1. RFC 1122
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.


   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
    ARP Cache Validation .........................   22
    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
    Version Number ...............................   29
    Checksum .....................................   29
    Addressing ...................................   29
    Fragmentation and Reassembly .................   32
    Identification ...............................   32
    Type-of-Service ..............................   33
    Time-to-Live .................................   34
    Options ......................................   35
         3.2.2 Internet Control Message Protocol -- ICMP ..........   38
    Destination Unreachable ......................   39
    Redirect .....................................   40
    Source Quench ................................   41
    Time Exceeded ................................   41
    Parameter Problem ............................   42
    Echo Request/Reply ...........................   42
    Information Request/Reply ....................   43
    Timestamp and Timestamp Reply ................   43
    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
    Local/Remote Decision ........................   47
    Gateway Selection ............................   48
    Route Cache ..................................   49
    Dead Gateway Detection .......................   51
    New Gateway Selection ........................   55
    Initialization ...............................   56
         3.3.2  Reassembly ........................................   56
         3.3.3  Fragmentation .....................................   58
         3.3.4  Local Multihoming .................................   60
    Introduction .................................   60
    Multihoming Requirements .....................   61
    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
    Ports ........................................   77
    IP Options ...................................   77
    ICMP Messages ................................   78
    UDP Checksums ................................   78
    UDP Multihoming ..............................   79
    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
    Well-Known Ports .............................   82
    Use of Push ..................................   82
    Window Size ..................................   83
    Urgent Pointer ...............................   84
    TCP Options ..................................   85
    Maximum Segment Size Option ..................   85
    TCP Checksum .................................   86
    TCP Connection State Diagram .................   86
    Initial Sequence Number Selection ............   87
    Simultaneous Open Attempts ..................   87
    Recovery from Old Duplicate SYN .............   87
    RST Segment .................................   87
    Closing a Connection ........................   87
    Data Communication ..........................   89
    Retransmission Timeout ......................   90
    Managing the Window .........................   91
    Probing Zero Windows ........................   92
    Passive OPEN Calls ..........................   92
    Time to Live ................................   93
    Event Processing ............................   93
    Acknowledging Queued Segments ...............   94
         4.2.3  SPECIFIC ISSUES ...................................   95
    Retransmission Timeout Calculation ...........   95
    When to Send an ACK Segment ..................   96
    When to Send a Window Update .................   97
    When to Send Data ............................   98

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    TCP Connection Failures ......................  100
    TCP Keep-Alives ..............................  101
    TCP Multihoming ..............................  103
    IP Options ...................................  103
    ICMP Messages ................................  103
    Remote Address Validation ...................  104
    TCP Traffic Patterns ........................  104
    Efficiency ..................................  105
         4.2.4  TCP/APPLICATION LAYER INTERFACE ...................  106
    Asynchronous Reports .........................  106
    Type-of-Service ..............................  107
    Flush Call ...................................  107
    Multihoming ..................................  108
         4.2.5  TCP REQUIREMENT SUMMARY ...........................  108

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

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   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

      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

         (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

         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

         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

      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

         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

         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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         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

   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

         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

         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

         (3)  Specific Issues -- discusses protocol design and
              implementation issues that were not included in the walk-

         (4)  Interfaces -- discusses the service interface to the next
              higher layer.

         (5)  Summary -- contains a summary of the requirements of the

         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

         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:

              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.

              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-

Internet Engineering Task Force                                [Page 17]
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.

              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.

              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

              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.

Internet Engineering Task Force                                [Page 18]
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

              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.

              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 |

                    <--------  Datagram ------------------>
                             <-------- Message ----------->
           or, for TCP:
                   | IP hdr |  TCP hdr | Application Data |

                    <--------  Datagram ------------------>
                             <-------- Segment ----------->

Internet Engineering Task Force                                [Page 19]
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

      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



      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.




      2.3.1  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.

