brian@ucsdUCSD.Edu (Brian Kantor) (09/04/89)
Network Working Group Internet Engineering Task Force Request for Comments: COMM R. Braden, Editor June 16, 1989 Requirements for Internet Hosts -- Communication Layers *** DRAFT *** Status of This Memo This is a draft of one RFC of a pair that defines and discusses the requirements for Internet host software. This RFC covers the communications protocol layers: link layer, IP layer, and transport layer; its companion RFC-APPL covers the application and support protocols. When complete, these two RFC's will form an official specification for the Internet community. It incorporates by reference, amends, corrects, and supplements the primary protocol standards documents relating to hosts. Distribution of this document is unlimited. This draft incorporates the changes agreed to at the Austin IETF meeting, January 1989, plus many minor changes suggested by Mike Karels and by others, plus major changes agreed to by the WG. Most recent changes are indicated with !, while earlier changes are marked with | or #. Minor improvements in wording or clarifications are marked with @. Table of Contents 1. INTRODUCTION ............................................... 5 1.1 The Internet Architecture .............................. 6 1.1.1 Internet Hosts .................................... 6 1.1.2 Architectural Assumptions ......................... 7 1.1.3 Internet Protocol Suite ........................... 8 1.1.4 Embedded Gateway Code ............................. 10 1.2 General Considerations ................................. 12 1.2.1 Continuing Internet Evolution ........................ 12 Internet Engineering Task Force [Page 1] ***DRAFT RFC*** INTRODUCTION June 16, 1989 1.2.2 Robustness Principle .............................. 12 1.2.3 Error Logging ..................................... 13 1.2.4 Configuration ..................................... 14 1.3 Reading this Document .................................. 15 1.3.1 Organization ...................................... 15 1.3.2 Requirements ...................................... 16 1.3.3 Terminology ....................................... 17 2. LINK LAYER .................................................. 20 2.1 INTRODUCTION ........................................... 20 2.2 PROTOCOL WALK-THROUGH .................................. 20 2.3 SPECIFIC ISSUES ........................................ 20 2.3.1 Trailer Protocol Negotiation ...................... 20 2.3.2 Address Resolution Protocol -- ARP ................ 21 2.3.2.1 ARP Cache Validation ......................... 21 2.3.2.2 ARP Packet Queue ............................. 23 2.3.3 Ethernet and IEEE 802 Encapsulation ............... 23 2.4 LINK/INTERNET LAYER INTERFACE .......................... 24 2.5 LINK LAYER REQUIREMENTS SUMMARY ........................ 25 3. INTERNET LAYER PROTOCOLS .................................... 26 3.1 INTRODUCTION ............................................ 26 3.2 PROTOCOL WALK-THROUGH .................................. 27 3.2.1 Internet Protocol -- IP ............................ 27 3.2.1.1 Version Number ............................... 27 3.2.1.2 Checksum ..................................... 28 3.2.1.3 Addressing ................................... 28 3.2.1.4 Fragmentation and Reassembly ................. 30 3.2.1.5 Identification ............................... 30 3.2.1.6 Type-of-Service .............................. 31 3.2.1.7 Time-to-Live ................................. 32 3.2.1.8 Options ...................................... 32 3.2.2 Internet Control Message Protocol -- ICMP .......... 36 3.2.2.1 Destination Unreachable ...................... 38 3.2.2.2 Redirect ..................................... 39 3.2.2.3 Source Quench ................................ 39 3.2.2.4 Time Exceeded ................................ 40 3.2.2.5 Parameter Problem ............................ 40 3.2.2.6 Echo Request/Reply ........................... 40 3.2.2.7 Information Request/Reply .................... 41 3.2.2.8 Timestamp and Timestamp Reply ................ 41 3.2.2.9 Address Mask Request/Reply ................... 42 3.3 SPECIFIC ISSUES ........................................ 44 3.3.1 Routing Outbound Datagrams ........................ 45 3.3.1.1 Local/Remote Decision ........................ 45 3.3.1.2 Gateway Selection ............................ 45 3.3.1.3 Route Cache .................................. 47 3.3.1.4 Dead Gateway Detection ....................... 48 Internet Engineering Task Force [Page 2] ***DRAFT RFC*** INTRODUCTION June 16, 1989 3.3.1.5 New Gateway Selection ........................ 52 3.3.1.6 Initialization ............................... 53 3.3.2 Reassembly ........................................ 54 3.3.3 Fragmentation ..................................... 55 3.3.4 Multihomed Hosts .................................. 57 3.3.4.1 Local Multihoming ............................ 57 3.3.4.2 Selecting a Logical Interface ................ 59 3.3.5 IP Source Address ................................. 61 3.3.6 Broadcasts ........................................ 62 3.3.7 Error Reporting ................................... 63 3.4 INTERNET/TRANSPORT LAYER INTERFACE ..................... 63 3.5 INTERNET LAYER REQUIREMENTS SUMMARY .................... 66 4. TRANSPORT PROTOCOLS ......................................... 71 4.1 USER DATAGRAM PROTOCOL -- UDP .......................... 71 4.1.1 INTRODUCTION ...................................... 71 4.1.2 PROTOCOL WALK-THROUGH ............................. 71 4.1.3 SPECIFIC ISSUES ................................... 71 4.1.3.1 Ports ........................................ 71 4.1.3.2 IP Options ................................... 71 4.1.3.3 ICMP Messages ................................ 72 4.1.3.4 UDP Checksums ................................ 72 4.1.3.5 UDP Multihoming .............................. 73 4.1.3.6 Invalid Addresses ............................ 73 4.1.4 UDP/APPLICATION LAYER INTERFACE ................... 73 4.1.5 UDP REQUIREMENTS SUMMARY .......................... 74 4.2 TRANSMISSION CONTROL PROTOCOL -- TCP ................... 76 4.2.1 INTRODUCTION ...................................... 76 4.2.2 PROTOCOL WALK-THROUGH ............................. 76 4.2.2.1 Well-Known Ports ............................. 76 4.2.2.2 Use of Push .................................. 76 4.2.2.3 Window Size .................................. 77 4.2.2.4 Urgent Pointer ............................... 78 4.2.2.5 TCP Options .................................. 78 4.2.2.6 Maximum Segment Size Option .................. 79 4.2.2.7 TCP Checksum ................................. 80 4.2.2.8 TCP Connection State Diagram ................. 80 4.2.2.9 Initial Sequence Number Selection ............ 81 4.2.2.10 Simultaneous Open Attempts .................. 81 4.2.2.11 Recovery from Old Duplicate SYN ............. 81 4.2.2.12 RST Segment ................................. 81 4.2.2.13 Closing a Connection ........................ 81 4.2.2.14 Data Communication .......................... 83 4.2.2.15 Retransmission Timeout ...................... 84 4.2.2.16 Managing the Window ......................... 85 4.2.2.17 Probing Zero Windows ........................ 85 4.2.2.18 Passive OPEN Calls .......................... 86 4.2.2.19 Queueing Out-of-Order Segments .............. 86 Internet Engineering Task Force [Page 3] ***DRAFT RFC*** INTRODUCTION June 16, 1989 4.2.2.20 Event Processing ............................ 87 4.2.2.21 Acknowledging Queued Segments ............... 88 4.2.3 SPECIFIC ISSUES ................................... 88 4.2.3.1 Retransmission Timeout Calculation ........... 88 4.2.3.2 When to Send an ACK Segment .................. 90 4.2.3.3 When to Send a Window Update ................. 90 4.2.3.4 When to Send Data ............................ 91 4.2.3.5 TCP Connection Liveness ...................... 93 4.2.3.6 TCP Open Failure ............................. 95 4.2.3.7 TCP Multihoming .............................. 96 4.2.3.8 IP Options ................................... 96 4.2.3.9 ICMP Messages ................................ 96 4.2.3.10 Remote Address Validation ................... 97 4.2.3.11 TCP Traffic Patterns ........................ 97 4.2.3.12 Efficiency .................................. 98 4.2.4 TCP/APPLICATION LAYER INTERFACE ................... 99 4.2.4.1 Asynchronous Reports ......................... 99 4.2.4.2 Type-of-Service .............................. 100 4.2.4.3 Flush Call ................................... 100 4.2.4.4 Multihoming .................................. 101 4.2.5 TCP REQUIREMENT SUMMARY ........................... 101 5. REFERENCES ................................................. 105 Internet Engineering Task Force [Page 4] ***DRAFT RFC*** INTRODUCTION June 16, 1989 1. INTRODUCTION This document is one of a pair of RFC's that defines and discusss the | requirements for host system implementations of the Internet protocol | suite. This RFC covers the communication protocol layers: link | layer, IP layer, and transport layer. Its companion RFC, | "Requirements for Internet Hosts -- Application and Support", RFC- | appl [INTRO:1], covers the application layer protocols. These two | RFC's should also be read in conjunction with "Requirements for | Internet Gateways," RFC-1009 [INTRO:2]. | This RFC enumerates standard protocols that a host connected to the Internet must use, and it incorporates by reference the RFCs and other documents describing the current specifications for these protocols. It corrects errors in the referenced documents and adds additional discussion and guidance for an implementor. For each protocol, this document contains an explicit set of requirements, recommendations, and options. The reader must understand that the list of requirements in this document is incomplete by itself; the complete set of requirements for an Internet host is primarily defined in the standard protocol specification document, with corrections, amendments, and supplements contained in this RFC. In many cases, the "requirements" in this RFC are already stated or implied in the standard protocol documents, so that their inclusion here is, in a sense, redundant. However, many of the requirements that have been listed here have been ignored by some set of implementors in the past, causing problems of interoperability, performance, and robustness. This document includes discussion and explanation of many of the requirements and recommendations. A simple list of requirements would be dangerous, because: o Some required features are more important than others, and some features are optional. o There may be valid reasons why particular vendor products that are designed for restricted contexts might choose to use different specifications. However, the specifications of this document must be followed to meet the general goal of arbitrary host interoperation across the diversity and complexity of the Internet system. Although most current implementations fail to meet these requirements in various ways, some minor and some major, this specification is the ideal towards which we need to move. Internet Engineering Task Force [Page 5] ***DRAFT RFC*** INTRODUCTION June 16, 1989 These requirements are based on the current level of Internet architecture. This document will be updated as required to provide additional clarifications or to include additional information in those areas in which specifications are still evolving. This introductory section begins with a brief overview of the Internet architecture as it relates to hosts, and then gives some general advice to host software vendors. Finally, there is some guidance on reading the rest of the document and general references. 1.1 The Internet Architecture General background and discussion on the Internet architecture and supporting protocol suite can be found in the DDN Protocol Handbook [INTRO:3]; for background see for example [INTRO:9], [INTRO:10], and [INTRO:11]. Reference [INTRO:5] describes the procedure for obtaining Internet protocol documents, while [INTRO:6] contains a list of the numbers assigned within Internet protocols. 1.1.1 Internet Hosts A host computer, or simply "host," is the ultimate consumer of communication services. A host generally executes application programs on behalf of user(s), employing network and/or Internet communication services in support of this function. An Internet host corresponds to the concept of an "End-System" used in the OSI protocol suite [INTRO:13]. An Internet communication system consists of interconnected packet networks supporting communications among host computers using the Internet protocols. The networks are interconnected using packet-switching computers called "gateways" or "IP routers" by the Internet community, and "Intermediate Systems" by the OSI world [INTRO:13]. The RFC "Requirements for Internet Gateways" [INTRO:2] contains the official specifications for Internet gateways. That RFC together with the present document and its companion [INTRO:1] define the rules for the current realization of the Internet architecture. Internet hosts span a wide range of size, speed, and function. They range in size from small microprocessors through workstations to mainframes and supercomputers. In function, they range from single-purpose hosts (such as terminal servers) to full-service hosts that support a variety of online network services, typically including remote login, file transfer, and electronic mail. Internet Engineering Task Force [Page 6] ***DRAFT RFC*** INTRODUCTION June 16, 1989 A host is generally said to be multihomed if it has more than one interface to the same or to different networks. A more precise definition will be given later. 1.1.2 Architectural Assumptions The current Internet architecture is based on a set of assumptions about the system; the assumptions most relevant to hosts are as follows: (1) The Internet is a network of networks. Each host is directly connected to some particular network(s); its connection to the Internet is only conceptual. Two hosts on the same network will communicate with each other using the same set of protocols that they would use to communicate with hosts on distant networks. (2) Gateways don't keep connection state information. To improve robustness of the communication system, gateways are designed to be stateless, forwarding each IP datagram independently of other datagrams. As a result, redundant paths can be exploited to provide robust service in spite of failures of intervening gateways and networks. All state information required for end-to-end flow control and reliability is implemented in the hosts, in the transport layer or in application programs. All connection control information is thus co-located with the end points of the communication, so it will be lost only if an end point fails. (3) Routing complexity should be in the gateways. Routing is a complex and difficult problem, and ought to be performed by the gateways, not the hosts. An important objective is to insulate host software from changes caused by the inevitable evolution of the Internet routing architecture. (4) The System must tolerate wide network variation. Internet Engineering Task Force [Page 7] ***DRAFT RFC*** INTRODUCTION June 16, 1989 A basic objective of the Internet design is to tolerate a wide range of network characteristics -- e.g., bandwidth, delay, packet loss, packet reordering, and maximum packet size. Another objective is robustness against failure of individual networks, gateways, and hosts, using whatever bandwidth is still available. Finally, the goal is full "open system interconnection": an Internet host must be able to interoperate robustly and effectively with any other Internet host, across diverse Internet paths. Sometimes host implementors have designed for less ambitious goals. For example, the LAN environment is typically much more benign than the Internet as a whole; LANs have low packet loss and delay and do not reorder packets. Some vendors have fielded host implementations that are adequate for a simple LAN environment, but work badly for general interoperation. The vendor justifies such a product as being economical within the restricted LAN market. However, isolated LANs seldom stay isolated for long; they are soon gatewayed to each other, to organization-wide internets, and eventually to the global Internet system. In the end, neither the customer nor the vendor is served by incomplete or substandard Internet host software. The requirements spelled out in this document are designed for a full-function Internet host, capable of full interoperation over an arbitrary Internet path. 1.1.3 Internet Protocol Suite To communicate using the Internet system, a host must implement the layered set of protocols comprising the Internet protocol suite. A host typically must implement at least one protocol from each layer. The protocol layers used in the Internet architecture are as follows [INTRO:4]: o Application Layer The application layer is the top layer of the Internet protocol suite. We distinguish two categories of application layer protocols: user protocols that provide service directly to users, and support protocols that provide common system functions. Internet Engineering Task Force [Page 8] ***DRAFT RFC*** INTRODUCTION June 16, 1989 The most common user protocols are: o Telnet (remote login) o FTP (file transfer) o SMTP (electronic mail delivery) There are a number of other standardized user protocols [INTRO:4] and many private user protocols. Support protocols, used for host name mapping, booting, and management, include SNMP, BOOTP, RARP, and the Domain Name System (DNS) protocols. Requirements for user and support protocols will be found | in the companion RFC [INTRO:1]. | The Internet suite does not further subdivide the application layer, although some of the Internet application layer protocols do contain some internal sub- layering. The application layer of the Internet suite essentially combines the functions of the top two layers -- Presentation and Application -- of the OSI reference model. o Transport Layer The transport layer provides end-to-end communication services for applications. There are two primary transport layer protocols at present: o Transmission Control Protocol (TCP) o User Datagram Protocol (UDP) TCP is a reliable connection-oriented transport service that provides end-to-end reliability, resequencing, and flow control. UDP is a connectionless ("datagram") transport service. Other transport protocols have been developed by the research community, and the set of official Internet transport protocols may be expanded in the future. Transport layer protocols are discussed in Chapter 4. o Internet Layer Internet Engineering Task Force [Page 9] ***DRAFT RFC*** INTRODUCTION June 16, 1989 All Internet transport protocols use the Internet Protocol (IP) to carry data from source host to destination host. IP is a connectionless or datagram internetwork service, providing no end-to-end delivery guarantees. Thus, IP datagrams may arrive at the destination host damaged, duplicated, out of order, or not at all. The layers above IP are responsible for reliable delivery service when it is required. The IP protocol includes provision for addressing, type-of-service specification, fragmentation and reassembly, and security information. The datagram or connectionless nature of the IP protocol is a fundamental and characteristic feature of the Internet architecture. Internet IP was the model for the ISO Connectionless Network Protocol [INTRO:12]. ICMP is a control protocol that is considered to be an integral part of IP, although it is architecturally layered upon IP, i.e., it uses IP to carry its data end- to-end just as a transport protocol like TCP or UDP does. ICMP provides error reporting, congestion reporting, and first-hop gateway redirection. The Internet layer protocols IP and ICMP are discussed in Chapter 3. o Link Layer To communicate on its directly-connected network, a host must implement the communication protocol used to interface to that network. We call this a link layer or media-access layer protocol. There is a wide variety of link layer protocols, corresponding to the many different types of networks. See Chapter 2. 1.1.4 Embedded Gateway Code Some Internet host software includes embedded gateway functionality, so that these hosts can forward packets as a gateway would, while still performing the application layer functions of a host. Such dual-purpose systems must follow the requirements of RFC- 1009 with respect to their gateway functions, and must follow Internet Engineering Task Force [Page 10] ***DRAFT RFC*** INTRODUCTION June 16, 1989 the present document with respect to their host functions. In all overlapping cases, the two specifications should be in agreement. There are varying opinions in the Internet community about whether embedded gateway functionality is a good idea. The main arguments are as follows: o Pro: in a local network environment where networking is informal, or in isolated internets, it may be convenient and economical to use existing host systems as gateways. There is also an architectural argument for embedded gateway functionality: multihoming is much more common than originally foreseen, and multihoming forces a host to make routing decisions as if it were a gateway. If the multihomed host contains an embedded gateway, it will have full routing knowledge and as a result will be able to make optimal routing decisions. o Con: Gateway algorithms and protocols are still changing, and they will continue to change as the Internet system grows larger. Attempting to include a general gateway function within the host IP layer will force the host system maintainer to track these (more frequent) changes. Also, a larger pool of gateway implementations will make coordinating the changes more difficult. Finally, the complexity of a gateway IP layer is somewhat greater than that of a host, making the implementation and operation tasks more complex. In addition, the style of operation of some hosts is not appropriate for providing stable and robust gateway service. There is considerable merit in both of these viewpoints. One conclusion can be drawn: any Internet host software that includes embedded gateway code must have a configuration switch to disable the gateway function, and THIS SWITCH MUST DEFAULT TO THE NON-GATEWAY MODE. In this mode, a datagram arriving through one interface will not be forwarded to another host or gateway (unless it is source-routed), regardless of whether the host is single-homed or multihomed. The host software must not automatically move into gateway mode if the host has more than one interface, as the operator of the machine may neither want to provide that service nor be competent to do so. Internet Engineering Task Force [Page 11] ***DRAFT RFC*** INTRODUCTION June 16, 1989 1.2 General Considerations There are two important lessons that vendors of Internet host software have learned and which a new vendor should consider seriously. 1.2.1 Continuing Internet Evolution The enormous growth of the Internet has revealed problems of management and scaling in a large datagram-based packet communication system. These problems are being addressed, and as a result there will be continuing evolution of the specifications described in this document. These changes will be carefully planned and controlled, since there is extensive participation in this planning by the vendors and by the organizations responsible for operations of the networks. Development, evolution, and revision are characteristic of computer network protocols today, and this situation will persist for some years. A vendor who develops computer communication software for the Internet protocol suite (or any other protocol suite!) and then fails to maintain and update that software for changing specifications is going to leave a trail of unhappy customers. The Internet is a large communication network, and the users are in constant contact through it. Experience has shown that knowledge of deficiencies in vendor software propagates quickly through the Internet technical community. 