RFC1122

Network Working Group Internet Engineering Task Force Request for Comments: 1122 R. Braden, Editor

                                                           October 1989

       Requirements for Internet Hosts -- Communication Layers

Status of This Memo

  This RFC is 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.

Summary

  This is 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-1123 covers the application and support protocols.

                          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
        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
     1.4  Acknowledgments ........................................   20

  2. LINK LAYER ..................................................   21
     2.1  INTRODUCTION ...........................................   21

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     2.2  PROTOCOL WALK-THROUGH ..................................   21
     2.3  SPECIFIC ISSUES ........................................   21
        2.3.1  Trailer Protocol Negotiation ......................   21
        2.3.2  Address Resolution Protocol -- ARP ................   22
           2.3.2.1  ARP Cache Validation .........................   22
           2.3.2.2  ARP Packet Queue .............................   24
        2.3.3  Ethernet and IEEE 802 Encapsulation ...............   24
     2.4  LINK/INTERNET LAYER INTERFACE ..........................   25
     2.5  LINK LAYER REQUIREMENTS SUMMARY ........................   26

  3. INTERNET LAYER PROTOCOLS ....................................   27
     3.1 INTRODUCTION ............................................   27
     3.2  PROTOCOL WALK-THROUGH ..................................   29
        3.2.1 Internet Protocol -- IP ............................   29
           3.2.1.1  Version Number ...............................   29
           3.2.1.2  Checksum .....................................   29
           3.2.1.3  Addressing ...................................   29
           3.2.1.4  Fragmentation and Reassembly .................   32
           3.2.1.5  Identification ...............................   32
           3.2.1.6  Type-of-Service ..............................   33
           3.2.1.7  Time-to-Live .................................   34
           3.2.1.8  Options ......................................   35
        3.2.2 Internet Control Message Protocol -- ICMP ..........   38
           3.2.2.1  Destination Unreachable ......................   39
           3.2.2.2  Redirect .....................................   40
           3.2.2.3  Source Quench ................................   41
           3.2.2.4  Time Exceeded ................................   41
           3.2.2.5  Parameter Problem ............................   42
           3.2.2.6  Echo Request/Reply ...........................   42
           3.2.2.7  Information Request/Reply ....................   43
           3.2.2.8  Timestamp and Timestamp Reply ................   43
           3.2.2.9  Address Mask Request/Reply ...................   45
        3.2.3  Internet Group Management Protocol IGMP ...........   47
     3.3  SPECIFIC ISSUES ........................................   47
        3.3.1  Routing Outbound Datagrams ........................   47
           3.3.1.1  Local/Remote Decision ........................   47
           3.3.1.2  Gateway Selection ............................   48
           3.3.1.3  Route Cache ..................................   49
           3.3.1.4  Dead Gateway Detection .......................   51
           3.3.1.5  New Gateway Selection ........................   55
           3.3.1.6  Initialization ...............................   56
        3.3.2  Reassembly ........................................   56
        3.3.3  Fragmentation .....................................   58
        3.3.4  Local Multihoming .................................   60
           3.3.4.1  Introduction .................................   60
           3.3.4.2  Multihoming Requirements .....................   61
           3.3.4.3  Choosing a Source Address ....................   64
        3.3.5  Source Route Forwarding ...........................   65

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        3.3.6  Broadcasts ........................................   66
        3.3.7  IP Multicasting ...................................   67
        3.3.8  Error Reporting ...................................   69
     3.4  INTERNET/TRANSPORT LAYER INTERFACE .....................   69
     3.5  INTERNET LAYER REQUIREMENTS SUMMARY ....................   72

  4. TRANSPORT PROTOCOLS .........................................   77
     4.1  USER DATAGRAM PROTOCOL -- UDP ..........................   77
        4.1.1  INTRODUCTION ......................................   77
        4.1.2  PROTOCOL WALK-THROUGH .............................   77
        4.1.3  SPECIFIC ISSUES ...................................   77
           4.1.3.1  Ports ........................................   77
           4.1.3.2  IP Options ...................................   77
           4.1.3.3  ICMP Messages ................................   78
           4.1.3.4  UDP Checksums ................................   78
           4.1.3.5  UDP Multihoming ..............................   79
           4.1.3.6  Invalid Addresses ............................   79
        4.1.4  UDP/APPLICATION LAYER INTERFACE ...................   79
        4.1.5  UDP REQUIREMENTS SUMMARY ..........................   80
     4.2  TRANSMISSION CONTROL PROTOCOL -- TCP ...................   82
        4.2.1  INTRODUCTION ......................................   82
        4.2.2  PROTOCOL WALK-THROUGH .............................   82
           4.2.2.1  Well-Known Ports .............................   82
           4.2.2.2  Use of Push ..................................   82
           4.2.2.3  Window Size ..................................   83
           4.2.2.4  Urgent Pointer ...............................   84
           4.2.2.5  TCP Options ..................................   85
           4.2.2.6  Maximum Segment Size Option ..................   85
           4.2.2.7  TCP Checksum .................................   86
           4.2.2.8  TCP Connection State Diagram .................   86
           4.2.2.9  Initial Sequence Number Selection ............   87
           4.2.2.10  Simultaneous Open Attempts ..................   87
           4.2.2.11  Recovery from Old Duplicate SYN .............   87
           4.2.2.12  RST Segment .................................   87
           4.2.2.13  Closing a Connection ........................   87
           4.2.2.14  Data Communication ..........................   89
           4.2.2.15  Retransmission Timeout ......................   90
           4.2.2.16  Managing the Window .........................   91
           4.2.2.17  Probing Zero Windows ........................   92
           4.2.2.18  Passive OPEN Calls ..........................   92
           4.2.2.19  Time to Live ................................   93
           4.2.2.20  Event Processing ............................   93
           4.2.2.21  Acknowledging Queued Segments ...............   94
        4.2.3  SPECIFIC ISSUES ...................................   95
           4.2.3.1  Retransmission Timeout Calculation ...........   95
           4.2.3.2  When to Send an ACK Segment ..................   96
           4.2.3.3  When to Send a Window Update .................   97
           4.2.3.4  When to Send Data ............................   98

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           4.2.3.5  TCP Connection Failures ......................  100
           4.2.3.6  TCP Keep-Alives ..............................  101
           4.2.3.7  TCP Multihoming ..............................  103
           4.2.3.8  IP Options ...................................  103
           4.2.3.9  ICMP Messages ................................  103
           4.2.3.10  Remote Address Validation ...................  104
           4.2.3.11  TCP Traffic Patterns ........................  104
           4.2.3.12  Efficiency ..................................  105
        4.2.4  TCP/APPLICATION LAYER INTERFACE ...................  106
           4.2.4.1  Asynchronous Reports .........................  106
           4.2.4.2  Type-of-Service ..............................  107
           4.2.4.3  Flush Call ...................................  107
           4.2.4.4  Multihoming ..................................  108
        4.2.5  TCP REQUIREMENT SUMMARY ...........................  108

  5.  REFERENCES .................................................  112

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1. INTRODUCTION

  This document is one of a pair that defines and discusses 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"
  [INTRO:1], covers the application layer protocols.  This document
  should also be read in conjunction with "Requirements for Internet
  Gateways" [INTRO:2].

  These documents are intended to provide guidance for vendors,
  implementors, and users of Internet communication software.  They
  represent the consensus of a large body of technical experience and
  wisdom, contributed by the members of the Internet research and
  vendor communities.

  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 also 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 documents, with the corrections, amendments, and
  supplements contained in this RFC.

  A good-faith implementation of the protocols that was produced after
  careful reading of the RFC's and with some interaction with the
  Internet technical community, and that followed good communications
  software engineering practices, should differ from the requirements
  of this document in only minor ways.  Thus, 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, they were included because some past
  implementation has made the wrong choice, causing problems of
  interoperability, performance, and/or 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.

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  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.

  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 some terminology.

  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 communication 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

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        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.

        A host is generally said to be multihomed if it has more than
        one interface to the same or to different networks.  See
        Section 1.1.3 on "Terminology".

     1.1.2  Architectural Assumptions

        The current Internet architecture is based on a set of
        assumptions about the communication system.  The assumptions
        most relevant to hosts are as follows:

        (a)  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 communicate
             with each other using the same set of protocols that they
             would use to communicate with hosts on distant networks.

        (b)  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.

        (c)  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

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             objective is to insulate host software from changes caused
             by the inevitable evolution of the Internet routing
             architecture.

        (d)  The System must tolerate wide network variation.

             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

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             The application layer is the top layer of the Internet
             protocol suite.  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.

             We distinguish two categories of application layer
             protocols:  user protocols that provide service directly
             to users, and support protocols that provide common system
             functions.  Requirements for user and support protocols
             will be found in the companion RFC [INTRO:1].

             The most common Internet 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.

        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.

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        o  Internet Layer

             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
             OSI 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.

             IGMP is an Internet layer protocol used for establishing
             dynamic host groups for IP multicasting.

             The Internet layer protocols IP, ICMP, and IGMP 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

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        gateway would, while still performing the application layer
        functions of a host.

        Such dual-purpose systems must follow the Gateway Requirements
        RFC [INTRO:2]  with respect to their gateway functions, and
        must follow 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
        embedded gateway functionality.  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 more 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 host system
             maintainers 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: an host administrator must have
        conscious control over whether or not a given host acts as a
        gateway.  See Section 3.1 for the detailed requirements.

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  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;

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        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 result elsewhere.  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
        Internet protocol software.  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 header(s) of a
        packet that caused an 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
        through the management protocol (see [INTRO:1]); and (2) allow

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        the logging of a great variety of events to be selectively
        enabled.  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 aggressive 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 may be insufficient.  In some cases,
        configurability is needed because of 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
        configuration file is only necessary to override those defaults

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RFC1122 INTRODUCTION October 1989

        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.  A 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.
        Protocols in different layers 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].

        This document describes the conceptual service interface
        between layers using a functional ("procedure call") notation,
        like that used in the TCP specification [TCP:1].  A host
        implementation must support the logical information flow

Internet Engineering Task Force [Page 15]

RFC1122 INTRODUCTION October 1989

        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
        giving them shared access to common data structures.  These
        data structures, rather than explicit procedure calls, are then
        the agency for passing much of the information that is
        required.

        In general, each major section of this document is organized
        into the following subsections:

        (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 protocol design and
             implementation issues 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.

        The summary sections are intended to be guides and indexes to
        the text, but are necessarily cryptic and incomplete.  The
        summaries should never be used or referenced separately from
        the complete RFC.

