The trade-offs between different transport protocols --
   especially when it comes to middlebox traversal -- deserve careful
   analysis.

   UDP applications that could be deployed in the Internet need to be
   designed understanding that there are many variants of middlebox
   behavior, and although UDP is connectionless, middleboxes often
   maintain state for each UDP flow.  Using multiple UDP flows can
   consume available state space and also can lead to changes in the way
   the middlebox handles subsequent packets (either to protect its
   internal resources, or to prevent perceived misuse).  The probability
   of path failure can increase when applications use multiple UDP flows
   in parallel (see Section 5.1.1 for recommendations on usage of
   multiple ports).

3.6.  Limited Applicability and Controlled Environments

   Two different types of applicability have been identified for the
   specification of IETF applications that utilize UDP:

   General Internet.  By default, IETF specifications target deployment
      on the general Internet.  Experience has shown that successful
      protocols developed in one specific context or for a particular
      application tends to become used in a wider range of contexts.
      For example, a protocol with an initial deployment within a local
      area network may subsequently be used over a virtual network that
      traverses the Internet, or in the Internet in general.
      Applications designed for general Internet use may experience a
      range of network device behaviors, and in particular should
      consider whether applications need to operate over paths that may
      include middleboxes.

   Controlled Environment  A protocol/encapsulation/tunnel could be
      designed to be used only within a controlled environment.  For

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      example, an application designed for use by a network operator
      might only be deployed within the network of that single network
      operator or on networks of an adjacent set of cooperating network
      operators.  The application traffic may then be managed to avoid
      congestion, rather than relying on built-in mechanisms, which are
      required when operating over the general.  Applications that
      target a limited applicability use case may be able to take
      advantage of specific hardware (e.g., carrier-grade equipment) or
      underlying protocol features of the subnetwork over which they are
      used.

   Specifications addressing a limited applicability use case SHOULD
   identify how in their restricted deployment a level of safety is
   provided that is equivalent to that of a protocol designed for
   operation over the general Internet (e.g., a design based on
   extensive experience with deployments of particular methods that
   provide features that cannot be expected in general Internet
   equipment and the robustness of the design of MPLS to corruption of
   headers both helped justify use of an alternate UDP integrity check
   [RFC7510].)  Mechanisms also need to specify how to prevent this type
   of application traffic from escaping to the public Internet.

4.  Multicast UDP Usage Guidelines

   This section complements Section 3 by providing additional guidelines
   that are applicable to multicast and broadcast usage of UDP.

   Multicast and broadcast transmission [RFC1112] usually employ the UDP
   transport protocol, although they may be used with other transport
   protocols (e.g., UDP-Lite).

   There are currently two models of multicast delivery: the Any-Source
   Multicast (ASM) model as defined in [RFC1112] and the Source-Specific
   Multicast (SSM) model as defined in [RFC4607].  ASM group members
   will receive all data sent to the group by any source, while SSM
   constrains the distribution tree to only one single source.

   Specialized classes of applications also use UDP for IP multicast or
   broadcast [RFC0919].  The design of such specialized applications
   requires expertise that goes beyond simple, unicast-specific
   guidelines, since these senders may transmit to potentially very many
   receivers across potentially very heterogeneous paths at the same
   time, which significantly complicates congestion control, flow
   control, and reliability mechanisms.

   This section provides guidance on multicast and broadcast UDP usage.
   Use of broadcast by an application is normally constrained by routers
   to the local subnetwork.  However, use of tunneling techniques and

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   proxies can and does result in some broadcast traffic traversing
   Internet paths.  These guidelines therefore also apply to broadcast
   traffic.

   The IETF has defined a reliable multicast framework [RFC3048] and
   several building blocks to aid the designers of multicast
   applications, such as [RFC3738] or [RFC4654].

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

   Most UDP tunnels that carry IP multicast traffic use a tunnel
   encapsulation with a unicast destination address.  These MUST follow
   the same requirements as a tunnel carrying unicast data (see
   Section 3.1.9).  There are deployment cases and solutions where the
   outer header of a UDP tunnel contains a multicast destination
   address, such as [RFC6513].  These cases are primarily deployed in
   controlled environments over reserved capacity, often operating
   within a single administrative domain, or between two domains over a
   bi-laterally agreed upon path with reserved capacity, and so
   congestion control is OPTIONAL, but circuit breaker techniques are
   still RECOMMENDED in order to restore some degree of service should
   the offered load exceed the reserved capacity (e.g., due to
   misconfiguration).

4.1.  Multicast Congestion Control Guidelines

   Unicast congestion-controlled transport mechanisms 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 implementing the congestion
   control.

   Applications using multicast SHOULD provide appropriate congestion
   control.  Multicast congestion control needs to be designed using
   mechanisms that are robust to the potential heterogeneity of both the
   multicast distribution tree and the receivers belonging to a group.
   Heterogeneity may manifest itself in some receivers experiencing more
   loss that others, higher delay, and/or less ability to respond to

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   network conditions.  Congestion control is particularly important for
   any multicast session were all or part of the multicast distribution
   tree spans an access network (e.g., a home gateway).  Two styles of
   congestion control have been defined in the RFC-series:

   o  Feedback-based congestion control, in which the sender receives
      multicast or unicast UDP messages from the receivers allowing it
      to assess the level of congestion and then adjust the sender
      rate(s) (e.g., [RFC5740],[RFC4654]).  Multicast methods may
      operate on longer timescales than for unicast (e.g., due to the
      higher group RTT of a heterogeneous group).  A control method
      could decide not to reduce the rate of the entire multicast group
      in response to a control message received from a single receiver
      (e.g., a sender could set a minimum rate and decide to request a
      congested receiver to leave the multicast group and could also
      decide to distribute content to these congested receivers at a
      lower rate using unicast congestion control).

   o  Receiver-driven congestion control, which does not require a
      receiver to send explicit UDP control messages for congestion
      control (e.g., [RFC3738], [RFC5775]).  Instead, the sender
      distributes the data across multiple IP multicast groups (e.g.,
      using a set of {S,G} channels).  Each receiver determines its own
      level of congestion and controls its reception rate using only
      multicast join/leave messages sent in the network control plane.
      This method scales to arbitrary large groups of receivers.

   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.

   Some classes of multicast applications support applications that can
   monitor the user-level quality of the transfer at the receiver.
   Applications that can detect a significant reduction in user quality
   SHOULD regard this as a congestion signal (e.g., to leave a group
   using layered multicast encoding) or, if not, SHOULD use this signal
   to provide a circuit breaker to terminate the flow by leaving the
   multicast group.

4.1.1.  Bulk Transfer Multicast Applications

   Applications that perform bulk transmission of data over a multicast
   distribution tree, i.e., applications that exchange more than a few
   UDP datagrams per RTT, SHOULD implement a method for congestion
   control.  The currently RECOMMENDED IETF methods are: Asynchronous
   Layered Coding (ALC) [RFC5775], TCP-Friendly Multicast Congestion

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   Control (TFMCC) [RFC4654], Wave and Equation Based Rate Control
   (WEBRC) [RFC3738], NACK-Oriented Reliable Multicast (NORM) transport
   protocol [RFC5740], File Delivery over Unidirectional Transport
   (FLUTE) [RFC6726], Real Time Protocol/Control Protocol (RTP/RTCP)
   [RFC3550].

   An application can alternatively implement another congestion control
   schemes following the guidelines of [RFC2887] and utilizing the
   framework of [RFC3048].  Bulk transfer applications that choose not
   to implement [RFC4654], [RFC5775], [RFC3738], [RFC5740], [RFC6726],
   or [RFC3550] SHOULD implement a congestion control scheme that
   results in bandwidth use that competes fairly with TCP within an
   order of magnitude.

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

4.1.2.  Low Data-Volume Multicast Applications

   All the recommendations in Section 3.1.2 are also applicable to low
   data-volume multicast applications.

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

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   common method, the remainder of this section discusses it in more
   detail.