1.2.2 Robustness Principle At every layer of the protocols, there is a general rule whose application can lead to enormous benefits in robustness and interoperability [IP:1]: "Be liberal in what you accept, and conservative in what you send" Software should be written to deal with every conceivable error, no matter how unlikely; sooner or later a packet will come in with that particular combination of errors and attributes, and unless the software is prepared, chaos can ensue. In general, it is best to assume that the network is filled with malevolent entities that will send in packets designed to have the worst possible effect. This assumption will lead to suitable protective design, although the most serious problems in the Internet have been caused by unenvisaged mechanisms triggered by low-probability events; Internet Engineering Task Force [Page 12] ***DRAFT RFC*** INTRODUCTION June 16, 1989 mere human malice would never have taken so devious a course! Adaptability to change must be designed into all levels of Internet host software. As a simple example, consider a protocol specification that contains an enumeration of values for a particular header field -- e.g., a type field, a port number, or an error code; this enumeration must be assumed to be incomplete. Thus, if a protocol specification defines four possible error codes, the software must not break when a fifth code shows up. An undefined code might be logged (see below), but it must not cause a failure. The second part of the principle is almost as important: software on other hosts may contain deficiencies that make it unwise to exploit legal but obscure protocol features. It is unwise to stray far from the obvious and simple, lest untoward effects elsewhere result. A corollary of this is "watch out for misbehaving hosts"; host software should be prepared, not just to survive other misbehaving hosts, but also to cooperate to limit the amount of disruption such hosts can cause to the shared communication facility. 1.2.3 Error Logging | The Internet includes a great variety of host and gateway | systems, each implementing many protocols and protocol layers, | and some of these contain bugs and mis-features in their | protocol processing. As a result of complexity, diversity, and | distribution of function, the diagnosis of Internet problems is | often very difficult. | Problem diagnosis will be aided if host implementations include | a carefully designed facility for logging erroneous or | "strange" protocol events. It is important to include as much | diagnostic information as possible when an error is logged. In | particular, it is often useful to record the entire header of | the packet that caused the error. However, care must be taken | to ensure that error logging does not consume prohibitive | amounts of resources or otherwise interfere with the operation | of the host. | There is a tendency for abnormal but harmless protocol events | to overflow error logging files; this can be avoided by using a | "circular" log, or by enabling logging only while diagnosing a | known failure. It may be useful to filter and count duplicate | successive messages. One strategy that seems to work well is: | (1) always count abnormalities and make such counts accessible | Internet Engineering Task Force [Page 13] ***DRAFT RFC*** INTRODUCTION June 16, 1989 through the management protocol (see RFC-app [INTRO:1]); and | (2) be able to selectively enable logging of a great variety of | events. For example, it might useful to be able to "log | everything" or to "log everything for host X". | Note that different managements may have differing policies | about the amount of error logging that they want normally | enabled in a host. Some will say, "if it doesn't hurt me, I | don't want to know about it", while others will want to take a | more watchful and agressive attitude about detecting and | removing protocol abnormalities. 1.2.4 Configuration It would be ideal if a host implementation of the Internet protocol suite could be entirely self-configuring. This would allow the whole suite to be implemented in ROM or cast into silicon, it would simplify diskless workstations, and it would be an immense boon to harried LAN administrators as well as system vendors. We have not reached this ideal; in fact, we are not even close. At many points in this document, you will find a requirement that a parameter be a configurable option. There are several different reasons behind such requirements. In a few cases, there is current uncertainty or disagreement about the best value, and it may be necessary to update the recommended value in the future. In other cases, the value really depends on external factors -- e.g., the size of the host and the distribution of its communication load, or the speeds and topology of nearby networks -- and self-tuning algorithms are unavailable and would probably be insufficient. In some cases, the configurability is needed because of observed administrative requirements. Finally, some configuration options are required to communicate with obsolete or incorrect implementations of the protocols, distributed without sources, that unfortunately persist in many parts of the Internet. To make correct systems coexist with these faulty systems, administrators often have to "mis- configure" the correct systems. This problem will correct itself gradually as the faulty systems are retired, but it cannot be ignored by vendors. When we say that a parameter must be configurable, we do not intend to require that its value be explicitly read from a configuration file at every boot time. We recommend that implementors set up a default for each parameter, so a Internet Engineering Task Force [Page 14] ***DRAFT RFC*** INTRODUCTION June 16, 1989 configuration file is only necessary to override those defaults that are inappropriate in a particular installation. Thus, the configurability requirement is an assurance that it will be POSSIBLE to override the default when necessary, even in a binary-only or ROM-based product. This document requires a particular value for such defaults in some cases. The choice of default is a sensitive issue when the configuration item controls the accommodation to existing faulty systems. If the Internet is to converge successfully to complete interoperability, the default values built into implementations must implement the official protocol, not "mis-configurations" to accommodate faulty implementations. Although marketing considerations have led some vendors to choose mis-configuration defaults, we urge vendors to choose defaults that will conform to the standard. Finally, we note that a vendor needs to provide adequate documentation on all configuration parameters, their limits and effects. 1.3 Reading this Document 1.3.1 Organization Protocol layering, which is generally used as an organizing principle in implementing network software, has also been used to organize this document. In describing the rules, we assume that an implementation does strictly mirror the layering of the protocols. Thus, the following three major sections specify the requirements for the link layer, the internet layer, and the transport layer, respectively. The companion RFC [INTRO:1] covers application level software. This layerist organization was chosen for simplicity and clarity. However, strict layering is an imperfect model, both for the protocol suite and for recommended implementation approaches. The layers of the protocols interact in complex and sometimes subtle ways, and particular functions often involve multiple layers. There are many design choices in an implementation, many of which involve creative "breaking" of strict layering. Every implementor is urged to read references [INTRO:7] and [INTRO:8]. In general, each major section is organized into the following subsections: Internet Engineering Task Force [Page 15] ***DRAFT RFC*** INTRODUCTION June 16, 1989 (1) Introduction (2) Protocol Walk-Through -- considers the protocol specification documents section-by-section, correcting errors, stating requirements that may be ambiguous or ill-defined, and providing further clarification or explanation. (3) Specific Issues -- discusses design and implementation issues in the protocols that were not included in the walk-through. (4) Interfaces -- discusses the service interface to the next higher layer. (5) Summary -- contains a summary of the requirements of the section. Under many of the individual topics in this document, there is parenthetical material labeled "DISCUSSION" or "IMPLEMENTATION." This material is intended to give clarification and explanation of the preceding requirements text. It also includes some suggestions on possible future directions or developments. The implementation material contains suggested approaches that an implementor may want to consider. 1.3.2 Requirements In this document, the words that are used to define the significance of each particular requirement are capitalized. These words are: * "MUST" This word or the adjective "REQUIRED" means that the item is an absolute requirement of the specification. * "SHOULD" This word or the adjective "RECOMMENDED" means that there may exist valid reasons in particular circumstances to ignore this item, but the full implications should be understood and the case carefully weighed before choosing a different course. * "MAY" Internet Engineering Task Force [Page 16] ***DRAFT RFC*** INTRODUCTION June 16, 1989 This word or the adjective "OPTIONAL" means that this item is truly optional. One vendor may choose to include the item because a particular marketplace requires it or because it enhances the product, for example; another vendor may omit the same item. An implementation is not compliant if it fails to satisfy one or more of the MUST requirements for the protocols it implements. An implementation that satisfies all the MUST and all the SHOULD requirements for its protocols is said to be "unconditionally compliant"; one that satisfies all the MUST requirements but not all the SHOULD requirements for its protocols is said to be "conditionally compliant". 1.3.3 Terminology This document uses the following technical terms: Segment A segment is the unit of end-to-end transmission in the TCP protocol. A segment consists of a TCP header followed by application data. A segment is transmitted as an IP datagram. Message In this description of the lower-layer protocols, a message is the unit of transmission in a transport layer protocol. It consists of a transport protocol header followed by application protocol data. To be transmitted end-to-end through the Internet, a message must be encapsulated inside a datagram. In particular, a TCP segment is a message. IP Datagram An IP datagram is the unit of end-to-end transmission in the IP protocol. An IP datagram consists of an IP header followed by transport layer data, i.e., of an IP header followed by a message. In the description of the internet layer (Section 3), the unqualified term "datagram" should be understood to refer to an IP datagram. Packet A packet is the unit of data passed across the interface between the internet layer and the link layer. It includes an IP header and data. A packet may be a Internet Engineering Task Force [Page 17] ***DRAFT RFC*** INTRODUCTION June 16, 1989 complete IP datagram or a fragment of an IP datagram. Frame A frame is the unit of transmission in a link layer protocol, and consists of a link-layer header followed by a packet. Connected Network A network to which a host is interfaced is often known as the "local network" or the "subnetwork" relative to that host. However, these terms can cause confusion, and therefore we use the term "connected network" in this document. Physical network interface This is a physical interface to a connected network and has a (possibly unique) link-layer address. Multiple physical network interfaces on a single host may share the same link-layer address, but the address must be unique for different hosts on the same physical network. Logical [network] interface A Logical [network] interface is a logical path to a connected network and is distinguished by a unique IP address. Multihomed A host is said to be multihomed if it has multiple logical interfaces, i.e., multiple IP addresses, on connected network(s). For more discussion of the logical/physical interface distinction and of multihoming, see Section 3.3.4 below. Path At a given moment, all the IP datagrams from a particular source host to a particular destination host will typically traverse the same sequence of gateways. We use the term "path" for this sequence. Note that a path is uni-directional; it is not unusual to have different paths in the two directions between a given host pair. @ MTU @ The maximum transmission unit, i.e., the size of the @ largest packet that can be transmitted. @ The terms frame, packet, datagram, message, and segment are illustrated by the following schematic diagrams: Internet Engineering Task Force [Page 18] ***DRAFT RFC*** INTRODUCTION June 16, 1989 A. Transmission on connected network: _______________________________________________ | LL hdr | IP hdr | (data) | |________|________|_____________________________| <---------- Frame -----------------------------> <----------Packet --------------------> B. Before IP fragmentation or after IP reassembly: ______________________________________ | IP hdr | transport| Application Data | |________|____hdr___|__________________| <-------- Datagram ------------------> <-------- Message -----------> or, for TCP: ______________________________________ | IP hdr | TCP hdr | Application Data | |________|__________|__________________| <-------- Datagram ------------------> <-------- Segment -----------> Internet Engineering Task Force [Page 19] ***DRAFT RFC*** LINK LAYER June 16, 1989 2. LINK LAYER 2.1 INTRODUCTION All Internet systems, both hosts and gateways, have the same requirements for link layer protocols. These requirements are given in Chapter 3 of "Requirements for Internet Gateways" [INTRO:2], with the addition of the material in this section. 2.2 PROTOCOL WALK-THROUGH None. 2.3 SPECIFIC ISSUES 2.3.1 Trailer Protocol Negotiation The trailer protocol [LINK:1] for link-level encapsulation MAY be used, but only if it has been verified that both systems (host or gateway) involved in the link-level communication implement trailers. If the system does not dynamically negotiate use of the trailer protocol on a per-destination basis, the default configuration MUST disable the protocol. DISCUSSION: The trailer protocol is a link-layer encapsulation technique that rearranges the data contents of packets sent on the physical network. In some cases, trailers improve the throughput of higher level protocols by reducing the amount of data copying within the operating system. Higher level protocols are unaware of trailer use, but both the sending and receiving host MUST understand the protocol if it is used. Improper use of trailers can result in very confusing symptoms. Only packets with specific size attributes are encapsulated using trailers, and typically only a small fraction of the packets being exchanged have these attributes. Thus, if a system using trailers exchanges packets with a system that does not, some packets disappear into a black hole while others are delivered successfully. IMPLEMENTATION: | On an Ethernet, packets encapsulated with trailers use a | distinct Ethernet type [LINK:1], and trailer negotiation | is performed at the time that ARP is used to discover the | link-layer address of a destination system. | Internet Engineering Task Force [Page 20] ***DRAFT RFC*** LINK LAYER June 16, 1989 Specifically, the ARP exchange is completed in the usual | manner using the normal IP protocol type, but a host that | wants to speak trailers will send an additional "trailer | ARP reply" packet, i.e., an ARP reply that specifies the | trailer encapsulation protocol type but otherwise has the | format of a normal ARP reply. If a host configured to use | trailers receives a trailer ARP reply message from a | remote machine, it can add that machine to the list of | machines that understand trailers, e.g., by marking the | corresponding entry in the ARP cache. | Hosts wishing to receive trailer encapsulations send | trailer ARP replies whenever they complete exchanges of | normal ARP messages for IP. Thus, a host that received an | ARP request for its IP protocol address would send a | trailer ARP reply in addition to the normal IP reply; a | host that sent the IP ARP request would send a trailer ARP | reply when it received the corresponding IP ARP reply. In | this way, either the requesting or responding host in an | IP ARP exchange may request that it receive trailer | encapsulations. | This scheme, using extra trailer ARP reply packets rather | than sending an ARP request for the trailer protocol type, | was designed to avoid a continuous exchange of ARP packets | with a misbehaving host that, contrary to any | specification or common sense, responded to an ARP reply | for trailers with another ARP reply for IP. This problem | is avoided by sending a trailer ARP reply in response to | an IP ARP reply only when the IP reply answers an | outstanding request; this is true when the hardware | address for the host is still unknown when the IP ARP | reply is received. A trailer ARP reply may always be sent | along with an IP ARP reply responding to an IP request. | 2.3.2 Address Resolution Protocol -- ARP 2.3.2.1 ARP Cache Validation An implementation of the Address Resolution Protocol (ARP) MUST provide a mechanism to flush out-of-date cache entries. If this mechanism involves a timeout, it SHOULD be possible to configure the timeout value. A mechanism to prevent ARP flooding (repeatedly sending an ARP Request for the same IP address, at a high rate) MUST be included. The recommended maximum rate is 1 per second per destination. Internet Engineering Task Force [Page 21] ***DRAFT RFC*** LINK LAYER June 16, 1989 DISCUSSION: The ARP specification [LINK:2] suggests but does not require a timeout mechanism to invalidate cache entries when hosts change their Ethernet addresses. The prevalence of proxy ARP (see Section 2.4 of [INTRO:1]) has significantly increased the likelihood that cache entries in hosts will become invalid, and therefore some cache-invalidation mechanism is now required for hosts. Even in the absence of proxy ARP, a long-period cache timeout is useful in order to automatically correct any bad ARP data that might have been cached. IMPLEMENTATION: Four mechanisms have been used, sometimes in combination, to flush out-of-date cache entries. (1) Timeout -- Periodically time out cache entries, even if they are in use. Note that this timeout should be restarted when the cache entry is "refreshed" (by observing the source fields, regardless of target address, of an ARP broadcast from the system in question). For proxy ARP situations, the timeout needs to be on the order of a minute. (2) Unicast Poll -- Actively poll the remote host by periodically sending a point-to-point ARP Request to it, and delete the entry if no ARP Reply is received from N successive polls. Again, the timeout should be on the order of a minute, and typically N is 2. (3) Link-Layer Advice -- If the link-layer driver detects a delivery problem, flush the corresponding ARP cache entry. (4) Higher-level Advice -- Provide a call from the Internet layer to the link layer to indicate a delivery problem. The effect of this call would be to invalidate the corresponding cache entry. This call would be analogous to the "ADVISE_DELIVPROB()" call from the transport layer to the Internet layer (see Section 3.4), and in fact the ADVISE_DELIVPROB routine would in turn call the link-layer advice routine to invalidate the cache entry. Internet Engineering Task Force [Page 22] ***DRAFT RFC*** LINK LAYER June 16, 1989 Approaches (1) and (2) involve ARP cache timeouts on the order of a minute or less. In the absence of proxy ARP, a timeout this short could create noticeable overhead traffic on a very large Ethernet. Therefore, it may be necessary to configure a host to lengthen the ARP cache timeout. 2.3.2.2 ARP Packet Queue The link layer SHOULD save (rather than discard) at least @ one (the latest) packet of each set of packets destined to @ the same unresolved IP address, and transmit the saved @ packet when the address has been resolved. @ 2.3.3 Ethernet and IEEE 802 Encapsulation ! The IP encapsulation for Ethernets is described in RFC-894 ! [LINK:3], while RFC-1042 [LINK:4] describes the IP ! encapsulation for IEEE 802 networks. RFC-1042 elaborates and ! replaces the discussion in Section 3.4 of [INTRO:1]. ! Every Internet host connected to a 10Mbps Ethernet cable: ! o MUST be able to send and receive packets using RFC-894 ! encapsulation; ! o SHOULD be able to receive RFC-1042 packets, intermixed ! with RFC-894 packets; and ! o MAY be able to send packets using RFC-1042 encapsulation. ! An Internet host that implements sending both the RFC-894 and ! the RFC-1042 encapsulations MUST provide a configuration switch ! to select which is sent, and this switch MUST default to RFC- ! 894. ! Note that the standard IP encapsulation in RFC-1042 does not ! use the protocol id value (K1=6) that IEEE reserved for IP; ! instead, it uses a value (K1=170) that implies an extension ! (the "SNAP") which can be used to hold the Ether-Type field. ! An Internet system MUST NOT send 802 packets using K1=6. ! Address translation from Internet addresses to link-level ! addresses on Ethernet and IEEE 802 networks MUST be managed by ! the Address Resolution Protocol (ARP). ! The MTU for an Ethernet is 1500 and for 802.3 is 1492. ! Internet Engineering Task Force [Page 23] ***DRAFT RFC*** LINK LAYER June 16, 1989 DISCUSSION: ! The IEEE 802.3 specification provides for operation over a ! 10Mbps Ethernet cable, in which case Ethernet and IEEE ! 802.3 frames can be physically intermixed. A receiver can ! distinguish Ethernet and 802.3 frames by the value of the ! 802.3 Length field; this two-octet field coincides in the ! header with the Ether-Type field of an Ethernet frame. In ! particular, the 802.3 Length field must be less than or ! equal to 1500, while all valid Ether-Type values are ! greater than 1500. ! Another compatibility problem arises with link-level ! broadcasts. A broadcast sent with one framing will not be ! seen by hosts that can receive only the other framing. ! The provisions of this section were designed to provide ! direct interoperation between 894-capable and 1042-capable ! systems on the same cable, to the maximum extent possible. ! It is intended to support the present situation where ! 894-only systems predominate, while providing an easy ! transition to a possible future in which 1042-capable ! systems become common. ! Note that there is no way that 894-only systems can ! interoperate directly with 1042-only systems; they could ! only communicate as different logical networks on the same ! cable, through a special bridge box that transformed from ! one frame format to the other. Furthermore, it is not ! useful or even possible for a dual-format host to discover ! automatically which format to send, because of the problem ! of link-layer broadcasts. ! 2.4 LINK/INTERNET LAYER INTERFACE The packet receive interface between the IP layer and the link layer MUST include a flag to indicate whether the incoming packet was addressed to a link-layer broadcast address. DISCUSSION Although the IP layer does not generally know link layer addresses (since every different network medium generally has a different address format), the broadcast address on a broadcast-capable medium is an important special case. See Section 3.2.2, especially the DISCUSSION concerning broadcast storms. The packet send interface between the IP and link layers MUST include a flag to indicate whether the packet is to be sent with a Internet Engineering Task Force [Page 24] ***DRAFT RFC*** LINK LAYER June 16, 1989 link-layer broadcast address, and also the 5-bit TOS field (see Section 3.2.1.6). The link layer MUST NOT report a Destination Unreachable error to | IP solely because there is no ARP cache entry for a destination. | 2.5 LINK LAYER REQUIREMENTS SUMMARY | | | | |S| | | | | | |H| |F | | | | |O|M|o | | |S| |U|U|o | | |H| |L|S|t | |M|O| |D|T|n | |U|U|M| | |o | |S|L|A|N|N|t | |T|D|Y|O|O|t FEATURE |SECTION| | | |T|T|e --------------------------------------------------|-------|-|-|-|-|-|-- | | | | | | | Trailer encapsulation |2.3.1 | | |x| | | Send Trailers by default without negotiation |2.3.1 | | | | |x| ARP |2.3.2 | | | | | | Flush out-of-date ARP cache entries |2.3.2.1|x| | | | | Prevent ARP floods |2.3.2.1|x| | | | | Configurable cache timeout |2.3.2.1| |x| | | | Save at least one (latest) unresolved pkt |2.3.2.2| |x| | | | Ethernet and IEEE 802 Encapsulation |2.3.3 | | | | | | Ethernet host able to: |2.3.3 | | | | | | Send & receive RFC-894 encapsulation |2.3.3 |x| | | | | Receive RFC-1042 encapsulation |2.3.3 | |x| | | | SendRFC-1042 encapsulation |2.3.3 | | |x| | | Then config. sw. to select, RFC-894 dflt |2.3.3 |x| | | | | Send K1=6 encapsulation |2.3.3 | | | | |x| Use ARP on Ethernet and IEEE 802 nets |2.3.3 |x| | | | | Link-layer report b'casts to IP layer |2.4 |x| | | | | Internet Engineering Task Force [Page 25] ***DRAFT RFC*** INTERNET LAYER June 16, 1989 3. INTERNET LAYER PROTOCOLS 3.1 INTRODUCTION The IP layer of host software MUST implement both the Internet Protocol IP and the Internet Control Message Protocol ICMP. RFC- 791 [IP:1] defines the IP protocol and gives an introduction to the architecture of the Internet. RFC-792 [IP:2] defines ICMP, which provides routing, diagnostic and error functionality to IP. Although ICMP messages are encapsulated within IP datagrams, ICMP processing is considered to be (and is typically implemented as) part of the IP layer. RFC-950 [IP:3] defines the mandatory subnet extension to the addressing architecture. RFC-1054 [IP:4], describing an extension to provide Internet-wide multicasting at the IP level, is currently a Draft Internet Standard. The target of an IP multicast may be an arbitrary group of Internet hosts. This facility provides a natural extension of the multicasting facility on particular networks, and also provides a standard way to access these local facilities. Many services that currently make use of broadcast are likely to be redefined to use IP multicast. Implementors are urged to read RFC-1054 and to include at least the "hooks" necessary to implement Internet multicasting. Other important references are listed in Section 5 of this document. The host IP layer has only two basic functions: (1) choose the "next hop" gateway or host for outgoing datagrams and (2) do IP reassembly of incoming datagrams. The IP layer may also (3) implement fragmentation of outgoing datagrams. Finally, the IP layer must include a small amount of (4) diagnostic and error functionality. We expect that IP layer functions may increase somewhat in the future, as further network control and management is developed. For normal datagrams, the processing is straightforward. For incoming datagrams, the IP layer: (1) verifies that the datagram is correctly formatted; (2) verifies that it is addressed to the local host, to a broadcast address, or to a multicast group that includes the local host; (3) processes options; Internet Engineering Task Force [Page 26] ***DRAFT RFC*** INTERNET LAYER June 16, 1989 (4) reassembles the datagram if necessary; and (5) passes the encapsulated message to the appropriate transport-layer protocol module. For outgoing datagrams, the IP layer: (6) sets any fields not set by the transport layer; (7) selects the correct first hop on the connected network; (8) fragments the datagram if necessary and if intentional fragmentation is implemented (see Section 3.3.3); and (9) passes the packet(s) to the appropriate link-layer module. Note that the IP layer in a host accepts datagrams only to process them within that host, and sends only datagrams that were constructed locally or are being source-routed through the host. Any implementation that is prepared to forward datagrams generated by another host is acting as a gateway and MUST meet the specifications laid out in the gateway requirements RFC [INTRO:1]. In the following, the action specified in certain cases for a ! received datagram is to "silently ignore". By "silently", we mean ! that the host will not send any ICMP error message (see Section ! 