              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

              On an Ethernet, packets encapsulated with trailers use a
              distinct Ethernet type [LINK:1], and trailer negotiation
              is performed at the time that ARP is used to discover the
              link-layer address of a destination system.

Internet Engineering Task Force                                [Page 21]
RFC1122                        LINK LAYER                   October 1989

              Specifically, the ARP exchange is completed in the usual
              manner using the normal IP protocol type, but a host that
              wants to speak trailers will send an additional "trailer
              ARP reply" packet, i.e., an ARP reply that specifies the
              trailer encapsulation protocol type but otherwise has the
              format of a normal ARP reply.  If a host configured to use
              trailers receives a trailer ARP reply message from a
              remote machine, it can add that machine to the list of
              machines that understand trailers, e.g., by marking the
              corresponding entry in the ARP cache.

              Hosts wishing to receive trailer encapsulations send
              trailer ARP replies whenever they complete exchanges of
              normal ARP messages for IP.  Thus, a host that received an
              ARP request for its IP protocol address would send a
              trailer ARP reply in addition to the normal IP 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

              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

      2.3.2  Address Resolution Protocol -- ARP
  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

Internet Engineering Task Force                                [Page 22]
RFC1122                        LINK LAYER                   October 1989


                 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.

                 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

Internet Engineering Task Force                                [Page 23]
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.
  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.

                 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

         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-

Internet Engineering Task Force                                [Page 24]
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.

              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.


      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.

Internet Engineering Task Force                                [Page 25]
RFC1122                        LINK LAYER                   October 1989

           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

      The packet send interface between the IP and link layers MUST
      include the 5-bit TOS field (see Section

      The link layer MUST NOT report a Destination Unreachable error to
      IP solely because there is no ARP cache entry for a destination.


                                                  |       | | | |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             ||x| | | | |
  Prevent ARP floods                              ||x| | | | |
  Cache timeout configurable                      || |x| | | |
  Save at least one (latest) unresolved pkt       || |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



      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

      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

Internet Engineering Task Force                                [Page 27]
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

Internet Engineering Task Force                                [Page 28]
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.

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


      3.2.1 Internet Protocol -- IP
  Version Number: RFC-791 Section 3.1

            A datagram whose version number is not 4 MUST be silently
  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
  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

Internet Engineering Task Force                                [Page 29]
RFC1122                      INTERNET LAYER                 October 1989

            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

                { <Network-number>, <Host-number> }

                { <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

                 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.

Internet Engineering Task Force                                [Page 30]
RFC1122                      INTERNET LAYER                 October 1989

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

                 Directed broadcast to all subnets of the specified
                 subnetted network.  It MUST NOT be used as a source

            (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

            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 and

            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

            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.

                 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.
  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.
  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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                 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.
  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

            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.

                 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

                 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.
  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.

                 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.
  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.

                 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

                 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

                 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-

                 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.

                      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 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
               Redirect                  (see Section
               Source Quench             (see Section
               Time Exceeded             (see Section
               Parameter Problem         (see Section

              ICMP query messages:

                Echo                     (see Section
                Information              (see Section
                Timestamp                (see Section
                Address Mask             (see Section

         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

         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

         *    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


              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

              This requires that the link layer inform the IP layer when
              a link-layer broadcast datagram has been received; see
              Section 2.4.
  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

            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

            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
  ,, 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).
  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 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).
  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.

                 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
  Time Exceeded: RFC-792

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

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                 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.
  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.

                 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
              Code 1 = required option is missing.

                 This variant is currently in use in the military
                 community for a missing security option.
  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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                 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
   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.
  Information Request/Reply: RFC-792

            A host SHOULD NOT implement these messages.

                 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.
  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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  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

            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

                 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, 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.

                 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.


      3.3.1  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.
  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

            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

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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.
  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.