     1.3.2  Requirements

        In this document, the words that are used to define the
        significance of each particular requirement are capitalized.

Internet Engineering Task Force [Page 16]

RFC1122 INTRODUCTION October 1989

        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"

             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 by
             encapsulation inside an IP datagram.

        Message
             In this description of the lower-layer protocols, a
             message is the unit of transmission in a transport layer
             protocol.  In particular, a TCP segment is a message.  A
             message consists of a transport protocol header followed
             by application protocol data.  To be transmitted end-to-

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RFC1122 INTRODUCTION October 1989

             end through the Internet, a message must be encapsulated
             inside a datagram.

        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
             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.

        Multihomed
             A host is said to be multihomed if it has multiple IP
             addresses.  For a discussion of multihoming, see Section
             3.3.4 below.

        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
             We define a logical [network] interface to be a logical
             path, distinguished by a unique IP address, to a connected
             network.  See Section 3.3.4.

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RFC1122 INTRODUCTION October 1989

        Specific-destination address
             This is the effective destination address of a datagram,
             even if it is broadcast or multicast; see Section 3.2.1.3.

        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:

        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 ----------->

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RFC1122 INTRODUCTION October 1989

  1.4  Acknowledgments

     This document incorporates contributions and comments from a large
     group of Internet protocol experts, including representatives of
     university and research labs, vendors, and government agencies.
     It was assembled primarily by the Host Requirements Working Group
     of the Internet Engineering Task Force (IETF).

     The Editor would especially like to acknowledge the tireless
     dedication of the following people, who attended many long
     meetings and generated 3 million bytes of electronic mail over the
     past 18 months in pursuit of this document: Philip Almquist, Dave
     Borman (Cray Research), Noel Chiappa, Dave Crocker (DEC), Steve
     Deering (Stanford), Mike Karels (Berkeley), Phil Karn (Bellcore),
     John Lekashman (NASA), Charles Lynn (BBN), Keith McCloghrie (TWG),
     Paul Mockapetris (ISI), Thomas Narten (Purdue), Craig Partridge
     (BBN), Drew Perkins (CMU), and James Van Bokkelen (FTP Software).

     In addition, the following people made major contributions to the
     effort: Bill Barns (Mitre), Steve Bellovin (AT&T), Mike Brescia
     (BBN), Ed Cain (DCA), Annette DeSchon (ISI), Martin Gross (DCA),
     Phill Gross (NRI), Charles Hedrick (Rutgers), Van Jacobson (LBL),
     John Klensin (MIT), Mark Lottor (SRI), Milo Medin (NASA), Bill
     Melohn (Sun Microsystems), Greg Minshall (Kinetics), Jeff Mogul
     (DEC), John Mullen (CMC), Jon Postel (ISI), John Romkey (Epilogue
     Technology), and Mike StJohns (DCA).  The following also made
     significant contributions to particular areas: Eric Allman
     (Berkeley), Rob Austein (MIT), Art Berggreen (ACC), Keith Bostic
     (Berkeley), Vint Cerf (NRI), Wayne Hathaway (NASA), Matt Korn
     (IBM), Erik Naggum (Naggum Software, Norway), Robert Ullmann
     (Prime Computer), David Waitzman (BBN), Frank Wancho (USA), Arun
     Welch (Ohio State), Bill Westfield (Cisco), and Rayan Zachariassen
     (Toronto).

     We are grateful to all, including any contributors who may have
     been inadvertently omitted from this list.

Internet Engineering Task Force [Page 20]

RFC1122 LINK LAYER October 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], augmented with 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-layer encapsulation MAY
        be used, but only when it has been verified that both systems
        (host or gateway) involved in the link-layer 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 layer protocols by
             reducing the amount of data copying within the operating
             system.  Higher layer 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.

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             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 ARP 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 ARP 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 ARP
             request.

     2.3.2  Address Resolution Protocol -- ARP

        2.3.2.1  ARP Cache Validation

           An implementation of the Address Resolution Protocol (ARP)
           [LINK:2] 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

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RFC1122 LINK LAYER October 1989

           destination.

           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:2])
                has significantly increased the likelihood that cache
                entries in hosts will become invalid, and therefore
                some ARP-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-layer 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 might in turn
                     call the link-layer advice routine to invalidate

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RFC1122 LINK LAYER October 1989

                     the ARP cache entry.

                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.

           DISCUSSION:
                Failure to follow this recommendation causes the first
                packet of every exchange to be lost.  Although higher-
                layer protocols can generally cope with packet loss by
                retransmission, packet loss does impact performance.
                For example, loss of a TCP open request causes the
                initial round-trip time estimate to be inflated.  UDP-
                based applications such as the Domain Name System are
                more seriously affected.

     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:2].

        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.

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RFC1122 LINK LAYER October 1989

        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-layer
        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.

        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-layer
             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 894-only systems cannot interoperate directly
             with 1042-only systems.  If the two system types are set
             up as two different logical networks on the same cable,
             they can communicate only through an IP gateway.
             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.

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RFC1122 LINK LAYER October 1989

     DISCUSSION
          Although the IP layer does not generally know link layer
          addresses (since every different network medium typically 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 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| | | | |
 Cache timeout configurable                      |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 | | | | | |

 Host able to:                                   |2.3.3  | | | | | |
   Send & receive RFC-894 encapsulation          |2.3.3  |x| | | | |
   Receive RFC-1042 encapsulation                |2.3.3  | |x| | | |
   Send RFC-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| | | | | IP layer pass TOS to link layer |2.4 |x| | | | | No ARP cache entry treated as Dest. Unreach. |2.4 | | | | |x|

Internet Engineering Task Force [Page 26]

RFC1122 INTERNET LAYER October 1989

3. INTERNET LAYER PROTOCOLS

  3.1 INTRODUCTION

     The Robustness Principle: "Be liberal in what you accept, and
     conservative in what you send" is particularly important in the
     Internet layer, where one misbehaving host can deny Internet
     service to many other hosts.

     The protocol standards used in the Internet layer are:

     o    RFC-791 [IP:1] defines the IP protocol and gives an
          introduction to the architecture of the Internet.

     o    RFC-792 [IP:2] defines ICMP, which provides routing,
          diagnostic and error functionality for 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.  See Section 3.2.2.

     o    RFC-950 [IP:3] defines the mandatory subnet extension to the
          addressing architecture.

     o    RFC-1112 [IP:4] defines the Internet Group Management
          Protocol IGMP, as part of a recommended extension to hosts
          and to the host-gateway interface to support Internet-wide
          multicasting at the IP level.  See Section 3.2.3.

          The target of an IP multicast may be an arbitrary group of
          Internet hosts.  IP multicasting is designed as a natural
          extension of the link-layer multicasting facilities of some
          networks, and it provides a standard means for local access
          to such link-layer multicasting facilities.

     Other important references are listed in Section 5 of this
     document.

     The Internet layer of host software MUST implement both IP and
     ICMP.  See Section 3.3.7 for the requirements on support of IGMP.

     The host IP layer has two basic functions:  (1) choose the "next
     hop" gateway or host for outgoing IP datagrams and (2) reassemble
     incoming IP datagrams.  The IP layer may also (3) implement
     intentional fragmentation of outgoing datagrams.  Finally, the IP
     layer must (4) provide diagnostic and error functionality.  We
     expect that IP layer functions may increase somewhat in the
     future, as further Internet control and management facilities are
     developed.

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RFC1122 INTERNET LAYER October 1989

     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 destined to the local host;

     (3)  processes options;

     (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:

     (1)  sets any fields not set by the transport layer;

     (2)  selects the correct first hop on the connected network (a
          process called "routing");

     (3)  fragments the datagram if necessary and if intentional
          fragmentation is implemented (see Section 3.3.3); and

     (4)  passes the packet(s) to the appropriate link-layer driver.

     A host is said to be multihomed if it 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.

     At present, remote multihoming MUST be handled at the application
     layer, as discussed in the companion RFC [INTRO:1].  A host MAY
     support local multihoming, which is discussed in this document,
     and in particular in Section 3.3.4.

     Any host that forwards datagrams generated by another host is
     acting as a gateway and MUST also meet the specifications laid out
     in the gateway requirements RFC [INTRO:2].  An Internet host that
     includes embedded gateway code MUST have a configuration switch to
     disable the gateway function, and this switch MUST default to the

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RFC1122 INTERNET LAYER October 1989

     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.

     In the following, the action specified in certain cases is to
     "silently discard" a received datagram.  This means that the
     datagram will be discarded without further processing and 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-discarded datagram, and SHOULD record
     the event in a statistics counter.

     DISCUSSION:
          Silent discard of erroneous datagrams is generally intended
          to prevent "broadcast storms".

  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
           discarded.

        3.2.1.2  Checksum: RFC-791 Section 3.1

           A host MUST verify the IP header checksum on every received
           datagram and silently discard 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.

           A multicast (Class D) address is a 28-bit logical address
           that stands for a group of hosts, and may be either
           permanent or transient.  Permanent multicast addresses are
           allocated by the Internet Assigned Number Authority
           [INTRO:6], while transient addresses may be allocated

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           dynamically to transient groups.  Group membership is
           determined dynamically using IGMP [IP:4].

           We now summarize the important special cases for Class A, B,
           and C 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.  It 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.  It 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.

           (d)  { <Network-number>, -1 }

                Directed broadcast to the specified network.  It MUST
                NOT be used as a source address.

           (e)  { <Network-number>, <Subnet-number>, -1 }

                Directed broadcast to the specified subnet.  It MUST
                NOT be used as a source address.

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           (f)  { <Network-number>, -1, -1 }

                Directed broadcast to all subnets of the specified
                subnetted network.  It 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 so that
           its value will be unique in the entire world.

           IP addresses are not permitted to 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.

           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).

           A host MUST silently discard an incoming datagram that is
           not destined for the host.  An incoming datagram is destined
           for the host if the datagram's destination address field is:

           (1)  (one of) the host's IP address(es); or

           (2)  an IP broadcast address valid for the connected
                network; or

           (3)  the address for a multicast group of which the host is
                a member on the incoming physical interface.

           For most purposes, a datagram addressed to a broadcast or
           multicast destination is processed as if it had been
           addressed to one of the host's IP addresses; we use the term
           "specific-destination address" for the equivalent local IP

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           address of the host.  The specific-destination address is
           defined to be the destination address in the IP header
           unless the header contains a broadcast or multicast address,
           in which case the specific-destination is an IP address
           assigned to the physical interface on which the datagram
           arrived.