3.2.2) as a result. However, for diagnosis of problems a host ! SHOULD provide the capability of logging the error (see Section ! 1.2.3), including the contents of the silently-ignored datagram, ! and SHOULD record the event in a statistics counter. ! DISCUSSION: This silence about errors is generally intended to prevent "broadcast storms" on broadcast LAN's. 3.2 PROTOCOL WALK-THROUGH 3.2.1 Internet Protocol -- IP 3.2.1.1 Version Number: RFC-791 Section 3.1 A datagram whose version number is not 4 MUST be silently ignored. Internet Engineering Task Force [Page 27] ***DRAFT RFC*** INTERNET LAYER June 16, 1989 3.2.1.2 Checksum: RFC-791 Section 3.1 A host MUST verify the IP header checksum on every received datagram and silently ignore every datagram that has a bad checksum. 3.2.1.3 Addressing: RFC-791 Section 3.2 There are now five classes of IP addresses: Class A through Class E. Class D addresses are used for IP multicasting [IP:4], while Class E addresses are reserved for experimental use. We now summarize the important special cases for IP addresses, using the following notation for an IP address: { <Network-number>, <Host-number> } or { <Network-number>, <Subnet-number>, <Host-number> } and the notation "-1" for a field that contains all 1 bits. This notation is not intended to imply that the 1-bits in an Address Mask need be contiguous. (a) { 0, 0 } This host on this network. MUST NOT be sent, except as a source address as part of an initialization procedure by which the host learns its own IP address. See also Section 3.3.6 for a non-standard use of {0,0}. (b) { 0, <Host-number> } Specified host on this network. MUST NOT be sent, except as a source address as part of an initialization procedure by which the host learns its full IP address. (c) { -1, -1 } Limited broadcast. MUST NOT be used as a source address. A datagram with this destination address will be received by every host on the connected physical network but will not be forwarded outside that network. Internet Engineering Task Force [Page 28] ***DRAFT RFC*** INTERNET LAYER June 16, 1989 (d) { <Network-number>, -1 } Directed broadcast to the specified network. MUST NOT be used as a source address. (e) { <Network-number>, <Subnet-number>, -1 } Directed broadcast to the specified subnet. MUST NOT be used as a source address. (f) { <Network-number>, -1, -1 } Directed broadcast to all subnets of the specified subnetted network. MUST NOT be used as a source address. (g) { 127, <any> } Internal host loopback address. Addresses of this form MUST NOT appear outside a host. The <Network-number> is administratively assigned by the Internet numbering authority so that its value will be unique in the entire world. IP addresses must MUST NOT have the value 0 or -1 for any of | the <Host-number>, <Network-number>, or <Subnet-number> | fields (except in the special cases listed above). This | implies that each of these fields will be at least two bits | long. | For further discussion of broadcast addresses, see Section 3.3.6. A host MUST support the subnet extensions to IP [IP:3]. As | a result, there will be an Address Mask of the form: | {-1, -1, 0} associated with each of the host's local IP | addresses; see Sections 3.2.2.9 and 3.3.1.1. A host MUST silently ignore an incoming datagram that is not addressed to it, i.e., whose destination address is not either (1) a local IP address for the host, (2) an IP broadcast address valid for the host, or (3) a multicast address for a group of which the host is a member. A host MAY silently ignore an incoming datagram whose ! destination address does not correspond to the physical ! interface through which it is received. ! Internet Engineering Task Force [Page 29] ***DRAFT RFC*** INTERNET LAYER June 16, 1989 A host MUST silently ignore an incoming datagram containing ! an invalid IP source address; this validation could be done ! in either the IP layer or by each protocol in the transport ! layer. ! DISCUSSION: A mis-addressed datagram might be caused by a) a link- layer broadcast of a unicast datagram, b) a gateway being confused, or c) another host being confused. Note that normal Internet routing mechanisms could not ! route a datagram to a physical interface that did not ! correspond to the destination address. ! An architectural goal for Internet host software was to @ allow IP addresses to be featureless 32-bit numbers, @ avoiding algorithms that required a knowledge of the IP @ address format. Otherwise, any future change in the @ format or interpretation of IP addresses will require @ host software changes. However, validation of @ broadcast and multicast addresses violates this goal; a @ few other violations are described elsewhere in this @ document. @ 3.2.1.4 Fragmentation and Reassembly: RFC-791 Section 3.2 The Internet model requires that every host MUST support reassembly. For further discussion of these topics, see Sections 3.3.2 and 3.3.3 below. 3.2.1.5 Identification: RFC-791 Section 3.2 When sending an identical copy of an earlier datagram, a host MAY optionally retain the same Identification field in the copy. DISCUSSION: Some Internet protocol experts have maintained that when a host sends an identical copy of an earlier datagram, the new copy should contain the same Identification value as the original. There are two suggested advantages: (1) if the datagrams are fragmented and some of the fragments are lost, the receiver may be able to reconstruct a complete datagram from fragments of the original and the copies; (2) a congested gateway might use the IP Identification field (and Fragment Offset) to discard duplicate datagrams from the queue. Internet Engineering Task Force [Page 30] ***DRAFT RFC*** INTERNET LAYER June 16, 1989 However, the observed patterns of datagram loss in the Internet do not favor the probability of retransmitted fragments filling in reassembly gaps, while other mechanisms (e.g., TCP repacketizing upon retransmission) tend to prevent retransmission of an identical datagram [IP:9]. Therefore, many believe that retransmitting the same Identification field is not useful. Also, a connectionless transport protocol like UDP would require the cooperation of the application programs to retain the same Identification value in identical datagrams. 3.2.1.6 Type-of-Service: RFC-791 Section 3.2 The "Type-of-Service" byte in the IP header is divided into two sections: the Precedence field (high-order 3 bits), and a field that is customarily called "Type-of-Service" or "TOS" (low-order 5 bits). In this document, all references to "TOS" or the "TOS field" refer to the low-order 5 bits only. The Precedence field is intended for Department of Defense applications of the Internet protocols, and is outside the scope of this document and the IP standard specification. Vendors should consult the Defense Communication Agency (DCA) for guidance on the IP Precedence field and its implications for other protocol layers. The IP layer MUST provide a means for the transport layer to set the TOS field of every datagram that is sent; the default is all zero bits. The IP layer SHOULD pass received TOS values up to the transport layer. The particular link-layer mappings of TOS contained in RFC- 795 SHOULD NOT be implemented. DISCUSSION: While the TOS field has been little used in the past, it is expected to play an increasing role in the near future. The TOS field is expected to be used to control two aspects of gateway operations: routing and queueing algorithms. See Section 2 of [INTRO:1] for the requirements on application programs to specify TOS values. The TOS field may also be mapped into link-layer service selectors. This has been applied to provide effective sharing of serial lines by different classes Internet Engineering Task Force [Page 31] ***DRAFT RFC*** INTERNET LAYER June 16, 1989 of TCP traffic, for example. However, the mappings suggested in RFC-795 for networks that were included in the Internet as of 1981 are now obsolete. 3.2.1.7 Time-to-Live: RFC-791 Section 3.2 A host MUST NOT send a datagram with a Time-to-Live (TTL) value of zero. A host MUST NOT discard a datagram just because it was received with TTL less than 2. The exception is a datagram that is being forwarded because of source routing, in which case the host MUST follow the gateway rules for TTL [INTRO:1]. The IP layer MUST provide a means for the transport layer to | set the TTL field of every datagram that is sent. This will | allow a higher-layer protocol to implement an "expanding | scope" search for some Internet resource; this is expected | to be useful to locate the "nearest" server of a given | class, using IP multicasting, for example. | When a fixed TTL value is used, its value MUST be | configurable. | DISCUSSION: The TTL field has two functions: limit the lifetime of TCP segments (see RFC-793 [TCP:1], p. 28), and reduce the Internet traffic caused by routing loops. Although TTL is a time in seconds, it also has some attributes of a hop-count, since each gateway is required to reduce the TTL field by at least one. The default value must be at least big enough for the Internet "diameter," i.e., the longest possible path. A reasonable value is about twice the diameter, to allow for continued Internet growth. The current suggested value will be published in the Assigned Numbers RFC [INTRO:5]. 3.2.1.8 Options: RFC-791 Section 3.2 There MUST be a means for the transport layer to specify IP options to be included in transmitted IP datagrams (see Section 3.4). All IP options that are received (except NOP or END-OF-LIST) MUST be passed to the transport layer or to ICMP processing (when the datagram is an ICMP message). Later sections of Internet Engineering Task Force [Page 32] ***DRAFT RFC*** INTERNET LAYER June 16, 1989 this document discuss specific IP option support required by each of ICMP, TCP, and UDP. The IP and transport layer MUST each interpret those IP options that they understand. DISCUSSION: @ Passing all received IP options to the transport layer @ is a deliberate "violation of strict layering" that is @ designed to ease the introduction of new transport- @ relevant IP options in the future. Each layer must @ pick out any options that are relevant to its own @ processing and ignore the rest. For this purpose, @ every IP option except NOP and END-OF-LIST will include @ a specification of its own length. @ IMPLEMENTATION: The IP layer must not crash as the result of an option length that is outside the possible range. For example, erroneous zero option lengths have been observed to put some IP implementations into infinite loops. Here are the requirements for specific IP options: (1) Security Option Some environments may require the Security option in every datagram; such a requirement is outside the scope of this document and the IP standard specification. Note, however, that the security option described in RFC-791 is obsolete. Vendors should consult the Defense Communication Agency (DCA) for guidance. (21) Stream Identifier Option This option is obsolete; it MUST NOT be sent, and it MUST be silently ignored if received. (3) Source Route Options A host MUST support originating a source route and MUST @ be able to act as the final destination of a source @ route. @ Subject to restrictions given below, a host MAY support @ being an intermediate address in a source route, @ forwarding the source-routed datagram to the next @ Internet Engineering Task Force [Page 33] ***DRAFT RFC*** INTERNET LAYER June 16, 1989 specified hop; however, in performing this gateway-like @ function, the host MUST obey all the relevant rules for @ a gateway forwarding source-routed datagrams [INTRO:1]; @ this includes updating the source route and the TTL @ fields. @ If a datagram is received with a source route that is @ completed (i.e., the pointer points beyond the last @ field), the datagram has reached its final destination; @ the option as received (the recorded route) MUST be @ passed up to ICMP or the transport layer. This | recorded route will be reversed and used to form a | return source route for reply datagrams (see discussion | of Source Route Options in Section 4). When a return | source route is built, it MUST be correctly formed even | if the recorded route included the source host (see | case (B) in following Discussion). | An IP header containing more than one Source Route # option MUST NOT be sent; the effect on routing of # multiple Source Route options is implementation- # specific. # To define the rules restricting host forwarding of # source-routed datagrams, we use the term "local # source-routing" if the next hop will be through the # same logical interface through which the datagram # arrived; otherwise, it is "non-local source-routing". # o A host is permitted to perform local source- # routing without restriction. # o There MUST be a configurable switch to disable # non-local source-routing, and this switch MUST # default to no forwarding. # o The host MUST satisfy all gateway requirements for # configurable policy filters [INTRO:1] restricting # non-local forwarding. # If a host receives a datagram with an incomplete source route but does not forward it for some reason, the host SHOULD return an ICMP Destination Unreachable (code 5, Source Route Failed) message, unless the datagram was itself an ICMP error message. DISCUSSION: If a source-routed datagram is fragmented, each Internet Engineering Task Force [Page 34] ***DRAFT RFC*** INTERNET LAYER June 16, 1989 fragment will contain a copy of the source route. Since processing IP options (including a source route) must precede reassembly, the original datagram will not be reassembled until the final destination is reached. Suppose a source routed datagram is to be routed from host S to host D via gateways G1, G2, ... Gn. There was an ambiguity in the specification over whether the source route option in a datagram sent out by S should be (A) or (B): (A): {>>G2, G3, ... Gn, D} <--- CORRECT (B): {S, >>G2, G3, ... Gn, D} <---- WRONG (where >> represents the pointer). If (A) is sent, the datagram received at D will contain the option: {G1, G2, ... Gn >>}, with S and D as the IP source and destination addresses. If (B) were sent, the datagram received at D would again contain S and D as the same IP source and destination addresses, but the option would be: {S, G1, ...Gn >>}; i.e., the originating host would be the first hop in the route. Since there are implementations that use the erroneous case (B), a host must be prepared to receive and build a non-redundant return route for this case. There is concern about the use of host source- | routing for circumventing Internet access | restrictions. Administrators are urged to enable | off-network source routing in a host only in | special circumstances and then only with vigilant | management oversight. IMPLEMENTATION: Some implementations reverse the order of the elements of an completed source route, i.e., form a return route, before passing the option to the higher layer. This avoids having similar code in many higher-layer modules. (4) Record Route Option Internet Engineering Task Force [Page 35] ***DRAFT RFC*** INTERNET LAYER June 16, 1989 Implementation of sending and receiving the Record ! Route option is OPTIONAL. ! (5) Timestamp Option Implementation of sending and receiving the Timestamp option is OPTIONAL. If it is implemented, the following rules apply: o The originating host SHOULD NOT record a timestamp ! in a Timestamp option, unless its interface ! address is explicitly specified in the first slot ! of the option. However, a host that is forwarding | a source-routed datagram MUST (if possible) add | the current timestamp to a Timestamp option in the | datagram. | o The destination host MUST (if possible) add the current timestamp to a Timestamp option before passing the option to the transport layer or to ICMP for processing. o A timestamp value MUST follow the rules given below for the ICMP Timestamp message. 3.2.2 Internet Control Message Protocol -- ICMP ICMP messages are grouped into two classes. * ICMP error messages: Destination Unreachable (see Section 3.2.2.1) Redirect (see Section 3.2.2.2) Source Quench (see Section 3.2.2.3) Time Exceeded (see Section 3.2.2.4) Parameter Problem (see Section 3.2.2.5) * ICMP query messages: Echo (see Section 3.2.2.6) Information (see Section 3.2.2.7) Timestamp (see Section 3.2.2.8) Address Mask (see Section 3.2.2.9) Internet Engineering Task Force [Page 36] ***DRAFT RFC*** INTERNET LAYER June 16, 1989 If an ICMP message of unknown type is received, it MUST be silently ignored. Every ICMP error message includes the Internet header and at least the first 8 data octets of the datagram that triggered the error. In those cases where the Internet layer is required to pass an ICMP error message to the transport layer, the IP protocol number MUST be extracted from the original header and used to select the appropriate transport protocol entity to handle the error. An ICMP error message SHOULD be sent with normal (i.e., zero) TOS bits. An ICMP error message MUST never be sent as the result of receiving: * an ICMP error message, or * a datagram destined to an IP broadcast or multicast address, or | * a datagram sent as a link-layer broadcast, or * a non-initial fragment, or * a datagram whose source address does not define a single host -- e.g., a zero address, a loopback address, a broadcast address, or a multicast address. NOTE: THESE RESTRICTIONS TAKE PRECEDENCE OVER ANY REQUIREMENT ELSEWHERE IN THIS DOCUMENT FOR SENDING ICMP ERROR MESSAGES. DISCUSSION: These rules will prevent the "broadcast storms" that have resulted from hosts returning ICMP error messages in response to broadcast datagrams. For example, a broadcast UDP segment to a non-existent port could trigger a flood of ICMP Destination Unreachable datagrams from all machines that do not have a client for that destination port. On a large Ethernet, the resulting collisions can render the network useless for a second or more. Every datagram that is broadcast on the connected network should have a valid IP broadcast address as its IP destination (see Section 3.3.6). However, some hosts violate this rule. To be certain to detect broadcast datagrams, therefore, hosts are required to check for a Internet Engineering Task Force [Page 37] ***DRAFT RFC*** INTERNET LAYER June 16, 1989 link-layer broadcast as well as an IP-layer broadcast address. IMPLEMENTATION: This requires that the link layer inform the IP layer when a link-layer broadcast datagram has been received; see Section 2.4. 3.2.2.1 Destination Unreachable: RFC-792 A host SHOULD generate Destination Unreachable messages with code: 1 (Host Unreachable) when a source-routed datagram cannot be forwarded because of a routing problem; 2 (Protocol Unreachable) when the designated transport protocol is not supported; 3 (Port Unreachable) when the designated transport protocol (e.g., UDP) is unable to demultiplex the datagram but has no protocol mechanism to inform the sender; 4 (Fragmentation Required but DF Set) when a source- routed datagram cannot be fragmented to fit into the target network; # 5 (Bad Source Route) when a source-routed datagram cannot # be forwarded, e.g., because of a routing problem or # because the next hop of a strict source route is not on # a connected network. # A Destination Unreachable message that is received MUST be reported to the transport layer. A Destination Unreachable message that is received with code # 0 (Net), 1 (Host), or 5 (Bad Source Route) may result from a # routing transient and MUST therefore be interpreted as only a hint, not proof, that the specified destination is unreachable [IP:11]. For example, it MUST NOT be used as proof of a dead gateway (see Section 3.3.1). DISCUSSION: The transport layer MUST use the information appropriately; see Sections 4.1.3.3, 4.2.3.9, and 4.2.4 below. A transport protocol that uses its own Internet Engineering Task Force [Page 38] ***DRAFT RFC*** INTERNET LAYER June 16, 1989 mechanism (e.g., TCP RST segments) for notifying the sender that a port is unreachable MUST nevertheless accept an ICMP Port Unreachable for the same purpose. 3.2.2.2 Redirect: RFC-792 Redirect messages are generated only by gateways. A host receiving a Redirect message MUST update its routing information accordingly. Every host MUST be prepared to accept both host and network Redirects and to process them as described in Section 3.3.1.2 below. A Redirect message SHOULD be ignored silently if the new gateway address it specifies is not on the same connected (sub-) net through which the Redirect arrived [INTRO:1, Appendix A], or if the source of the Redirect is not the | current route to the specified destination (see Section | 3.3.1). | 3.2.2.3 Source Quench: RFC-792 A host MAY send a Source Quench message if it is approaching, or has reached, the point at which it is forced to discard incoming datagrams due to a shortage of reassembly buffers or other resources. See Section 2.2.3 of [INTRO:1] for suggestions on when to send Source Quench. If a Source Quench message is received, the IP layer MUST report it to the transport layer. In general, the transport or application layer SHOULD implement a mechanism to respond to Source Quench for any protocol that can send a sequence of datagrams to the same destination and which can reasonably be expected to maintain enough state information to make this feasible. See Section 4 for the handling of Source Quench by TCP and UDP. DISCUSSION: A Source Quench may be generated by the target host or by some gateway in the path of a datagram. The host receiving a Source Quench should throttle itself back for a period of time, then gradually increase the transmission rate again. The mechanism to respond to Source Quench may be in the transport layer (for connection-oriented protocols like TCP) or in the application layer (for protocols that are built on top of UDP). Internet Engineering Task Force [Page 39] ***DRAFT RFC*** INTERNET LAYER June 16, 1989 Although a mechanism has been proposed [IP:14] to make the IP layer respond directly to Source Quench by controlling the rate at which datagrams are sent, this proposal is controversial and untried. Therefore, the higher-layer protocol mechanisms just described are required. 