                 This recommendation is to protect against gateways that
                 erroneously send Network Redirects for a subnetted
                 network, in violation of the gateway requirements

            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

                 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

                 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.
  Route Cache

            Each route cache entry needs to include the following

            (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 for a discussion of the implications of
            multihoming for the lookup procedure in this cache.

                 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

                 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, Redirect messages
                      will generally result in entries keyed on
                      destination host addresses; the simplest and most
                      general scheme would be to use host addresses

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

                 (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.

                 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.
  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

            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

            *    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

                 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

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

                 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

                 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
  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.

                 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.

                 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.

            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].

                 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).

              A fixed EMTU_R limit should not be built into the code
              because some application layer protocols require EMTU_R
              values larger than 576.

              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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              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).

              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

              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

         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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         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

              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

              (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

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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

            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

                 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.
  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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            (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 and 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

            (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.

                 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

                 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

Internet Engineering Task Force                                [Page 63]
RFC1122                      INTERNET LAYER                 October 1989

  Choosing a Source Address

                 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

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

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

                 (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
             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.

                 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

Internet Engineering Task Force                                [Page 64]
RFC1122                      INTERNET LAYER                 October 1989

                 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

              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

              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

              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

              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

              A host that is forwarding a source-routed datagram

Internet Engineering Task Force                                [Page 65]
RFC1122                      INTERNET LAYER                 October 1989

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

         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

         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 defined the four standard IP broadcast address

           Limited Broadcast:  {-1, -1}

           Directed Broadcast:  {<Network-number>,-1}

           Subnet Directed Broadcast:

           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.

Internet Engineering Task Force                                [Page 66]
RFC1122                      INTERNET LAYER                 October 1989

         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)

         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.

              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, 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.

Internet Engineering Task Force                                [Page 67]
RFC1122                      INTERNET LAYER                 October 1989

              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 ( when the IP layer is initialized and
         remain a member for as long as the IP layer is active.

              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

         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.

Internet Engineering Task Force                                [Page 68]
RFC1122                      INTERNET LAYER                 October 1989

      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.

              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.


      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.

           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

      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

      *    Receive Datagram

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

Internet Engineering Task Force                                [Page 69]
RFC1122                      INTERNET LAYER                 October 1989

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

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

           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

      *    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 The other parameters were
           defined earlier.

      *    Send ICMP Message

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

Internet Engineering Task Force                                [Page 70]
RFC1122                      INTERNET LAYER                 October 1989

                (Parameters defined in RFC-791).

           Passing an Id parameter is optional; see Section
           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

           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

           In the future, there may be additions to this interface to
           pass path data (see Section between the IP and
           transport layers.

Internet Engineering Task Force                                [Page 71]
RFC1122                      INTERNET LAYER                 October 1989


                                                 |        | | | |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                    | |x| | | | |
Verify IP checksum, silently discard bad dgram   | |x| | | | |
Addressing:                                      |        | | | | | |
  Subnet addressing (RFC-950)                    | |x| | | | |
  Src address must be host's own IP address      | |x| | | | |
  Silently discard datagram with bad dest addr   | |x| | | | |
  Silently discard datagram with bad src addr    | |x| | | | |
Support reassembly                               | |x| | | | |
Retain same Id field in identical datagram       | | | |x| | |
                                                 |        | | | | | |
TOS:                                             |        | | | | | |
  Allow transport layer to set TOS               | |x| | | | |
  Pass received TOS up to transport layer        | | |x| | | |
  Use RFC-795 link-layer mappings for TOS        | | | | |x| |
TTL:                                             |        | | | | | |
  Send packet with TTL of 0                      | | | | | |x|
  Discard received packets with TTL < 2          | | | | | |x|
  Allow transport layer to set TTL               | |x| | | | |
  Fixed TTL is configurable                      | |x| | | | |
                                                 |        | | | | | |
IP Options:                                      |        | | | | | |
  Allow transport layer to send IP options       | |x| | | | |
  Pass all IP options rcvd to higher layer       | |x| | | | |