   Although the sockets API was developed for UNIX in the early 1980s, a
   wide variety of non-UNIX operating systems also implement it.  The
   sockets API supports both IPv4 and IPv6 [RFC3493].  The UDP sockets
   API differs from that for TCP in several key ways.  Because
   application programmers are typically more familiar with the TCP
   sockets API, this section discusses these differences.  [STEVENS]
   provides usage examples of the UDP sockets API.

   UDP datagrams may be directly sent and received, without any
   connection setup.  Using the sockets API, applications can receive
   packets from more than one IP source address on a single UDP socket.
   Some servers use this to exchange data with more than one remote host
   through a single UDP socket at the same time.  Many applications need
   to ensure that they receive packets from a particular source address;
   these applications MUST implement corresponding checks at the
   application layer or explicitly request that the operating system
   filter the received packets.

   Many operating systems also allow a UDP socket to be connected, i.e.,
   to bind a UDP socket to a specific pair of addresses and ports.  This
   is similar to the corresponding TCP sockets API functionality.
   However, for UDP, this is only a local operation that serves to
   simplify the local send/receive functions and to filter the traffic
   for the specified addresses and ports.  Binding a UDP socket does not
   establish a connection -- UDP does not notify the remote end when a
   local UDP socket is bound.  Binding a socket also allows configuring
   options that affect the UDP or IP layers, for example, use of the UDP
   checksum or the IP Timestamp option.  On some stacks, a bound socket
   also allows an application to be notified when ICMP error messages
   are received for its transmissions [RFC1122].

   If a client/server application executes on a host with more than one
   IP interface, the application SHOULD send any UDP responses with an
   IP source address that matches the IP destination address of the UDP
   datagram that carried the request (see [RFC1122], Section 4.1.3.5).
   Many middleboxes expect this transmission behavior and drop replies
   that are sent from a different IP address, as explained in
   Section 3.5.

   A UDP receiver can receive a valid UDP datagram with a zero-length
   payload.  Note that this is different from a return value of zero
   from a read() socket call, which for TCP indicates the end of the
   connection.

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   UDP provides no flow-control, i.e., the sender at any given time does
   not know whether the receiver is able to handle incoming
   transmissions.  This is another reason why UDP-based applications
   need to be robust in the presence of packet loss.  This loss can also
   occur within the sending host, when an application sends data faster
   than the line rate of the outbound network interface.  It can also
   occur at the destination, where receive calls fail to return all the
   data that was sent when the application issues them too infrequently
   (i.e., such that the receive buffer overflows).  Robust flow control
   mechanisms are difficult to implement, which is why applications that
   need this functionality SHOULD consider using a full-featured
   transport protocol such as TCP.

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

5.1.  Using UDP Ports

   The rules procedures for the management of the Service Name and
   Transport Protocol Port Number Registry are specified in [RFC6335].
   Recommendations for use of UDP ports are provided in [RFC7605].

   A UDP sender SHOULD NOT use a source port value of zero.  A source
   port number that cannot be easily determined from the address or
   payload type provides protection at the receiver from data injection
   attacks by off-path devices.  A UDP receiver SHOULD NOT bind to port
   zero.

   Applications SHOULD implement receiver port and address checks at the
   application layer or explicitly request that the operating system
   filter the received packets to prevent receiving packets with an
   arbitrary port.  This measure is designed to provide additional
   protection from data injection attacks from an off-path source (where
   the port values may not be known).

   Applications SHOULD provide a check that protects from off-path data
   injection, avoiding an application receiving packets that were
   created by an unauthorized third party.  TCP stacks commonly use a
   randomized source port to provide this protection [RFC6056]; UDP

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   applications should follow the same technique.  Middleboxes and end
   systems often make assumptions about the system ports or user ports,
   hence it is recommended to use randomized ports in the Dynamic and/or
   Private Port range.  Setting a "randomized" source port also provides
   greater assurance that reported ICMP errors originate from network
   systems on the path used by a particular flow.  Some UDP applications
   choose to use a predetermined value for the source port (including
   some multicast applications), these applications need to therefore
   employ a different technique.  Protection from off-path data attacks
   can also be provided by randomizing the initial value of another
   protocol field within the datagram payload, and checking the validity
   of this field at the receiver (e.g., RTP has random initial sequence
   number and random media timestamp offsets [RFC3550]).

   When using multicast, IP routers perform a reverse-path forwarding
   (RPF) check for each multicast packet.  This provides protection from
   off-path data injection.  When a receiver joins a multicast group and
   filters based on the source address the filter verifies the sender's
   IP address.  This is always the case when using a SSM {S,G} channel.

   The UDP source port number field has been used as a basis to design
   load-balancing solutions for IPv4.  This approach has also been
   leveraged for IPv6 [RFC6438], but for IPv6 the "flow label" [RFC6437]
   may also be used as entropy for load balancing.  This use of the flow
   label for load balancing is consistent with the definition of the
   field, although further clarity was needed to ensure the field can be
   consistently used for this purpose.  Therefore, an updated IPv6 flow
   label [RFC6437] and ECMP routing [RFC6438] usage were specified.
   Router vendors are encouraged to start using the flow label as a part
   of the flow hash, providing support for IP-level ECMP without
   requiring use of UDP.  The end-to-end use of flow labels for load
   balancing is a long-term solution.  Even if the usage of the flow
   label has been clarified, there will be a transition time before a
   significant proportion of endpoints start to assign a good quality
   flow label to the flows that they originate.  The use of load
   balancing using the transport header fields will likely continue
   until widespread deployment is finally achieved.

5.1.1.  Applications using Multiple UDP Ports

   A single application may exchange several types of data.  In some
   cases, this may require multiple UDP flows (e.g., multiple sets of
   flows, identified by different five-tuples).  [RFC6335] recommends
   application developers not to apply to IANA to be assigned multiple
   well-known ports (user or system).  This does not discuss the
   implications of using multiple flows with the same well-known port or
   pairs of dynamic ports (e.g., identified by a service name or
   signaling protocol).

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   Use of multiple flows can affect the network in several ways:

   o  Starting a series of successive connections can increase the
      number of state bindings in middleboxes (e.g., NAPT or Firewall)
      along the network path.  UDP-based middlebox traversal usually
      relies on timeouts to remove old state, since middleboxes are
      unaware when a particular flow ceases to be used by an
      application.

   o  Using several flows at the same time may result in seeing
      different network characteristics for each flow.  It cannot be
      assumed both follow the same path (e.g., when ECMP is used,
      traffic is intentionally hashed onto different parallel paths
      based on the port numbers).

   o  Using several flows can also increase the occupancy of a binding
      or lookup table in a middlebox (e.g., NAPT or Firewall), which may
      cause the device to change the way it manages the flow state.

   o  Further, using excessive numbers of flows can degrade the ability
      of a unicast congestion control to react to congestion events,
      unless the congestion state is shared between all flows in a
      session.  A receiver-driven multicast congestion control requires
      the sending application to distribute its data over a set of IP
      multicast groups, each receiver is therefore expected to receive
      data from a modest number of simultaneously active UDP ports.

   Therefore, applications MUST NOT assume consistent behavior of
   middleboxes when multiple UDP flows are used; many devices respond
   differently as the number of used ports increases.  Using multiple
   flows with different QoS requirements requires applications to verify
   that the expected performance is achieved using each individual flow
   (five-tuple), see Section 3.1.7.

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

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   validation internally before passing ICMP information to the
   application.

   Any application response to ICMP error messages SHOULD be robust to
   temporary routing failures (sometimes called "soft errors"), e.g.,
   transient ICMP "unreachable" messages ought to not normally cause a
   communication abort.

   As mentioned above, ICMP messages are being increasingly filtered by
   middleboxes.  A UDP application therefore SHOULD NOT rely on their
   delivery for correct and safe operation.

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.

   UDP applications SHOULD provide protection from off-path data
   injection attacks using a randomized source port or equivalent
   technique (see Section 5.1).

