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) | ||
Formats | |||
Additional resources | Mailing list discussion | ||
IESG | Responsible AD | Alex D. Zinin | |
Send notices to | (None) |
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 Eggert, et al. Expires May 5, 2016 [Page 2] 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. Clausen & Jacquet Experimental [Page 21] RFC 3626 Optimized Link State Routing October 2003 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". Clausen & Jacquet Experimental [Page 22] RFC 3626 Optimized Link State Routing October 2003 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". Clausen & Jacquet Experimental [Page 23] RFC 3626 Optimized Link State Routing October 2003 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. Clausen & Jacquet Experimental [Page 24] RFC 3626 Optimized Link State Routing October 2003 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. Clausen & Jacquet Experimental [Page 25] RFC 3626 Optimized Link State Routing October 2003 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, Clausen & Jacquet Experimental [Page 26] RFC 3626 Optimized Link State Routing October 2003 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: Clausen & Jacquet Experimental [Page 27] RFC 3626 Optimized Link State Routing October 2003 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] Clausen & Jacquet Experimental [Page 28] RFC 3626 Optimized Link State Routing October 2003 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 | +-------+-------+-------+-------+-------+-------+-------+-------+ Clausen & Jacquet Experimental [Page 29] RFC 3626 Optimized Link State Routing October 2003 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, Eggert, et al. Expires May 5, 2016 [Page 3] Internet-Draft UDP Usage Guidelines November 2015 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]. Eggert, et al. Expires May 5, 2016 [Page 4] Internet-Draft UDP Usage Guidelines November 2015 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 Eggert, et al. Expires May 5, 2016 [Page 5] Internet-Draft UDP Usage Guidelines November 2015 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 Eggert, et al. Expires May 5, 2016 [Page 6] Internet-Draft UDP Usage Guidelines November 2015 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. Eggert, et al. Expires May 5, 2016 [Page 7] Internet-Draft UDP Usage Guidelines November 2015 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 Clausen & Jacquet Experimental [Page 30] RFC 3626 Optimized Link State Routing October 2003 - 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 Clausen & Jacquet Experimental [Page 31] RFC 3626 Optimized Link State Routing October 2003 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. Clausen & Jacquet Experimental [Page 32] RFC 3626 Optimized Link State Routing October 2003 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 Clausen & Jacquet Experimental [Page 33] RFC 3626 Optimized Link State Routing October 2003 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: Clausen & Jacquet Experimental [Page 34] RFC 3626 Optimized Link State Routing October 2003 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: Clausen & Jacquet Experimental [Page 35] RFC 3626 Optimized Link State Routing October 2003 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. Clausen & Jacquet Experimental [Page 36] RFC 3626 Optimized Link State Routing October 2003 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: Clausen & Jacquet Experimental [Page 37] RFC 3626 Optimized Link State Routing October 2003 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. Clausen & Jacquet Experimental [Page 38] RFC 3626 Optimized Link State Routing October 2003 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. Clausen & Jacquet Experimental [Page 39] RFC 3626 Optimized Link State Routing October 2003 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 Eggert, et al. Expires May 5, 2016 [Page 8] Internet-Draft UDP Usage Guidelines November 2015 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 Eggert, et al. Expires May 5, 2016 [Page 9] Internet-Draft UDP Usage Guidelines November 2015 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. Eggert, et al. Expires May 5, 2016 [Page 10] Internet-Draft UDP Usage Guidelines November 2015 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). Eggert, et al. Expires May 5, 2016 [Page 11] Internet-Draft UDP Usage Guidelines November 2015 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. Eggert, et al. Expires May 5, 2016 [Page 12] Internet-Draft UDP Usage Guidelines November 2015 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. Eggert, et al. Expires May 5, 2016 [Page 13] Internet-Draft UDP Usage Guidelines November 2015 [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. Eggert, et al. Expires May 5, 2016 [Page 14] Internet-Draft UDP Usage Guidelines November 2015 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 Eggert, et al. Expires May 5, 2016 [Page 15] Internet-Draft UDP Usage Guidelines November 2015MPR set of an interface 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: Clausen & Jacquet Experimental [Page 40] RFC 3626 Optimized Link State Routing October 2003 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: Clausen & Jacquet Experimental [Page 41] RFC 3626 Optimized Link State Routing October 2003 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. Clausen & Jacquet Experimental [Page 42] RFC 3626 Optimized Link State Routing October 2003 - 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. Clausen & Jacquet Experimental [Page 43] RFC 3626 Optimized Link State Routing October 2003 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. Clausen & Jacquet Experimental [Page 44] RFC 3626 Optimized Link State Routing October 2003 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. Clausen & Jacquet Experimental [Page 45] RFC 3626 Optimized Link State Routing October 2003 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. Clausen & Jacquet Experimental [Page 46] RFC 3626 Optimized Link State Routing October 2003 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 Clausen & Jacquet Experimental [Page 47] RFC 3626 Optimized Link State Routing October 2003 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. Clausen & Jacquet Experimental [Page 48] RFC 3626 Optimized Link State Routing October 2003 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 Eggert, et al. Expires May 5, 2016 [Page 16] Internet-Draft UDP Usage Guidelines November 2015 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]. Eggert, et al. Expires May 5, 2016 [Page 17] Internet-Draft UDP Usage Guidelines November 2015 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. Eggert, et al. Expires May 5, 2016 [Page 18] Internet-Draft UDP Usage Guidelines November 2015 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. Eggert, et al. Expires May 5, 2016 [Page 19] Internet-Draft UDP Usage Guidelines November 2015 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 Eggert, et al. Expires May 5, 2016 [Page 20] Internet-Draft UDP Usage Guidelines November 2015 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. Eggert, et al. Expires May 5, 2016 [Page 21] Internet-Draft