Internet Engineering Task Force                                [Page 72]
RFC1122                      INTERNET LAYER                 October 1989

  IP layer silently ignore unknown options       | |x| | | | |
  Security option                                || | |x| | |
  Send Stream Identifier option                  || | | |x| |
  Silently ignore Stream Identifer option        ||x| | | | |
  Record Route option                            || | |x| | |
  Timestamp option                               || | |x| | |
Source Route Option:                             |        | | | | | |
  Originate & terminate Source Route options     ||x| | | | |
  Datagram with completed SR passed up to TL     ||x| | | | |
  Build correct (non-redundant) return route     ||x| | | | |
  Send multiple SR options in one header         || | | | |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)         | | |x| | | |
    Pass ICMP Dest Unreachable to higher layer   | |x| | | | |
    Higher layer act on Dest Unreach             | | |x| | | |
      Interpret Dest Unreach as only hint        | |x| | | | |
  Redirect:                                      |        | | | | | |
    Host send Redirect                           | | | | |x| |
    Update route cache when recv Redirect        | |x| | | | |
    Handle both Host and Net Redirects           | |x| | | | |
    Discard illegal Redirect                     | | |x| | | |
  Source Quench:                                 |        | | | | | |
    Send Source Quench if buffering exceeded     | | | |x| | |
    Pass Source Quench to higher layer           | |x| | | | |
    Higher layer act on Source Quench            | | |x| | | |
  Time Exceeded: pass to higher layer            | |x| | | | |
  Parameter Problem:                             |        | | | | | |
    Send Parameter Problem messages              | | |x| | | |
    Pass Parameter Problem to higher layer       | |x| | | | |
    Report Parameter Problem to user             | | | |x| | |
                                                 |        | | | | | |
  ICMP Echo Request or Reply:                    |        | | | | | |
    Echo server and Echo client                  | |x| | | | |

Internet Engineering Task Force                                [Page 73]
RFC1122                      INTERNET LAYER                 October 1989

    Echo client                                  | | |x| | | |
    Discard Echo Request to broadcast address    | | | |x| | |
    Discard Echo Request to multicast address    | | | |x| | |
    Use specific-dest addr as Echo Reply src     | |x| | | | |
    Send same data in Echo Reply                 | |x| | | | |
    Pass Echo Reply to higher layer              | |x| | | | |
    Reflect Record Route, Time Stamp options     | | |x| | | |
    Reverse and reflect Source Route option      | |x| | | | |
                                                 |        | | | | | |
  ICMP Information Request or Reply:             | | | | |x| |
  ICMP Timestamp and Timestamp Reply:            | | | |x| | |
    Minimize delay variability                   | | |x| | | |1
    Silently discard b'cast Timestamp            | | | |x| | |1
    Silently discard m'cast Timestamp            | | | |x| | |1
    Use specific-dest addr as TS Reply src       | |x| | | | |1
    Reflect Record Route, Time Stamp options     | | |x| | | |1
    Reverse and reflect Source Route option      | |x| | | | |1
    Pass Timestamp Reply to higher layer         | |x| | | | |1
    Obey rules for "standard value"              | |x| | | | |1
                                                 |        | | | | | |
  ICMP Address Mask Request and Reply:           |        | | | | | |
    Addr Mask source configurable                | |x| | | | |
    Support static configuration of addr mask    | |x| | | | |
    Get addr mask dynamically during booting     | | | |x| | |
    Get addr via ICMP Addr Mask Request/Reply    | | | |x| | |
      Retransmit Addr Mask Req if no Reply       | |x| | | | |3
      Assume default mask if no Reply            | | |x| | | |3
      Update address mask from first Reply only  | |x| | | | |3
    Reasonableness check on Addr Mask            | | |x| | | |
    Send unauthorized Addr Mask Reply msgs       | | | | | |x|
      Explicitly configured to be agent          | |x| | | | |
    Static config=> Addr-Mask-Authoritative flag | | |x| | | |
      Broadcast Addr Mask Reply when init.       | |x| | | | |3
                                                 |        | | | | | |
ROUTING OUTBOUND DATAGRAMS:                      |        | | | | | |
  Use address mask in local/remote decision      | |x| | | | |
  Operate with no gateways on conn network       | |x| | | | |
  Maintain "route cache" of next-hop gateways    | |x| | | | |
  Treat Host and Net Redirect the same           | | |x| | | |
  If no cache entry, use default gateway         | |x| | | | |
    Support multiple default gateways            | |x| | | | |
  Provide table of static routes                 | | | |x| | |
    Flag: route overridable by Redirects         | | | |x| | |
  Key route cache on host, not net address       | | | |x| | |
  Include TOS in route cache                     | | |x| | | |
                                                 |        | | | | | |
  Able to detect failure of next-hop gateway     | |x| | | | |
  Assume route is good forever                   | | | | |x| |