           A host MUST silently discard an incoming datagram containing
           an IP source address that is invalid by the rules of this
           section.  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 link-
                layer broadcast of a unicast datagram or by a gateway
                or host that is confused or mis-configured.

                An architectural goal for Internet hosts 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.

                Implementers should be aware that applications
                depending upon the all-subnets directed broadcast
                address (f) may be unusable on some networks.  All-
                subnets broadcast is not widely implemented in vendor
                gateways at present, and even when it is implemented, a
                particular network administration may disable it in the
                gateway configuration.

        3.2.1.4  Fragmentation and Reassembly: RFC-791 Section 3.2

           The Internet model requires that every host support
           reassembly.  See Sections 3.3.2 and 3.3.3 for the
           requirements on fragmentation and reassembly.

        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.

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           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.

                However, the observed patterns of datagram loss in the
                Internet do not favor the probability of retransmitted
                fragments filling reassembly gaps, while other
                mechanisms (e.g., TCP repacketizing upon
                retransmission) tend to prevent retransmission of an
                identical datagram [IP:9].  Therefore, we 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.  The use of non-zero
           values in this field 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.  However, vendors should note that the use of
           precedence will most likely require that its value be passed
           between protocol layers in just the same way as the TOS
           field is passed.

           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

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           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
                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 IP layer MUST provide a means for the transport layer to
           set the TTL field of every datagram that is sent.  When a
           fixed TTL value is used, it MUST be configurable.  The
           current suggested value will be published in the "Assigned
           Numbers" RFC.

           DISCUSSION:
                The TTL field has two functions: limit the lifetime of
                TCP segments (see RFC-793 [TCP:1], p. 28), and
                terminate Internet 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 intent is that TTL expiration will cause a datagram
                to be discarded by a gateway but not by the destination
                host; however, hosts that act as gateways by forwarding
                datagrams must follow the gateway rules for TTL.

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                A higher-layer protocol may want to set the TTL in
                order to implement an "expanding scope" search for some
                Internet resource.  This is used by some diagnostic
                tools, and is expected to be useful for locating the
                "nearest" server of a given class using IP
                multicasting, for example.  A particular transport
                protocol may also want to specify its own TTL bound on
                maximum datagram lifetime.

                A fixed 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.

        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 (except NOP or END-OF-LIST) received in
           datagrams MUST be passed to the transport layer (or to ICMP
           processing when the datagram is an ICMP message).  The IP
           and transport layer MUST each interpret those IP options
           that they understand and silently ignore the others.

           Later sections of this document discuss specific IP option
           support required by each of ICMP, TCP, and UDP.

           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.

                This document does not define the order in which a
                receiver must process multiple options in the same IP
                header.  Hosts sending multiple options must be aware
                that this introduces an ambiguity in the meaning of
                certain options when combined with a source-route
                option.

           IMPLEMENTATION:
                The IP layer must not crash as the result of an option

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                length that is outside the possible range.  For
                example, erroneous option lengths have been observed to
                put some IP implementations into infinite loops.

           Here are the requirements for specific IP options:

           (a)  Security Option

                Some environments 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 options described in RFC-791
                and RFC-1038 are obsolete.  For DoD applications,
                vendors should consult [IP:8] for guidance.

           (b)  Stream Identifier Option

                This option is obsolete; it SHOULD NOT be sent, and it
                MUST be silently ignored if received.

           (c)  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.

                If host receives a datagram containing a completed
                source route (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 the transport layer (or to ICMP message
                processing).  This recorded route will be reversed and
                used to form a return source route for reply datagrams
                (see discussion of IP 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 the discussion below).

                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.

                Section 3.3.5 presents the rules for a host acting as
                an intermediate hop in a source route, i.e., forwarding

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                a source-routed datagram.

                DISCUSSION:
                     If a source-routed datagram is fragmented, each
                     fragment will contain a copy of the source route.
                     Since the processing of 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.

           (d)  Record Route Option

                Implementation of originating and processing the Record
                Route option is OPTIONAL.

           (e)  Timestamp Option

                Implementation of originating and processing the
                Timestamp option is OPTIONAL.  If it is implemented,
                the following rules apply:

                o    The originating host MUST record a timestamp in a
                     Timestamp option whose Internet address fields are
                     not pre-specified or whose first pre-specified
                     address is the host's interface address.

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                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 in
                     Section 3.2.2.8 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)

        If an ICMP message of unknown type is received, it MUST be
        silently discarded.

        Every ICMP error message includes the Internet header and at
        least the first 8 data octets of the datagram that triggered
        the error; more than 8 octets MAY be sent; this header and data
        MUST be unchanged from the received datagram.

        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.

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        An ICMP error message MUST NOT be sent as the result of
        receiving:

        *    an ICMP error message, or

        *    a datagram destined to an IP broadcast or IP 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, a multicast address, or a Class E
             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
             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

           The following additional codes are hereby defined:

                   6 = destination network unknown

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                   7 = destination host unknown

                   8 = source host isolated

                   9 = communication with destination network
                           administratively prohibited

                  10 = communication with destination host
                           administratively prohibited

                  11 = network unreachable for type of service

                  12 = host unreachable for type of service

           A host SHOULD generate Destination Unreachable messages with
           code:

           2    (Protocol Unreachable), when the designated transport
                protocol is not supported; or

           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.

           A Destination Unreachable message that is received MUST be
           reported to the transport layer.  The transport layer SHOULD
           use the information appropriately; for example, see Sections
           4.1.3.3, 4.2.3.9, and 4.2.4 below.  A transport protocol
           that has its own mechanism for notifying the sender that a
           port is unreachable (e.g., TCP, which sends RST segments)
           MUST nevertheless accept an ICMP Port Unreachable for the
           same purpose.

           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).

        3.2.2.2  Redirect: RFC-792

           A host SHOULD NOT send an ICMP Redirect message; Redirects
           are to be sent only by gateways.

           A host receiving a Redirect message MUST update its routing
           information accordingly.  Every host MUST be prepared to

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           accept both Host and Network Redirects and to process them
           as described in Section 3.3.1.2 below.

           A Redirect message SHOULD be silently discarded if the new
           gateway address it specifies is not on the same connected
           (sub-) net through which the Redirect arrived [INTRO:2,
           Appendix A], or if the source of the Redirect is not the
           current first-hop gateway for 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:2] 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 (or ICMP processing). 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).

                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, however, this
                proposal is currently experimental and not currently
                recommended.

        3.2.2.4  Time Exceeded: RFC-792

           An incoming Time Exceeded message MUST be passed to the
           transport layer.

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           DISCUSSION:
                A gateway will send a Time Exceeded Code 0 (In Transit)
                message when it discards a datagram due to an expired
                TTL field.  This indicates either a gateway routing
                loop or too small an initial TTL value.

                A host may receive a Time Exceeded Code 1 (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 be part of 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.

           A new variant on the Parameter Problem message is hereby
           defined:
             Code 1 = required option is missing.

           DISCUSSION:
                This variant is currently in use in the military
                community for a missing security option.

        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.

           An ICMP Echo Request destined to an IP broadcast or IP
           multicast address MAY be silently discarded.

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           DISCUSSION:
                This neutral provision results from a passionate debate
                between those who feel that ICMP Echo to a broadcast
                address provides a valuable diagnostic capability and
                those who feel that misuse of this feature can too
                easily create packet storms.

           The IP source address in an ICMP Echo Reply MUST be the same
           as the specific-destination address (defined in Section
           3.2.1.3) 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 to the ICMP user
           interface, unless the corresponding Echo Request originated
           in the IP layer.

           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 for the Echo Reply message.

        3.2.2.7  Information Request/Reply: RFC-792

           A host SHOULD 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.

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           o    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).

           The following cases for Timestamp are to be handled
           according to the corresponding rules for ICMP Echo:

           o    An ICMP Timestamp Request message to an IP broadcast or
                IP multicast address MAY be silently discarded.

           o    The IP source address in an ICMP Timestamp Reply MUST
                be the same as the specific-destination address of the
                corresponding Timestamp Request message.

           o    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 for the Timestamp Reply message.

           o    If a Record Route and/or Timestamp option is received
                in a Timestamp Request, this (these) option(s) SHOULD
                be updated to include the current host and included in
                the IP header of the Timestamp Reply message.

           o    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":

           (a)  A "standard value" MUST be updated at least 15 times
                per second (i.e., at most the six low-order bits of the
                value may be undefined).

           (b)  The accuracy of a "standard value" MUST approximate
                that of operator-set CPU clocks, i.e., correct within a
                few minutes.

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        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)
           corresponding to its IP address(es):

           (1)  static configuration information;

           (2)  obtaining the address mask(s) dynamically as a side-
                effect of the system initialization process (see
                [INTRO:1]); and

           (3)  sending ICMP Address Mask Request(s) and receiving ICMP
                Address Mask Reply(s).

           The choice of method to be used in a particular host MUST be
           configurable.

           When method (3), the use of Address Mask messages, is
           enabled, then:

           (a)  When it initializes, the host MUST broadcast an Address
                Mask Request message on the connected network
                corresponding to the IP address.  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 a mask appropriate for the address class
                of the IP address, i.e., assume that the connected
                network is not subnetted.

           (c)  The first Address Mask Reply message received MUST be
                used to set the address mask corresponding to the
                particular local IP address.  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
                (silently) ignored.

           Conversely, if Address Mask messages are disabled, then no
           ICMP Address Mask Requests will be sent, and any ICMP
           Address Mask Replies received for that local IP address MUST
           be (silently) ignored.

           A host SHOULD make some reasonableness check on any address

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           mask it installs; see IMPLEMENTATION section below.

           A system MUST NOT send an Address Mask Reply unless it is an
           authoritative agent for address masks.  An authoritative
           agent may be a host or a gateway, but it MUST be explicitly
           configured as a address mask agent.  Receiving an address
           mask 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 act as an authoritative agent for this mask,
           i.e., whether it will answer Address Mask Request messages
           using this mask.

           If it is configured as an agent, the host MUST broadcast an
           Address Mask Reply for the mask on the appropriate interface
           when it initializes.

           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 agents that 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 source IP address.  If the network part of
                this address is zero (see (a) and (b) in 3.2.1.3), the
                Reply will be broadcast.