   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 for securing UDP communications is with IPsec [RFC4301],
   which can provide authentication for flows of IP packets through the
   Authentication Header (AH) [RFC4302] and encryption and/or
   authentication through the Encapsulating Security Payload (ESP)
   [RFC4303].  Applications use the Internet Key Exchange (IKE)
   [RFC7296] to configure IPsec for their sessions.  Depending on how
   IPsec is configured for a flow, it can authenticate or encrypt the
   UDP headers as well as UDP payloads.  If an application only requires
   authentication, ESP with no encryption but with authentication is
   often a better option than AH, because ESP can operate across
   middleboxes.  An application that uses IPsec requires the support of
   an operating system that implements the IPsec protocol suite.

   Although it is possible to use IPsec to secure UDP communications,
   not all operating systems support IPsec or allow applications to
   easily configure it for their flows.  A second option for securing
   UDP communications is through Datagram Transport Layer Security
   (DTLS) [RFC6347].  DTLS provides communication privacy by encrypting

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   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.  There exist a number of security options for RTP
   [RFC3550] over UDP, especially to accomplish key-management, see
   [RFC7201].  These options covers many usages, including point-to-
   point, centralized group communication as well as multicast.  In some
   applications, a better solution is to protect larger stand-alone
   objects, such as files or messages, instead of individual UDP
   payloads.  In these situations, CMS [RFC5652], S/MIME [RFC5751] or
   OpenPGP [RFC4880] could be used.  In addition, there are many non-
   IETF protocols in this area.

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

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

   In terms of congestion control, [RFC2309] and [RFC2914] discuss the
   dangers of congestion-unresponsive flows to the Internet.
   [I-D.ietf-tsvwg-circuit-breaker] describes methods that can be used
   to set a performance envelope that can assist in preventing
   congestion collapse in the absence of congestion control or when the
   congestion control fails to react to congestion events.  This
   document provides guidelines to designers of UDP-based applications
   to congestion-control their transmissions, and does not raise any
   additional security concerns.

   Some network operators have experienced surges of UDP attack traffic
   that are multiple orders of magnitude above the baseline traffic rate
   for UDP.  This can motivate operators to limit the data rate or
   packet rate of UDP traffic.  This may in turn limit the throughput
   that an application can achieve using UDP and could also result in
   higher packet loss for UDP traffic that would not be experienced if
   other transport protocols had been used.

   A UDP application with a long-lived association between the sender
   and receiver, ought to be designed so that the sender periodically
   checks that the receiver still wants ("consents") to receive traffic
   and need to be designed to stop if there is no explicit confirmation

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   of this [RFC7675].  Applications that require communications in two
   directions to implement protocol functions (such as reliability or
   congestion control) will need to independently check both directions
   of communication, and may have to exchange keep-alive packets to
   traverse middleboxes (see Section 3.5).

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    |         |
   | SHOULD back-off retransmission timers following loss    |         |
   |                                                         |         |
   | SHOULD provide mechanisms to regulate the bursts of     | 3.1.4   |
   | transmission                                            |         |
   |                                                         |         |
   | MAY implement ECN; a specific set of application        | 3.1.5   |
   | mechanisms are REQUIRED if ECN is used.                 |         |
   |                                                         |         |
   | for DiffServ, SHOULD NOT rely on implementation of PHBs | 3.1.6   |
   |                                                         |         |
   | for QoS-enabled paths, MAY choose not to use CC         | 3.1.7   |
   |                                                         |         |
   | SHOULD NOT rely solely on QoS for their capacity        | 3.1.8   |
   | non-CC controlled flows SHOULD implement a transport    |         |
   | circuit breaker                                         |         |
   | MAY implement a circuit breaker for other applications  |         |
   |                                                         |         |
   | for tunnels carrying IP Traffic,                        | 3.1.9   |
   | SHOULD NOT perform congestion control                   |         |
   | MUST correctly process the IP ECN field                 |         |
   |                                                         |         |

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   | for non-IP tunnels or rate not determined by traffic,   |         |
   | SHOULD perform CC or use circuit breaker                | 3.1.9   |
   | SHOULD restrict types of traffic transported by the     |         |
   | tunnel                                                  |         |
   |                                                         |         |
   | SHOULD NOT send datagrams that exceed the PMTU, i.e.,   | 3.2     |
   | SHOULD discover PMTU or send datagrams < minimum PMTU;  |         |
   | Specific application mechanisms are REQUIRED if PLPMTUD |         |
   | is used.                                                |         |
   |                                                         |         |
   | SHOULD handle datagram loss, duplication, reordering    | 3.3     |
   | SHOULD be robust to delivery delays up to 2 minutes     |         |
   |                                                         |         |
   | SHOULD enable IPv4 UDP checksum                         | 3.4     |
   | SHOULD enable IPv6 UDP checksum; Specific application   |         |
   | mechanisms are REQUIRED if a zero IPv6 UDP checksum is  |         |
   | used.                                                   |         |
   |                                                         |         |
   | SHOULD provide protection from off-path attacks         | 5.1     |
   | else, MAY use UDP-Lite with suitable checksum coverage  | 3.4.1   |
   |                                                         |         |
   | SHOULD NOT always send middlebox keep-alives            | 3.5     |
   | MAY use keep-alives when needed (min. interval 15 sec)  |         |
   |                                                         |         |
   | Bulk multicast apps SHOULD implement congestion control | 4.1.1   |
   |                                                         |         |
   | Low volume multicast apps SHOULD implement congestion   | 4.1.2   |
   | control                                                 |         |
   |                                                         |         |
   | Multicast apps SHOULD use a safe PMTU                   | 4.2     |
   |                                                         |         |
   | SHOULD avoid using multiple ports                       | 5.1     |
   | MUST check received IP source address                   |         |
   |                                                         |         |
   | SHOULD use a randomized source port or equivalent       | 5.2     |
   | technique, and, for client/server applications, SHOULD  |         |
   | send responses from source address matching request     |         |
   |                                                         |         |
   | SHOULD validate payload in ICMP messages                | 5.2     |
   |                                                         |         |
   | SHOULD use standard IETF security protocols when needed | 6       |
   +---------------------------------------------------------+---------+

                    Table 1: Summary of recommendations

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8.  IANA Considerations

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

   This document raises no IANA considerations.

9.  Acknowledgments

   The middlebox traversal guidelines in Section 3.5 incorporate ideas
   from Section 5 of [I-D.ford-behave-app] by Bryan Ford, Pyda
   Srisuresh, and Dan Kegel.  G.  Fairhurst received funding from the
   European Union's Horizon 2020 research and innovation program
   2014-2018 under grant agreement No. 644334 (NEAT).  Lars Eggert has
   received funding from the European Union's Horizon 2020 research and
   innovation program 2014-2018 under grant agreement No. 644866
   ("SSICLOPS").  This document reflects only the authors' views and the
   European Commission is not responsible for any use that may be made
   of the information it contains.

10.  References

10.1.  Normative References

   [I-D.ietf-tsvwg-circuit-breaker]
              Fairhurst, G., "Network Transport Circuit Breakers",
              draft-ietf-tsvwg-circuit-breaker-07 (work in progress),
              October 2015.

   [RFC0768]  Postel, J., "User Datagram Protocol", STD 6, RFC 768,
              DOI 10.17487/RFC0768, August 1980,
              <http://www.rfc-editor.org/info/rfc768>.

   [RFC0793]  Postel, J., "Transmission Control Protocol", STD 7,
              RFC 793, DOI 10.17487/RFC0793, September 1981,
              <http://www.rfc-editor.org/info/rfc793>.

   [RFC1122]  Braden, R., Ed., "Requirements for Internet Hosts -
              Communication Layers", STD 3, RFC 1122,
              DOI 10.17487/RFC1122, October 1989,
              <http://www.rfc-editor.org/info/rfc1122>.

   [RFC1191]  Mogul, J. and S. Deering, &Clausen & Jacquet             Experimental                     [Page 63]
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   'validity time' ("Vtime" and "Htime" of packets) to compensate for
   coarser timer resolution, at least in the case where "validity time"
   could be shorter than the sum of emission interval and maximum
   expected timer error.