Internet Engineering Task Force                                [Page 74]
RFC1122                      INTERNET LAYER                 October 1989

  Ping gateways continuously                     | | | | | |x|
  Ping only when traffic being sent              | |x| | | | |
  Ping only when no positive indication          | |x| | | | |
  Higher and lower layers give advice            | | |x| | | |
  Switch from failed default g'way to another    | |x| | | | |
  Manual method of entering config info          | |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         | | |x| | | |
  Allow application to choose local IP addr      | |x| | | | |
  Silently discard d'gram in "wrong" interface   | | | |x| | |
  Only send d'gram through "right" interface     | | | |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               | | | | | |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| | | |


(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



      4.1.1  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.


         There are no known errors in the specification of UDP.

      4.1.3  SPECIFIC ISSUES

            UDP well-known ports follow the same rules as TCP well-known
            ports; see Section 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.
  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.

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RFC1122                  TRANSPORT LAYER -- UDP             October 1989

                 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.
  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

                 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
  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.

                 Some applications that normally run only across local
                 area networks have chosen to turn off UDP checksums for

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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.

                 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.
  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

                 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].
  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

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


         The application interface to UDP MUST provide the full services
         of the IP/transport interface described in Section 3.4 of this

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         document.  Thus, an application using UDP needs the functions
         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.


                                                 |        | | | |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                        | | |x| | | |
                                                 |        | | | | | |
IP Options in UDP                                |        | | | | | |
 - Pass rcv'd IP options to applic layer         | |x| | | | |
 - Applic layer can specify IP options in Send   | |x| | | | |
 - UDP passes IP options down to IP layer        | |x| | | | |
                                                 |        | | | | | |
Pass ICMP msgs up to applic layer                | |x| | | | |
                                                 |        | | | | | |
UDP checksums:                                   |        | | | | | |
 - Able to generate/check checksum               | |x| | | | |
 - Silently discard bad checksum                 | |x| | | | |
 - Sender Option to not generate checksum        | | | |x| | |
   - Default is to checksum                      | |x| | | | |
 - Receiver Option to require checksum           | | | |x| | |
                                                 |        | | | | | |
UDP Multihoming                                  |        | | | | | |
 - Pass spec-dest addr to application            | |x| | | | |

Internet Engineering Task Force                                [Page 80]
RFC1122                  TRANSPORT LAYER -- UDP             October 1989

 - Applic layer can specify Local IP addr        | |x| | | | |
 - Applic layer specify wild Local IP addr       | |x| | | | |
 - Applic layer notified of Local IP addr used   | | |x| | | |
                                                 |        | | | | | |
Bad IP src addr silently discarded by UDP/IP     | |x| | | | |
Only send valid IP source address                | |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


      4.2.1  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].

  Well-Known Ports: RFC-793 Section 2.7

                 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.
  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

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            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

                 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.
  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

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            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.

                 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.
  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.

                 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.