                Getting no reply to its Address Mask Request messages,
                a host will assume there is no agent and use an
                unsubnetted mask, but the agent may be only temporarily
                unreachable.  An agent will broadcast an unsolicited
                Address Mask Reply whenever it initializes, in order to
                update the masks of all hosts that have initialized in
                the meantime.

           IMPLEMENTATION:
                The following reasonableness check on an address mask
                is suggested: the mask is not all 1 bits, and it is

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                either zero or else the 8 highest-order bits are on.

     3.2.3  Internet Group Management Protocol IGMP

        IGMP [IP:4] is a protocol used between hosts and gateways on a
        single network to establish hosts' membership in particular
        multicast groups.  The gateways use this information, in
        conjunction with a multicast routing protocol, to support IP
        multicasting across the Internet.

        At this time, implementation of IGMP is OPTIONAL; see Section
        3.3.7 for more information.  Without IGMP, a host can still
        participate in multicasting local to its connected networks.

  3.3  SPECIFIC ISSUES

     3.3.1  Routing Outbound Datagrams

        The IP layer chooses the correct next hop for each datagram it
        sends.  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 a 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 (particular to a local IP address for
                a multihomed host) 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 match the 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 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 special-case destination address is handled as follows:

           *    For a limited broadcast or a multicast address, simply
                pass the datagram to the link layer for the appropriate
                interface.

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           *    For a (network or subnet) directed broadcast, the
                datagram can use the standard routing algorithms.

           The host IP layer MUST operate correctly in a minimal
           network environment, and in particular, when there are 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.

        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 uses the following
           basic algorithm on this cache to route a datagram; this
           algorithm is designed to put the 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.

           (b)  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.

           (c)  When it receives a Redirect, the host updates the
                next-hop gateway in the appropriate route cache entry,
                so later datagrams to the same destination will go
                directly to the best gateway.

           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; i.e., the
           cache entry for the destination host (only) would be updated
           (or created, if an entry for that host did not exist) for
           the new gateway.

           DISCUSSION:
                This recommendation is to protect against gateways that
                erroneously send Network Redirects for a subnetted
                network, in violation of the gateway requirements
                [INTRO:2].

           When there is no route cache entry for the destination host
           address (and the destination is not on the connected

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           network), the IP layer MUST pick 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".  Each such static route MAY include a
           flag specifying whether it may be overridden by ICMP
           Redirects.

           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
                possibly useful in some circumstances, may cause
                problems in other cases (e.g., gateways are not all
                equal), and it is not recommended.

                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 Type-of-
                Service (see next section).  Static routes would be set
                up by system administrators to override the normal
                automatic routing mechanism, to handle exceptional
                situations.  However, any static routing information is
                a potential source of failure as configurations change
                or equipment fails.

        3.3.1.3  Route Cache

           Each route cache entry needs to include the following
           fields:

           (1)  Local IP address (for a multihomed host)

           (2)  Destination IP address

           (3)  Type(s)-of-Service

           (4)  Next-hop gateway IP address

           Field (2) MAY be the full IP address of the destination

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           host, or only the destination network number.  Field (3),
           the TOS, SHOULD be included.

           See Section 3.3.4.2 for a discussion of the implications of
           multihoming for the lookup procedure in this cache.

           DISCUSSION:
                Including the Type-of-Service field in the route cache
                and considering it in the host route algorithm will
                provide the necessary mechanism for the future when
                Type-of-Service routing is commonly used in the
                Internet.  See Section 3.2.1.6.

                Each route cache entry defines the endpoints of an
                Internet path.  Although the connecting 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.  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.
                This data will generally be both gathered and used by a
                higher layer protocol, e.g., by TCP, or by an
                application using UDP.  Experiments are currently in
                progress on caching path properties in this manner.

                There is no consensus on whether the route cache should
                be keyed on destination host addresses alone, or allow
                both host and network addresses.  Those who favor the
                use of only host addresses argue that:

                (1)  As required in Section 3.3.1.2, Redirect messages
                     will generally result in entries keyed on
                     destination host addresses; the simplest and most
                     general scheme would be to use host addresses
                     always.

                (2)  The IP layer may not always know the address mask
                     for a network address in a complex subnetted
                     environment.

                (3)  The use of only host addresses allows the
                     destination address to be used as a pure 32-bit
                     number, which may allow the Internet architecture
                     to be more easily extended in the future without

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                     any change to the hosts.

                The opposing view is that allowing a mixture of
                destination hosts and networks in the route cache:

                (1)  Saves memory space.

                (2)  Leads to a simpler data structure, easily
                     combining the cache with the tables of default and
                     static routes (see below).

                (3)  Provides a more useful place to cache path
                     properties, as discussed earlier.

           IMPLEMENTATION:
                The cache needs to be large enough to include entries
                for the maximum number of destination hosts that may be
                in use at one time.

                A route cache entry may also include control
                information used 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.

                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.

                Although we have described the route cache, the lists
                of default gateways, and a table of static routes as
                conceptually distinct, in practice they may be combined
                into a single "routing table" data structure.

        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 route 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

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           algorithm which is totally satisfactory, though it has
           identified several forbidden paths and promising techniques.

           *    A particular gateway SHOULD NOT be used indefinitely in
                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 continuously.

           *    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 re-routing,
                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

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                architecture, but it is expected to come primarily from
                the transport and link layers.  Here are some possible
                sources for gateway advice:

                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, an ACK for new data proves that the
                     acknowleged data must have been transmitted
                     successfully.

                o    An ICMP Redirect message from a particular gateway
                     should be used as positive advice about that
                     gateway.

                o    Link-layer information that reliably detects and
                     reports host failures (e.g., ARPANET Destination
                     Dead messages) should be used as negative advice.

                o    Failure to ARP or to re-validate 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 route cache.  This is
                     probably prohibitively inefficient.

                Note that positive advice that is given for 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 therefore allow a
                neutral value for advice, since either always-positive
                or always-negative advice leads to incorrect behavior.

                There is another technique for dead gateway detection
                that has been commonly used but is not recommended.

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                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
                needs to recognize all the interior gateway protocols
                that gateways may use (see [INTRO:2]).  In addition, it
                only works on a broadcast network.

                At present, pinging (i.e., using ICMP Echo messages) is
                the mechanism for gateway probing when absolutely
                required.  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 discards packets that 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 route 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 would reset the reroute timer to
                     Tr.  Negative advice would 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 ping to the
                     gateway, followed immediately by the datagram.

                o    The ping (ICMP Echo) would be sent again if
                     necessary, up to N times.  If no ping 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.

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                Note that the size of Tr is inversely related to the
                amount of advice available.  Tr should be large enough
                to insure that:

                *    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 route 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 not the current default, the IP
           layer can immediately switch to a default gateway.  If it is
           the current default that failed, the IP layer MUST select a
           different default gateway (assuming more than one default is
           known) for the failed route and for 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 send a Redirect back
                to the host pointing to the failed gateway (!).  The
                result is likely to be a rapid oscillation in the
                contents of the host's route cache 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 default
                gateway is to simply round-robin among the default
                gateways in the host's list.  Another is to rank the

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                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.

        3.3.1.6  Initialization

           The following information MUST be configurable:

           (1)  IP address(es).

           (2)  Address mask(s).

           (3)  A list of default gateways, with a preference level.

           A manual method of entering this configuration data MUST be
           provided.  In addition, a variety of methods can be used to
           determine this information dynamically; see the section on
           "Host Initialization" in [INTRO:1].

           DISCUSSION:
                Some host implementations use "wiretapping" of gateway
                protocols on a broadcast network to learn what gateways
                exist.  A standard method for default gateway discovery
                is under development.

     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 sometimes
        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.

        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 reassembled
             datagram size; in the latter case, EMTU_R is said to be

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             "indefinite".

             Logically, reassembly is performed by simply copying each
             fragment into the packet buffer at the proper offset.
             Note that fragments may overlap if successive
             retransmissions use different packetizing but the same
             reassembly Id.

             The tricky part of reassembly 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.
             However, note 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.  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.  If this timeout expires, the partially-reassembled
        datagram MUST be discarded and an ICMP Time Exceeded message
        sent to the source host (if fragment zero has been received).

        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

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             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 sets an upper
             limit on a reasonable reassembly timeout value.

     3.3.3  Fragmentation

        Optionally, the IP layer MAY implement a mechanism to fragment
        outgoing datagrams intentionally.

        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
        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 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

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        MTU.

        The MTU of each physical interface MUST be configurable.

        A host IP layer implementation MAY have a configuration flag
        "All-Subnets-MTU", indicating that the MTU of the connected
        network is to 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, an "All-Subnets-MTU" flag is needed for each network
        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 than 576 bytes (including header and
                  data), so a host must not send a larger datagram
                  without explicit knowledge or prior arrangement with
                  the destination host.  Thus, MMS_S is only an upper
                  bound on the datagram size that a transport protocol
                  may send; even when MMS_S exceeds 556, the transport
                  layer must limit its messages to 556 bytes in the
                  absence of other knowledge about the destination
                  host.

             (b)  Some transport protocols (e.g., TCP) provide a way to
                  explicitly inform the sender about the largest
                  datagram the other end can receive and reassemble
                  [IP:7].  There is no corresponding mechanism in the
                  IP layer.

                  A transport protocol that assumes an EMTU_R larger
                  than 576 (see Section 3.3.2), can send a datagram of
                  this larger size to another host that implements the
                  same protocol.

             (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].

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             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 non-local networks.

             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 would 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).

     3.3.4  Local Multihoming

        3.3.4.1  Introduction

           A multihomed host has multiple IP addresses, which we may
           think of as "logical interfaces".  These logical interfaces
           may be associated with one or more physical interfaces, and
           these physical interfaces may be connected to the same or
           different networks.

           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.

           (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

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                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.
                The choice of interface (i.e., the choice of first-hop
                network) may significantly affect performance or even
                reachability of remote parts of the Internet.

           Finally, we note another possibility that is NOT
           multihoming:  one logical interface may be 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.

           In the Internet protocol architecture, a transport protocol
           instance ("entity") has no address of its own, but instead
           uses a single 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.

        3.3.4.2  Multihoming Requirements

           The following general rules apply to the selection of an IP
           source address for sending a datagram from a multihomed

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           host.

           (1)  If the datagram is sent in response to a received
                datagram, the source address for the response SHOULD be
                the specific-destination address of the request.  See
                Sections 4.1.3.5 and 4.2.3.7 and the "General Issues"
                section of [INTRO:1] for more specific requirements on
                higher layers.

                Otherwise, a source address must be selected.