18.2.  Emission Intervals

          HELLO_INTERVAL        = 2 seconds

          REFRESH_INTERVAL      = 2 seconds

          TC_INTERVAL           = 5 seconds

          MID_INTERVAL          = TC_INTERVAL

          HNA_INTERVAL          = TC_INTERVAL

18.3.  Holding Time

          NEIGHB_HOLD_TIME      = 3 x REFRESH_INTERVAL

          TOP_HOLD_TIME         = 3 x TC_INTERVAL

          DUP_HOLD_TIME         = 30 seconds

          MID_HOLD_TIME         = 3 x MID_INTERVAL

          HNA_HOLD_TIME         = 3 x HNA_INTERVAL

   The Vtime in the message header (see section 3.3.2), and the Htime in
   the HELLO message (see section 6.1) are the fields which hold
   information about the above values in mantissa and exponent format
   (rounded up).  In other words:

     value = C*(1+a/16)*2^b [in seconds]

   where a is the integer represented by the four highest bits of the
   field and b the integer represented by the four lowest bits of the
   field.

   Notice, that for the previous proposed value of C, (1/16 seconds),
   the values, in seconds, expressed by the formula above can be stored,
   without loss of precision, in binary fixed point or floating point
   numbers with at least 8 bits of fractional part.  This corresponds
   with NTP time-stamps and single precision IEEE Standard 754 floating
   point numbers.

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   Given one of the above holding times, a way of computing the
   mantissa/exponent representation of a number T (of seconds) is the
   following:

     -    find the largest integer 'b' such that: T/C >= 2^b

     -    compute the expression 16*(T/(C*(2^b))-1), which may not be a
          integer, and round it up.  This results in the value for 'a'

     -    if 'a' is equal to 16: increment 'b' by one, and set 'a' to 0

     -    now, 'a' and 'b' should be integers between 0 and 15, and the
          field will be a byte holding the value a*16+b

   For instance, for values of 2 seconds, 6 seconds, 15 seconds, and 30
   seconds respectively, a and b would be: (a=0,b=5), (a=8,b=6),
   (a=14,b=7) and (a=14,b=8) respectively.

18.4.  Message Types

          HELLO_MESSAGE         = 1

          TC_MESSAGE            = 2

          MID_MESSAGE           = 3

          HNA_MESSAGE           = 4

18.5.  Link Types

          UNSPEC_LINK           = 0

          ASYM_LINK             = 1

          SYM_LINK              = 2

          LOST_LINK             = 3

18.6.  Neighbor Types

          NOT_NEIGH             = 0

          SYM_NEIGH             = 1

          MPR_NEIGH             = 2

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18.7.  Link Hysteresis

          HYST_THRESHOLD_HIGH   = 0.8

          HYST_THRESHOLD_LOW    = 0.3

          HYST_SCALING          = 0.5

18.8.  Willingness

          WILL_NEVER            = 0

          WILL_LOW              = 1

          WILL_DEFAULT          = 3

          WILL_HIGH             = 6

          WILL_ALWAYS           = 7

   The willingness of a node may be set to any integer value from 0 to
   7, and specifies how willing a node is to be forwarding traffic on
   behalf of other nodes.  Nodes will, by default, have a willingness
   WILL_DEFAULT.  WILL_NEVER indicates a node which does not wish to
   carry traffic for other nodes, for example due to resource
   constraints (like being low on battery).  WILL_ALWAYS indicates that
   a node always should be selected to carry traffic on behalf of other
   nodes, for example due to resource abundance (like permanent power
   supply, high capacity interfaces to other nodes).

   A node may dynamically change its willingness as its conditions
   change.

   One possible application would, for example, be for a node, connected
   to a permanent power supply and with fully charged batteries, to
   advertise a willingness of WILL_ALWAYS.  Upon being disconnected from
   the permanent power supply (e.g., a PDA being taken out of its
   charging cradle), a willingness of WILL_DEFAULT is advertised.  As
   battery capacity is drained, the willingness would be further
   reduced.  First to the intermediate value between WILL_DEFAULT and
   WILL_LOW, then to WILL_LOW and finally to WILL_NEVER, when the
   battery capacity of the node does no longer support carrying foreign
   traffic.

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18.9.  Misc. Constants

          TC_REDUNDANCY         = 0

          MPR COVERAGE          = 1

          MAXJITTER             = HELLO_INTERVAL / 4

19.  Sequence Numbers

   Sequence numbers are used in OLSR with the purpose of discarding
   "old" information, i.e., messages received out of order.  However
   with a limited number of bits for representing sequence numbers,
   wrap-around (that the sequence number is incremented from the maximum
   possible value to zero) will occur.  To prevent this from interfering
   with the operation of the protocol, the following MUST be observed.

   The term MAXVALUE designates in the following the largest possible
   value for a sequence number.

   The sequence number S1 is said to be "greater than" the sequence
   number S2 if:

          S1 > S2 AND S1 - S2 <= MAXVALUE/2 OR

          S2 > S1 AND S2 - S1 > MAXVALUE/2

   Thus when comparing two messages, it is possible - even in the
   presence of wrap-around - to determine which message contains the
   most recent information.

20.  Security Considerations

   Currently, OLSR does not specify any special security measures.  As a
   proactive routing protocol, OLSR makes a target for various attacks.
   The various possible vulnerabilities are discussed in this section.

20.1.  Confidentiality

   Being a proactive protocol, OLSR periodically diffuses topological
   information.  Hence, if used in an unprotected wireless network, the
   network topology is revealed to anyone who listens to OLSR control
   messages.

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   In situations where the confidentiality of the network topology is of
   importance, regular cryptographic techniques such as exchange of OLSR
   control traffic messages encrypted by PGP [9] or encrypted by some
   shared secret key can be applied to ensure that control traffic can
   be read and interpreted by only those authorized to do so.

20.2.  Integrity

   In OLSR, each node is injecting topological information into the
   network through transmitting HELLO messages and, for some nodes, TC
   messages.  If some nodes for some reason, malicious or malfunction,
   inject invalid control traffic, network integrity may be compromised.
   Therefore, message authentication is recommended.

   Different such situations may occur, for instance:

     1    a node generates TC (or HNA) messages, advertising links to
          non-neighbor nodes:

     2    a node generates TC (or HNA) messages, pretending to be
          another node,

     3    a node generates HELLO messages, advertising non-neighbor
          nodes,

     4    a node generates HELLO messages, pretending to be another
          node.

     5    a node forwards altered control messages,

     6    a node does not broadcast control messages,

     7    a node does not select multipoint relays correctly.

     8    a node forwards broadcast control messages unaltered, but does
          not forward unicast data traffic;

     9    a node "replays" previously recorded control traffic from
          another node.

   Authentication of the originator node for control messages (for
   situation 2, 4 and 5) and on the individual links announced in the
   control messages (for situation 1 and 3) may be used as a
   countermeasure.  However to prevent nodes from repeating old (and
   correctly authenticated) information (situation 9) temporal
   information is required, allowing a node to positively identify such
   delayed messages.

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   In general, digital signatures and other required security
   information may be transmitted as a separate OLSR message type,
   thereby allowing that "secured" and "unsecured" nodes can coexist in
   the same network, if desired.

   Specifically, the authenticity of entire OLSR control messages can be
   established through employing IPsec authentication headers, whereas
   authenticity of individual links (situation 1 and 3) require
   additional security information to be distributed.

   An important consideration is, that all control messages in OLSR are
   transmitted either to all nodes in the neighborhood (HELLO messages)
   or broadcast to all nodes in the network (e.g., TC messages).

   For example, a control message in OLSR is always a point-to-
   multipoint transmission.  It is therefore important that the
   authentication mechanism employed permits that any receiving node can
   validate the authenticity of a message.  As an analogy, given a block
   of text, signed by a PGP private key, then anyone with the
   corresponding public key can verify the authenticity of the text.