3.2.2.4 Time Exceeded: RFC-792 An incoming Time Exceeded message with code 0 (In Transit) MUST be passed to the transport layer. An incoming Time Exceeded message with code 1 (Reassembly Timeout) SHOULD be silently ignored. DISCUSSION: A gateway will send a Time Exceeded (In Transit) message when it discards a datagram due to an expired TTL field. This indicates either a gateway routing loop or that the initial TTL value was too small. A host may receive a Time Exceeded (Reassembly Timeout) message from a destination host that has timed out and discarded an incomplete datagram; see Section 3.3.2 below. In the future, receipt of this message might @ trigger some "MTU discovery" procedure, to discover the @ maximum datagram size that can be sent on the path @ without fragmentation. @ 3.2.2.5 Parameter Problem: RFC-792 A host SHOULD generate Parameter Problem messages. An incoming Parameter Problem message MUST be passed to the transport layer and it MAY be reported to the user. DISCUSSION: The ICMP Parameter Problem message is sent to the source host for any problem not specifically covered by another ICMP message. Receipt of a Parameter Problem message generally indicates some local or remote implementation error. 3.2.2.6 Echo Request/Reply: RFC-792 Every host MUST implement an ICMP Echo server function that receives Echo Requests and sends corresponding Echo Replies. A host SHOULD also implement an application-layer interface for sending an Echo Request and receiving an Echo Reply, for diagnostic purposes. Internet Engineering Task Force [Page 40] ***DRAFT RFC*** INTERNET LAYER June 16, 1989 An ICMP Echo Request to an IP broadcast or multicast address | MAY be silently ignored. | The IP source address in an ICMP Echo Reply MUST be the same | as the IP destination address of the corresponding ICMP Echo | Request message. | Data received in an ICMP Echo Request MUST be entirely included in the resulting Echo Reply. However, if sending the Echo Reply requires intentional fragmentation that is not implemented, the datagram MUST be truncated to maximum transmission size (see Section 3.3.3) and sent. Echo Reply messages MUST be passed up to the ICMP user interface. If a Record Route and/or Time Stamp option is received in an ICMP Echo Request, this option (these options) SHOULD be updated to include the current host and included in the IP ! header of the Echo Reply message, without "truncation". ! Thus, the recorded route will be for the entire round trip. ! If a Source Route option is received in an ICMP Echo Request, the return route MUST be reversed and used as a Source Route option in the Echo Reply message. 3.2.2.7 Information Request/Reply: RFC-792 A host SHOULD NOT not implement these messages. DISCUSSION: The Information Request/Reply pair was intended to support self-configuring systems such as diskless workstations, to allow them to discover their IP network numbers at boot time. However, the RARP and BOOTP protocols provide better mechanisms for a host to discover its own IP address. 3.2.2.8 Timestamp and Timestamp Reply: RFC-792 A host MAY implement Timestamp and Timestamp Reply. If they are implemented, the following rules MUST be followed. * The ICMP Timestamp server function returns a Timestamp Reply to every Timestamp message that is received. If this function is implemented, it SHOULD be designed for minimum variability in delay (e.g., implemented in the kernel to avoid delay in scheduling a user process). Internet Engineering Task Force [Page 41] ***DRAFT RFC*** INTERNET LAYER June 16, 1989 * The following cases for Timestamp are to be handled according to the corresponding rules for Echo (Section 3.2.2.6): - Timestamp message to broadcast or multicast address. - Timestamp message to a multihomed host. - Source-routed Timestamp message. | - A Timestamp message containing a Timestamp option | or a Record-Route option. | * Incoming Timestamp Reply messages MUST be passed up to the ICMP user interface. * The preferred form for a timestamp value (the "standard value") is in units of milliseconds since midnight Universal Time. However, it may be difficult to provide this value with millisecond resolution. For example, many systems use clocks that update only at line frequency, 50 or 60 times per second. Therefore, some latitude is allowed in a "standard value": o A "standard value" be updated at least 15 times | per second (i.e., at most the six low-order bits | of the value may be undefined). | o The accuracy of a "standard value" MUST | approximate that of operator-set CPU clocks, i.e., | correct within a few minutes. | 3.2.2.9 Address Mask Request/Reply: RFC-950 A host MUST support the first, and MAY implement all three, of the following methods for determining the address mask(s) for its logical interface(s): (1) static configuration information; (2) sending ICMP Address Mask Request and receiving ICMP Address Mask Reply messages; and (3) obtaining the address mask dynamically as a side-effect Internet Engineering Task Force [Page 42] ***DRAFT RFC*** INTERNET LAYER June 16, 1989 of the system initialization process (see [INTRO:1]). The choice of method to be used in a particular host MUST be configurable for each logical interface. When method (2), the use of Address Mask messages, is enabled for a particular interface, then: (a) When it initializes, the host MUST broadcast an Address Mask Request message on the connected network. It MUST retransmit this message a small number of times if it does not receive an immediate Address Mask Reply. (b) Until it has received an Address Mask Reply, the host SHOULD assume an all-zero address mask (i.e., pretend that all possible destinations are on the local net). (c) The first Address Mask Reply message received MUST be used to set the address mask for the logical interface. This is true even if the first Address Mask Reply message is "unsolicited", in which case it will have been broadcast and may arrive after the host has ceased to retransmit Address Mask Requests. Once the mask has been set by an Address Mask Reply, later Address Mask Reply messages MUST be ignored. Conversely, if Address Mask messages are disabled for an interface, then no ICMP Address Mask Requests will be sent on that interface, and any ICMP Address Mask Replies received on that interface MUST be ignored. A host SHOULD make at least the following "sanity check" on | any address mask it installs: the mask MUST NOT be all 1 | bits, and it MUST be either zero or else the 8 highest-order | bits MUST be on. | A system MUST NOT send an Address Mask Reply unless it is an authoritative agent for address masks. An address mask received via an Address Mask Reply does not give the receiver authority and MUST NOT be used as the basis for issuing Address Mask Replies. With a statically configured address mask, there SHOULD be | an additional configuration flag that determines whether the | host is to be considered authoritative with this mask, i.e., | whether it will itself answer Address Mask Request messages. | Internet Engineering Task Force [Page 43] ***DRAFT RFC*** INTERNET LAYER June 16, 1989 See "System Initialization" in [INTRO:1] for more information about the use of Address Mask Request/Reply messages. DISCUSSION Hosts that casually send Address Mask Replies with invalid address masks have often been a serious nuisance. To prevent this, Address Mask Replies ought to be sent only by authoritative servers which have been selected by explicit administrative action. When an authoritative agent receives an Address Mask Request message, it will send a unicast Address Mask Reply to the host with the address mask of the corresponding network interface. If the network part of the host address is zero (see (a) and (b) in 3.2.1.3), the Reply will be broadcast. Of course, agents MUST carefully avoid sending spurious mask information at any time. There have been serious problems with systems sending incorrect Address Mask Replies, often because they have sent the Reply before they have finished loading their own configuration information. Getting no reply to its Address Mask Request messages, a host will assume there is no agent and use mask zero, when the agent may be only temporarily unreachable. An agent will broadcast an unsolicited Address Mask Reply whenever it initializes; this SHOULD update the masks of all hosts that have initialized in the meantime. The requirement to use a default address mask of zero differs from the suggestion in RFC-950 [IP:3] that the default mask SHOULD correspond to the network part of the address. One advantage of using a zero mask is that it avoids requiring the IP layer to parse its own address into class A, B or C. 3.3 SPECIFIC ISSUES The general principle: "Be liberal in what you accept, and conservative in what you send" is particularly important in the IP layer, where one misbehaving host can deny Internet service to many other hosts. Many of the following rules were learned from disasters, many of which would have been avoided if these "good citizen" principles had been followed. Internet Engineering Task Force [Page 44] ***DRAFT RFC*** INTERNET LAYER June 16, 1989 3.3.1 Routing Outbound Datagrams The IP layer chooses the correct next hop for each datagram it sends; this process is called "routing." If the destination is on a connected network, the datagram is sent directly to the destination host; otherwise, it has to be routed to a gateway on the connected network. 3.3.1.1 Local/Remote Decision To decide if the destination is on a connected network, the following algorithm MUST be used [see IP:3]: (a) The "Address Mask" for a logical interface is a 32-bit @ mask that selects the network number and subnet number @ fields of the corresponding IP address. @ (b) If the IP destination address bits extracted by the @ Address Mask matches IP source address bits extracted @ by the same mask, then the destination is on the @ corresponding connected network, and the datagram is to @ be transmitted through that directly to the destination @ host. (c) If not, then the destination is accessible only through a gateway. Selection of a gateway is described below (3.3.1.2). A datagram whose destination is a broadcast or multicast address (see [IP:4]) MUST be handled specially: * For a limited broadcast or a multicast address, simply pass the datagram to the link layer; no routing is necessary. * For a (network or subnet) directed broadcast, the datagram can use the standard routing algorithms that we are describing, with host number -1. 3.3.1.2 Gateway Selection To efficiently route a series of datagrams to the same destination, the source host MUST keep a "route cache" of mappings to next-hop gateways. A host MUST use the following basic algorithm to route a datagram destined to a remote network; this algorithm is designed to put the Internet Engineering Task Force [Page 45] ***DRAFT RFC*** INTERNET LAYER June 16, 1989 primary routing burden on the gateways [IP:11]: (a) If the route cache contains no information for a particular destination, the host chooses a "default" gateway and sends the datagram to it. It also builds a corresponding Route Cache entry. (c) If that gateway is not the best next hop to the destination, the gateway will forward the datagram to the best next-hop gateway and return an ICMP Redirect message to the source host. (d) When it receives a Redirect, the host will update the next-hop gateway in the appropriate routing cache entry, so later datagrams to the same destination will go directly to the best gateway. It is recommended that a host perform a "sanity" check on an ICMP Redirect before applying it; see Section 3.2.2.2. Since the subnet mask appropriate to the destination address is generally not known, a Network Redirect message SHOULD be treated identically to a Host Redirect message: the cache entry for the destination host (only) SHOULD be updated with the new gateway. When there is no route cache entry for the destination host address (and the destination is not on the connected network), the IP layer picks a gateway from its list of "default" gateways. The IP layer MUST support multiple default gateways. As an extra feature, a host IP layer MAY implement a table of "static routes." Static routes would be set up by system administrators to override the normal automatic routing mechanism, to handle exceptional situations. DISCUSSION: A host generally needs to know at least one default gateway to get started. This information can be obtained from a configuration file or else from the host startup sequence, e.g., the BOOTP protocol (see [INTRO:1]). It has been suggested that a host can augment its list of default gateways by recording any new gateways it learns about. For example, it can record every gateway to which it is ever redirected. Such a feature, while Internet Engineering Task Force [Page 46] ***DRAFT RFC*** INTERNET LAYER June 16, 1989 possibly useful in some circumstances, may cause problems in other cases; gateways are not all equal. A static route is typically a particular preset mapping from destination host or network into a particular next-hop gateway; it might also depend on the logical network interface and the Type-of-Service (see next section). Each route may include a flag specifying whether it may be overridden by ICMP Redirects. 3.3.1.3 Route Cache Each route cache entry MUST include the following fields: # (1) Destination IP address # (2) Type(s)-of-Service # (3) Next-hop gateway IP address # (4) Control information # Fields (1) and (2) form the argument used to retrieve the # cache entry containing the required gateway address (3). We # RECOMMEND that field (1) be the full IP address of the # destination host, not the destination network. In any case, # the value of field MUST be used as a featureless 32-bit # number in this match. # Field (4), the control information, is needed to choose an # entry for replacement. This might take the form of a # "recently used" bit, a use count, or a last-used timestamp, # for example. It is RECOMMENDed that it include the time of # last modification of the entry, for diagnostic purposes. # The cache SHOULD be large enough to include entries for the maximum number of destination hosts that may be in use at one time. DISCUSSION: A route cache has sometimes been keyed on destination network addresses rather than destination host addresses. We recommend the choice of destination host addresses because: (1) the IP layer does not always know the Address Mask for an address on a remote (sub-) net; and (2) it may allow the Internet architecture to be extended in the future without any change to the Internet Engineering Task Force [Page 47] ***DRAFT RFC*** INTERNET LAYER June 16, 1989 hosts. For example, the route cache may enable solutions to the problems of partitioned nets and mobile hosts with only additions to gateway mechanisms. Including the Type-of-Service field in the routing cache and considering it in the host routing algorithm will provide the necessary mechanism for the future when Type-of-Service routing is commonly used in the Internet. Each route cache entry corresponds to the endpoints of an Internet path. Although the intervening path may change dynamically in an arbitrary way, the transmission characteristics of the path tend to remain approximately constant over a time period longer than a single typical host-host transport connection. Therefore, a route cache entry is a natural place to cache data on the properties of the path. This data will generally be both gathered and used by a higher layer protocol, e.g., by TCP, or by an application using UDP. Examples of such properties might be the maximum unfragmented datagram size (see Section 3.3.3), or the average round-trip delay measured by a transport protocol. Experiments are currently in progress on caching path properties in the routing cache in this manner. IMPLEMENTATION: An implementation may wish to reduce the overhead of scanning the route cache for every datagram to be transmitted. This may be accomplished with a hash table to speed the lookup, or by giving a connection- oriented transport protocol a "hint" or temporary handle on the appropriate cache entry, to be passed to the IP layer with each subsequent datagram. 3.3.1.4 Dead Gateway Detection The IP layer MUST be able to detect the failure of a "next- hop" gateway that is listed in its routing cache and to choose an alternate gateway (see Section 3.3.1.5). Dead gateway detection is covered in some detail in RFC-816 [IP:11]. Experience to date has not produced a complete algorithm which is totally satisfactory, though it has identified several forbidden paths and promising techniques. * A particular gateway should not be used indefinitely in Internet Engineering Task Force [Page 48] ***DRAFT RFC*** INTERNET LAYER June 16, 1989 the absence of positive indications that it is functioning. * Active probes such as "pinging" (i.e., using an ICMP Echo Request/Reply exchange) are expensive and scale poorly. In particular, hosts MUST NOT actively check the status of a first-hop gateway by simply pinging the gateway continually. * Even when it is the only effective way to verify a gateway's status, pinging MUST be used only when traffic is being sent to the gateway and when there is no other positive indication to suggest that the gateway is functioning. * To avoid pinging, the layers above and/or below the Internet layer SHOULD be able to give "advice" on the status of route cache entries when either positive (gateway OK) or negative (gateway dead) information is available. DISCUSSION: If an implementation does not include an adequate mechanism for detecting a dead gateway and rerouting, a gateway failure may cause datagrams to apparently vanish into a "black hole." This failure can be extremely confusing for users and difficult for network personnel to debug. The dead-gateway detection mechanism must not cause unacceptable load on the host, on connected networks, or on first-hop gateway(s). The exact constraints on the timeliness of dead gateway detection and on acceptable load may vary somewhat depending on the nature of the host's mission, but a host generally needs to detect a failed first-hop gateway quickly enough that transport-layer connections will not break before an alternate gateway can be selected. Passing advice from other layers of the protocol stack complicates the interfaces between the layers, but it is the preferred approach to dead gateway detection. Advice can come from almost any part of the IP/TCP architecture, but it is expected to come primarily from the transport and link layers. Here are some possible sources for gateway advice: Internet Engineering Task Force [Page 49] ***DRAFT RFC*** INTERNET LAYER June 16, 1989 o TCP or any connection-oriented transport protocol should be able to give negative advice, e.g., triggered by excessive retransmissions. o TCP may give positive advice when (new) data is acknowledged. Even though the route may be asymmetric, the receipt of an ACK for new data proves that the data must have been sent. o An ICMP Redirect message from a particular gateway should be used as positive advice about that gateway. o Link-level information which reliably detects and reports host failures (e.g., ARPANET Destination Dead messages) should be used as negative advice. o Failure to ARP or to revalidate ARP mappings may be used as negative advice for the corresponding IP address. o Packets arriving from a particular link-layer address are evidence that the system at this address is alive. However, turning this information into advice about gateways requires mapping the link-layer address into an IP address, and then checking that IP address against the gateways pointed to by the routing cache. This is probably prohibitively inefficient. Note that positive advice, which will be given for nearly every datagram received, may cause unacceptable overhead in the implementation. While advice might be passed using required arguments in all interfaces to the IP layer, some transport and application layer protocols cannot deduce the correct advice. These interfaces must allow a neutral value for advice, since either always-positive or always- negative advice leads to incorrect behaviour. There is another technique for dead gateway detection that has been commonly used but is not recommended. This technique depends upon the host passively receiving ("wiretapping") the Interior Gateway Protocol (IGP) datagrams that the gateways are broadcasting to each other. This approach has the drawback that a host Internet Engineering Task Force [Page 50] ***DRAFT RFC*** INTERNET LAYER June 16, 1989 would need to recognize all the interior gateway protocols that gateways may use (see [INTRO:1]). In addition, it only works on a broadcast network. At present, pinging (i.e., using ICMP Echo messages) is the mechanism for gateway probing. A successful ping guarantees that the addressed interface and its associated machine are up, but it does not guarantee that the machine is a gateway as opposed to a host. The normal inference is that if a Redirect or other evidence indicates that a machine was a gateway, successful pings will indicate that the machine is still up and hence still a gateway. However, since a host silently ignores packets which a gateway would forward or redirect, this assumption could sometimes fail. To avoid this problem, a new ICMP message under development will ask "are you a gateway?" IMPLEMENTATION: The following specific algorithm has been suggested: o Associate a "reroute timer" with each gateway pointed to by the routing cache. Initialize the timer to a value Tr, which must be small enough to allow detection of a dead gateway before transport connections time out. o Positive advice might reset the reroute timer to Tr. Negative advice might reduce or zero the reroute timer. o Whenever the IP layer used a particular gateway to route a datagram, it would check the corresponding reroute timer. If the timer had expired (reached zero), the IP layer would send a probe to the gateway, followed immediately by the datagram. o The probe would be sent again if necessary, up to N times. If no probe reply was received in N tries, the gateway would be assumed to have failed, and a new first-hop gateway would be chosen for all cache entries pointing to the failed gateway. Note that the size of Tr is inversely related to the amount of advice available. Tr should be large enough to insure that: Internet Engineering Task Force [Page 51] ***DRAFT RFC*** INTERNET LAYER June 16, 1989 * Any pinging will be at a low level (e.g. <10%) of all packets sent to a gateway from the host, AND * pinging is infrequent (e.g. every 3 minutes) Since the recommended algorithm is concerned with the gateways pointed to by routing cache entries, rather than the cache entries themselves, a two level data structure (perhaps coordinated with ARP or similar caches) may be desirable for implementing a route cache. 