           (2)  An application MUST be able to explicitly specify the
                source address for initiating a connection or a
                request.

           (3)  In the absence of such a specification, the networking
                software MUST choose a source address.  Rules for this
                choice are described below.

           There are two key requirement issues related to multihoming:

           (A)  A host MAY silently discard an incoming datagram whose
                destination address does not correspond to the physical
                interface through which it is received.

           (B)  A host MAY restrict itself to sending (non-source-
                routed) IP datagrams only through the physical
                interface that corresponds to the IP source address of
                the datagrams.

           DISCUSSION:
                Internet host implementors have used two different
                conceptual models for multihoming, briefly summarized
                in the following discussion.  This document takes no
                stand on which model is preferred; each seems to have a
                place.  This ambivalence is reflected in the issues (A)
                and (B) being optional.

                o    Strong ES Model

                     The Strong ES (End System, i.e., host) model
                     emphasizes the host/gateway (ES/IS) distinction,
                     and would therefore substitute MUST for MAY in
                     issues (A) and (B) above.  It tends to model a
                     multihomed host as a set of logical hosts within
                     the same physical host.

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                     With respect to (A), proponents of the Strong ES
                     model note that automatic Internet routing
                     mechanisms could not route a datagram to a
                     physical interface that did not correspond to the
                     destination address.

                     Under the Strong ES model, the route computation
                     for an outgoing datagram is the mapping:

                        route(src IP addr, dest IP addr, TOS)
                                                       -> gateway

                     Here the source address is included as a parameter
                     in order to select a gateway that is directly
                     reachable on the corresponding physical interface.
                     Note that this model logically requires that in
                     general there be at least one default gateway, and
                     preferably multiple defaults, for each IP source
                     address.

                o    Weak ES Model

                     This view de-emphasizes the ES/IS distinction, and
                     would therefore substitute MUST NOT for MAY in
                     issues (A) and (B).  This model may be the more
                     natural one for hosts that wiretap gateway routing
                     protocols, and is necessary for hosts that have
                     embedded gateway functionality.

                     The Weak ES Model may cause the Redirect mechanism
                     to fail.  If a datagram is sent out a physical
                     interface that does not correspond to the
                     destination address, the first-hop gateway will
                     not realize when it needs to send a Redirect.  On
                     the other hand, if the host has embedded gateway
                     functionality, then it has routing information
                     without listening to Redirects.

                     In the Weak ES model, the route computation for an
                     outgoing datagram is the mapping:

                        route(dest IP addr, TOS) -> gateway, interface

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        3.3.4.3  Choosing a Source Address

           DISCUSSION:
                When it sends an initial connection request (e.g., a
                TCP "SYN" segment) or a datagram service request (e.g.,
                a UDP-based query), the transport layer on a multihomed
                host needs to know which source address to use.  If the
                application does not specify it, the transport layer
                must ask the IP layer to perform the conceptual
                mapping:

                    GET_SRCADDR(remote IP addr, TOS)
                                              -> local IP address

                Here TOS is the Type-of-Service value (see Section
                3.2.1.6), and the result is the desired source address.
                The following rules are suggested for implementing this
                mapping:

                (a)  If the remote Internet address lies on one of the
                     (sub-) nets to which the host is directly
                     connected, a corresponding source address may be
                     chosen, unless the corresponding interface is
                     known to be down.

                (b)  The route cache may be consulted, to see if there
                     is an active route to the specified destination
                     network through any network interface; if so, a
                     local IP address corresponding to that interface
                     may be chosen.

                (c)  The table of static routes, if any (see Section
                     3.3.1.2) may be similarly consulted.

                (d)  The default gateways may be consulted.  If these
                     gateways are assigned to different interfaces, the
                     interface corresponding to the gateway with the
                     highest preference may be chosen.

                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.

           IMPLEMENTATION:
                It will be noted that this process is essentially the
                same as datagram routing (see Section 3.3.1), and
                therefore hosts may be able to combine the

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                implementation of the two functions.

     3.3.5  Source Route Forwarding

        Subject to restrictions given below, a host MAY be able to act
        as an intermediate hop in a source route, forwarding a source-
        routed datagram to the next 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:2].  This includes the following
        specific provisions, which override the corresponding host
        provisions given earlier in this document:

        (A)  TTL (ref. Section 3.2.1.7)

             The TTL field MUST be decremented and the datagram perhaps
             discarded as specified for a gateway in [INTRO:2].

        (B)  ICMP Destination Unreachable (ref. Section 3.2.2.1)

             A host MUST be able to generate Destination Unreachable
             messages with the following codes:

             4    (Fragmentation Required but DF Set) when a source-
                  routed datagram cannot be fragmented to fit into the
                  target network;

             5    (Source Route Failed) 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.

        (C)  IP Source Address (ref. Section 3.2.1.3)

             A source-routed datagram being forwarded MAY (and normally
             will) have a source address that is not one of the IP
             addresses of the forwarding host.

        (D)  Record Route Option (ref. Section 3.2.1.8d)

             A host that is forwarding a source-routed datagram
             containing a Record Route option MUST update that option,
             if it has room.

        (E)  Timestamp Option (ref. Section 3.2.1.8e)

             A host that is forwarding a source-routed datagram

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             containing a Timestamp Option MUST add the current
             timestamp to that option, according to the rules for this
             option.

        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 physical 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    A host that supports non-local source-routing MUST have a
             configurable switch to disable forwarding, and this switch
             MUST default to disabled.

        o    The host MUST satisfy all gateway requirements for
             configurable policy filters [INTRO:2] 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.

     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

_________________________

• 4.2BSD Unix and its derivatives, but not 4.3BSD.

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        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 of broadcast address, for each physical
        interface, but this option SHOULD default to the standard (-1)
        form.

        When a host sends a datagram to a link-layer broadcast address,
        the IP destination address MUST be a legal IP broadcast or IP
        multicast address.

        A host SHOULD silently discard 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.

        Hosts SHOULD use the Limited Broadcast address to broadcast to
        a connected network.

        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 multihomed host.  This specification
             takes no stand on the issue.

     3.3.7  IP Multicasting

        A host SHOULD support local IP multicasting on all connected
        networks for which a mapping from Class D IP addresses to
        link-layer addresses has been specified (see below).  Support
        for local IP multicasting includes sending multicast datagrams,
        joining multicast groups and receiving multicast datagrams, and
        leaving multicast groups.  This implies support for all of
        [IP:4] except the IGMP protocol itself, which is OPTIONAL.

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        DISCUSSION:
             IGMP provides gateways that are capable of multicast
             routing with the information required to support IP
             multicasting across multiple networks.  At this time,
             multicast-routing gateways are in the experimental stage
             and are not widely available.  For hosts that are not
             connected to networks with multicast-routing gateways or
             that do not need to receive multicast datagrams
             originating on other networks, IGMP serves no purpose and
             is therefore optional for now.  However, the rest of
             [IP:4] is currently recommended for the purpose of
             providing IP-layer access to local network multicast
             addressing, as a preferable alternative to local broadcast
             addressing.  It is expected that IGMP will become
             recommended at some future date, when multicast-routing
             gateways have become more widely available.

        If IGMP is not implemented, a host SHOULD still join the "all-
        hosts" group (224.0.0.1) when the IP layer is initialized and
        remain a member for as long as the IP layer is active.

        DISCUSSION:
             Joining the "all-hosts" group will support strictly local
             uses of multicasting, e.g., a gateway discovery protocol,
             even if IGMP is not implemented.

        The mapping of IP Class D addresses to local addresses is
        currently specified for the following types of networks:

        o    Ethernet/IEEE 802.3, as defined in [IP:4].

        o    Any network that supports broadcast but not multicast,
             addressing: all IP Class D addresses map to the local
             broadcast address.

        o    Any type of point-to-point link (e.g., SLIP or HDLC
             links): no mapping required.  All IP multicast datagrams
             are sent as-is, inside the local framing.

        Mappings for other types of networks will be specified in the
        future.

        A host SHOULD provide a way for higher-layer protocols or
        applications to determine which of the host's connected
        network(s) support IP multicast addressing.

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     3.3.8  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 datagram 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
     extension of the information in Section 3.3 of RFC-791 [IP:1].

     *    Send Datagram

               SEND(src, dst, prot, TOS, TTL, BufPTR, len, Id, DF, opt
                    => result )

          where the parameters are defined in RFC-791.  Passing an Id
          parameter is optional; see Section 3.2.1.5.

     *    Receive Datagram

               RECV(BufPTR, prot
                    => result, src, dst, SpecDest, TOS, len, opt)

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          All the parameters are defined in RFC-791, except for:

               SpecDest = specific-destination address of datagram
                           (defined in Section 3.2.1.3)

          The result parameter dst contains the datagram's destination
          address.  Since this may be a broadcast or multicast address,
          the SpecDest parameter (not shown in RFC-791) MUST be passed.
          The parameter opt contains all the IP options received in the
          datagram; these MUST also be passed to the transport layer.

     *    Select Source Address

               GET_SRCADDR(remote, TOS)  -> local

               remote = remote IP address
               TOS = Type-of-Service
               local = local IP address

          See Section 3.3.4.3.

     *    Find Maximum Datagram Sizes

               GET_MAXSIZES(local, remote, TOS) -> MMS_R, MMS_S

               MMS_R = maximum receive transport-message size.
               MMS_S = maximum send transport-message size.
              (local, remote, TOS defined above)

          See Sections 3.3.2 and 3.3.3.

     *    Advice on Delivery Success

               ADVISE_DELIVPROB(sense, local, remote, TOS)

          Here the parameter sense is a 1-bit flag indicating whether
          positive or negative advice is being given; see the
          discussion in Section 3.3.1.4. The other parameters were
          defined earlier.

     *    Send ICMP Message

               SEND_ICMP(src, dst, TOS, TTL, BufPTR, len, Id, DF, opt)
                    -> result

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               (Parameters defined in RFC-791).

          Passing an Id parameter is optional; see Section 3.2.1.5.
          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.

     *    Receive ICMP Message

               RECV_ICMP(BufPTR ) -> result, src, dst, len, opt

               (Parameters defined in RFC-791).

          The 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.

          For an ICMP error message, the data that is passed up MUST
          include the original Internet header plus all the octets of
          the original message that are included in the ICMP message.
          This data 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    Echo Reply (to ICMP user interface, unless the Echo
               Request originated in the IP layer)

          o    Timestamp Reply (to ICMP user interface)

          o    Time Exceeded

     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.