                 The generic ERROR-REPORT() upcall described in Section
        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

  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.
  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


            *    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

            *    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

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            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.

                 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.
  TCP Checksum: RFC-793 Section 3.1

            Unlike the UDP checksum (see Section, the TCP
            checksum is never optional.  The sender MUST generate it and
            the receiver MUST check it.
  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).

Internet Engineering Task Force                                [Page 86]
RFC1122                  TRANSPORT LAYER -- TCP             October 1989

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

  Initial Sequence Number Selection: RFC-793 Section
            3.3, page 27

            A TCP MUST use the specified clock-driven selection of
            initial sequence numbers.
  Simultaneous Open Attempts: RFC-793 Section 3.4, page

            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.

                 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.
  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.
  RST Segment: RFC-793 Section 3.4

            A TCP SHOULD allow a received RST segment to include data.

                 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
  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,

            (2)  returns to TIME-WAIT state if the SYN turns out to be
                 an old duplicate.

                 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.
  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, when to send an acknowledgment (Section
  , and when to update the window (Section

                 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
        and the receiving side (Section

                 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
                 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,
  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 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

                 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.
  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 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).

                 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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  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.

                 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

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

                 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.
  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.


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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.

                 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.
  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 for discussion.
  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

                 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

            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:


            (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

            (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.

            (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

  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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                 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

                 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
  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.

                 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

            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

                 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.
  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


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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

                 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].
  When to Send a Window Update

            A TCP MUST include a SWS avoidance algorithm in the receiver

                 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 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

                 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               ^

             1 - RCV.USER =  data received but not yet consumed;
             2 - RCV.WND =   space advertised to sender;
             3 - Reduction = space available but not yet

                 The suggested SWS avoidance algorithm for the receiver
                 is to keep RCV.NXT+RCV.WND fixed until the reduction

                      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  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.
  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

                 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

                 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.

                 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

                 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

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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

                 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

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

            (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, 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

                 Some Internet paths have significant setup times, and
                 the number of such paths is likely to increase in the
  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.

                 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

                 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

                 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

                 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

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  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
  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.
  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

                      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).

  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

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

                 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.


                 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

                 (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.

  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

            *    Excessive retransmissions (see

            *    Urgent pointer advance (see

            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.

                 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.

            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

                 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

                 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.
  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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            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.

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

                 A passive OPEN call with a specified "local IP address"
                 parameter will await an incoming connection request to
                 that address.  If the parameter is unspecified, a
                 passive OPEN will await an incoming connection request
                 to any local IP address, and then bind the local IP
                 address of the connection to the particular address
                 that is used.

                 For an active OPEN call, a specified "local IP address"
                 parameter will be used for opening the connection.  If
                 the parameter is unspecified, the networking software
                 will choose an appropriate local IP address (see
                 Section for the connection


                                                 |        | | | |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              | | | |x| | |
  Sender collapse successive PSH flags           | | |x| | | |
  SEND call can specify PUSH                     | | | |x| | |