3.3.1.5 New Gateway Selection If the failed gateway is the current default, the IP layer MUST select a different default gateway (assuming more than one default is known), for use in establishing new routes. DISCUSSION: When a gateway does fail, the other gateways on the connected network will learn of the failure through some inter-gateway routing protocol. However, this will not happen instantaneously, since gateway routing protocols typically have a settling time of 30-60 seconds. If the host switches to an alternative gateway before the gateways have agreed on the failure, the new target gateway will probably forward the datagram to the failed gateway and return a Redirect to the host (!). The result is likely to be the rapid oscillation in the host's route cache entry during the gateway settling period. It has been proposed that the dead gateway logic should include some hysteresis mechanism to prevent such oscillations. However, experience has not shown any harm from such oscillations, since service cannot be restored to the host until the gateways' routing information does settle down. IMPLEMENTATION: One implementation technique for choosing a new gateway is to simply round-robin among the default gateways in the host's list. Another is to rank the gateways in priority order, and when the current default gateway is not the highest priority one, to "ping" the higher- priority gateways slowly to detect when they return to service. This pinging can be at a very low rate, e.g., 0.005 per second. Internet Engineering Task Force [Page 52] ***DRAFT RFC*** INTERNET LAYER June 16, 1989 3.3.1.6 Initialization The following information MUST be configurable for each logical network interface: (1) IP address of the logical network interface. (2) Address Mask corresponding to that IP address. (3) MTU (Maximum Transmission Unit): maximum packet size of the network. (4) Relative preference order of the interface (see Section 3.3.4.2). In addition, there there MAY be a subnets-are-local flag (see Section 3.3.3) for each interface. Finally, the host MUST have a list of default gateways. | A variety of methods can be used to determine this information dynamically. A manual method of entering this data MUST be provided for use on networks that do not support broadcast. The host IP layer MUST operate correctly in a minimal network environment, and in particular, on one with no gateways. For example, if the IP layer of a host insists on finding at least one gateway to initialize, the host will be unable to operate on a single isolated broadcast net with no gateways. DISCUSSION: Even though some of these quantities (e.g., the MTU) would seem to be fixed by the interface hardware, experience has shown a requirement for possibly overriding implementation choices that are bad or just different. Some host implementations use "wiretapping" of gateway protocols on a broadcast network to learn what gateways exist. For the same reasons cited in Section 3.3.1.4, this approach is less than general, but a host may implement such a mechanism. Internet Engineering Task Force [Page 53] ***DRAFT RFC*** INTERNET LAYER June 16, 1989 3.3.2 Reassembly The IP layer MUST implement reassembly of IP datagrams. We designate the largest datagram size that can be reassembled by EMTU_R ("Effective MTU to receive"); this is also called the "reassembly buffer size." EMTU_R MUST be greater than or equal to 576, SHOULD be either configurable or indefinite, and SHOULD be greater than or equal to the MTU of the connected network(s). DISCUSSION: A fixed EMTU_R limit should not be built into the code because some application layer protocols require EMTU_R values larger than 576. Therefore, installing a new application protocol on a system could require that the EMTU_R value be increased. IMPLEMENTATION: An implementation may use a contiguous reassembly buffer for each datagram, or it may use a more complex data structure that places no definite limit on the the reassembled datagram size; in the latter case, EMTU_R is said to be "indefinite." Reassembly is basically performed by copying each fragment into the buffer at the proper offset. Note that fragments may overlap, if successive retransmissions use different packetizing. The tricky part is the bookkeeping to determine when all bytes of the datagram have been reassembled. We recommend Clark's algorithm [IP:10] that requires no additional data space for the bookkeeping. | Note however, that contrary to [IP:10], the first fragment | header needs to be saved for inclusion in a possible ICMP | Time Exceeded (Reassembly Timeout) message. There MUST be a mechanism by which the transport layer can # learn MMS_R, the maximum message size that can be received and # reassembled in an IP datagram (see GET_MAXSIZES calls in # Section 3.4). If EMTU_R is not indefinite, then the value of # MMS_R is given by: # MMS_R = EMTU_R - 20 # since 20 is the minimum size of an IP header. # There MUST be a reassembly timeout. If this timeout expires, the partially-reassembled datagram MUST be discarded and an Internet Engineering Task Force [Page 54] ***DRAFT RFC*** INTERNET LAYER June 16, 1989 ICMP Time Exceeded message sent to the source host (if fragment zero has been received). The reassembly timeout value SHOULD be a fixed value, not set from the remaining TTL. It is recommended that the value lie between 60 seconds and 120 seconds. DISCUSSION: The IP specification says that the reassembly timeout should be the remaining TTL from the IP header, but this does not work well because gateways generally treat TTL as a simple hop count rather than an elapsed time. If the reassembly timeout is too small, datagrams will be discarded unnecessarily, and communication may fail. The timeout needs to be at least as large as the typical maximum delay across the Internet. A realistic minimum reassembly timeout would be 60 seconds. It has been suggested that a cache might be kept of round-trip times measured by transport protocols for various destinations, and that these values might be used to dynamically determine a reasonable reassembly timeout value. Further investigation of this approach is required. If the reassembly timeout is set too high, buffer resources in the receiving host will be tied up too long, and the MSL (Maximum Segment Lifetime) [TCP:1] will be larger than necessary. The MSL controls the maximum rate at which (fragmented) datagrams can be sent using distinct values of the 16-bit Ident field; a larger MSL lowers the maximum rate. The TCP specification [TCP:1] arbitrarily assumes a value of 2 minutes for MSL. This is an upper limit on a reasonable reassembly timeout value. 3.3.3 Fragmentation Optionally, the IP layer MAY implement a mechanism to locally ! fragment outgoing datagrams. ! We designate by EMTU_S ("Effective MTU for sending") the ! maximum IP datagram size that may be sent, for a particular ! combination of IP source and destination addresses and perhaps ! TOS. ! A host MUST implement a mechanism to allow the transport layer ! to learn MMS_S, the maximum transport-layer message size that ! may be sent for a given {source, destination, TOS} triplet (see ! GET_MAXSIZES call in Section 3.4). If no local fragmentation ! Internet Engineering Task Force [Page 55] ***DRAFT RFC*** INTERNET LAYER June 16, 1989 is performed, the value of MMS_S will be: ! MMS_S = EMTU_S - <IP header size> ! and EMTU_S must be less than or equal to the MTU of the network ! interface corresponding to the source address of the datagram. ! Note that <IP header size> in this equation will be 20, unless ! the IP reserves space to insert IP options for its own purposes ! in addition to any options inserted by the transport layer. ! A host that does not implement local fragmentation MUST ensure ! that the transport layer (for TCP) or the application layer ! (for UDP) obtains MMS_S from the IP layer and does not try to ! send a datagram exceeding MMS_S in size. ! It is generally desirable to avoid local fragmentation and to ! choose EMTU_S low enough to avoid fragmentation in any gateway ! along the path. In the absence of actual knowledge of the ! minimum MTU along the path, the IP layer SHOULD use EMTU_S <= ! 576 whenever the destination address is not on a connected ! network, and otherwise use the connected network's MTU. ! A host IP layer implementation MAY have a configuration flag "Subnets-are-local," which indicates that the MTU of the connected network should be used for destinations on different subnets within the same network, but not for other networks. Thus, this flag causes the network class mask, rather than the subnet Address Mask, to be used to choose an EMTU_S. For a multihomed host, a "Subnets-are-local" flag is needed for each logical interface. DISCUSSION: Picking the correct datagram size to use when sending data is a complex topic [IP:9]. (a) In general, no host is required to accept an IP ! datagram larger that 576 bytes (including header and ! data), so a host must not send a larger datagram ! without explicit prior arrangement with the ! destination host. Thus, MMS_S is only an upper bound ! on the datagram that a transport protocol can send; ! even if MMS_S exceeds 576, the transport layer must ! limit its datagrams to 576 in the absence of other ! knowledge about the destination host. (b) Some transport protocols (e.g., TCP) do provide a way to explicitly inform the sender about the largest datagram the other end can receive and reassemble Internet Engineering Task Force [Page 56] ***DRAFT RFC*** INTERNET LAYER June 16, 1989 [IP:7]. There is no corresponding mechanism in the IP layer. (c) Hosts should ideally limit their EMTU_S for a given destination to the minimum MTU of all the networks along the path, to avoid any fragmentation. IP fragmentation, while formally correct, can create a serious transport protocol performance problem, because loss of a single fragment means all the fragments in the segment must be retransmitted [IP:9]. It has been suggested that a host could determine the MTU over a given path by sending a zero-offset datagram fragment and waiting for the receiver to time out the reassembly (which cannot complete!) and return an ICMP Time Exceeded message. This message will include the largest remaining fragment header in its body. More direct mechanisms are being experimented with, but have not yet been adopted (see e.g. RFC-1063). Since nearly all networks in the Internet currently support an MTU of 576 or greater, we strongly recommend the use of 576 for datagrams sent to other networks. 3.3.4 Multihomed Hosts A multihomed host has multiple IP addresses. Multihoming introduces considerable confusion and complexity into the protocol suite, and it is an area in which the Internet architecture falls seriously short of solving all problems. There are two distinct problem areas in multihoming: (1) Local multihoming -- the host itself is multihomed; or (2) Remote multihoming -- the local host needs to communicate with a remote multihomed host. This section discusses local multihoming. At present, remote multihoming MUST be handled at the application layer, as discussed in the companion RFC [INTRO:1]. 3.3.4.1 Local Multihoming A multihomed host has multiple IP addresses and therefore multiple logical interfaces (since each IP address corresponds uniquely to a logical address). These logical Internet Engineering Task Force [Page 57] ***DRAFT RFC*** INTERNET LAYER June 16, 1989 interfaces may be mapped onto one or more physical interfaces, and these physical interfaces may be connected to the same or different networks. Note that a host using link-layer multiplexing (see Section 2.3.2) may have multiple physical interfaces but only one logical interface; such a host is not multihomed. In the Internet protocol architecture, a transport protocol instance ("entity") has no address of its own, but instead uses an Internet Protocol (IP) address. This has implications for the IP, transport, and application layers, and for the interfaces between them. In particular, the application software may have to be aware of the multiple IP addresses of a multihomed host; in other cases, the choice can be made within the network software. From the Internet viewpoint, a multihomed host may be modelled as a set of logical hosts within the same physical host. For example, a request/response application protocol built on UDP may require that a response come from the same logical host to which the request was sent. Here are some important cases of multihoming: (a) Multiple Logical Networks The Internet architects envisioned that each physical network would have a single unique IP network (or subnet) number. However, LAN administrators have sometimes found it useful to violate this assumption, operating a LAN with multiple logical networks per physical connected network. If a host connected to such a physical network is configured to handle traffic for each of N different logical networks, then the host will have N logical interfaces. These could share a single physical interface, or might use N physical interfaces to the same network, for example. (b) Multiple Logical Hosts When a host has multiple IP addresses that all have the same <Network-number> part (and the same <Subnet- number> part, if any), the logical interfaces are known as "logical hosts." These logical interfaces might share a single physical interface or might use separate Internet Engineering Task Force [Page 58] ***DRAFT RFC*** INTERNET LAYER June 16, 1989 physical interfaces to the same physical network. (c) Simple Multihoming In this case, each logical interface is mapped into a separate physical interface and each physical interface is connected to a different physical network. The term "multihoming" was originally applied only to this case, but it is now applied more generally. A host with embedded gateway functionality will typically fall into the simple multihoming case. Note, however, that a host may be simply multihomed without containing an embedded gateway, i.e., without forwarding datagrams from one connected network to another. This case presents the most difficult routing problems. It is possible for the choice of interface (i.e., the choice of first-hop network) to significantly affect performance or even reachability of remote parts of the Internet. Finally, we note another possibility that is NOT @ multihoming: one logical interface bound to multiple @ physical interfaces, in order to increase the reliability or @ throughput between directly connected machines by providing @ alternative physical paths between them. For instance, two @ systems might be connected by multiple point-to-point links. @ We call this "link-layer multiplexing." With link-layer @ multiplexing, the protocols above the link layer are unaware @ that multiple physical interfaces are present; the link- @ layer device driver is responsible for multiplexing and @ routing packets across the physical interfaces. @ 3.3.4.2 Selecting a Logical Interface The following general rules apply to the selection of ! logical interface (i.e., local IP source address) for ! sending a datagram from a multihomed host. ! (1) If the datagram is sent in response to a received ! datagram, the source IP address for the response SHOULD ! generally match the IP address to which the request was ! sent; see Sections 4.1.3.5 and 4.2.3.7 and the ! Introduction section of [INTRO:1] for more specific ! Internet Engineering Task Force [Page 59] ***DRAFT RFC*** INTERNET LAYER June 16, 1989 requirements on higher layers. ! Otherwise, a logical interface must be selected. ! (2) An application MUST be able to explicitly specify the ! logical interface for initiating a connection or a ! request. ! (3) In the absence of such a specification, the networking ! software MUST choose the logical interface. Rules for ! this choice will now be described. ! Under the model of a strictly-layered implementation, we ! assume that a transport-layer protocol like TCP will call an ! IP-layer routine to choose a logical network interface for ! an outgoing connection on a multi-homed host. The choice of ! network interface belongs in the IP layer since it involves ! Internet routing and may be required for any transport ! protocol. This implies the following generic call in the ! transport/IP interface (see Section 3.4): ! GET_INTERFACE(remote IP addr, TOS)-> logical interface Here TOS is the Type-of-Service value; see Section 3.2.1.6, and the logical interface implies the local IP address to be used. The following techniques are recommended for implementing the GET_INTERFACE function: (a) If the remote Internet address lies on one of the (sub-) nets to which the host is directly connected, the corresponding logical interface SHOULD be chosen, unless it is down. (b) Configuration data for the host MAY include a list of (network, interface, TOS) triples. If the given remote network matches one of these entries, the corresponding logical interface is to be used, unless it is down. (c) Configuration data for the host MUST include a list of logical interfaces with a definition of preference order, for use when (a) and (b) fail. DISCUSSION: Some implementations of the Internet protocols choose a Internet Engineering Task Force [Page 60] ***DRAFT RFC*** INTERNET LAYER June 16, 1989 logical network interface for the simple multihoming case (b) by letting the host eavesdrop on ("wiretap") the routing update datagrams that the gateways are interchanging. This approach has two disadvantages: (1) it only works on a broadcast network, and (2) a host would need to implement all the interior gateway protocols that gateways may use [INTRO:1]. A suggested technique is for the IP layer to send ICMP Echo Request messages to the given remote Internet address through all the logical interfaces, and to choose the interface through which an ICMP Echo Reply first arrives. The Echo Request would be sent with the requested TOS, to measure the appropriate route. This approach has not yet been tried in practice. In the future, there may be a defined way for a multihomed host to ask the gateways on all connected networks for advice about the best network to use for a given destination. 3.3.5 IP Source Address When a host sends an IP datagram through a particular network interface, the source address field of the IP header MUST correspond to that interface, unless it is a source-routed datagram that is being forwarded. DISCUSSION: There has been some controversy about this requirement for the case of multihomed hosts, as it implies that the binding between logical and physical interfaces is fixed. Several observations need to be made about this restriction: o It is consistent with the current Internet model with respect to multihoming (see previous Section). o It is necessary to make the Redirect mechanism work. If a datagram were sent out a physical interface that did not correspond to the IP address (logical interface) in its header, the first-hop gateway would not realize that it might need to send a Redirect. o It is relevant only for hosts without embedded gateway functionality. If the host is capable of acting as a gateway, then effectively the first-hop gateway is internal; as a gateway, it will be Internet Engineering Task Force [Page 61] ***DRAFT RFC*** INTERNET LAYER June 16, 1989 participating in the IGP with other gateways and will therefore have the necessary routing information without Redirects. 3.3.6 Broadcasts Section 3.2.1.3 defined the four standard IP broadcast address forms: Limited Broadcast: {-1, -1} Directed Broadcast: {<Network-number>,-1} Subnet Directed Broadcast: {<Network-number>,<Subnet-number>,-1} All-Subnets Directed Broadcast: {<Network-number>,-1,-1} A host MUST recognize any of these forms in the destination @ address of an incoming datagram. @ There is a class of hosts* that use non-standard broadcast @ address forms, substituting 0 for -1. All hosts SHOULD @ recognize and accept any of these non-standard broadcast @ addresses as the destination address of an incoming datagram. @ A host MAY optionally have a configuration option to choose the @ 0 or the -1 form for sending a broadcast address, for each @ logical interface. @ When a host sends a datagram to a link-layer broadcast address, the IP destination address MUST be a legal IP broadcast or multicast address. A host SHOULD silently ignore a datagram that is received via a link-layer broadcast (see Section 2.4) but does not specify an IP multicast or broadcast destination address. When a host sends any datagram, the IP source address MUST be one of its own IP addresses (but not a broadcast or multicast address), except when a source-routed datagram is forwarded. Hosts SHOULD use the Limited Broadcast address to broadcast to a connected network. _________________________ *4.2BSD Unix and its derivatives. Internet Engineering Task Force [Page 62] ***DRAFT RFC*** INTERNET LAYER June 16, 1989 DISCUSSION: Using the Limited Broadcast address instead of a Directed Broadcast address may improve system robustness. Problems are often caused by machines that do not understand the plethora of broadcast addresses (see Section 3.2.1.3), or that may have different ideas about which broadcast addresses are in use. The prime example of the latter is machines that do not understand subnetting but are attached to a subnetted net. Sending a Subnet Broadcast for the connected network will confuse those machines, which will see it as a message to some other host. There has been discussion on whether a datagram addressed to the Limited Broadcast address ought to be sent from all the interfaces of a multi-homed host. This specification takes no stand on this issue. 3.3.7 Error Reporting Wherever practical, hosts MUST return ICMP error datagrams on detection of an error, except in those cases where returning an ICMP error message is specifically prohibited. DISCUSSION: A common phenomenon in networks is the "black hole disease"; datagrams are sent out, but nothing comes back. Without any error datagrams, it is difficult for the user to figure out what the problem is. 3.4 INTERNET/TRANSPORT LAYER INTERFACE The interface between the IP layer and the transport layer MUST provide full access to all the mechanisms of the IP layer, including options, Type-of-Service, and Time-to-Live. The transport layer MUST either have mechanisms to set these interface parameters, or provide a path to pass them through from an application, or both. DISCUSSION: Applications are urged to make use of these mechanisms where applicable, even when the mechanisms are not currently effective in the Internet (e.g., TOS). This will allow these mechanisms to be immediately useful when they do become effective, without a large amount of retrofitting of host software. We now describe a conceptual interface between the transport layer and the IP layer, as a set of procedure calls. This is an Internet Engineering Task Force [Page 63] ***DRAFT RFC*** INTERNET LAYER June 16, 1989 extension of the information in Section 3.3 of RFC-791 [IP:1]. A host implementation MUST support the logical information flow implied by these calls, but need not literally implement the calls themselves. For example, many implementations reflect the coupling between the transport layer and the IP layer by letting the two layers have shared access to common data structures; these data structures are then the agency for passing much of the information that is required. * Send Datagram SEND(src, dst, prot, TOS, TTL, BufPTR, len, Id, DF, opt => result ) where the parameters are defined in RFC-791. * Receive Datagram RECV(BufPTR, prot => result, src, dst, interface, TOS, len, opt) All the parameters are defined in RFC-791, except for: interface = handle on logical network interface; implies local IP address. The result parameter dst contains the datagram's destination address. Since this may be a broadcast or multicast address, the interface parameter (not shown in RFC-791) MUST be passed, to support multihoming. The parameter opt contains all the IP options received in the datagram; these MUST also be passed to the transport layer. * Select Local Logical Interface GET_INTERFACE(remote, TOS) -> interface interface = handle on logical network interface; implies local IP address remote = remote Internet address TOS = TOS field (low 5 bits of TOS byte) See Section 3.3.4.2. Internet Engineering Task Force [Page 64] ***DRAFT RFC*** INTERNET LAYER June 16, 1989 * Find Maximum Datagram Sizes GET_MAXSIZES(interface, remote, TOS) -> MMS_R, MMS_S MMS_R = maximum receive transport-message size. MMS_S = maximum send transport-message size. (interface, remote, TOS defined above) See Sections 3.3.2 and 3.3.3. * Advise of Delivery Problem ADVISE_DELIVPROB(interface, remote, TOS) (Parameters defined above) The transport protocol MUST call this routine when repeated timeouts raise the suspicion that segments are not being delivered by IP. It will be a signal to the IP layer to try a different gateway, for example. * Send ICMP Message | SEND_ICMP(src, dst, TOS, TTL, BufPTR, len, Id, DF, opt | => result ) | (Parameters defined in RFC-791). | The transport layer MUST be able to send certain ICMP | messages: Port Unreachable or any of the query-type | messages. This function could be considered to be a special | case of the SEND() call, of course; we describe it separately | for clarity. | For an ICMP error message, the data to be passed MUST include | the "Internet Header + 64 bits of Data Datagram" [IP:2] of | the offending datagram. | * Receive ICMP Message RECV_ICMP(BufPTR => result, src, dst, len, opt) (Parameters defined in RFC-791). Internet Engineering Task Force [Page 65] ***DRAFT RFC*** INTERNET LAYER June 16, 1989 IP layer MUST pass certain ICMP messages up to the appropriate transport-layer routine. This function could be considered to be a special case of the RECV() call, of course; we describe it separately for clarity. The data that is passed up MUST include the "Internet Header + 64 bits of Data Datagram" [IP:2] contained in the ICMP message. It will be used by the transport layer to locate the connection state information, if any. In particular, the following ICMP messages are to be passed up: o Destination Unreachable o Source Quench o Timestamp Reply (to ICMP user interface) o Echo Reply (to ICMP user interface) o Time Exceeded (code 0). DISCUSSION: In the future, there may be additions to this interface to pass path data (see Section 3.3.1.3) between the IP and transport layers. 