20.3.  Interaction with External Routing Domains

   OLSR does, through the HNA messages specified in section 12, provide
   a basic mechanism for injecting external routing information to the
   OLSR domain.  Section 12 also specifies that routing information can
   be extracted from the topology table or the routing table of OLSR
   and, potentially, injected into an external domain if the routing
   protocol governing that domain permits.

   Other than as described in the section 20.2, when operating nodes,
   connecting OLSR to an external routing domain, care MUST be taken not
   to allow potentially insecure and un-trustworthy information to be
   injected from the OLSR domain to external routing domains.  Care MUST
   be taken to validate the correctness of information prior to it being
   injected as to avoid polluting routing tables with invalid
   information.

   A recommended way of extending connectivity from an existing routing
   domain to an OLSR routed MANET is to assign an IP prefix (under the
   authority of the nodes/gateways connecting the MANET with the exiting
   routing domain) exclusively to the OLSR MANET area, and to configure
   the gateways statically to advertise routes to that IP sequence to
   nodes in the existing routing domain.

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20.4.  Node Identity

   OLSR does not make any assumption about node addresses, other than
   that each node is assumed to have a unique IP address.

21.  Flow and congestion control

   Due to its proactive nature, the OLSR protocol has a natural control
   over the flow of its control traffic.  Nodes transmits control
   message at predetermined rates fixed by predefined refresh intervals.
   Furthermore the MPR optimization greatly saves on control overhead,
   and this is done on two sides.  First, the packets that advertise the
   topology are much shorter since only MPR selectors may be advertised.
   Second, the cost of flooding this information is greatly reduced
   since only MPR nodes forward the broadcast packets.  In dense
   networks, the reduction of control traffic can be of several orders
   of magnitude compared to routing protocols using classical flooding
   (such as OSPF) [10].  This feature naturally provides more bandwidth
   for useful data traffic and pushes further the frontier of
   congestion.  Since the control traffic is continuous and periodic, it
   keeps more stable the quality of the links used in routing, where
   reactive protocols, with bursty floodings for route discoveries and
   repairs, may damage the link qualities for short times by causing
   numerous collisions on those links, possibly provoking route repair
   cascades.  However, in certain OLSR options, some control messages
   may be intentionally sent in advance of their deadline(TC or Hello
   messages) in order to increase the reactiveness of the protocol
   against topology changes.  This may cause a small, temporary and
   local increase of control traffic.

22.  IANA Considerations

   OLSR defines a "Message Type" field for control messages.  A new
   registry has been created for the values for this Message Type field,
   and the following values assigned:

       Message Type             Value
      --------------------      -----
       HELLO_MESSAGE              1
       TC_MESSAGE                 2
       MID_MESSAGE                3
       HNA_MESSAGE                4

   Future values in the range 5-127 of the Message Type can be allocated
   using standards action [7].

   Additionally, values in the range 128-255 are reserved for
   private/local use.

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23.  Acknowledgments

   The authors would like to thank Joseph Macker
   <macker@itd.nrl.navy.mil> and his team, including Justin Dean
   <jdean@itd.nrl.navy.mil>, for their valuable suggestions on the
   advanced neighbor sensing mechanism and other various aspects of the
   protocol, including careful review of the protocol specification.

   The authors would also like to thank Christopher Dearlove
   <chris.dearlove@baesystems.com> for valuable input on the MPR
   selection heuristics and for careful reviews of the protocol
   specification.

24.  Contributors

   During the development of this specification, the following list of
   people contributed.  The contributors are listed alphabetically.

   Cedric Adjih
   Project HIPERCOM
   INRIA Rocquencourt, BP 105
   78153 Le Chesnay Cedex, France

   Phone: +33 1 3963 5215
   EMail: Cedric.Adjih@inria.fr

   Thomas Heide Clausen
   Project HIPERCOM
   INRIA Rocquencourt, BP 105
   78153 Le Chesnay Cedex, France

   Phone: +33 1 3963 5133
   EMail: T.Clausen@computer.org

   Philippe Jacquet
   Project HIPERCOM
   INRIA Rocquencourt, BP 105
   78153 Le Chesnay Cedex, France

   Phone: +33 1 3963 5263
   EMail: Philippe.Jacquet@inria.fr

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   Anis Laouiti
   Project HIPERCOM
   INRIA Rocquencourt, BP 105
   78153 Le Chesnay Cedex, France

   Phone: +33 1 3963 5088
   EMail: Anis.Laouiti@inria.fr

   Pascale Minet
   Project HIPERCOM
   INRIA Rocquencourt, BP 105
   78153 Le Chesnay Cedex, France

   Phone: +33 1 3963 5233
   EMail: Pascale.Minet@inria.fr

   Paul Muhlethaler
   Project HIPERCOM
   INRIA Rocquencourt, BP 105
   78153 Le Chesnay Cedex, France

   Phone: +33 1 3963 5278
   EMail: Paul.Muhlethaler@inria.fr

   Amir Qayyum
   Center for Advanced Research in Engineering Pvt. Ltd.
   19 Ataturk Avenue
   Islamabad, Pakistan

   Phone: +92-51-2874115
   EMail: amir@carepvtltd.com

   Laurent Viennot
   Project HIPERCOM
   INRIA Rocquencourt, BP 105
   78153 Le Chesnay Cedex, France

   Phone: +33 1 3963 5225
   EMail: Laurent.Viennot@inria.fr

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25.  References

25.1.  Normative References

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

   [7]   T.  Clausen, P.  Jacquet, A.  Laouiti, P.  Muhlethaler, A.
         Qayyum and L.  Viennot.  Optimized Link State Routing Protocol.
         IEEE INMIC Pakistan 2001.

25.2.  Informative References

   [1]   P. Jacquet, P. Minet, P. Muhlethaler, N. Rivierre.  Increasing
         reliability in cable free radio LANs: Low level forwarding in
         HIPERLAN.  Wireless Personal Communications, 1996.

   [2]   A.  Qayyum, L.  Viennot, A.  Laouiti.  Multipoint relaying: An
         efficient technique for flooding in mobile wireless networks.
         35th Annual Hawaii International Conference on System Sciences
         (HICSS'2001).

   [3]   ETSI STC-RES10 Committee.  Radio equipment and systems:
         HIPERLAN type 1, functional specifications ETS 300-652, ETSI,
         June 1996.

   [4]   P. Jacquet and L. Viennot, Overhead in Mobile Ad-hoc Network
         Protocols, INRIA research report RR-3965, 2000.

   [6]   T. Clausen, G. Hansen, L. Christensen and G. Behrmann.  The
         Optimized Link State Routing Protocol, Evaluation through
         Experiments and Simulation.  IEEE Symposium on "Wireless
         Personal Mobile Communications", September 2001.

   [8]   Narten, T. and H. Alvestrand, "Guidelines for Writing an IANA
         Considerations Section in RFCs"Path MTU discovery", RFC 1191,
              DOI 10.17487/RFC1191, November 1990,
              <http://www.rfc-editor.org/info/rfc1191>.

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   [RFC1981]  McCann, J., Deering, S., and J. Mogul, "Path MTU Discovery
              for IP version 6", RFC 1981, DOI 10.17487/RFC1981, August
              1996, <http://www.rfc-editor.org/info/rfc1981>.

   [RFC2119]  Bradner, S., "Key words for use in RFCs to Indicate
              Requirement Levels", BCP 14, RFC 2119,
              DOI 10.17487/RFC2119, March 1997,
              <http://www.rfc-editor.org/info/rfc2119>.

   [RFC2460]  Deering, S. and R. Hinden, "Internet Protocol, Version 6
              (IPv6) Specification", RFC 2460, DOI 10.17487/RFC2460,
              December 1998, <http://www.rfc-editor.org/info/rfc2460>.

   [RFC2914]  Floyd, S., "Congestion Control Principles", BCP 41,
              RFC 2914, DOI 10.17487/RFC2914, September 2000,
              <http://www.rfc-editor.org/info/rfc2914>.

   [RFC3828]  Larzon, L-A., Degermark, M., Pink, S., Jonsson, L-E., Ed.,
              and G. Fairhurst, Ed., "The Lightweight User Datagram
              Protocol (UDP-Lite)", RFC 3828, DOI 10.17487/RFC3828, July
              2004, <http://www.rfc-editor.org/info/rfc3828>.