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  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

---

|--------|-|-|-|-|-|--

                                                |        | | | | | |

Implement IP and ICMP |3.1 |x| | | | | Handle remote multihoming in application layer |3.1 |x| | | | | Support local multihoming |3.1 | | |x| | | Meet gateway specs if forward datagrams |3.1 |x| | | | | Configuration switch for embedded gateway |3.1 |x| | | | |1

  Config switch default to non-gateway          |3.1     |x| | | | |1
  Auto-config based on number of interfaces     |3.1     | | | | |x|1

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 dgram |3.2.1.2 |x| | | | | Addressing: | | | | | | |

 Subnet addressing (RFC-950)                    |3.2.1.3 |x| | | | |
 Src address must be host's own IP address      |3.2.1.3 |x| | | | |
 Silently discard datagram with bad dest addr   |3.2.1.3 |x| | | | |
 Silently discard datagram with bad src addr    |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 TTL < 2          |3.2.1.7 | | | | |x|
 Allow transport layer to set TTL               |3.2.1.7 |x| | | | |
 Fixed TTL is configurable                      |3.2.1.7 |x| | | | |
                                                |        | | | | | |

IP Options: | | | | | | |

 Allow transport layer to send IP options       |3.2.1.8 |x| | | | |
 Pass all IP options rcvd to higher layer       |3.2.1.8 |x| | | | |

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 IP layer silently ignore unknown options       |3.2.1.8 |x| | | | |
 Security option                                |3.2.1.8a| | |x| | |
 Send Stream Identifier option                  |3.2.1.8b| | | |x| |
 Silently ignore Stream Identifer option        |3.2.1.8b|x| | | | |
 Record Route option                            |3.2.1.8d| | |x| | |
 Timestamp option                               |3.2.1.8e| | |x| | |

Source Route Option: | | | | | | |

 Originate & terminate Source Route options     |3.2.1.8c|x| | | | |
 Datagram with completed SR passed up to TL     |3.2.1.8c|x| | | | |
 Build correct (non-redundant) return route     |3.2.1.8c|x| | | | |
 Send multiple SR options in one header         |3.2.1.8c| | | | |x|
                                                |        | | | | | |

ICMP: | | | | | | |

 Silently discard ICMP msg with unknown type    |3.2.2   |x| | | | |
 Include more than 8 octets of orig datagram    |3.2.2   | | |x| | |
     Included octets same as received           |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 IP m'cast                      |3.2.2   | | | | |x|
  - Link-layer b'cast                           |3.2.2   | | | | |x|
  - Non-initial fragment                        |3.2.2   | | | | |x|
  - Datagram with non-unique src address        |3.2.2   | | | | |x|
 Return ICMP error msgs (when not prohibited)   |3.3.8   |x| | | | |
                                                |        | | | | | |
 Dest Unreachable:                              |        | | | | | |
   Generate Dest Unreachable (code 2/3)         |3.2.2.1 | |x| | | |
   Pass ICMP Dest Unreachable to higher layer   |3.2.2.1 |x| | | | |
   Higher layer act on Dest Unreach             |3.2.2.1 | |x| | | |
     Interpret Dest Unreach as only hint        |3.2.2.1 |x| | | | |
 Redirect:                                      |        | | | | | |
   Host send Redirect                           |3.2.2.2 | | | |x| |
   Update route cache when recv Redirect        |3.2.2.2 |x| | | | |
   Handle both Host and Net Redirects           |3.2.2.2 |x| | | | |
   Discard illegal Redirect                     |3.2.2.2 | |x| | | |
 Source Quench:                                 |        | | | | | |
   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| | | |
 Time Exceeded: pass to higher layer            |3.2.2.4 |x| | | | |
 Parameter Problem:                             |        | | | | | |
   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 Request or Reply:                    |        | | | | | |
   Echo server and Echo client                  |3.2.2.6 |x| | | | |

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   Echo client                                  |3.2.2.6 | |x| | | |
   Discard Echo Request to broadcast address    |3.2.2.6 | | |x| | |
   Discard Echo Request to multicast address    |3.2.2.6 | | |x| | |
   Use specific-dest addr as Echo Reply src     |3.2.2.6 |x| | | | |
   Send same data in Echo Reply                 |3.2.2.6 |x| | | | |
   Pass Echo Reply to higher layer              |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 Information Request or Reply:             |3.2.2.7 | | | |x| |
 ICMP Timestamp and Timestamp Reply:            |3.2.2.8 | | |x| | |
   Minimize delay variability                   |3.2.2.8 | |x| | | |1
   Silently discard b'cast Timestamp            |3.2.2.8 | | |x| | |1
   Silently discard m'cast Timestamp            |3.2.2.8 | | |x| | |1
   Use specific-dest addr as TS Reply src       |3.2.2.8 |x| | | | |1
   Reflect Record Route, Time Stamp options     |3.2.2.6 | |x| | | |1
   Reverse and reflect Source Route option      |3.2.2.8 |x| | | | |1
   Pass Timestamp Reply to higher layer         |3.2.2.8 |x| | | | |1
   Obey rules for "standard value"              |3.2.2.8 |x| | | | |1
                                                |        | | | | | |
 ICMP Address Mask Request and Reply:           |        | | | | | |
   Addr Mask source configurable                |3.2.2.9 |x| | | | |
   Support static configuration of addr mask    |3.2.2.9 |x| | | | |
   Get addr mask dynamically during booting     |3.2.2.9 | | |x| | |
   Get addr via ICMP Addr Mask Request/Reply    |3.2.2.9 | | |x| | |
     Retransmit Addr Mask Req if no Reply       |3.2.2.9 |x| | | | |3
     Assume default mask if no Reply            |3.2.2.9 | |x| | | |3
     Update address mask from first Reply only  |3.2.2.9 |x| | | | |3
   Reasonableness check on Addr Mask            |3.2.2.9 | |x| | | |
   Send unauthorized Addr Mask Reply msgs       |3.2.2.9 | | | | |x|
     Explicitly configured to be agent          |3.2.2.9 |x| | | | |
   Static config=> Addr-Mask-Authoritative flag |3.2.2.9 | |x| | | |
     Broadcast Addr Mask Reply when init.       |3.2.2.9 |x| | | | |3
                                                |        | | | | | |

ROUTING OUTBOUND DATAGRAMS: | | | | | | |

 Use address mask in local/remote decision      |3.3.1.1 |x| | | | |
 Operate with no gateways on conn network       |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| | | |
 If no cache entry, use default gateway         |3.3.1.2 |x| | | | |
   Support multiple default gateways            |3.3.1.2 |x| | | | |
 Provide table of static routes                 |3.3.1.2 | | |x| | |
   Flag: route overridable by 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| | | |
                                                |        | | | | | |
 Able to detect failure of next-hop gateway     |3.3.1.4 |x| | | | |
 Assume route is good forever                   |3.3.1.4 | | | |x| |

Internet Engineering Task Force [Page 74]

RFC1122 INTERNET LAYER October 1989

 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| | | | |
                                                |        | | | | | |

REASSEMBLY and FRAGMENTATION: | | | | | | |

 Able to reassemble incoming datagrams          |3.3.2   |x| | | | |
   At least 576 byte datagrams                  |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| | | |
 All-Subnets-MTU configuration flag             |3.3.3   | | |x| | |
                                                |        | | | | | |

MULTIHOMING: | | | | | | |

 Reply with same addr as spec-dest addr         |3.3.4.2 | |x| | | |
 Allow application to choose local IP addr      |3.3.4.2 |x| | | | |
 Silently discard d'gram in "wrong" interface   |3.3.4.2 | | |x| | |
 Only send d'gram through "right" interface     |3.3.4.2 | | |x| | |4
                                                |        | | | | | |

SOURCE-ROUTE FORWARDING: | | | | | | |

 Forward datagram with Source Route option      |3.3.5   | | |x| | |1
   Obey corresponding gateway rules             |3.3.5   |x| | | | |1
     Update TTL by gateway rules                |3.3.5   |x| | | | |1
     Able to generate ICMP err code 4, 5        |3.3.5   |x| | | | |1
     IP src addr not local host                 |3.3.5   | | |x| | |1
     Update Timestamp, Record Route options     |3.3.5   |x| | | | |1
   Configurable switch for non-local SRing      |3.3.5   |x| | | | |1
     Defaults to OFF                            |3.3.5   |x| | | | |1
   Satisfy gwy access rules for non-local SRing |3.3.5   |x| | | | |1
   If not forward, send Dest Unreach (cd 5)     |3.3.5   | |x| | | |2
                                                |        | | | | | |

BROADCAST: | | | | | | |

 Broadcast addr as IP source addr               |3.2.1.3 | | | | |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| | |
   Default to -1 broadcast                      |3.3.6   | |x| | | |
 Recognize all broadcast address formats        |3.3.6   |x| | | | |
 Use IP b'cast/m'cast addr in link-layer b'cast |3.3.6   |x| | | | |
 Silently discard link-layer-only b'cast dg's   |3.3.6   | |x| | | |
 Use Limited Broadcast addr for connected net   |3.3.6   | |x| | | |

Internet Engineering Task Force [Page 75]

RFC1122 INTERNET LAYER October 1989

                                                |        | | | | | |

MULTICAST: | | | | | | |

 Support local IP multicasting (RFC-1112)       |3.3.7   | |x| | | |
 Support IGMP (RFC-1112)                        |3.3.7   | | |x| | |
 Join all-hosts group at startup                |3.3.7   | |x| | | |
 Higher layers learn i'face m'cast capability   |3.3.7   | |x| | | |
                                                |        | | | | | |

INTERFACE: | | | | | | |

 Allow transport layer to use all IP mechanisms |3.4     |x| | | | |
 Pass interface ident up to transport layer     |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 transp. layer  |3.4     |x| | | | |
    Include IP hdr+8 octets or more from orig.  |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 implemented and is configured "on".

(4) Unless has embedded gateway functionality or is source routed.

Internet Engineering Task Force [Page 76]

RFC1122 TRANSPORT LAYER -- UDP October 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 datagram delivery -- and
        gives 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 to the
           IP layer.

Internet Engineering Task Force [Page 77]

RFC1122 TRANSPORT LAYER -- UDP October 1989

           DISCUSSION:
                At present, the only options that need be passed
                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 format or content of options it
                passes to or from the application; an exception to this
                might be an IP-layer security option.

                An application based on UDP will need to obtain a
                source route from a request datagram and supply a
                reversed route for sending the corresponding reply.