3.5 INTERNET LAYER REQUIREMENTS SUMMARY | | | | |S| | | | | | |H| |F | | | | |O|M|o | | |S| |U|U|o | | |H| |L|S|t | |M|O| |D|T|n | |U|U|M| | |o | |S|L|A|N|N|t | |T|D|Y|O|O|t FEATURE |SECTION | | | |T|T|e -------------------------------------------------|--------|-|-|-|-|-|-- | | | | | | | Able to log discarded datagrams |3.1 | |x| | | | Record in counter |3.1 | |x| | | | Silently discard Version != 4 |3.2.1.1 |x| | | | | Verify IP checksum, silently discard bad pkt |3.2.1.2 |x| | | | | Internet Engineering Task Force [Page 66] ***DRAFT RFC*** INTERNET LAYER June 16, 1989 Addressing: |3.2.1.3 | | | | | | Subnet addressing (RFC-950) |3.2.1.3 |x| | | | | Silently ignore datagram with bad dest addr |3.2.1.3 |x| | | | | Silently ignore d'gram to wrong phys. i'face |3.2.1.3 | | |x| | | Silently ignore datagram with bad src address |3.2.1.3 |x| | | | | Support reassembly |3.2.1.4 |x| | | | | Retain same Id field in identical datagram |3.2.1.5 | | |x| | | | | | | | | | TOS: | | | | | | | Allow transport layer to set TOS |3.2.1.6 |x| | | | | Pass received TOS up to transport layer |3.2.1.6 | |x| | | | Use RFC-795 link-layer mappings for TOS |3.2.1.6 | | | |x| | TTL: | | | | | | | Send packet with TTL of 0 |3.2.1.7 | | | | |x| Discard received packets with low TTL |3.2.1.7 | | | | |x| Allow transport layer to set TTL |3.2.1.7 |x| | | | | Configurable default TTL if possible |3.2.1.7 |x| | | | | | | | | | | | IP Options: | | | | | | | Allow transport layer to send IP options |3.2.1.8 |x| | | | | Pass all IP options rcv'd to higher layer |3.2.1.8 |x| | | | | IP layer silently ignore unknown options |3.2.1.8 |x| | | | | Security option |3.2.1.8 | | |x| | | Stream Identifier option |3.2.1.8 | | | | |x| Record Route option |3.2.1.8 | | |x| | | Timestamp option |3.2.1.8 | | |x| | | Source Route Option: | | | | | | | Send and receive Source Route options |3.2.1.8 |x| | | | | Datagram with completed SR passed up to TL |3.2.1.8 |x| | | | | Build correct (non-redundant) return route |3.2.1.8 |x| | | | | Send multiple SR options in one header |3.2.1.8 | | | | |x| Forward datagrams with Source Route option |3.2.1.8 | | |x| | | Obey corresponding gateway rules |3.2.1.8 |x| | | | | Configurable switch for non-local SRing |3.2.1.8 |x| | | | | Defaults to OFF |3.2.1.8 |x| | | | | Satisfy gwy access rules for non-local SRing |3.2.1.8 |x| | | | | If not forward, send Dest Unreach (cd 5) |3.2.1.8 | |x| | | |2 | | | | | | | ICMP: | | | | | | | Silently ignore unknown type ICMP message |3.2.2 |x| | | | | Demux ICMP Error to transport protocol |3.2.2 |x| | | | | Send ICMP error message with TOS=0 |3.2.2 | |x| | | | Send ICMP error message for: | | | | | | | - ICMP error msg |3.2.2 | | | | |x| - IP b'cast or m'cast addressed datagram |3.2.2 | | | | |x| - link-layer b'cast addressed datagram |3.2.2 | | | | |x| - non-initial fragment |3.2.2 | | | | |x| - datagram with non-unique src address |3.2.2 | | | | |x| Internet Engineering Task Force [Page 67] ***DRAFT RFC*** INTERNET LAYER June 16, 1989 | | | | | | | Generate Dest Unreachable (code 1/2/3/4/5) |3.2.2.1 | |x| | | | Pass ICMP Dest Unreachable to higher layer |3.2.2.1 |x| | | | | Interpret Dest Unreachable as only hint |3.2.2.1 |x| | | | | Update route cache when recv Redirect |3.2.2.2 |x| | | | | Handle both Host and Net Redirect |3.2.2.2 |x| | | | | Discard Redirect to non-connected gateway |3.2.2.2 | |x| | | | Send Source Quench if buffering exceeded |3.2.2.3 | | |x| | | Pass Source Quench to higher layer |3.2.2.3 |x| | | | | Higher layer act on Source Quench |3.2.2.3 | |x| | | | Pass Time Exceeded (code 0) to higher layer |3.2.2.4 |x| | | | | Silently ignore Time Exceeded (code 1) |3.2.2.4 | |x| | | | Send Parameter Problem messages |3.2.2.5 | |x| | | | Pass Parameter Problem to higher layer |3.2.2.5 |x| | | | | Report Parameter Problem to user |3.2.2.5 | | |x| | | | | | | | | | ICMP Echo server |3.2.2.6 |x| | | | | Ignore Echo Request to broadcast address |3.2.2.6 | | |x| | | Ignore Echo Request to multicast address |3.2.2.6 | | |x| | | Answer Echo Request from same IP address |3.2.2.6 |x| | | | | Answer Echo Request with same data |3.2.2.6 |x| | | | | Reflect Record Route, Time Stamp options |3.2.2.6 | |x| | | | Reverse and reflect Source Route option |3.2.2.6 |x| | | | | ICMP Echo Client |3.2.2.6 | |x| | | | Pass Echo Reply to higher layer |3.2.2.6 |x| | | | | | | | | | | | ICMP Information Request or Reply |3.2.2.7 | | | |x| | | | | | | | | ICMP Timestamp and Timestamp Reply |3.2.2.8 | | |x| | | Ignore b'cast Timestamp |3.2.2.8 |x| | | | |1 Ignore m'cast Timestamp |3.2.2.8 | | |x| | |1 Answer from same IP address |3.2.2.8 |x| | | | |1 Reverse and reflect Source Route option |3.2.2.8 |x| | | | |1 Pass recvd Timestamp Reply to higher layer |3.2.2.8 |x| | | | |1 Update Timestamp at least 15 times/sec |3.2.2.8 |x| | | | |1 | | | | | | | ICMP Address Mask Request and Reply | | | | | | | Support static configuration of addr mask |3.2.2.9 |x| | | | | Support sending ICMP Addr Mask Request |3.2.2.9 | | |x| | | Get addr mask dynamically during booting |3.2.2.9 | | |x| | | Retransmit Addr Mask Req if no Reply |3.2.2.9 |x| | | | |3 Assume address mask = 0 if no Reply |3.2.2.9 |x| | | | |3 Update address mask from first Reply only |3.2.2.9 |x| | | | |3 Send unauthorized Addr Mask Reply msgs |3.2.2.9 | | | | |x| Static config=> Addr-Mask-Authoritative flag |3.2.2.9 | |x| | | | | | | | | | | ROUTING OUTBOUND DATRAGRAMS: | | | | | | | Use Address Mask in local/remote decision |3.3.1.1 |x| | | | | Internet Engineering Task Force [Page 68] ***DRAFT RFC*** INTERNET LAYER June 16, 1989 Special code for broadcasts/multicasts |3.3.1.1 |x| | | | | Maintain "route cache" of next-hop gateways |3.3.1.2 |x| | | | | Treat Host and Net Redirect the same |3.3.1.2 |x| | | | | Support multiple default gateways |3.3.1.2 |x| | | | | Provide table of static routes |3.3.1.2 | | |x| | | Allow static routes to override Redirects |3.3.1.2 | | |x| | | Key route cache on host, not net address |3.3.1.3 | |x| | | | Include TOS in route cache |3.3.1.3 | |x| | | | Include non-IP path info (eg MTU or RTT) |3.3.1.3 | | |x| | | | | | | | | | Detect failure of next-hop gateway |3.3.1.4 |x| | | | | Assume route is good forever |3.3.1.4 | | | |x| | Ping gateways continuously |3.3.1.4 | | | | |x| Ping only when traffic being sent |3.3.1.4 |x| | | | | Ping only when no positive indication |3.3.1.4 |x| | | | | Higher and lower layers give advice |3.3.1.4 | |x| | | | Switch from failed default g'way to another |3.3.1.5 |x| | | | | Manual method of entering config info |3.3.1.6 |x| | | | | Operate with no gateways on a network |3.3.1.6 |x| | | | | | | | | | | | REASSEMBLY and FRAGMENTATION: | | | | | | | Able to reassemble incoming datagrams |3.3.2 |x| | | | | Able to reassemble 576 byte datagrams |3.3.2 |x| | | | | Able to reassemble >576 bytes |3.3.2 | |x| | | | EMTU_R configurable or indefinite |3.3.2 | |x| | | | Transport layer able to learn MMS_R |3.3.2 |x| | | | | Send ICMP Time Exceeded on reassembly timeout |3.3.2 |x| | | | | Fixed reassembly timeout value |3.3.2 | |x| | | | | | | | | | | Pass MMS_S to higher layers |3.3.3 |x| | | | | Local fragmentation of outgoing packets |3.3.3 | | |x| | | Else don't send bigger than MMS_S |3.3.3 |x| | | | | Send max 576 to off-net destination |3.3.3 | |x| | | | Subnets-are-local configuration flag |3.3.3 | | |x| | | | | | | | | | Reply through same logical interface |3.3.4.2 | |x| | | | Allow application to choose logical i'face |3.3.4.2 |x| | | | | Mechanism to choose logical interface |3.3.4.2 |x| | | | | Choose interface for dest on connected network |3.3.4.2 | |x| | | | Configure a list of <net,iface,TOS> triples |3.3.4.2 | | |x| | | Order the list of logical interfaces |3.3.4.2 |x| | | | | Send d'gram with IP src addr of interface |3.3.5 |x| | | | |4 | | | | | | | Recognize all broadcast address formats |3.3.6 |x| | | | | Use IP b'cast/m'cast addr in link-level b'cast |3.3.6 |x| | | | | Silently ignore link-layer-only broadcast dg's |3.3.6 | |x| | | | Receive 0 or -1 broadcast formats OK |3.3.6 | |x| | | | Config'ble option to send 0 or -1 b'cast |3.3.6 | | |x| | | Internet Engineering Task Force [Page 69] ***DRAFT RFC*** INTERNET LAYER June 16, 1989 Broadcast or multicast addr as IP source addr |3.3.6 | | | | |x| Use Limited Broadcast addr for connected net |3.3.6 | |x| | | | Return ICMP error msgs (when not prohibited) |3.3.7 |x| | | | | | | | | | | | Allow transport layer to use all IP mechanisms |3.4 |x| | | | | Pass interface ident in RECV() call |3.4 |x| | | | | Pass all IP options up to transport layer |3.4 |x| | | | | Transport layer can send certain ICMP messages |3.4 |x| | | | | Pass spec'd ICMP messages up to transport layer |3.4 |x| | | | | Able to leap tall buildings at a single bound |3.5 |x| | | | | Footnotes: (1) Only if feature is implemented. (2) This requirement is overruled if datagram is an ICMP error message. (3) Only if feature is configured "on". (4) Unless has embedded gateway functionality or is source routed. Internet Engineering Task Force [Page 70] ***DRAFT RFC*** TRANSPORT LAYER -- UDP June 16, 1989 4. TRANSPORT PROTOCOLS 4.1 USER DATAGRAM PROTOCOL -- UDP 4.1.1 INTRODUCTION The User Datagram Protocol UDP [UDP:1] offers only a minimal transport service -- non-guaranteed delivery of datagrams -- and is designed to give applications direct access to the datagram service of the IP layer. UDP is used by applications that do not require the level of service of TCP or that wish to use communications services (e.g., multicast or broadcast delivery) not available from TCP. UDP is almost a null protocol; the only services it provides over IP are checksumming of data and multiplexing by port number. Therefore, an application program running over UDP must deal directly with end-to-end communication problems that a connection-oriented protocol would have handled -- e.g., retransmission for reliable delivery, packetization and | reassembly, flow control, congestion avoidance, etc., when | these are required. The fairly complex coupling between IP and TCP will be mirrored in the coupling between UDP and many applications using UDP. 4.1.2 PROTOCOL WALK-THROUGH There are no known errors in the specification of UDP. 4.1.3 SPECIFIC ISSUES 4.1.3.1 Ports UDP well-known ports follow the same rules as TCP well-known ports; see Section 4.2.2.1 below. If a datagram arrives addressed to a UDP port for which | there is no pending LISTEN call, UDP SHOULD send an ICMP | Port Unreachable message. | 4.1.3.2 IP Options UDP MUST pass any IP option that it receives from the IP layer transparently to the application layer. An application MUST be able to specify IP options to be sent | in its UDP datagrams, and UDP MUST pass these options | transparently to the IP layer. | Internet Engineering Task Force [Page 71] ***DRAFT RFC*** TRANSPORT LAYER -- UDP June 16, 1989 DISCUSSION: At present, the only options that need be passed transparently through UDP are Source Route, Record Route, and Time Stamp. However, new options may be defined in the future, and UDP need not and should not make any assumptions about the options it passes transparently. An application that uses UDP will need to save source routes from request datagrams and reverse them to send the corresponding reply datagrams. 4.1.3.3 ICMP Messages UDP MUST pass ICMP error messages that it receives from the IP layer transparently up to the application layer. Conceptually at least, this may be accomplished with an upcall to the ERROR_REPORT routine (see Section 4.2.4.1). DISCUSSION: ! Note that ICMP error messages resulting from sending a ! UDP datagram are received asynchronously. A UDP-based ! application that wants to receive ICMP error messages ! is responsible for maintaining the state necessary to ! demultiplex these mesages when they arrive; for ! example, the application may keep a pending receive ! operation for this purpose. The application is also ! responsible to avoid confusion from a delayed ICMP ! error message resulting from an earlier use of the same ! port(s). ! 4.1.3.4 UDP Checksums A host MUST implement the facility to generate and validate UDP checksums. An application MAY optionally be able to control whether a UDP checksum will be generated, but it MUST default to checksumming on. If a UDP datagram is received with a checksum that is non- zero and invalid, UDP MUST silently discard the datagram. An application MAY optionally be able to control whether UDP datagrams without checksums should be discarded or passed to the application. DISCUSSION: Some applications that normally run only across local area networks have chosen to turn off UDP checksums for efficiency. As a result, numerous cases of undetected Internet Engineering Task Force [Page 72] ***DRAFT RFC*** TRANSPORT LAYER -- UDP June 16, 1989 errors have been reported. The advisability of ever turning off UDP checksumming is very controversial. IMPLEMENTATION: There is a common implementation error in UDP checksums. Unlike the TCP checksum, the UDP checksum is optional; the value zero is transmitted in the checksum field of a UDP header to indicate the absence of a checksum. If the transmitter really calculates a UDP checksum of zero, it must transmit the checksum as all 1's (65535). No special action is required at the receiver, since zero and 65535 are equivalent in 1's complement arithmetic. 4.1.3.5 UDP Multihoming When a UDP datagram is received, the local IP address to which it was directed MUST be passed up to the application layer. An application program MUST be able to specify the logical ! interface (local IP address) to be used when it sends a UDP ! datagram. It MUST allow the local IP address to be ! unspecified (value zero), in which case the networking ! software will choose an appropriate interface. There SHOULD then be a way to communicate the resulting choice back to the application layer (e.g, so that the application could receive a reply datagram only from the corresponding interface). DISCUSSION: Since UDP is a pure datagram protocol with no retained connection state, knowledge of the local IP address cannot be retained in the transport layer. A request/response application that uses UDP should make the response through the same logical interface through which the request arrived. 4.1.3.6 Invalid Addresses UDP MUST silently discard a datagram received with an invalid IP source address (e.g., a broadcast or multicast address). 4.1.4 UDP/APPLICATION LAYER INTERFACE The application interface to UDP MUST provide the full services of the IP/transport interface described in Section 3.4 of this Internet Engineering Task Force [Page 73] ***DRAFT RFC*** TRANSPORT LAYER -- UDP June 16, 1989 document. For example, an application using UDP needs the functions of the GET_INTERFACE, GET_MAXSIZES, ADVISE_DELIVPROB, and RECV_ICMP calls described in Section 3.4. For example, GET_MAXSIZES can be used to learn the effective maximum UDP maximum datagram size for a particular {interface,remote host,TOS} triplet. An application-layer program MUST be able to set the TTL and TOS values as well as IP options for sending a UDP datagram, and these values must be passed transparently to the IP layer. UDP MAY pass the received TOS up to the application layer. 4.1.5 UDP REQUIREMENTS SUMMARY | | | | |S| | | | | | |H| |F | | | | |O|M|o | | |S| |U|U|o | | |H| |L|S|t | |M|O| |D|T|n | |U|U|M| | |o | |S|L|A|N|N|t | |T|D|Y|O|O|t FEATURE |SECTION | | | |T|T|e -------------------------------------------------|--------|-|-|-|-|-|-- | | | | | | | UDP | | | | | | | -------------------------------------------------|--------|-|-|-|-|-|-- UDP send Port Unreachable |4.1.3.1 | |x| | | | IP Options in UDP | | | | | | | - Pass rcv'd IP options to applic layer |4.1.3.2 |x| | | | | - Applic layer can specify IP options in Send |4.1.3.2 |x| | | | | - UDP passes IP options down to IP layer |4.1.3.2 |x| | | | | Pass ICMP msgs up to applic layer |4.1.3.3 |x| | | | | UDP checksums: | | | | | | | - Able to generate/check checksum |4.1.3.4 |x| | | | | - Silently discard bad checksum |4.1.3.4 |x| | | | | - Sender Option to not generate checksum |4.1.3.4 | | |x| | | - Default is to checksum |4.1.3.4 |x| | | | | - Receiver Option to require checksum |4.1.3.4 | | |x| | | UDP Multihoming | | | | | | | - Pass dest IP addr for rcv'd UDP dg to applic |4.1.3.5 |x| | | | | - Applic layer can specify Local IP addr |4.1.3.5 |x| | | | | - Applic layer specify wild Local IP addr |4.1.3.5 |x| | | | | - Applic layer notified of Local IP addr used |4.1.3.5 | |x| | | | Silently discard bad IP source address |4.1.3.6 |x| | | | | Only send valid IP source address |4.1.3.6 |x| | | | | Internet Engineering Task Force [Page 74] ***DRAFT RFC*** TRANSPORT LAYER -- UDP June 16, 1989 UDP Application Interface Services | | | | | | | Full IP interface of 3.4 |4.1.4 |x| | | | | - Able to spec TTL, TOS, IP when send dg |4.1.4 |x| | | | | - Pass received TOS up to applic layer |4.1.4 | | |x| | | Internet Engineering Task Force [Page 75] ***DRAFT RFC*** TRANSPORT LAYER -- TCP June 16, 1989 4.2 TRANSMISSION CONTROL PROTOCOL -- TCP 4.2.1 INTRODUCTION The Transmission Control Protocol TCP [TCP:1] is the primary virtual-circuit transport protocol for the Internet suite. TCP provides reliable, in-sequence delivery of a full-duplex stream of octets (8-bit bytes). TCP is used by those applications needing reliable, connection-oriented transport service, e.g., mail (SMTP) file transfer (FTP), and virtual terminal service (Telnet); requirements for these protocols are described [INTRO:1]. 4.2.2 PROTOCOL WALK-THROUGH 4.2.2.1 Well-Known Ports: RFC-793 Section 2.7 DISCUSSION: TCP reserves port numbers in the range 1-255 for "well-known" ports, used to access services that are standardized across the Internet. The remainder of the port space can be freely allocated to application processes. Current well-known port definitions are listed in the RFC entitled "Assigned Numbers" [INTRO:5]. A prerequisite for defining a new well- known port is an RFC documenting the proposed service in enough detail to allow new implementations. Some systems extend this notion by adding a third subdivision of the TCP port space: reserved ports, which are generally used for operating-system-specific services. For example, reserved ports might fall between 256 and some system-dependent upper limit. Some systems further choose to protect well-known and reserved ports by permitting only privileged users to open TCP connections with those port values. This is perfectly reasonable as long as the host does not assume that all hosts protect their low-numbered ports in this manner. 4.2.2.2 Use of Push: RFC-793 Section 2.8 When an application issues a series of SEND calls without setting the PUSH flag, the TCP MAY aggregate the data internally without sending it. Similarly, when a series of segments is received without the PSH bit, a TCP MAY queue the data internally without passing it to the receiving application. Internet Engineering Task Force [Page 76] ***DRAFT RFC*** TRANSPORT LAYER -- TCP June 16, 1989 The PSH bit is not a record marker and is independent of segment boundaries. The transmitter SHOULD collapse successive PSH bits when it packetizes data, to send the largest possible segment. A TCP MAY implement PUSH flags on SEND calls. If they are not implemented, then the sending TCP must not buffer data indefinitely. Furthermore, such a TCP MUST set the PSH bit in the last buffered segment (i.e., when there is no more queued data to be sent). The discussion in RFC-793 on pages 48, 50, and 74 erroneously implies that a received PSH flag must be passed to the application layer. Passing a received PSH flag to the application layer is now OPTIONAL. An application program is logically required to set the PUSH flag in a SEND call whenever it needs to force delivery of the data to avoid a communication deadlock. However, a TCP SHOULD send a maximum-sized segment whenever possible, to improve performance (see Section 4.2.3.4). DISCUSSION: When the PUSH flag is not implemented on SEND calls, i.e., when the application/TCP interface uses a pure streaming model, responsibility for aggregating any | tiny data fragments to form reasonable sized segments | is partially borne by the application layer. | Generally, an interactive application protocol must set the PUSH flag at least in the last SEND call in each command or response sequence. A bulk transfer protocol like FTP should set the PUSH flag on the last segment of a file or when necessary to prevent buffer deadlock. At the receiver, the PSH bit forces buffered data to be delivered to the application. Conversely, the lack of a PSH bit can be used to avoid unnecessary wakeup calls to the application process; this can be an important performance optimization for large timesharing hosts. Passing the PSH bit to the receiving application allows an analogous optimization within the application. 4.2.2.3 Window Size: RFC-793 Section 3.1 The window size MUST be treated as an unsigned number, or else large window sizes will appear like negative windows and TCP will not work. Internet Engineering Task Force [Page 77] ***DRAFT RFC*** TRANSPORT LAYER -- TCP June 16, 1989 It is known that the window field in the TCP header is too @ small for high-speed, long-delay paths. An experimental TCP @ option has been defined to extend the window size [TCP:11]. @ In anticipation of the adoption of such an extension, it is @ RECOMMENDED that implementations reserve 32-bit fields for @ the send and receive window sizes in the connection record @ and do all window computations with 32 bits. @ 4.2.2.4 Urgent Pointer: RFC-793 Section 3.1 The second sentence is in error: the urgent pointer points to the sequence number of the LAST octet (not LAST+1) in a sequence of urgent data. The description on page 56 (last sentence) is correct. A TCP MUST support a sequence of urgent data of any length. | A TCP MUST inform the application layer asynchronously | whenever it receives an Urgent pointer and there was | previously no pending urgent data, or whenever the Urgent | pointer advances in the data stream. There MUST be a way | for the application to learn how much urgent data remains to | be read from the connection, or at least to determine | whether or not more urgent data remains to be read. | DISCUSSION: | Although the Urgent mechanism may be used for any | application, it is normally used to send "interrupt"- | type commands to a Telnet program (see "Using Telnet | Synch Sequence" section in HRUL). | The asynchronous or "out-of-band" notification will | allow the application to go into "urgent mode", reading | data from the TCP connection. This allows control | commands to be sent to an application whose normal | input buffers are full of unprocessed data. | IMPLEMENTATION: | The generic ERROR-REPORT() upcall described in Section | 4.2.4.1 is a possible mechanism for informing the | application of the arrival of urgent data. | 4.2.2.5 TCP Options: RFC-793 Section 3.1 A TCP MUST be able to receive a TCP option in a non-SYN segment. A TCP MUST be able to ignore without error any option it does not implement, assuming that the option has a length field (all TCP options defined in the future will Internet Engineering Task Force [Page 78] ***DRAFT RFC*** TRANSPORT LAYER -- TCP June 16, 1989 have length fields). TCP MUST be prepared to handle an illegal option length (e.g., zero) without crashing; a suggested procedure is to reset the connection and log the reason. 