   [RFC4787]  Audet, F., Ed. and C. Jennings, "Network Address
              Translation (NAT) Behavioral Requirements for Unicast
              UDP", BCP 127, RFC 4787, DOI 10.17487/RFC4787, January
              2007, <http://www.rfc-editor.org/info/rfc4787>.

   [RFC4821]  Mathis, M. and J. Heffner, "Packetization Layer Path MTU
              Discovery", RFC 4821, DOI 10.17487/RFC4821, March 2007,
              <http://www.rfc-editor.org/info/rfc4821>.

   [RFC5348]  Floyd, S., Handley, M., Padhye, J., and J. Widmer, "TCP
              Friendly Rate Control (TFRC): Protocol Specification",
              RFC 5348, DOI 10.17487/RFC5348, September 2008,
              <http://www.rfc-editor.org/info/rfc5348>.

   [RFC5405]  Eggert, L. and G. Fairhurst, "Unicast UDP Usage Guidelines
              for Application Designers", BCP 145, RFC 5405,
              DOI 10.17487/RFC5405, November 2008,
              <http://www.rfc-editor.org/info/rfc5405>.

   [RFC6040]  Briscoe, B., "Tunnelling of Explicit Congestion
              Notification", RFC 6040, DOI 10.17487/RFC6040, November
              2010, <http://www.rfc-editor.org/info/rfc6040>.

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   [RFC6298]  Paxson, V., Allman, M., Chu, J., and M. Sargent,
              "Computing TCP's Retransmission Timer", RFC 6298,
              DOI 10.17487/RFC6298, June 2011,
              <http://www.rfc-editor.org/info/rfc6298>.

10.2.  Informative References

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

   [FABER]    Faber, T., Touch, J., and W. Yue, "The TIME-WAIT State in
              TCP and Its Effect on Busy Servers", Proc. IEEE Infocom,
              March 1999.

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

   [I-D.ietf-aqm-ecn-benefits]
              Fairhurst, G. and M. Welzl, "The Benefits of using
              Explicit Congestion Notification (ECN)", draft-ietf-aqm-
              ecn-benefits-06 (work in progress), July 2015.

   [I-D.ietf-avtcore-rtp-circuit-breakers]
              Perkins, C. and V. Singh, "Multimedia Congestion Control:
              Circuit Breakers for Unicast RTP Sessions", draft-ietf-
              avtcore-rtp-circuit-breakers-11 (work in progress),
              October 2015.

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

   [I-D.ietf-intarea-tunnels]
              Touch, J. and W. Townsley, "IP Tunnels in the Internet
              Architecture", draft-ietf-intarea-tunnels-01 (work in
              progress), July 2015.

   [I-D.ietf-rtgwg-dt-encap]
              Nordmark, E., Tian, A., Gross, J., Hudson, J., Kreeger,
              L., Garg, P., Thaler, P., and T. Herbert, "Encapsulation
              Considerations", draft-ietf-rtgwg-dt-encap-00 (work in
              progress), July 2015.

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   [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, DOI 10.17487/RFC0896, January 1984,
              <http://www.rfc-editor.org/info/rfc896>.

   [RFC0919]  Mogul, J., "Broadcasting Internet Datagrams", STD 5,
              RFC 919, DOI 10.17487/RFC0919, October 1984,
              <http://www.rfc-editor.org/info/rfc919>.

   [RFC1112]  Deering, S., "Host extensions for IP multicasting", STD 5,
              RFC 1112, DOI 10.17487/RFC1112, August 1989,
              <http://www.rfc-editor.org/info/rfc1112>.

   [RFC1536]  Kumar, A., Postel, J., Neuman, C., Danzig, P., and S.
              Miller, "Common DNS Implementation Errors and Suggested
              Fixes", RFC 1536, DOI 10.17487/RFC1536, October 1993,
              <http://www.rfc-editor.org/info/rfc1536>.

   [RFC1546]  Partridge, C., Mendez, T., and W. Milliken, "Host
              Anycasting Service", RFC 1546, DOI 10.17487/RFC1546,
              November 1993, <http://www.rfc-editor.org/info/rfc1546>.

   [RFC2309]  Braden, B., Clark, D., Crowcroft, J., Davie, B., Deering,
              S., Estrin, D., Floyd, S., Jacobson, V., Minshall, G.,
              Partridge, C., Peterson, L., Ramakrishnan, K., Shenker,
              S., Wroclawski, J., and L. Zhang, "Recommendations on
              Queue Management and Congestion Avoidance in the
              Internet", RFC 2309, DOI 10.17487/RFC2309, April 1998,
              <http://www.rfc-editor.org/info/rfc2309>.

   [RFC2475]  Blake, S., Black, D., Carlson, M., Davies, E., Wang, Z.,
              and W. Weiss, "An Architecture for Differentiated
              Services", RFC 2475, DOI 10.17487/RFC2475, December 1998,
              <http://www.rfc-editor.org/info/rfc2475>.

   [RFC2675]  Borman, D., Deering, S., and R. Hinden, "IPv6 Jumbograms",
              RFC 2675, DOI 10.17487/RFC2675, August 1999,
              <http://www.rfc-editor.org/info/rfc2675>.

   [RFC2743]  Linn, J., "Generic Security Service Application Program
              Interface Version 2, Update 1", RFC 2743,
              DOI 10.17487/RFC2743, January 2000,
              <http://www.rfc-editor.org/info/rfc2743>.

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   [RFC2887]  Handley, M., Floyd, S., Whetten, B., Kermode, R.,
              Vicisano, L., and M. Luby, "The Reliable Multicast Design
              Space for Bulk Data Transfer", RFC 2887,
              DOI 10.17487/RFC2887, August 2000,
              <http://www.rfc-editor.org/info/rfc2887>.

   [RFC3048]  Whetten, B., Vicisano, L., Kermode, R., Handley, M.,
              Floyd, S., and M. Luby, "Reliable Multicast Transport
              Building Blocks for One-to-Many Bulk-Data Transfer",
              RFC 3048, DOI 10.17487/RFC3048, January 2001,
              <http://www.rfc-editor.org/info/rfc3048>.

   [RFC3124]  Balakrishnan, H. and S. Seshan, "The Congestion Manager",
              RFC 3124, DOI 10.17487/RFC3124, June 2001,
              <http://www.rfc-editor.org/info/rfc3124>.

   [RFC3168]  Ramakrishnan, K., Floyd, S., and D. Black, "The Addition
              of Explicit Congestion Notification (ECN) to IP",
              RFC 3168, DOI 10.17487/RFC3168, September 2001,
              <http://www.rfc-editor.org/info/rfc3168>.

   [RFC3261]  Rosenberg, J., Schulzrinne, H., Camarillo, G., Johnston,
              A., Peterson, J., Sparks, R., Handley, M., and E.
              Schooler, "SIP: Session Initiation Protocol", RFC 3261,
              DOI 10.17487/RFC3261, June 2002,
              <http://www.rfc-editor.org/info/rfc3261>.

   [RFC3303]  Srisuresh, P., Kuthan, J., Rosenberg, J., Molitor, A., and
              A. Rayhan, "Middlebox communication architecture and
              framework", RFC 3303, DOI 10.17487/RFC3303, August 2002,
              <http://www.rfc-editor.org/info/rfc3303>.

   [RFC3493]  Gilligan, R., Thomson, S., Bound, J., McCann, J., and W.
              Stevens, "Basic Socket Interface Extensions for IPv6",
              RFC 3493, DOI 10.17487/RFC3493, February 2003,
              <http://www.rfc-editor.org/info/rfc3493>.

   [RFC3550]  Schulzrinne, H., Casner, S., Frederick, R., and V.
              Jacobson, "RTP: A Transport Protocol for Real-Time
              Applications", STD 64, RFC 3550, DOI 10.17487/RFC3550,
              July 2003, <http://www.rfc-editor.org/info/rfc3550>.