        4.1.3.3  ICMP Messages

           UDP MUST pass to the application layer all ICMP error
           messages that it receives from the IP 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 messages 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

Internet Engineering Task Force [Page 78]

RFC1122 TRANSPORT LAYER -- UDP October 1989

                efficiency.  As a result, numerous cases of undetected
                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, its specific-destination
           address MUST be passed up to the application layer.

           An application program MUST be able to specify the IP source
           address to be used for sending a UDP datagram or to leave it
           unspecified (in which case the networking software will
           choose an appropriate source address).  There SHOULD be a
           way to communicate the chosen source address up to the
           application layer (e.g, so that the application can later
           receive a reply datagram only from the corresponding
           interface).

           DISCUSSION:
                A request/response application that uses UDP should use
                a source address for the response that is the same as
                the specific destination address of the request.  See
                the "General Issues" section of [INTRO:1].

        4.1.3.6  Invalid Addresses

           A UDP datagram received with an invalid IP source address
           (e.g., a broadcast or multicast address) must be discarded
           by UDP or by the IP layer (see Section 3.2.1.3).

           When a host sends a UDP datagram, the source address MUST be
           (one of) the IP address(es) of the host.

     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 79]

RFC1122 TRANSPORT LAYER -- UDP October 1989

        document.  Thus, an application using UDP needs the functions
        of the GET_SRCADDR(), 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 spec-dest addr to application            |4.1.3.5 |x| | | | |

Internet Engineering Task Force [Page 80]

RFC1122 TRANSPORT LAYER -- UDP October 1989

- 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| | | |
                                                |        | | | | | |

Bad IP src addr silently discarded by UDP/IP |4.1.3.6 |x| | | | | Only send valid IP source address |4.1.3.6 |x| | | | | UDP Application Interface Services | | | | | | | Full IP interface of 3.4 for application |4.1.4 |x| | | | |

- Able to spec TTL, TOS, IP opts when send dg   |4.1.4   |x| | | | |
- Pass received TOS up to applic layer          |4.1.4   | | |x| | |

Internet Engineering Task Force [Page 81]

RFC1122 TRANSPORT LAYER -- TCP October 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 application-layer protocols
        are described in [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 0-255 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:6].  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 82]

RFC1122 TRANSPORT LAYER -- TCP October 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 PUSH flags
           are not implemented, then the sending TCP: (1) must not
           buffer data indefinitely, and (2) 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 (even if less than a full
                buffer has been received). 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

Internet Engineering Task Force [Page 83]

RFC1122 TRANSPORT LAYER -- TCP October 1989

           and TCP will not work.  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.

           DISCUSSION:
                It is known that the window field in the TCP header is
                too small for high-speed, long-delay paths.
                Experimental TCP options have been defined to extend
                the window size; see for example [TCP:11].  In
                anticipation of the adoption of such an extension, TCP
                implementors should treat windows as 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 [INTRO:1]).

                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.

Internet Engineering Task Force [Page 84]

RFC1122 TRANSPORT LAYER -- TCP October 1989

        4.2.2.5  TCP Options: RFC-793 Section 3.1

           A TCP MUST be able to receive a TCP option in any segment.
           A TCP MUST ignore without error any TCP option it does not
           implement, assuming that the option has a length field (all
           TCP options defined in the future will 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 must be less than

Internet Engineering Task Force [Page 85]

RFC1122 TRANSPORT LAYER -- TCP October 1989

           or equal to:

              MMS_R - 20

           where MMS_R is the maximum size for a transport-layer
           message that can be received (and reassembled).  TCP obtains
           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.  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 to the IP layer the task of
                determining a suitable MTU for the Internet path.  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).

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           (c)  It is possible to go directly from FIN-WAIT-1 to the
                TIME-WAIT state (see page 75 of the spec).

        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

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           connection is closed by the remote site, the local
           application MUST be informed whether it closed normally or
           was aborted.

           The normal TCP close sequence delivers buffered data
           reliably in both directions.  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, and a host is permitted to continue sending data
           in the open direction on a half-closed connection.

           A host MAY implement a "half-duplex" TCP close sequence, so
           that 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 2xMSL (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 larger 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:
                TCP's full-duplex data-preserving close is a feature
                that is not included in the analogous ISO transport
                protocol TP4.

                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 2xMSL, i.e.,
                4 minutes.  During this period, the (remote socket,

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                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, 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 to character-at-
                a-time hosts, tend to send streams of one-octet data
                segments.  To avoid deadlocks, every TCP SEND call from
                such applications must be "pushed", either explicitly
                by the application or else implicitly by TCP.  The
                result may be a stream of TCP segments that contain one
                data octet each, which 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

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                characters over Telnet connections; this may initially
                surprise users accustomed to single-character echo, but
                user acceptance has not been a problem.

                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.  For
                example, 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

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                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
                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 SHOULD 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 4.2.3.4) to become negative.

           If this happens, the sender SHOULD NOT send new data, but
           SHOULD retransmit normally the old unacknowledged data
           between SND.UNA and SND.UNA+SND.WND.  The sender MAY also
           retransmit old data beyond SND.UNA+SND.WND, but SHOULD NOT
           time out the connection if data beyond the right window edge
           is not acknowledged.  If the window shrinks to zero, the TCP
           MUST probe it in the standard way (see next Section).

           DISCUSSION:
                Many TCP implementations become confused if the window
                shrinks from the right after data has been sent into a
                larger window.  Note that TCP has a heuristic to select
                the latest window update despite possible datagram
                reordering; as a result, it may ignore a window update
                with a smaller window than previously offered if
                neither the sequence number nor the acknowledgment
                number is increased.

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        4.2.2.17  Probing Zero Windows: RFC-793 Section 3.7, page 42

           Probing of zero (offered) windows MUST be supported.

           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.

                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.

           The transmitting host SHOULD send the first zero-window
           probe when a zero window has existed for the retransmission
           timeout period (see Section 4.2.2.15), and SHOULD increase
           exponentially the interval between successive probes.

           DISCUSSION:
                This procedure minimizes delay if the zero-window
                condition is due to a lost ACK segment containing a
                window-opening update.  Exponential backoff is
                recommended, possibly with some maximum interval not
                specified here.  This procedure is similar to that of
                the retransmission algorithm, and it may be possible to
                combine the two procedures in the implementation.

        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:

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                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 they may allow
                "cloning" of LISTEN-state connections from a single
                passive OPEN call.

        4.2.2.19  Time to Live: RFC-793 Section 3.9, page 52

           RFC-793 specified that TCP was to request the IP layer to
           send TCP segments with TTL = 60.  This is obsolete; the TTL
           value used to send TCP segments MUST be configurable.  See
           Section 3.2.1.7 for discussion.

        4.2.2.20  Event Processing: 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".

           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 of RFC-793.

           (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

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                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 =<
                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 when a
           valid segment arrives that is in the window but not at the
           left window edge.

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           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".   With this algorithm, the sender 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 chosen to compensate for the maximum
                likely segment reordering in the Internet.  There is
                not yet enough experience with the fast retransmit
                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
                that ambiguous round-trip times will not corrupt 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.

           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

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                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.

           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 is to 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 2*MSL, i.e., 240
           seconds.

           DISCUSSION:
                Experience has shown that these initialization values
                are reasonable, and that in any case the Karn and
                Jacobson algorithms make TCP behavior reasonably
                insensitive to the initial parameter choices.

        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 should not
           be excessively delayed; in particular, the delay MUST be
           less than 0.5 seconds, and in a stream of full-sized
           segments there SHOULD be an ACK for at least every second
           segment.

           DISCUSSION:

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                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 ACK's
                can disturb the round-trip timing and packet "clocking"
                algorithms [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].

           IMPLEMENTATION:
                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 to receiver SWS 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.

                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 requires that the receiver offer less
                than its full buffer space, i.e., the receiver must
                specify a RCV.WND that keeps RCV.NXT+RCV.WND constant
                as RCV.NXT increases.  Thus, the total buffer space
                RCV.BUFF is generally divided into three parts:

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                |<------- 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:  Eff.snd.MSS < RCV.BUFF/2).
                Note also that the receiver must use its own
                Eff.snd.MSS, assuming it is the same as the sender's.

        4.2.3.4  When to Send Data

           A TCP MUST include a SWS avoidance algorithm in the sender.

           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

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                     data (regardless of the PSH bit), until the
                     outstanding data has been acknowledged or until
                     the TCP can send a full-sized segment (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 the
                maximum rate.

           IMPLEMENTATION:
                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 is recommended.

                Send data:

                (1)  if a maximum-sized segment can be sent, i.e, if:

                          min(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);

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                (3)  or if at least a fraction Fs of the maximum window
                     can be sent, i.e., if:

                         [SND.NXT = SND.UNA and]

                                 min(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.  It may be convenient to combine this timer
                with the timer used to probe zero windows (Section
                4.2.2.17).

                Finally, note that the SWS avoidance algorithm just
                specified is to be used instead of the sender-side
                algorithm contained in [TCP:5].

        4.2.3.5  TCP Connection Failures

           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.  R1 and R2 might be measured in time units or
                as a count of retransmissions.

           (b)  When the number of transmissions of the same segment
                reaches or exceeds threshold R1, pass negative advice
                (see Section 3.3.1.4) to the IP layer, to trigger
                dead-gateway diagnosis.

           (c)  When the number of transmissions of the same segment
                reaches a threshold R2 greater than R1, close the
                connection.

           (d)  An application MUST be able to set the value for R2 for
                a particular connection.  For example, an interactive
                application might set R2 to "infinity," giving the user
                control over when to disconnect.

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           (d)  TCP SHOULD inform the application of the delivery
                problem (unless such information has been disabled by
                the application; see Section 4.2.4.1), when R1 is
                reached and before R2.  This will allow a remote login
                (User Telnet) application program to inform the user,
                for example.

           The value of R1 SHOULD correspond to at least 3
           retransmissions, at the current RTO.  The value of R2 SHOULD
           correspond to at least 100 seconds.

           An attempt to open a TCP connection could fail with
           excessive retransmissions 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 SYN
           and data segments.  In particular, R2 for a SYN segment MUST
           be set large enough to provide retransmission of the segment
           for at least 3 minutes.  The application can close the
           connection (i.e., give up on the 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.

        4.2.3.6  TCP Keep-Alives

           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 only be sent when no data or
           acknowledgement packets have been received for the
           connection within an interval.  This interval MUST be
           configurable and MUST default to no less than two hours.