4.2.2.6 Maximum Segment Size Option: RFC-793 Section 3.1 TCP MUST implement both sending and receiving the Maximum Segment Size option [TCP:4]. TCP SHOULD send an MSS (Maximum Segment Size) option in every SYN segment when its receive MSS differs from the default 536, and may send it always. If an MSS option is not received at connection setup, TCP MUST assume a default send MSS of 536 (576-40) [TCP:4]. The maximum size of a segment that TCP really sends, the "effective send MSS," MUST be the smaller of the send MSS (which reflects the available reassembly buffer size at the remote host) and the largest size permitted by the IP layer: Eff.snd.MSS = min(SendMSS+20, MMS_S) - TCPhdrsize - IPoptionsize where: * SendMSS is the MSS value received from the remote host, or the default 536 if no MSS option is received. * MMS_S is the maximum size for a transport-layer message that TCP may send. * TCPhdrsize is the size of the TCP header; this is normally 20, but may be larger if TCP options are to be sent. * IPoptionsize is the size of any IP options that TCP will pass to the IP layer with the current message. The MSS value to be sent in an MSS option is: MMS_R - 20 where MMS_R is the maximum size for a transport-layer message that can be received (and reassembled). TCP obtains Internet Engineering Task Force [Page 79] ***DRAFT RFC*** TRANSPORT LAYER -- TCP June 16, 1989 MMS_R and MMS_S from the IP layer; see the generic call GET_MAXSIZES in Section 3.4. DISCUSSION: The choice of TCP segment size has a strong effect on performance. There are two competing effects: larger segments increase throughput by amortizing header size and per-datagram processing overhead over more data bytes; however, if the packet is so large that it causes IP fragmentation, efficiency drops sharply if any fragments are lost [IP:9]. Some TCP implementations send an MSS option only if the destination host is on a non-connected network. However, in general the TCP layer may not have the appropriate information to make this decision, so it is preferable to leave the task of determining a suitable MTU for the Internet path to the IP layer. We @ therefore recommend that TCP always send the option (if @ not 536) and that the IP layer determine MMS_R as @ specified in 3.3.3 and 3.4. A proposed IP-layer mechanism to measure the MTU would then modify the IP layer without changing TCP. 4.2.2.7 TCP Checksum: RFC-793 Section 3.1 | Unlike the UDP checksum (see Section 4.1.3.4), the TCP | checksum is never optional. The sender MUST generate it and | the receiver MUST check it. | 4.2.2.8 TCP Connection State Diagram: RFC-793 Section 3.2, page 23 There are several problems with this diagram: (a) The arrow from SYN-SENT to SYN-RCVD should be labeled with "snd SYN,ACK", to agree with the text on page 68 and with Figure 8. (b) There could be an arrow from SYN-RCVD state to LISTEN state, conditioned on receiving a RST after a passive open (see text page 70). (c) It is possible to go directly from FIN-WAIT-1 to the TIME-WAIT state (see page 75 of the spec). Internet Engineering Task Force [Page 80] ***DRAFT RFC*** TRANSPORT LAYER -- TCP June 16, 1989 4.2.2.9 Initial Sequence Number Selection: RFC-793 Section | 3.3, page 27 | A TCP MUST use the specified clock-driven selection of | initial sequence numbers. | 4.2.2.10 Simultaneous Open Attempts: RFC-793 Section 3.4, page 32 There is an error in Figure 8: the packet on line 7 should be identical to the packet on line 5. A TCP MUST support simultaneous open attempts. DISCUSSION: It sometimes surprises implementors that if two applications attempt to simultaneously connect to each other, only one connection is generated instead of two. This was an intentional design decision; don't try to "fix" it. 4.2.2.11 Recovery from Old Duplicate SYN: RFC-793 Section 3.4, page 33 Note that a TCP implementation MUST keep track of whether a connection has reached SYN_RCVD state as the result of a passive OPEN or an active OPEN. 4.2.2.12 RST Segment: RFC-793 Section 3.4 A TCP SHOULD allow a received RST segment to include data. | DISCUSSION It has been suggested that a RST segment could contain ASCII text that encoded and explained the cause of the RST. No standard has yet been established for such data. 4.2.2.13 Closing a Connection: RFC-793 Section 3.5 A TCP connection may terminate in two ways: (1) the normal | TCP Close sequence using a FIN handshake, and (2) an "abort" | in which one or more RST segments are sent and the | connection state is immediately discarded. If a TCP | connection is closed by the remote site, the local | application MUST be informed whether it closed normally or | was aborted. | Internet Engineering Task Force [Page 81] ***DRAFT RFC*** TRANSPORT LAYER -- TCP June 16, 1989 The normal TCP close sequence delivers data in the pipeline | in both directions. Some hosts implement only a half-duplex | TCP close sequence, i.e., an application that has called | CLOSE cannot continue to read data from the connection. If | such a host issues a CLOSE call while received data is still | pending in TCP, or if new data is received after CLOSE is | called, its TCP SHOULD send a RST to show that data was | lost. | When a connection is closed actively, it MUST linger in # TIME-WAIT state for a time 2*MSL (Maximum Segment Lifetime). # However, it MAY accept a new SYN from the remote TCP to # reopen the connection directly from TIME-WAIT state, if it: # (1) assigns its initial sequence number for the new # connection to be 1 greater than the largest sequence # number it used on the previous connection incarnation, # and # (2) returns to TIME-WAIT state if the SYN turns out to be # an old duplicate. # DISCUSSION: This full-duplex data-preserving close is a feature of TCP that is not included in the analogous ISO transport protocol TP4. Since the two directions of a TCP connection are closed independently, it is possible for a connection to be "half closed," i.e., closed in only one direction. It is legal for a host to continue sending data in the open direction on a half-closed connection. However, some systems have not implemented half-closed connections, presumably because they do not fit into the I/O model of their particular operating system. On these systems, once an application has called CLOSE, it can no longer read input data from the connection; this is referred to as a "half-duplex" TCP close sequence. The graceful close algorithm of TCP requires that the @ connection state remain defined on (at least) one end @ of the connection, for a timeout period of twice MSL, @ i.e., 4 minutes. During this period, the (remote socket, local socket) pair that defines the connection is busy and cannot be reused. To shorten the time that a given port pair is tied up, Internet Engineering Task Force [Page 82] ***DRAFT RFC*** TRANSPORT LAYER -- TCP June 16, 1989 some TCPs allow a new SYN to be accepted in TIME-WAIT state. 4.2.2.14 Data Communication: RFC-793 Section 3.7, page 40 Since RFC-793 was written, there has been extensive work on TCP algorithms to achieve efficient data communication. Later sections of the present document describe required and recommended TCP algorithms to determine when to send data (Section 4.2.3.4), when to send an acknowledgment (Section 4.2.3.2), and when to update the window (Section 4.2.3.3). DISCUSSION: One important performance issue is "Silly Window Syndrome" or "SWS" [TCP:5], a stable pattern of small incremental window movements resulting in extremely poor TCP performance. Algorithms to avoid SWS are described below for both the sending side (Section 4.2.3.4) and the receiving side (Section 4.2.3.3). In brief, SWS is caused by the receiver advancing the right window edge whenever it has any new buffer space available to receive data and by the sender using any incremental window, no matter how small, to send more data [TCP:5]. The result can be a stable pattern of sending tiny data segments, even though both sender and receiver have a large total buffer space for the connection. SWS can only occur during the transmission of a large amount of data; if the connection goes quiescent, the problem will disappear. It is caused by typical straightforward implementation of window management, but the sender and receiver algorithms given below will avoid it. Another important TCP performance issue is that some applications, especially remote login for character- at-a-time hosts, tend to send streams of one-octet data segments. To avoid deadlocks, these applications must specify the "push" flag in every send call to TCP, and the result may be a stream of TCP segments each containing one data octet. This makes very inefficient use of the Internet and contributes to Internet congestion. The Nagle Algorithm described in Section 4.2.3.4 provides a simple and effective solution to this problem. It does have the effect of clumping characters over Telnet connections; this may initially surprise users accustomed to single-character echo, but user acceptance has not been a problem. Internet Engineering Task Force [Page 83] ***DRAFT RFC*** TRANSPORT LAYER -- TCP June 16, 1989 Note that the Nagle algorithm and the send SWS avoidance algorithm play complementary roles in improving performance. The Nagle algorithm discourages sending tiny segments when the data to be sent increases in small increments, while the SWS avoidance algorithm discourages small segments resulting from the right window edge advancing in small increments. A careless implementation can send two or more acknowledgment segments per data segment received. Thus, suppose the receiver acknowledges every data segment immediately. When the application program subsequently consumes the data and increases the available receive buffer space again, the receiver may send a second acknowledgment segment to update the window at the sender. The extreme case occurs with single-character segments on TCP connections using the Telnet protocol for remote login service. Some implementations have been observed in which each incoming 1-character segment generates three return segments: (1) the acknowledgment, (2) a one byte increase in the window, and (3) the echoed character, respectively. 4.2.2.15 Retransmission Timeout: RFC-793 Section 3.7, page 41 The algorithm suggested in RFC-793 for calculating the retransmission timeout is now known to be inadequate; see Section 4.2.3.1 below. Recent work by Jacobson [TCP:7] on Internet congestion and TCP retransmission stability has produced a transmission algorithm combining "slow start" with "congestion avoidance." A TCP MUST implement this algorithm. If a retransmitted packet is identical to the original packet (which implies not only that the data boundaries have not changed, but also that the window and acknowledgment fields of the header have not changed), then the same IP Identification field MAY be used (see Section 3.2.1.5). IMPLEMENTATION: Some TCP implementors have chosen to "packetize" the data stream, i.e., to pick segment boundaries when segments are originally sent and to queue these segments in a "retransmission queue" until they are acknowledged. Another design (which may be simpler) is to defer packetizing until each time data is Internet Engineering Task Force [Page 84] ***DRAFT RFC*** TRANSPORT LAYER -- TCP June 16, 1989 transmitted or retransmitted, so there will be no segment retransmission queue. In an implementation with a segment retransmission queue, TCP performance may be enhanced by repacketizing the segments awaiting acknowledgment when the first retransmission timeout occurs. That is, the outstanding segments that fitted would be combined into one maximum-sized segment, with a new IP Identification value. The TCP would then retain this combined segment in the retransmit queue until it was acknowledged. However, if the first two segments in the retransmission queue totalled more than one maximum- sized segment, the TCP would retransmit only the first segment using the original IP Identification field. 4.2.2.16 Managing the Window: RFC-793 Section 3.7, page 41 A TCP receiver MUST NOT shrink the window, i.e., move the right window edge to the left. However, a sending TCP MUST be robust against window shrinking, which may cause the "useable window" (see Section 4.2.3.3) to become negative. If this happens, the sender SHOULD continue retransmitting but not send any new data, until the useable window again becomes positive. However, if the window shrinks to zero, the TCP MUST probe it in the standard way (see next Section). DISCUSSION: RFC-793 said that shrinking the window is "strongly discouraged." Later experience has led to the conclusion that it should be banned altogether. 4.2.2.17 Probing Zero Windows: RFC-793 Section 3.7, page 42 Probing of zero (offered) windows MUST be supported. | Note that a TCP MAY keep its offered receive window closed | indefinitely. As long as the receiving TCP continues to | send acknowledgments in response to the probe segments, the | sending TCP MUST allow the connection to stay open. | DISCUSSION: It is extremely important to remember that ACK (acknowledgment) segments that contain no data are not reliably transmitted by TCP. If zero window probing is not supported, a connection may hang forever when an ACK segment that re-opens the window is lost. Internet Engineering Task Force [Page 85] ***DRAFT RFC*** TRANSPORT LAYER -- TCP June 16, 1989 The delay in opening a zero window generally occurs when the receiving application stops taking data from its TCP. For example, consider a printer daemon application, stopped because the printer ran out of paper. 4.2.2.18 Passive OPEN Calls: RFC-793 Section 3.8 Every passive OPEN call either creates a new connection record in LISTEN state, or it returns an error; it MUST NOT affect any previously created connection record. A TCP that supports multiple concurrent users MUST provide an OPEN call that will functionally allow an application to LISTEN on a port while a connection block with the same local port is in SYN-SENT or SYN-RECEIVED state. DISCUSSION: Some applications (e.g., SMTP servers) may need to handle multiple connection attempts at about the same time. The probability of a connection attempt failing is reduced by giving the application some means of listening for a new connection at the same time that an earlier connection attempt is going through the three- way handshake. IMPLEMENTATION: Acceptable implementations of concurrent opens may permit multiple passive OPEN calls or may allow this feature to be selected in a single passive OPEN call. 4.2.2.19 Queueing Out-of-Order Segments: RFC-793 Section 3.9 While it is not strictly required, a TCP SHOULD be capable of queueing out-of-order TCP segments. Change the "may" in the last sentence of the first paragraph on page 70 to "should." DISCUSSION: Some small-host implementations have omitted segment queueing because of limited buffer space. This omission may be expected to adversely affect TCP throughput, since loss of a single segment causes all later segments to appear to be "out of sequence." Internet Engineering Task Force [Page 86] ***DRAFT RFC*** TRANSPORT LAYER -- TCP June 16, 1989 4.2.2.20 Event Processing: RFC-793 Section 3.9 In general, the processing of received segments MUST be implemented to aggregate ACK segments whenever possible. For example, if the TCP is processing a series of queued segments, it MUST process them all before sending any ACK segments. Here are some detailed error corrections and notes on the event processing section. (a) CLOSE Call, CLOSE-WAIT state, p. 61: enter LAST-ACK state, not CLOSING. (b) LISTEN state, check for SYN (pp. 65, 66): With a SYN bit, if the security/compartment or the precedence is wrong for the segment, a reset is sent. The wrong form of reset is shown in the text; it should be: <SEQ=0><ACK=SEG.SEQ+SEG.LEN><CTL=RST,ACK> (c) SYN-SENT state, Check for SYN, p. 68: When the connection enters ESTABLISHED state, the following variables must be set: SND.WND <- SEG.WND SND.WL1 <- SEG.SEQ SND.WL2 <- SEG.ACK (d) Check security and precedence, p. 71: The first heading "ESTABLISHED STATE" should really be a list of all states other than SYN-RECEIVED: ESTABLISHED, FIN-WAIT- 1, FIN-WAIT-2, CLOSE-WAIT, CLOSING, LAST-ACK, and TIME-WAIT. (e) Check SYN bit, p. 71: In SYN-RECEIVED state and if the connection was initiated with a passive OPEN, then return this connection to the LISTEN state and return. Otherwise... (f) Check ACK field, SYN-RECEIVED state, p. 72: When the connection enters ESTABLISHED state, the variables listed in (c) must be set. (g) Check ACK field, ESTABLISHED state, p. 72: The ACK is a duplicate if SEG.ACK =< SND.UNA (the = was omitted). Similarly, the window should be updated if: SND.UNA =< Internet Engineering Task Force [Page 87] ***DRAFT RFC*** TRANSPORT LAYER -- TCP June 16, 1989 SEG.ACK =< SND.NXT. (h) USER TIMEOUT, p. 77: It would be better to notify the application of the timeout rather than letting TCP force the connection closed. However, see also Section 4.2.3.5. 4.2.2.21 Acknowledging Queued Segments: RFC-793 Section 3.9 A TCP MAY send an ACK segment acknowledging RCV.NXT for a valid segment that is in the window but not at the left window edge. DISCUSSION: RFC-793 (see page 74) was ambiguous about whether or @ not an ACK segment should be sent when an out-of-order @ segment was received, i.e., when SEG.SEQ was unequal to @ RCV.NXT. @ One reason for ACKing out-of-order segments might be to @ support an experimental algorithm known as "fast @ retransmit." This algorithm uses the "redundant" @ ACK's to deduce that a segment has been lost before the @ retransmission timer has expired. It counts the number @ of times an ACK has been received with the same value @ of SEG.ACK and with the same right window edge. If @ more than a threshold number of such ACK's is received, @ then the segment containing the octets starting at @ SEG.ACK is assumed to have been lost and is @ retransmitted, without awaiting a timeout. The @ threshold is intended to compensate for reordering of @ segments. There is not yet enough experience with this @ algorithm to determine how useful it is. @ 4.2.3 SPECIFIC ISSUES 4.2.3.1 Retransmission Timeout Calculation A host TCP MUST implement Karn's algorithm and Jacobson's algorithm for computing the retransmission timeout ("RTO"). o Jacobson's algorithm for computing the smoothed round- trip ("RTT") time incorporates a simple measure of the variance [TCP:7]. o Karn's algorithm for selecting RTT measurements ensures Internet Engineering Task Force [Page 88] ***DRAFT RFC*** TRANSPORT LAYER -- TCP June 16, 1989 that corrupted round-trip times will not be used in the calculation of the smoothed round-trip time [TCP:6]. This implementation also MUST include "exponential backoff" for successive RTO values for the same segment. Retransmission of SYN segments SHOULD use the same algorithm as data segments. The following values SHOULD be used to initialize the estimation parameters for a new connection: (a) RTT = 0 seconds. (b) RTO = 3 seconds. (The smoothed variance should be initialized to the value that will result in this RTO). The recommended upper and lower bounds on the RTO are known to be inadequate on large internets. The lower bound SHOULD be measured in fractions of a second (to accommodate high speed LANs) and the upper bound should be MSL, i.e., 120 seconds. DISCUSSION: There were two known problems with the RTO calculations specified in RFC-793. First, the accurate measurement of RTTs is difficult when there are retransmissions. Second, the algorithm to compute the smoothed round- trip time is inadequate [TCP:7], because it incorrectly assumed that the variance in RTT values would be small and constant. These problems were solved by Karn's and Jacobson's algorithm, respectively. The performance increase resulting from the use of these improvements varies from noticeable to dramatic. Jacobson's algorithm for incorporating the measured RTT variance is especially important on a low-speed link, where the natural variation of packet sizes causes a large variation in RTT. One vendor found link utilization on a 9.6kb line went from 10% to 90% as a result of implementing Jacobson's variance algorithm in TCP. Experience has shown that the specified initialization values are reasonable, and that the Karn and Jacobson algorithms make TCP behavior reasonably insensitive to the initial parameter choices. Internet Engineering Task Force [Page 89] ***DRAFT RFC*** TRANSPORT LAYER -- TCP June 16, 1989 4.2.3.2 When to Send an ACK Segment A host that is receiving a stream of TCP data segments can increase efficiency in both the Internet and the hosts by sending fewer than one ACK (acknowledgment) segment per data segment received; this is known as a "delayed ACK" [TCP:5]. A TCP SHOULD implement a delayed ACK, but an ACK MUST be | sent when the useable window is equal to or exceeds twice | the effective send MSS, and the delay MUST be 0.5 seconds or | less. | DISCUSSION: A delayed ACK gives the application an opportunity to update the window and perhaps to send an immediate response. In particular, in the case of character-mode remote login, a delayed ACK can reduce the number of segments sent by the server by a factor of 3 (ACK, window update, and echo character all combined in one segment). In addition, on some large multi-user hosts, a delayed ACK can substantially reduce protocol processing overhead by reducing the total number of packets to be processed [TCP:5]. However, excessive delays on ACKs can disturb the round-trip timing and packet "clocking" algorithms that are necessary to handle congestion [TCP:7]. 4.2.3.3 When to Send a Window Update A TCP MUST include a SWS avoidance algorithm in the receiver [TCP:5]. DISCUSSION: The receiver's SWS avoidance algorithm determines when the right window edge may be advanced; this is customarily known as "updating" the window. This algorithm combines with the delayed ACK algorithm (see Section 4.2.3.2) to determine when an ACK segment containing the current window will really be sent to the receiver. We use the notation of RFC-793; see Figures 4 and 5 in that document. The solution is to avoid advancing the right window edge RCV.NXT+RCV.WND in small increments, even if data is received from the network in small segments. Internet Engineering Task Force [Page 90] ***DRAFT RFC*** TRANSPORT LAYER -- TCP June 16, 1989 Suppose the total receive buffer space is RCV.BUFF. At any given moment, RCV.USER octets of this total may be tied up with data that has been received and acknowledged but which the user process has not yet consumed. When the connection is quiescent, RCV.WND = RCV.BUFF and RCV.USER = 0. Keeping the right window edge fixed as data arrives and is acknowledged, i.e., as RCV.NXT increases, requires that we "artificially reduce the offered window" [TCP:5] RCV.WND to keep RCV.NXT+RCV.WND constant. Thus, the total buffer space RCV.BUFF is generally divided into three parts: |<------- RCV.BUFF ---------------->| 1 2 3 ----|---------|------------------|------|---- RCV.NXT ^ (Fixed) 1 - RCV.USER = data received but not yet consumed; 2 - RCV.WND = space advertised to sender; 3 - Reduction = space available but not yet advertised. The suggested SWS avoidance algorithm for the receiver is to keep RCV.NXT+RCV.WND fixed until the reduction satisfies: RCV.BUFF - RCV.USER - RCV.WND >= min( Fr * RCV.BUFF, Eff.snd.MSS ) where Fr is a fraction whose recommended value is 1/2, and Eff.snd.MSS is the effective send MSS for the connection (see Section 4.2.2.6). When the inequality is satisfied, RCV.WND is set to RCV.BUFF-RCV.USER. Note that the general effect of this algorithm is to @ advance RCV.WND in increments of Eff.snd.MSS (for @ realistic receive buffers: RCV.BUFF >= Eff.snd.MSS/2). @ 4.2.3.4 When to Send Data A TCP MUST include a SWS avoidance algorithm in the sender. Internet Engineering Task Force [Page 91] ***DRAFT RFC*** TRANSPORT LAYER -- TCP June 16, 1989 A TCP SHOULD implement the Nagle Algorithm [TCP:9] to coalesce short segments. However, there MUST be a way for an application to disable the Nagle algorithm on an individual connection. In all cases, sending data is also subject to the limitation imposed by the Slow Start algorithm (Section 4.2.2.15). DISCUSSION: The Nagle algorithm is generally as follows: If there is unacknowledged data (i.e., SND.NXT > SND.UNA), then the sending TCP buffers all user data (regardless of the PSH bit), until the outstanding data has been acknowledged or until the TCP can send a full-sized segment (effective send MSS or Eff.snd.MSS bytes; see Section 4.2.2.6). Some applications (e.g., real-time display window updates) require that the Nagle algorithm be turned off, so small data segments can be streamed out at maximum rate. The sender's SWS avoidance algorithm is more difficult than the receivers's, because the sender does not know (directly) the receiver's total buffer space RCV.BUFF. An approach which has been found to work well is for the sender to calculate Max(SND.WND), the maximum send window it has seen so far on the connection, and to use this value as an estimate of RCV.BUFF. Unfortunately, this can only be an estimate; the receiver may at any time reduce the size of RCV.BUFF. To avoid a resulting deadlock, it is necessary to have a timeout to force transmission of data, overriding the SWS avoidance algorithm. In practice, this timeout should seldom occur. The "useable window" [TCP:5] is: U = SND.UNA + SND.WND - SND.NXT i.e., the offered window less the amount of data sent but not acknowledged. If D is the amount of data queued in the sending TCP but not yet sent, then the following set of rules are recommended. Send data: Internet Engineering Task Force [Page 92] ***DRAFT RFC*** TRANSPORT LAYER -- TCP June 16, 1989 (1) if a maximum-sized segment can be sent, i.e, if: D >= U >= Eff.snd.MSS; (2) or if the data is pushed and all queued data can be sent now, i.e., if: [SND.NXT = SND.UNA and] PUSHED and D <= U (the bracketed condition is imposed by the Nagle algorithm); (3) or if the useable window is at least a fraction Fs of the maximum window, i.e., if: D >= U >= Fs * Max(SND.WND); (4) or if data is PUSHed and the override timeout occurs. Here Fs is a fraction whose recommended value is 1/2. The override timeout should be in the range 0.1 - 1.0 seconds. Finally, note that the SWS avoidance algorithm just | specified is to be used instead of the sender-side | algorithm contained in [TCP:5]. | IMPLEMENTATION: | It may be convenient to combine this timer with the timer used to probe zero windows (Section 4.2.2.17). 