Internet Engineering Task Force                               [Page 108]
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    If cannot: sender buffer indefinitely        | | | | | |x|
    If cannot: PSH last segment                  | |x| | | | |
  Notify receiving ALP of PSH                    | | | |x| | |1
  Send max size segment when possible            | | |x| | | |
                                                 |        | | | | | |
Window                                           |        | | | | | |
  Treat as unsigned number                       | |x| | | | |
  Handle as 32-bit number                        | | |x| | | |
  Shrink window from right                       || | | |x| |
  Robust against shrinking window                ||x| | | | |
  Receiver's window closed indefinitely          || | |x| | |
  Sender probe zero window                       ||x| | | | |
    First probe after RTO                        || |x| | | |
    Exponential backoff                          || |x| | | |
  Allow window stay zero indefinitely            ||x| | | | |
  Sender timeout OK conn with zero wind          || | | | |x|
                                                 |        | | | | | |
Urgent Data                                      |        | | | | | |
  Pointer points to last octet                   | |x| | | | |
  Arbitrary length urgent data sequence          | |x| | | | |
  Inform ALP asynchronously of urgent data       | |x| | | | |1
  ALP can learn if/how much urgent data Q'd      | |x| | | | |1
                                                 |        | | | | | |
TCP Options                                      |        | | | | | |
  Receive TCP option in any segment              | |x| | | | |
  Ignore unsupported options                     | |x| | | | |
  Cope with illegal option length                | |x| | | | |
  Implement sending & receiving MSS option       | |x| | | | |
  Send MSS option unless 536                     | | |x| | | |
  Send MSS option always                         | | | |x| | |
  Send-MSS default is 536                        | |x| | | | |
  Calculate effective send seg size              | |x| | | | |
                                                 |        | | | | | |
TCP Checksums                                    |        | | | | | |
  Sender compute checksum                        | |x| | | | |
  Receiver check checksum                        | |x| | | | |
                                                 |        | | | | | |
Use clock-driven ISN selection                   | |x| | | | |
                                                 |        | | | | | |
Opening Connections                              |        | | | | | |
  Support simultaneous open attempts             ||x| | | | |
  SYN-RCVD remembers last state                  ||x| | | | |
  Passive Open call interfere with others        || | | | |x|
  Function: simultan. LISTENs for same port      ||x| | | | |
  Ask IP for src address for SYN if necc.        | |x| | | | |
    Otherwise, use local addr of conn.           | |x| | | | |
  OPEN to broadcast/multicast IP Address         || | | | |x|
  Silently discard seg to bcast/mcast addr       ||x| | | | |

Internet Engineering Task Force                               [Page 109]
RFC1122                  TRANSPORT LAYER -- TCP             October 1989

                                                 |        | | | | | |
Closing Connections                              |        | | | | | |
  RST can contain data                           || |x| | | |
  Inform application of aborted conn             ||x| | | | |
  Half-duplex close connections                  || | |x| | |
    Send RST to indicate data lost               || |x| | | |
  In TIME-WAIT state for 2xMSL seconds           ||x| | | | |
    Accept SYN from TIME-WAIT state              || | |x| | |
                                                 |        | | | | | |
Retransmissions                                  |        | | | | | |
  Jacobson Slow Start algorithm                  ||x| | | | |
  Jacobson Congestion-Avoidance algorithm        ||x| | | | |
  Retransmit with same IP ident                  || | |x| | |
  Karn's algorithm                               | |x| | | | |
  Jacobson's RTO estimation alg.                 | |x| | | | |
  Exponential backoff                            | |x| | | | |
  SYN RTO calc same as data                      | | |x| | | |
  Recommended initial values and bounds          | | |x| | | |
                                                 |        | | | | | |
Generating ACK's:                                |        | | | | | |
  Queue out-of-order segments                    || |x| | | |
  Process all Q'd before send ACK                ||x| | | | |
  Send ACK for out-of-order segment              || | |x| | |
  Delayed ACK's                                  | | |x| | | |
    Delay < 0.5 seconds                          | |x| | | | |
    Every 2nd full-sized segment ACK'd           | |x| | | | |
  Receiver SWS-Avoidance Algorithm               | |x| | | | |
                                                 |        | | | | | |
Sending data                                     |        | | | | | |
  Configurable TTL                               ||x| | | | |
  Sender SWS-Avoidance Algorithm                 | |x| | | | |
  Nagle algorithm                                | | |x| | | |
    Application can disable Nagle algorithm      | |x| | | | |
                                                 |        | | | | | |
Connection Failures:                             |        | | | | | |
  Negative advice to IP on R1 retxs              | |x| | | | |
  Close connection on R2 retxs                   | |x| | | | |
  ALP can set R2                                 | |x| | | | |1
  Inform ALP of  R1<=retxs<R2                    | | |x| | | |1
  Recommended values for R1, R2                  | | |x| | | |
  Same mechanism for SYNs                        | |x| | | | |
    R2 at least 3 minutes for SYN                | |x| | | | |
                                                 |        | | | | | |
Send Keep-alive Packets:                         | | | |x| | |
  - Application can request                      | |x| | | | |
  - Default is "off"                             | |x| | | | |
  - Only send if idle for interval               | |x| | | | |
  - Interval configurable                        | |x| | | | |