   [RFC3551]  Schulzrinne, H. and S. Casner, "RTP Profile for Audio and
              Video Conferences with Minimal Control", STD 65, RFC 3551,
              DOI 10.17487/RFC3551, July 2003,
              <http://www.rfc-editor.org/info/rfc3551>.

Eggert, et al.             Expires May 5, 2016                 [Page 41]
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   [RFC3738]  Luby, M. and V. Goyal, "Wave and Equation Based Rate
              Control (WEBRC) Building Block", RFC 3738,
              DOI 10.17487/RFC3738, April 2004,
              <http://www.rfc-editor.org/info/rfc3738>.

   [RFC3758]  Stewart, R., Ramalho, M., Xie, Q., Tuexen, M., and P.
              Conrad, "Stream Control Transmission Protocol (SCTP)
              Partial Reliability Extension", RFC 3758,
              DOI 10.17487/RFC3758, May 2004,
              <http://www.rfc-editor.org/info/rfc3758>.

   [RFC3819]  Karn, P., Ed., Bormann, C., Fairhurst, G., Grossman, D.,
              Ludwig, R., Mahdavi, J., Montenegro, G., Touch, J., and L.
              Wood, "Advice for Internet Subnetwork Designers", BCP 89,
              RFC 3819, DOI 10.17487/RFC3819, July 2004,
              <http://www.rfc-editor.org/info/rfc3819>.

   [RFC4301]  Kent, S. and K. Seo, "Security Architecture for the
              Internet Protocol", RFC 4301, DOI 10.17487/RFC4301,
              December 2005, <http://www.rfc-editor.org/info/rfc4301>.

   [RFC4302]  Kent, S., "IP Authentication Header", RFC 4302,
              DOI 10.17487/RFC4302, December 2005,
              <http://www.rfc-editor.org/info/rfc4302>.

   [RFC4303]  Kent, S., "IP Encapsulating Security Payload (ESP)",
              RFC 4303, DOI 10.17487/RFC4303, December 2005,
              <http://www.rfc-editor.org/info/rfc4303>.

   [RFC4340]  Kohler, E., Handley, M., and S. Floyd, "Datagram
              Congestion Control Protocol (DCCP)", RFC 4340,
              DOI 10.17487/RFC4340, March 2006,
              <http://www.rfc-editor.org/info/rfc4340>.

   [RFC4341]  Floyd, S. and E. Kohler, "Profile for Datagram Congestion
              Control Protocol (DCCP) Congestion Control ID 2: TCP-like
              Congestion Control", RFC 4341, DOI 10.17487/RFC4341, March
              2006, <http://www.rfc-editor.org/info/rfc4341>.

   [RFC4342]  Floyd, S., Kohler, E., and J. Padhye, "Profile for
              Datagram Congestion Control Protocol (DCCP) Congestion
              Control ID 3: TCP-Friendly Rate Control (TFRC)", RFC 4342,
              DOI 10.17487/RFC4342, March 2006,
              <http://www.rfc-editor.org/info/rfc4342>.

   [RFC4607]  Holbrook, H. and B. Cain, "Source-Specific Multicast for
              IP", RFC 4607, DOI 10.17487/RFC4607, August 2006,
              <http://www.rfc-editor.org/info/rfc4607>.

Eggert, et al.             Expires May 5, 2016                 [Page 42]
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   [RFC4654]  Widmer, J. and M. Handley, "TCP-Friendly Multicast
              Congestion Control (TFMCC): Protocol Specification",
              RFC 4654, DOI 10.17487/RFC4654, August 2006,
              <http://www.rfc-editor.org/info/rfc4654>.

   [RFC4880]  Callas, J., Donnerhacke, L., Finney, H., Shaw, D., and R.
              Thayer, "OpenPGP Message Format", RFC 4880,
              DOI 10.17487/RFC4880, November 2007,
              <http://www.rfc-editor.org/info/rfc4880>.

   [RFC4960]  Stewart, R., Ed., "Stream Control Transmission Protocol",
              RFC 4960, DOI 10.17487/RFC4960, September 2007,
              <http://www.rfc-editor.org/info/rfc4960>.

   [RFC4963]  Heffner, J., Mathis, M., and B. Chandler, "IPv4 Reassembly
              Errors at High Data Rates", RFC 4963,
              DOI 10.17487/RFC4963, July 2007,
              <http://www.rfc-editor.org/info/rfc4963>.

   [RFC4987]  Eddy, W., "TCP SYN Flooding Attacks and Common
              Mitigations", RFC 4987, DOI 10.17487/RFC4987, August 2007,
              <http://www.rfc-editor.org/info/rfc4987>.

   [RFC5082]  Gill, V., Heasley, J., Meyer, D., Savola, P., Ed., and C.
              Pignataro, "The Generalized TTL Security Mechanism
              (GTSM)", RFC 5082, DOI 10.17487/RFC5082, October 2007,
              <http://www.rfc-editor.org/info/rfc5082>.

   [RFC5245]  Rosenberg, J., "Interactive Connectivity Establishment
              (ICE): A Protocol for Network Address Translator (NAT)
              Traversal for Offer/Answer Protocols", RFC 5245,
              DOI 10.17487/RFC5245, April 2010,
              <http://www.rfc-editor.org/info/rfc5245>.

   [RFC5622]  Floyd, S. and E. Kohler, "Profile for Datagram Congestion
              Control Protocol (DCCP) Congestion ID 4: TCP-Friendly Rate
              Control for Small Packets (TFRC-SP)", RFC 5622,
              DOI 10.17487/RFC5622, August 2009,
              <http://www.rfc-editor.org/info/rfc5622>.

   [RFC5652]  Housley, R., "Cryptographic Message Syntax (CMS)", STD 70,
              RFC 5652, DOI 10.17487/RFC5652, September 2009,
              <http://www.rfc-editor.org/info/rfc5652>.

   [RFC5740]  Adamson, B., Bormann, C., Handley, M., and J. Macker,
              "NACK-Oriented Reliable Multicast (NORM) Transport
              Protocol", RFC 5740, DOI 10.17487/RFC5740, November 2009,
              <http://www.rfc-editor.org/info/rfc5740>.

Eggert, et al.             Expires May 5, 2016                 [Page 43]
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   [RFC5751]  Ramsdell, B. and S. Turner, "Secure/Multipurpose Internet
              Mail Extensions (S/MIME) Version 3.2 Message
              Specification", RFC 5751, DOI 10.17487/RFC5751, January
              2010, <http://www.rfc-editor.org/info/rfc5751>.

   [RFC5775]  Luby, M., Watson, M., and L. Vicisano, "Asynchronous
              Layered Coding (ALC) Protocol Instantiation", RFC 5775,
              DOI 10.17487/RFC5775, April 2010,
              <http://www.rfc-editor.org/info/rfc5775>.

   [RFC5971]  Schulzrinne, H. and R. Hancock, "GIST: General Internet
              Signalling Transport", RFC 5971, DOI 10.17487/RFC5971,
              October 2010, <http://www.rfc-editor.org/info/rfc5971>.

   [RFC5973]  Stiemerling, M., Tschofenig, H., Aoun, C., and E. Davies,
              "NAT/Firewall NSIS Signaling Layer Protocol (NSLP)",
              RFC 5973, DOI 10.17487/RFC5973, October 2010,
              <http://www.rfc-editor.org/info/rfc5973>.

   [RFC6056]  Larsen, M. and F. Gont, "Recommendations for Transport-
              Protocol Port Randomization", BCP 156, RFC 6056,
              DOI 10.17487/RFC6056, January 2011,
              <http://www.rfc-editor.org/info/rfc6056>.

   [RFC6335]  Cotton, M., Eggert, L., Touch, J., Westerlund, M., and S.
              Cheshire, "Internet Assigned Numbers Authority (IANA)
              Procedures for the Management of the Service Name and
              Transport Protocol Port Number Registry", BCP 165,
              RFC 6335, DOI 10.17487/RFC6335, August 2011,
              <http://www.rfc-editor.org/info/rfc6335>.