UDP Usage Guidelines November 2015 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 Clausen & Jacquet Experimental [Page 49] RFC 3626 Optimized Link State Routing October 2003 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. Clausen & Jacquet Experimental [Page 50] RFC 3626 Optimized Link State Routing October 2003 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. Clausen & Jacquet Experimental [Page 51] RFC 3626 Optimized Link State Routing October 2003 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). Clausen & Jacquet Experimental [Page 52] RFC 3626 Optimized Link State Routing October 2003 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: Clausen & Jacquet Experimental [Page 53] RFC 3626 Optimized Link State Routing October 2003 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 Clausen & Jacquet Experimental [Page 54] RFC 3626 Optimized Link State Routing October 2003 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). Clausen & Jacquet Experimental [Page 55] RFC 3626 Optimized Link State Routing October 2003 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. Clausen & Jacquet Experimental [Page 56] RFC 3626 Optimized Link State Routing October 2003 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 Clausen & Jacquet Experimental [Page 57] RFC 3626 Optimized Link State Routing October 2003 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 Clausen & Jacquet Experimental [Page 58] RFC 3626 Optimized Link State Routing October 2003 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. Clausen & Jacquet Experimental [Page 59] RFC 3626 Optimized Link State Routing October 2003 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 Clausen & Jacquet Experimental [Page 60] RFC 3626 Optimized Link State Routing October 2003 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. Clausen & Jacquet Experimental [Page 61] RFC 3626 Optimized Link State Routing October 2003 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. Clausen & Jacquet Experimental [Page 62] RFC 3626 Optimized Link State Routing October 2003 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- Eggert, et al. Expires May 5, 2016 [Page 22] Internet-Draft UDP Usage Guidelines November 2015 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. Eggert, et al. Expires May 5, 2016 [Page 23] Internet-Draft UDP Usage Guidelines November 2015 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 Eggert, et al. Expires May 5, 2016 [Page 24] Internet-Draft UDP Usage Guidelines November 2015 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 Eggert, et al. Expires May 5, 2016 [Page 25] Internet-Draft UDP Usage Guidelines November 2015 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 Eggert, et al. Expires May 5, 2016 [Page 26] Internet-Draft UDP Usage Guidelines November 2015 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 Eggert, et al. Expires May 5, 2016 [Page 27] Internet-Draft UDP Usage Guidelines November 2015 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 Eggert, et al. Expires May 5, 2016 [Page 28] Internet-Draft UDP Usage Guidelines November 2015 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. Eggert, et al. Expires May 5, 2016 [Page 29] Internet-Draft UDP Usage Guidelines November 2015 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 Eggert, et al. Expires May 5, 2016 [Page 30] Internet-Draft UDP Usage Guidelines November 2015 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). Eggert, et al. Expires May 5, 2016 [Page 31] Internet-Draft UDP Usage Guidelines November 2015 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 Eggert, et al. Expires May 5, 2016 [Page 32] Internet-Draft UDP Usage Guidelines November 2015 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 Eggert, et al. Expires May 5, 2016 [Page 33] Internet-Draft UDP Usage Guidelines November 2015 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 Eggert, et al. Expires May 5, 2016 [Page 34] Internet-Draft UDP Usage Guidelines November 2015 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 | | | | | Eggert, et al. Expires May 5, 2016 [Page 35] Internet-Draft UDP Usage Guidelines November 2015 | 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 Eggert, et al. Expires May 5, 2016 [Page 36] Internet-Draft UDP Usage Guidelines November 2015 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] RFC 3626 Optimized Link State Routing October 2003 '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. Clausen & Jacquet Experimental [Page 64] RFC 3626 Optimized Link State Routing October 2003 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 Clausen & Jacquet Experimental [Page 65] RFC 3626 Optimized Link State Routing October 2003 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. Clausen & Jacquet Experimental [Page 66] RFC 3626 Optimized Link State Routing October 2003 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. Clausen & Jacquet Experimental [Page 67] RFC 3626 Optimized Link State Routing October 2003 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. Clausen & Jacquet Experimental [Page 68] RFC 3626 Optimized Link State Routing October 2003 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. Clausen & Jacquet Experimental [Page 69] RFC 3626 Optimized Link State Routing October 2003 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. Clausen & Jacquet Experimental [Page 70] RFC 3626 Optimized Link State Routing October 2003 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 Clausen & Jacquet Experimental [Page 71] RFC 3626 Optimized Link State Routing October 2003 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 Clausen & Jacquet Experimental [Page 72] RFC 3626 Optimized Link State Routing October 2003 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>. Eggert, et al. Expires May 5, 2016 [Page 37] Internet-Draft UDP Usage Guidelines November 2015 [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>. Eggert, et al. Expires May 5, 2016 [Page 38] Internet-Draft UDP Usage Guidelines November 2015 [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. Eggert, et al. Expires May 5, 2016 [Page 39] Internet-Draft UDP Usage Guidelines November 2015 [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>. Eggert, et al. Expires May 5, 2016 [Page 40] Internet-Draft UDP Usage Guidelines November 2015 [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] Internet-Draft UDP Usage Guidelines November 2015 [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] Internet-Draft UDP Usage Guidelines November 2015 [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] Internet-Draft UDP Usage Guidelines November 2015 [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>. Eggert, et al. Expires May 5, 2016 [Page 44] Internet-Draft UDP Usage Guidelines November 2015 [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>. Eggert, et al. Expires May 5, 2016 [Page 45] Internet-Draft UDP Usage Guidelines November 2015 [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] Internet-Draft UDP Usage Guidelines November 2015 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. Clausen & Jacquet Experimental [Page 73] 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. Clausen & Jacquet Experimental [Page 75]