           It is extremely important to remember that ACK segments that
           contain no data are not reliably transmitted by TCP.
           Consequently, if a keep-alive mechanism is implemented it
           MUST NOT interpret failure to respond to any specific probe
           as a dead connection.

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           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 compatibility with
           erroneous TCP implementations.

           DISCUSSION:
                A "keep-alive" mechanism periodically probes the other
                end of a connection when the connection is otherwise
                idle, even when there is no data to be sent.  The TCP
                specification does not include a keep-alive 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 and may or may not contain one garbage octet
                of data.  Note that on a quiet connection SND.NXT =
                RCV.NXT, so that this 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 RST instead of an acknowledgment
                segment.

                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
                server applications that might otherwise hang
                indefinitely and consume resources unnecessarily if a
                client crashes or aborts a connection during a network
                failure.

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        4.2.3.7  TCP Multihoming

           If an application on a multihomed host does not specify the
           local IP address when actively opening a TCP connection,
           then the TCP MUST ask the IP layer to select a local IP
           address before sending the (first) SYN.  See the function
           GET_SRCADDR() in Section 3.4.

           At all other times, a previous segment has either been sent
           or received on this connection, and TCP MUST use the same
           local address is used that 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
                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

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                conditions, TCP MUST NOT abort the connection, and it
                SHOULD make the information available to the
                application.

                DISCUSSION:
                     TCP could report the soft error condition directly
                     to the application layer with an upcall to the
                     ERROR_REPORT routine, or it could 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, 5 (see above).

           o    Parameter Problem

                This should be handled the same way as Destination
                Unreachable codes 0, 1, 5 (see above).

        4.2.3.10  Remote Address Validation

           A TCP implementation MUST reject as an error a local OPEN
           call for an invalid remote IP address (e.g., a broadcast or
           multicast address).

           An incoming SYN with an invalid source address must be
           ignored either by TCP or by the IP layer (see Section
           3.2.1.3).

           A TCP implementation MUST silently discard an incoming SYN
           segment that is addressed to a broadcast or multicast
           address.

        4.2.3.11  TCP Traffic Patterns

           IMPLEMENTATION:
                The TCP protocol specification [TCP:1] gives the
                implementor much freedom in designing the algorithms
                that control the message flow over the connection --
                packetizing, managing the window, sending

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                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

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                     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).

           However, an application program that does not want to
           receive such ERROR_REPORT calls SHOULD be able to

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           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.

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        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,] ... )

           to allow the specification of the local IP address.

           DISCUSSION:
                Some TCP-based applications need to specify the local
                IP address to be used to open a particular connection;
                FTP is an example.

           IMPLEMENTATION:
                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| | |

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   If cannot: sender buffer indefinitely        |4.2.2.2 | | | | |x|
   If cannot: PSH last segment                  |4.2.2.2 |x| | | | |
 Notify receiving ALP of PSH                    |4.2.2.2 | | |x| | |1
 Send max size segment when possible            |4.2.2.2 | |x| | | |
                                                |        | | | | | |

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| | | | |
 Receiver's window closed indefinitely          |4.2.2.17| | |x| | |
 Sender probe zero window                       |4.2.2.17|x| | | | |
   First probe after RTO                        |4.2.2.17| |x| | | |
   Exponential backoff                          |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| | | | |1
 ALP can learn if/how much urgent data Q'd      |4.2.2.4 |x| | | | |1
                                                |        | | | | | |

TCP Options | | | | | | |

 Receive TCP option in any segment              |4.2.2.5 |x| | | | |
 Ignore unsupported options                     |4.2.2.5 |x| | | | |
 Cope with illegal option length                |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 option always                         |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| | | | |
 Ask IP for src address for SYN if necc.        |4.2.3.7 |x| | | | |
   Otherwise, use local addr of conn.           |4.2.3.7 |x| | | | |
 OPEN to broadcast/multicast IP Address         |4.2.3.14| | | | |x|
 Silently discard seg to bcast/mcast addr       |4.2.3.14|x| | | | |

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                                                |        | | | | | |

Closing Connections | | | | | | |

 RST can contain data                           |4.2.2.12| |x| | | |
 Inform application of aborted conn             |4.2.2.13|x| | | | |
 Half-duplex close connections                  |4.2.2.13| | |x| | |
   Send RST to indicate data lost               |4.2.2.13| |x| | | |
 In TIME-WAIT state for 2xMSL seconds           |4.2.2.13|x| | | | |
   Accept SYN from TIME-WAIT state              |4.2.2.13| | |x| | |
                                                |        | | | | | |

Retransmissions | | | | | | |

 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| | | |
 Recommended initial values and bounds          |4.2.3.1 | |x| | | |
                                                |        | | | | | |

Generating ACK's: | | | | | | |

 Queue out-of-order segments                    |4.2.2.20| |x| | | |
 Process all Q'd before send ACK                |4.2.2.20|x| | | | |
 Send ACK for out-of-order segment              |4.2.2.21| | |x| | |
 Delayed ACK's                                  |4.2.3.2 | |x| | | |
   Delay < 0.5 seconds                          |4.2.3.2 |x| | | | |
   Every 2nd full-sized segment ACK'd           |4.2.3.2 |x| | | | |
 Receiver SWS-Avoidance Algorithm               |4.2.3.3 |x| | | | |
                                                |        | | | | | |

Sending data | | | | | | |

 Configurable TTL                               |4.2.2.19|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| | | | |
                                                |        | | | | | |

Connection Failures: | | | | | | |

 Negative advice to IP on R1 retxs              |4.2.3.5 |x| | | | |
 Close connection on R2 retxs                   |4.2.3.5 |x| | | | |
 ALP can set R2                                 |4.2.3.5 |x| | | | |1
 Inform ALP of  R1<=retxs<R2                    |4.2.3.5 | |x| | | |1
 Recommended values for R1, R2                  |4.2.3.5 | |x| | | |
 Same mechanism for SYNs                        |4.2.3.5 |x| | | | |
   R2 at least 3 minutes for SYN                |4.2.3.5 |x| | | | |
                                                |        | | | | | |

Send Keep-alive Packets: |4.2.3.6 | | |x| | |

 - Application can request                      |4.2.3.6 |x| | | | |
 - Default is "off"                             |4.2.3.6 |x| | | | |
 - Only send if idle for interval               |4.2.3.6 |x| | | | |
 - Interval configurable                        |4.2.3.6 |x| | | | |

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 - Default at least 2 hrs.                      |4.2.3.6 |x| | | | |
 - Tolerant of lost ACK's                       |4.2.3.6 |x| | | | |
                                                |        | | | | | |

IP Options | | | | | | |

 Ignore options TCP doesn't understand          |4.2.3.8 |x| | | | |
 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| | | | |1
     Overrides src rt in datagram               |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 |4.2.3.9 |x| | | | |

 Dest. Unreach (0,1,5) => inform ALP            |4.2.3.9 | |x| | | |
 Dest. Unreach (0,1,5) => abort conn            |4.2.3.9 | | | | |x|
 Dest. Unreach (2-4) => abort conn              |4.2.3.9 | |x| | | |
 Source Quench => slow start                    |4.2.3.9 | |x| | | |
 Time Exceeded => tell ALP, don't abort         |4.2.3.9 | |x| | | |
 Param Problem => tell ALP, don't abort         |4.2.3.9 | |x| | | |
                                                |        | | | | | |

Address Validation | | | | | | |

 Reject OPEN call to invalid IP address         |4.2.3.10|x| | | | |
 Reject SYN from invalid IP address             |4.2.3.10|x| | | | |
 Silently discard SYN to bcast/mcast addr       |4.2.3.10|x| | | | |
                                                |        | | | | | |

TCP/ALP Interface Services | | | | | | |

 Error Report mechanism                         |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| | | | |
   Passed unchanged to IP                       |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| | |
 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 111]

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5. REFERENCES

INTRODUCTORY REFERENCES

[INTRO:1] "Requirements for Internet Hosts -- Application and Support,"

    IETF Host Requirements Working Group, R. Braden, Ed., RFC-1123,
    October 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,

Internet Engineering Task Force [Page 112]

RFC1122 TRANSPORT LAYER -- TCP October 1989

    March 1985.  Also in: IEEE Communications Magazine, March 1985.
    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-1112,

    August 1989.

[IP:5] "Military Standard Internet Protocol," MIL-STD-1777, Department

    of Defense, August 1983.

    This specification, as amended by RFC-963, is intended to describe

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    the Internet Protocol but has some serious omissions (e.g., the
    mandatory subnet extension [IP:3] and the optional multicasting
    extension [IP:4]).  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] "Internet Protocol Security Options," B. Schofield, RFC-1108,

    October 1989.

[IP:9] "Fragmentation Considered Harmful," C. Kent and J. Mogul, ACM

    SIGCOMM-87, 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.

[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

    Introduced Delay (SQUID)," W. Prue and J. Postel, RFC-1016, July
    1987.

    This RFC first described directed broadcast addresses.  However,
    the bulk of the RFC is concerned with gateways, not hosts.

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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:

[TCP:8] "Modularity and Efficiency in Protocol Implementation," D.

    Clark, RFC-817, July 1982.

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RFC1122 TRANSPORT LAYER -- TCP October 1989

[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.

Security Considerations

  There are many security issues in the communication layers of host
  software, but a full discussion is beyond the scope of this RFC.

  The Internet architecture generally provides little protection
  against spoofing of IP source addresses, so any security mechanism
  that is based upon verifying the IP source address of a datagram
  should be treated with suspicion.  However, in restricted
  environments some source-address checking may be possible.  For
  example, there might be a secure LAN whose gateway to the rest of the
  Internet discarded any incoming datagram with a source address that
  spoofed the LAN address.  In this case, a host on the LAN could use
  the source address to test for local vs. remote source.  This problem
  is complicated by source routing, and some have suggested that
  source-routed datagram forwarding by hosts (see Section 3.3.5) should
  be outlawed for security reasons.

  Security-related issues are mentioned in sections concerning the IP
  Security option (Section 3.2.1.8), the ICMP Parameter Problem message
  (Section 3.2.2.5), IP options in UDP datagrams (Section 4.1.3.2), and
  reserved TCP ports (Section 4.2.2.1).

Author's Address

  Robert Braden
  USC/Information Sciences Institute
  4676 Admiralty Way
  Marina del Rey, CA 90292-6695

  Phone: (213) 822 1511

  EMail: [email protected]

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