4.2.3.5 TCP Connection Liveness Excessive retransmission of the same segment by TCP indicates some failure of the remote host or the Internet path. This failure may be of short or long duration. The following procedure MUST be used to handle excessive retransmissions of data segments [IP:11]: (a) There are two thresholds R1 and R2 measuring the amount of retransmission that has occurred for the same segment. It is RECOMMENDED that R1 and R2 be measured in terms of a count of retransmissions, although time Internet Engineering Task Force [Page 93] ***DRAFT RFC*** TRANSPORT LAYER -- TCP June 16, 1989 could also be used. (b) When the number of transmissions of the same segment reaches or exceeds a threshold R1, call the ADVISE_DELIVPROB entry to inform the IP layer, triggering dead-gateway diagnosis. Assuming R1 is a count, its value SHOULD be 3 or greater. (c) When the number of transmissions of the same segment reaches a threshold R2, close the connection. Here R2 is greater than R1. (d) An application MUST be able to set the value for R2 for a particular connection. For example, an interactive application might set R2 = "infinity," giving the user the control over when to disconnect. | (d) TCP MUST inform the application of the delivery problem | (unless such information has been disabled by the | application; see Section 4.2.4.1), after R1 and before | R2 is reached. This will allow a remote login (User | Telnet) application program to inform the user, for | example. | Implementors MAY include "keep-alives" in their TCP | implementations, although this practice is not universally | accepted. If keep-alives are included, the application MUST | be able to turn them on or off for each TCP connection, and | they MUST default to off. | Keep-alive packets MUST NOT be sent when any data or | acknowledgement packets have been received for the | connection within a configurable interval; this interval | MUST default to no less than two hours. | An implementation SHOULD send a keep-alive segment with no | data; however, it MAY be configurable to send a keep-alive | segment containing one garbage octet, for compatibililty | with erroneous TCP implementations. | DISCUSSION: | A "keep-alive" mechanism would periodically probe the | other end of a connection when the connection was | otherwise idle, even when there was no data to be sent. | The TCP specification does not include a keep- alive | Internet Engineering Task Force [Page 94] ***DRAFT RFC*** TRANSPORT LAYER -- TCP June 16, 1989 mechanism because it could: (1) cause perfectly good | connections to break during transient Internet | failures; (2) consume unnecessary bandwidth ("if no one | is using the connection, who cares if it is still | good?"); and (3) cost money for an Internet path that | charges for packets. | Some TCP implementations, however, have included a | keep-alive mechanism. To confirm that an idle | connection is still active, these implementations send | a probe segment designed to elicit a response from the | peer TCP. Such a segment generally contains SEG.SEQ = | SND.NXT-1. The segment may or may not contain one | garbage octet of data. Note that on a quiet | connection, SND.NXT = RCV.NXT and SEG.SEQ will be | outside the window. Therefore, the probe causes the | receiver to return an acknowledgment segment, | confirming that the connection is still live. If the | peer has dropped the connection due to a network | partition or a crash, it will respond with a reset | instead of an acknowledgement. | Unfortunately, some misbehaved TCP implementations fail | to respond to a segment with SEG.SEQ = SND.NXT-1 unless | the segment contains data. Alternatively, an | implementation could determine whether a peer responded | correctly to keep-alive packets with no garbage data | octet. | A TCP keep-alive mechanism should only be invoked in | network servers that might otherwise hang indefinitely | and consume resources unnecessarily if a client crashes | or aborts a connection during a network partition. | 4.2.3.6 TCP Open Failure An attempt to open a TCP connection could fail with | excessive transmissions of the SYN segment or by receipt of | a RST segment or an ICMP Port Unreachable. SYN | retransmissions MUST be handled in the general way just | described for data retransmissions, including notification | of the application layer. | However, the values of R1 and R2 may be different for retransmissions of SYN segments and data segments. In particular, R2 for a SYN segment MUST be set large enough to provide retransmission of the segment for 2-3 minutes. The application can close the connection (i.e., give up on the Internet Engineering Task Force [Page 95] ***DRAFT RFC*** TRANSPORT LAYER -- TCP June 16, 1989 open attempt) sooner, of course. DISCUSSION: Some Internet paths have significant setup times, and the number of such paths is likely to increase in the future. 4.2.3.7 TCP Multihoming DISCUSSION: | A TCP in a multihomed host needs to select a local IP | address when it is sending the (first) SYN for an | active connection request by a local user. At all | other times, a previous segment has either been sent or | received on this connection and the same local address | is used as was used in those previous segments. | 4.2.3.8 IP Options When received options are passed up to TCP from the IP | layer, TCP MUST ignore options that it does not understand. | A TCP MAY support the Time Stamp and Record Route options. | An application MUST be able to specify a source route when | it actively opens a TCP connection, and this MUST take | precedence over a source route received in a datagram. | When a TCP connection is OPENed passively and a packet | arrives with a completed IP Source Route option (containing | a return route), TCP MUST save the return route and use it | for all segments sent on this connection. If a different | source route arrives in a later segment, the later | definition SHOULD override the earlier one. | 4.2.3.9 ICMP Messages TCP MUST act on an ICMP error message passed up from the IP layer, directing it to the connection that created the error. The necessary demultiplexing information can be found in the IP header contained within the ICMP message. o Source Quench TCP MUST react to a Source Quench by slowing transmission on the connection. The recommended Internet Engineering Task Force [Page 96] ***DRAFT RFC*** TRANSPORT LAYER -- TCP June 16, 1989 procedure is for a Source Quench to trigger a "slow start," as if a retransmission timeout had occurred. | o Destination Unreachable -- codes 0, 1, 5 | Since these Unreachable messages indicate soft error | conditions, TCP MUST NOT abort the connection, and it | SHOULD make the information available to the | application. | It MAY report it directly to the application layer with | an upcall to the ERROR_REPORT routine; alternatively, | it MAY merely note the message and report it to the | application only when and if the TCP connection times | out. | o Destination Unreachable -- codes 2-4 | These are hard error conditions, so TCP SHOULD abort | the connection. o Time Exceeded (Codes 0, 1) This should be handled the same way as Destination Unreachable codes 0, 1 (see above). o Parameter Problem This should be handled the same way as Destination Unreachable codes 0, 1 (see above). 4.2.3.10 Remote Address Validation A TCP implementation MUST reject as an error an OPEN request | to an invalid IP address (e.g., a broadcast or multicast | address). It MUST also silently ignore an incoming SYN with | an invalid source address. | A TCP implementation MUST silently ignore an incoming SYN segment that is addressed to a broadcast or multicast address. 4.2.3.11 TCP Traffic Patterns The TCP protocol specification [TCP:1] gives the implementor much freedom in designing the algorithms that control the Internet Engineering Task Force [Page 97] ***DRAFT RFC*** TRANSPORT LAYER -- TCP June 16, 1989 message flow over the connection -- packetizing, managing the window, sending acknowledgments, etc. These design decisions are difficult because a TCP must adapt to a wide range of traffic patterns. Experience has shown that a TCP implementor needs to verify the design on two extreme traffic patterns: o Single-character Segments Even if the sender is using the Nagle Algorithm, when a TCP connection carries remote login traffic across a low-delay LAN the receiver will generally get a stream of single-character segments. If remote terminal echo mode is in effect, the receiver's system will generally echo each character as it is received. o Bulk Transfer When TCP is used for bulk transfer, the data stream should be made up (almost) entirely of segments of the size of the effective MSS. Although TCP uses a sequence number space with byte (octet) granularity, in bulk-transfer mode its operation should be as if TCP used a sequence space that counted only segments. Experience has furthermore shown that a single TCP can effectively and efficiently handle these two extremes. The most important tool for verifying a new TCP implementation is a packet trace program. There is a large volume of experience showing the importance of tracing a variety of traffic patterns with other TCP implementations and studying the results carefully. 4.2.3.12 Efficiency IMPLEMENTATION: Extensive experience has led to the following suggestions for efficient implementation of TCP: (a) Don't Copy Data In bulk data transfer, the primary CPU-intensive @ tasks are copying data from one place to another @ and checksumming the data. It is vital to @ minimize the number of copies of TCP data. Since @ Internet Engineering Task Force [Page 98] ***DRAFT RFC*** TRANSPORT LAYER -- TCP June 16, 1989 the ultimate speed limitation may be fetching data @ across the memory bus, it may be useful to combine @ the copy with checksumming, doing both with a @ single memory fetch. (b) Hand-Craft the Checksum Routine A good TCP checksumming routine is typically two to five times faster than a simple and direct implementation of the definition. Great care and clever coding are often required and advisable to make the checksumming code "blazing fast." See [TCP:10]. (c) Code for the Common Case TCP protocol processing can be complicated, but for most segments there are only a few simple decisions to be made. Per-segment processing will be greatly speeded up by coding the main line to minimize the number of decisions in the most common case. 4.2.4 TCP/APPLICATION LAYER INTERFACE 4.2.4.1 Asynchronous Reports There MUST be a mechanism for reporting soft TCP error conditions to the application. Generically, we assume this takes the form of an application-supplied ERROR_REPORT routine that may be upcalled [INTRO:7] asynchronously from the transport layer: ERROR_REPORT(local connection name, reason, subreason) The precise encoding of the reason and subreason parameters is not specified here. However, the conditions that are reported asynchronously to the application MUST include: * ICMP error message arrived (see 4.2.3.9) * Excessive retransmissions (see 4.2.3.5) # * Urgent pointer advance (see 4.2.2.4). # This same upcall MAY be used to report the existence of Internet Engineering Task Force [Page 99] ***DRAFT RFC*** TRANSPORT LAYER -- TCP June 16, 1989 pending urgent data (see Section 4.2.2.4). However, an application program that does not want to receive such ERROR_REPORT calls SHOULD be able to effectively disable these calls. DISCUSSION: These error reports generally reflect soft errors that can be ignored without harm by many applications. It has been suggested that these error report calls should default to "disabled," but this is not required. 4.2.4.2 Type-of-Service The application layer MUST be able to specify the Type-of- Service (TOS) for segments that are sent on a connection. It not required, but the application SHOULD be able to change the TOS during the connection lifetime. TCP SHOULD pass the current TOS value without change to the IP layer, when it sends segments on the connection. The TOS will be specified independently in each direction on the connection, so that the receiver application will specify the TOS used for ACK segments. TCP MAY pass the most recently received TOS up to the application. DISCUSSION Some applications (e.g., SMTP) change the nature of their communication during the lifetime of a connection, and therefore would like to change the TOS specification. Note also that the OPEN call specified in RFC-793 includes a parameter ("options") in which the caller can specify IP options such as source route, record route, or timestamp. 4.2.4.3 Flush Call Some TCP implementations have included a FLUSH call, which will empty the TCP send queue of any data for which the user has issued SEND calls but which is still to the right of the current send window. That is, it flushes as much queued send data as possible without losing sequence number synchronization. This is useful for implementing the "abort output" function of Telnet. Internet Engineering Task Force [Page 100] ***DRAFT RFC*** TRANSPORT LAYER -- TCP June 16, 1989 4.2.4.4 Multihoming ! The user interface outlined in sections 2.7 and 3.8 of RFC- ! 793 needs to be extended for multihoming. The OPEN call ! MUST have an optional parameter: ! OPEN( ... [local IP address,] ... ) ! that can specify a logical network interface on the local ! host. ! DISCUSSION: ! Some TCP-based applications need to specify the logical ! interface to be used to open a particular connection; ! FTP is an example. ! A passive OPEN call with a specified "local IP address" ! parameter will await an incoming connection request to ! that address. If the parameter is unspecified, a ! passive OPEN will await an incoming connection request ! to any local IP address, and then bind the local IP ! address of the connection to the particular address ! that is used. ! For an active OPEN call, a specified "local IP address" ! parameter will be used for opening the connection. If ! the parameter is unspecified, the networking software ! will choose an appropriate local IP address (see ! Section 3.3.4.2) for the connection ! 4.2.5 TCP REQUIREMENT SUMMARY | | | | |S| | | | | | |H| |F | | | | |O|M|o | | |S| |U|U|o | | |H| |L|S|t | |M|O| |D|T|n | |U|U|M| | |o | |S|L|A|N|N|t | |T|D|Y|O|O|t FEATURE |SECTION | | | |T|T|e -------------------------------------------------|--------|-|-|-|-|-|-- | | | | | | | Push flag | | | | | | | Aggregate or queue un-pushed data |4.2.2.2 | | |x| | | Sender collapse successive PSH flags |4.2.2.2 | |x| | | | SEND call can specify PUSH |4.2.2.2 | | |x| | | Internet Engineering Task Force [Page 101] ***DRAFT RFC*** TRANSPORT LAYER -- TCP June 16, 1989 Set PSH bits in segment as needed |4.2.2.2 |x| | | | | Notify receiving ALP of PSH |4.2.2.2 | | |x| | |1 | | | | | | | Window | | | | | | | Treat as unsigned number |4.2.2.3 |x| | | | | Handle as 32-bit number |4.2.2.3 | |x| | | | Shrink window from right |4.2.2.16| | | | |x| Robust against shrinking window |4.2.2.16|x| | | | | Sender probe zero window |4.2.2.17|x| | | | | Allow window stay zero indefinitely |4.2.2.17|x| | | | | Sender timeout OK conn with zero wind |4.2.2.17| | | | |x| | | | | | | | Urgent Data | | | | | | | Pointer points to last octet |4.2.2.4 |x| | | | | Arbitrary length urgent data sequence |4.2.2.4 |x| | | | | Inform ALP asynchronously of urgent data |4.2.2.4 |x| | | | | ALP can learn if/how much urgent data Q'd |4.2.2.4 |x| | | | | | | | | | | | TCP Options | | | | | | | Receive TCP option in a non-SYN segment |4.2.2.5 |x| | | | | Ignore unsupported options |4.2.2.5 |x| | | | | Implement sending & receiving MSS option |4.2.2.6 |x| | | | | Send MSS option unless 536 |4.2.2.6 | |x| | | | Send-MSS default is 536 |4.2.2.6 |x| | | | | Calculate effective send seg size |4.2.2.6 |x| | | | | | | | | | | | TCP Checksums | | | | | | | Sender compute checksum |4.2.2.7 |x| | | | | Receiver check checksum |4.2.2.7 |x| | | | | Use clock-driven ISN selection |4.2.2.9 |x| | | | | | | | | | | | Opening Connections | | | | | | | Support simultaneous open attempts |4.2.2.10|x| | | | | SYN-RCVD remembers last state |4.2.2.11|x| | | | | Passive Open call interfere with others |4.2.2.18| | | | |x| Function: simultan. LISTENs for same port |4.2.2.18|x| | | | | Persistent SYN retransmission |4.2.3.6 |x| | | | | OPEN to broadcast/multicast IP Address |4.2.3.14| | | | |x| Silently discard seg to bcast/mcast addr |4.2.3.14|x| | | | | | | | | | | | Closing Connections | | | | | | | RST can contain data |4.2.2.10| |x| | | | Inform application of aborted conn |4.2.2.13|x| | | | | Support half-closed connections |4.2.2.13| | |x| | | Send RST to indicate data lost |4.2.2.13| |x| | | | Graceful Close waits 2xMSL seconds |4.2.2.13|x| | | | | | | | | | | | Retransmissions | | | | | | | Internet Engineering Task Force [Page 102] ***DRAFT RFC*** TRANSPORT LAYER -- TCP June 16, 1989 Jacobson Slow Start algorithm |4.2.2.15|x| | | | | Jacobson Congestion-Avoidance algorithm |4.2.2.15|x| | | | | Retransmit with same IP ident |4.2.2.15| | |x| | | Karn's algorithm |4.2.3.1 |x| | | | | Jacobson's RTO estimation alg. |4.2.3.1 |x| | | | | Exponential backoff |4.2.3.1 |x| | | | | SYN RTO calc same as data |4.2.3.1 | |x| | | | | | | | | | | Generating ACK's: | | | | | | | Queue out-of-order segments |4.2.2.19| |x| | | | Process all Q'd before send ACK |4.2.2.20|x| | | | | Send ACK for queued segment |4.2.2.21| | |x| | | Delayed ACK's |4.2.3.2 | |x| | | | Limit on Delaying of ACK's |4.2.3.2 |x| | | | | Receiver SWS-Avoidance Algorithm |4.2.3.3 |x| | | | | | | | | | | | Sending data | | | | | | | Send max-size seg even if not PSH |4.2.2.2 | |x| | | | Sender SWS-Avoidance Algorithm |4.2.3.4 |x| | | | | Nagle algorithm |4.2.3.4 | |x| | | | Application can disable Nagle algorithm |4.2.3.4 |x| | | | | | | | | | | | Retransmission Thresholds: | | | | | | | Use counts, not time |4.2.3.5 | |x| | | | Advise IP on R1 retxs |4.2.3.5 |x| | | | | Close connection on R2 retxs |4.2.3.5 |x| | | | | User-settable R2 |4.2.3.5 |x| | | | | Inform ALP of R1<retxs<R2 |4.2.3.5 | |x| | | |1 | | | | | | | Keep-alive Packets: | | | | | | | - Support for... |4.2.3.5 | | | |x| | - Application can request |4.2.3.5 | | |x| | | - Default is "off" |4.2.3.5 |x| | | | | | | | | | | | IP Options | | | | | | | Time Stamp support |4.2.3.8 | | |x| | | Record Route support |4.2.3.8 | | |x| | | Source Route: | | | | | | | ALP can specify |4.2.3.8 |x| | | | | Build return route from src rt |4.2.3.8 |x| | | | | Later src route overrides |4.2.3.8 | |x| | | | | | | | | | | Receiving ICMP Messages from IP | | | | | | | Dest. Unreach (0,1) => inform ALP |4.2.3.9 | |x| | | | Dest. Unreach (0,1) => abort conn |4.2.3.9 | | | | |x| Dest. Unreach (2-5) => abort conn |4.2.3.9 | |x| | | | Source Quench => slow start |4.2.3.9 |x| | | | | Time Exceeded => tell ALP |4.2.3.9 | |x| | | | Internet Engineering Task Force [Page 103] ***DRAFT RFC*** TRANSPORT LAYER -- TCP June 16, 1989 Parameter Problem => tell ALP |4.2.3.9 | |x| | | | | | | | | | | Address Validation | | | | | | | Reject OPEN to invalid IP address |4.2.3.10|x| | | | | Reject SYN from invalid IP address |4.2.3.10|x| | | | | Silently ignore SYN to bcast/mcast addr |4.2.3.10|x| | | | | | | | | | | | TCP/ALP Interface Services | | | | | | | Error Report Routine |4.2.4.1 |x| | | | | ALP can disable Error Report Routine |4.2.4.1 | |x| | | | ALP can specify TOS for sending |4.2.4.2 |x| | | | | ALP can change TOS during connection |4.2.4.2 | |x| | | | Pass received TOS up to ALP |4.2.4.2 | | |x| | | IP Options in OPEN |4.2.4.2 |x| | | | | FLUSH call |4.2.4.3 | | |x| | | Optional local IP addr parm. in OPEN |4.2.4.4 |x| | | | | -------------------------------------------------|--------|-|-|-|-|-|-- -------------------------------------------------|--------|-|-|-|-|-|-- FOOTNOTES: (1) "ALP" means Application-Layer program. Internet Engineering Task Force [Page 104] ***DRAFT RFC*** TRANSPORT LAYER -- TCP June 16, 1989 5. REFERENCES INTRODUCTORY REFERENCES [INTRO:1] "Requirements for Internet Hosts -- Application and Support," IETF Host Requirements Working Group, R. Braden, Ed., RFC-app 1989. [INTRO:2] "Requirements for Internet Gateways," R. Braden and J. Postel, RFC-1009, June 1987. [INTRO:3] "DDN Protocol Handbook," NIC-50004, NIC-50005, NIC-50006, (three volumes), SRI International, December 1985. [INTRO:4] "Official Internet Protocols," J. Reynolds and J. Postel, RFC-1011, May 1987. This document is republished periodically with new RFC numbers; the latest version must be used. [INTRO:5] "Protocol Document Order Information," O. Jacobsen and J. Postel, RFC-980, March 1986. [INTRO:6] "Assigned Numbers," J. Reynolds and J. Postel, RFC-1010, May 1987. This document is republished periodically with new RFC numbers; the latest version must be used. [INTRO:7] "Modularity and Efficiency in Protocol Implementations," D. Clark, RFC-817, July 1982. [INTRO:8] "The Structuring of Systems Using Upcalls," D. Clark, 10th ACM SOSP, Orcas Island, Washington, December 1985. Secondary References: [INTRO:9] "A Protocol for Packet Network Intercommunication," V. Cerf and R. Kahn, IEEE Transactions on Communication, May 1974. [INTRO:10] "The ARPA Internet Protocol," J. Postel, C. Sunshine, and D. Cohen, Computer Networks, Vol. 5, No. 4, July 1981. [INTRO:11] "The DARPA Internet Protocol Suite," B. Leiner, J. Postel, R. Cole and D. Mills, Proceedings INFOCOM 85, IEEE, Washington DC, March 1985. Also in: IEEE Communications Magazine, March 1985. Internet Engineering Task Force [Page 105] ***DRAFT RFC*** TRANSPORT LAYER -- TCP June 16, 1989 Also available as ISI-RS-85-153. [INTRO:12] "Final Text of DIS8473, Protocol for Providing the Connectionless Mode Network Service," ANSI, published as RFC-994, March 1986. [INTRO:13] "End System to Intermediate System Routing Exchange Protocol," ANSI X3S3.3, published as RFC-995, April 1986. LINK LAYER REFERENCES [LINK:1] "Trailer Encapsulations," S. Leffler and M. Karels, RFC-893, April 1984. [LINK:2] "An Ethernet Address Resolution Protocol," D. Plummer, RFC-826, November 1982. [LINK:3] "A Standard for the Transmission of IP Datagrams over Ethernet Networks," C. Hornig, RFC-894, April 1984. [LINK:4] "A Standard for the Transmission of IP Datagrams over IEEE 802 "Networks," J. Postel and J. Reynolds, RFC-1042, February 1988. This RFC contains a great deal of information of importance to Internet implementers planning to use IEEE 802 networks. IP LAYER REFERENCES [IP:1] "Internet Protocol (IP)," J. Postel, RFC-791, September 1981. [IP:2] "Internet Control Message Protocol (ICMP)," J. Postel, RFC-792, September 1981. [IP:3] "Internet Standard Subnetting Procedure," J. Mogul and J. Postel, RFC-950, August 1985. [IP:4] "Host Extensions for IP Multicasting," S. Deering, RFC-1054, May 1988. This is a Draft Internet Standard for the host implementation of IP multicasting. [IP:5] "Military Standard Internet Protocol," MIL-STD-1777, Department of Defense, August 1983. Internet Engineering Task Force [Page 106] ***DRAFT RFC*** TRANSPORT LAYER -- TCP June 16, 1989 This specification, as amended by RFC-963, is intended to describe the Internet Protocol but has some serious omissions (e.g., the mandatory subnet extension of RFC-950). It is also out of date. If there is a conflict, RFC-791, RFC-792, and RFC-950 must be taken as authoritative, while the present document is authoritative over all. [IP:6] "Some Problems with the Specification of the Military Standard Internet Protocol," D. Sidhu, RFC-963, November 1985. [IP:7] "The TCP Maximum Segment Size and Related Topics," J. Postel, RFC-879, November 1983. Discusses and clarifies the relationship between the TCP Maximum Segment Size option and the IP datagram size. [IP:8] "Comments on the IP Source Route Option," J. Postel and J. Reynolds, RFC to be published. [IP:9] "Fragmentation Considered Harmful," C. Kent and J. Mogul, Proc. SIGCOMM '87, ACM, August 1987. Published as ACM Comp Comm Review, Vol. 17, no. 5. This useful paper discusses the problems created by Internet fragmentation and presents alternative solutions. [IP:10] "IP Datagram Reassembly Algorithms," D Clark, RFC-815, July 1982. This and the following paper should be read by every implementor. [IP:11] "Fault Isolation and Recovery," D. Clark, RFC-816, July 1982. SECONDARY IP REFERENCES: [IP:12] "Broadcasting Internet Datagrams in the Presence of Subnets," J. Mogul, RFC-922, October 1984. This RFC first described directed broadcast addresses. However, the bulk of the RFC is concerned with gateways, not hosts. [IP:13] "Name, Addresses, Ports, and Routes," D. Clark, RFC-814, July 1982. [IP:14] "Something a Host Could Do with Source Quench: The Source Quench Internet Engineering Task Force [Page 107] ***DRAFT RFC*** TRANSPORT LAYER -- TCP June 16, 1989 Introduced Delay (SQUID)," W. Prue and J. Postel, RFC-1016, July 1987. UDP REFERENCES: [UDP:1] "User Datagram Protocol," J. Postel, RFC-768, August 1980. TCP REFERENCES: [TCP:1] "Transmission Control Protocol," J. Postel, RFC-793, September 1981. [TCP:2] "Transmission Control Protocol," MIL-STD-1778, US Department of Defense, August 1984. This specification as amended by RFC-964 is intended to describe the same protocol as RFC-793 [TCP:1]. If there is a conflict, RFC-793 takes precedence, and the present document is authoritative over both. [TCP:3] "Some Problems with the Specification of the Military Standard Transmission Control Protocol," D. Sidhu and T. Blumer, RFC-964, November 1985. [TCP:4] "The TCP Maximum Segment Size and Related Topics," J. Postel, RFC-879, November 1983. [TCP:5] "Window and Acknowledgment Strategy in TCP," D. Clark, RFC-813, July 1982. [TCP:6] "Round Trip Time Estimation," P. Karn & C. Partridge, ACM SIGCOMM '87, August 1987. [TCP:7] "Congestion Avoidance and Control," V. Jacobson, ACM SIGCOMM '88, August 1988. SECONDARY TCP REFERENCES: Internet Engineering Task Force [Page 108] ***DRAFT RFC*** TRANSPORT LAYER -- TCP June 16, 1989 [TCP:8] "Modularity and Efficiency in Protocol Implementation," D. Clark, RFC-817, July 1982. [TCP:9] "Congestion Control in IP/TCP," J. Nagle, RFC-896, January 1984. [TCP:10] "Computing the Internet Checksum," R. Braden, D. Borman, and C. Partridge, RFC-1071, September 1988. [TCP:11] "TCP Extensions for Long-Delay Paths," V. Jacobson & R. Braden, RFC-1072, October 1988. Internet Engineering Task Force [Page 109]