   [RFC6347]  Rescorla, E. and N. Modadugu, "Datagram Transport Layer
              Security Version 1.2", RFC 6347, DOI 10.17487/RFC6347,
              January 2012, <http://www.rfc-editor.org/info/rfc6347>.

   [RFC6396]  Blunk, L., Karir, M., and C. Labovitz, "Multi-Threaded
              Routing Toolkit (MRT) Routing Information Export Format",
              RFC 6396, DOI 10.17487/RFC6396, October 2011,
              <http://www.rfc-editor.org/info/rfc6396>.

   [RFC6437]  Amante, S., Carpenter, B., Jiang, S., and J. Rajahalme,
              "IPv6 Flow Label Specification", RFC 6437,
              DOI 10.17487/RFC6437, November 2011,
              <http://www.rfc-editor.org/info/rfc6437>.

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   [RFC6438]  Carpenter, B. and S. Amante, "Using the IPv6 Flow Label
              for Equal Cost Multipath Routing and Link Aggregation in
              Tunnels", RFC 6438, DOI 10.17487/RFC6438, November 2011,
              <http://www.rfc-editor.org/info/rfc6438>.

   [RFC6513]  Rosen, E., Ed. and R. Aggarwal, Ed., "Multicast in MPLS/
              BGP IP VPNs", RFC 6513, DOI 10.17487/RFC6513, February
              2012, <http://www.rfc-editor.org/info/rfc6513>.

   [RFC6679]  Westerlund, M., Johansson, I., Perkins, C., O'Hanlon, P.,
              and K. Carlberg, "Explicit Congestion Notification (ECN)
              for RTP over UDP", RFC 6679, DOI 10.17487/RFC6679, August
              2012, <http://www.rfc-editor.org/info/rfc6679>.

   [RFC6726]  Paila, T., Walsh, R., Luby, M., Roca, V., and R. Lehtonen,
              "FLUTE - File Delivery over Unidirectional Transport",
              RFC 6726, DOI 10.17487/RFC6726, November 2012,
              <http://www.rfc-editor.org/info/rfc6726>.

   [RFC6807]  Farinacci, D., Shepherd, G., Venaas, S., and Y. Cai,
              "Population Count Extensions to Protocol Independent
              Multicast (PIM)", RFC 6807, DOI 10.17487/RFC6807, December
              2012, <http://www.rfc-editor.org/info/rfc6807>.

   [RFC6887]  Wing, D., Ed., Cheshire, S., Boucadair, M., Penno, R., and
              P. Selkirk, "Port Control Protocol (PCP)", RFC 6887,
              DOI 10.17487/RFC6887, April 2013,
              <http://www.rfc-editor.org/info/rfc6887>.

   [RFC6935]  Eubanks, M., Chimento, P., and M. Westerlund, "IPv6 and
              UDP Checksums for Tunneled Packets", RFC 6935,
              DOI 10.17487/RFC6935, April 2013,
              <http://www.rfc-editor.org/info/rfc6935>.

   [RFC6936]  Fairhurst, G. and M. Westerlund, "Applicability Statement
              for the Use of IPv6 UDP Datagrams with Zero Checksums",
              RFC 6936, DOI 10.17487/RFC6936, April 2013,
              <http://www.rfc-editor.org/info/rfc6936>.

   [RFC7143]  Chadalapaka, M., Satran, J., Meth, K., and D. Black,
              "Internet Small Computer System Interface (iSCSI) Protocol
              (Consolidated)", RFC 7143, DOI 10.17487/RFC7143, April
              2014, <http://www.rfc-editor.org/info/rfc7143>.

   [RFC7201]  Westerlund, M. and C. Perkins, "Options for Securing RTP
              Sessions", RFC 7201, DOI 10.17487/RFC7201, April 2014,
              <http://www.rfc-editor.org/info/rfc7201>.

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   [RFC7296]  Kaufman, C., Hoffman, P., Nir, Y., Eronen, P., and T.
              Kivinen, "Internet Key Exchange Protocol Version 2
              (IKEv2)", STD 79, RFC 7296, DOI 10.17487/RFC7296, October
              2014, <http://www.rfc-editor.org/info/rfc7296>.

   [RFC7510]  Xu, X., Sheth, N., Yong, L., Callon, R., and D. Black,
              "Encapsulating MPLS in UDP", RFC 7510,
              DOI 10.17487/RFC7510, April 2015,
              <http://www.rfc-editor.org/info/rfc7510>.

   [RFC7560]  Kuehlewind, M., Ed., Scheffenegger, R., and B. Briscoe,
              "Problem Statement and Requirements for Increased Accuracy
              in Explicit Congestion Notification (ECN) Feedback",
              RFC 7560, DOI 10.17487/RFC7560, August 2015,
              <http://www.rfc-editor.org/info/rfc7560>.

   [RFC7605]  Touch, J., "Recommendations on Using Assigned Transport
              Port Numbers", BCP 165, RFC 7605, DOI 10.17487/RFC7605,
              August 2015, <http://www.rfc-editor.org/info/rfc7605>.

   [RFC7675]  Perumal, M., Wing, D., Ravindranath, R., Reddy, T., and M.
              Thomson, "Session Traversal Utilities for NAT (STUN) Usage
              for Consent Freshness", RFC 7675, DOI 10.17487/RFC7675,
              October 2015, <http://www.rfc-editor.org/info/rfc7675>.

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

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

Eggert, et al.             Expires May 5, 2016                 [Page 46]
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quot;,  BCP 26, RFC 2434, October
         1998.

   [9]   Atkins, D., Stallings, W. and P. Zimmermann, "PGP Message
         Exchange Formats", RFC 1991, August 1996.

   [10]  P. Jacquet, A. Laouiti, P. Minet, L. Viennot.  Performance
         analysis of OLSR multipoint relay flooding in two ad hoc
         wireless network models, INRIA research report RR-4260, 2001.

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RFC 3626              Optimized Link State Routing          October 2003

26.  Authors' Addresses

   Thomas Heide Clausen
   Project HIPERCOM
   INRIA Rocquencourt, BP 105
   78153 Le Chesnay Cedex, France

   Phone: +33 1 3963 5133
   EMail: T.Clausen@computer.org

   Philippe Jacquet,
   Project HIPERCOM,
   INRIA Rocquencourt, BP 105
   78153 Le Chesnay Cedex, France

   Phone: +33 1 3963 5263,
   EMail: Philippe.Jacquet@inria.fr

Clausen & Jacquet             Experimental                     [Page 74]
RFC 3626              Optimized Link State Routing          October 2003

27.  Full Copyright Statement

   Copyright (C) The Internet Society (2003).  All Rights Reserved.

   This document and translations of it may be copied and furnished to
   others, and derivative works that comment on or otherwise explain it
   or assist in its implementation may be prepared, copied, published
   and distributed, in whole or in part, without restriction of any
   kind, provided that the above copyright notice and this paragraph are
   included on all such copies and derivative works.  However, this
   document itself may not be modified in any way, such as by removing
   the copyright notice or references to the Internet Society or other
   Internet organizations, except as needed for the purpose of
   developing Internet standards in which case the procedures for
   copyrights defined in the Internet Standards process must be
   followed, or as required to translate it into languages other than
   English.

   The limited permissions granted above are perpetual and will not be
   revoked by the Internet Society or its successors or assignees.

   This document and the information contained herein is provided on an
   "AS IS" basis and THE INTERNET SOCIETY AND THE INTERNET ENGINEERING
   TASK FORCE DISCLAIMS ALL WARRANTIES, EXPRESS OR IMPLIED, INCLUDING
   BUT NOT LIMITED TO ANY WARRANTY THAT THE USE OF THE INFORMATION
   HEREIN WILL NOT INFRINGE ANY RIGHTS OR ANY IMPLIED WARRANTIES OF
   MERCHANTABILITY OR FITNESS FOR A PARTICULAR PURPOSE.

Acknowledgement

   Funding for the RFC Editor function is currently provided by the
   Internet Society.

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