Internet Engineering Task Force                               [Page 110]
RFC1122                  TRANSPORT LAYER -- TCP             October 1989

  - Default at least 2 hrs.                      | |x| | | | |
  - Tolerant of lost ACK's                       | |x| | | | |
                                                 |        | | | | | |
IP Options                                       |        | | | | | |
  Ignore options TCP doesn't understand          | |x| | | | |
  Time Stamp support                             | | | |x| | |
  Record Route support                           | | | |x| | |
  Source Route:                                  |        | | | | | |
    ALP can specify                              | |x| | | | |1
      Overrides src rt in datagram               | |x| | | | |
    Build return route from src rt               | |x| | | | |
    Later src route overrides                    | | |x| | | |
                                                 |        | | | | | |
Receiving ICMP Messages from IP                  | |x| | | | |
  Dest. Unreach (0,1,5) => inform ALP            | | |x| | | |
  Dest. Unreach (0,1,5) => abort conn            | | | | | |x|
  Dest. Unreach (2-4) => abort conn              | | |x| | | |
  Source Quench => slow start                    | | |x| | | |
  Time Exceeded => tell ALP, don't abort         | | |x| | | |
  Param Problem => tell ALP, don't abort         | | |x| | | |
                                                 |        | | | | | |
Address Validation                               |        | | | | | |
  Reject OPEN call to invalid IP address         ||x| | | | |
  Reject SYN from invalid IP address             ||x| | | | |
  Silently discard SYN to bcast/mcast addr       ||x| | | | |
                                                 |        | | | | | |
TCP/ALP Interface Services                       |        | | | | | |
  Error Report mechanism                         | |x| | | | |
  ALP can disable Error Report Routine           | | |x| | | |
  ALP can specify TOS for sending                | |x| | | | |
    Passed unchanged to IP                       | | |x| | | |
  ALP can change TOS during connection           | | |x| | | |
  Pass received TOS up to ALP                    | | | |x| | |
  FLUSH call                                     | | | |x| | |
  Optional local IP addr parm. in OPEN           | |x| | | | |


(1)  "ALP" means Application-Layer program.

Internet Engineering Task Force                               [Page 111]
RFC1122                  TRANSPORT LAYER -- TCP             October 1989



[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

     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: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: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

Internet Engineering Task Force                               [Page 113]
RFC1122                  TRANSPORT LAYER -- TCP             October 1989

     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

[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

     This and the following paper should be read by every implementor.

[IP:11] "Fault Isolation and Recovery," D. Clark, RFC-816, July 1982.


[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

[IP:14] "Something a Host Could Do with Source Quench: The Source Quench
     Introduced Delay (SQUID)," W. Prue and J. Postel, RFC-1016, July

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

Internet Engineering Task Force                               [Page 114]
RFC1122                  TRANSPORT LAYER -- TCP             October 1989


[UDP:1] "User Datagram Protocol," J. Postel, RFC-768, August 1980.


[TCP:1] "Transmission Control Protocol," J. Postel, RFC-793, September

[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.


[TCP:8] "Modularity and Efficiency in Protocol Implementation," D.
     Clark, RFC-817, July 1982.

Internet Engineering Task Force                               [Page 115]
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, the ICMP Parameter Problem message
   (Section, IP options in UDP datagrams (Section, and
   reserved TCP ports (Section

Author's Address

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

   Phone: (213) 822 1511

   EMail: Braden@ISI.EDU

Internet Engineering Task Force                               [Page 116]
  1. RFC 1122