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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

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       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.

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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-

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RFC1122 INTRODUCTION October 1989

            end through the Internet, a message must be encapsulated
            inside a datagram.
       IP Datagram
            An IP datagram is the unit of end-to-end transmission in
            the IP protocol.  An IP datagram consists of an IP header
            followed by transport layer data, i.e., of an IP header
            followed by a message.
            In the description of the internet layer (Section 3), the
            unqualified term "datagram" should be understood to refer
            to an IP datagram.
            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.

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

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RFC1122 INTRODUCTION October 1989

 1.4  Acknowledgments
    This document incorporates contributions and comments from a large
    group of Internet protocol experts, including representatives of
    university and research labs, vendors, and government agencies.
    It was assembled primarily by the Host Requirements Working Group
    of the Internet Engineering Task Force (IETF).
    The Editor would especially like to acknowledge the tireless
    dedication of the following people, who attended many long
    meetings and generated 3 million bytes of electronic mail over the
    past 18 months in pursuit of this document: Philip Almquist, Dave
    Borman (Cray Research), Noel Chiappa, Dave Crocker (DEC), Steve
    Deering (Stanford), Mike Karels (Berkeley), Phil Karn (Bellcore),
    John Lekashman (NASA), Charles Lynn (BBN), Keith McCloghrie (TWG),
    Paul Mockapetris (ISI), Thomas Narten (Purdue), Craig Partridge
    (BBN), Drew Perkins (CMU), and James Van Bokkelen (FTP Software).
    In addition, the following people made major contributions to the
    effort: Bill Barns (Mitre), Steve Bellovin (AT&T), Mike Brescia
    (BBN), Ed Cain (DCA), Annette DeSchon (ISI), Martin Gross (DCA),
    Phill Gross (NRI), Charles Hedrick (Rutgers), Van Jacobson (LBL),
    John Klensin (MIT), Mark Lottor (SRI), Milo Medin (NASA), Bill
    Melohn (Sun Microsystems), Greg Minshall (Kinetics), Jeff Mogul
    (DEC), John Mullen (CMC), Jon Postel (ISI), John Romkey (Epilogue
    Technology), and Mike StJohns (DCA).  The following also made
    significant contributions to particular areas: Eric Allman
    (Berkeley), Rob Austein (MIT), Art Berggreen (ACC), Keith Bostic
    (Berkeley), Vint Cerf (NRI), Wayne Hathaway (NASA), Matt Korn
    (IBM), Erik Naggum (Naggum Software, Norway), Robert Ullmann
    (Prime Computer), David Waitzman (BBN), Frank Wancho (USA), Arun
    Welch (Ohio State), Bill Westfield (Cisco), and Rayan Zachariassen
    We are grateful to all, including any contributors who may have
    been inadvertently omitted from this list.

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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.

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            Specifically, the ARP exchange is completed in the usual
            manner using the normal IP protocol type, but a host that
            wants to speak trailers will send an additional "trailer
            ARP reply" packet, i.e., an ARP reply that specifies the
            trailer encapsulation protocol type but otherwise has the
            format of a normal ARP reply.  If a host configured to use
            trailers receives a trailer ARP reply message from a
            remote machine, it can add that machine to the list of
            machines that understand trailers, e.g., by marking the
            corresponding entry in the ARP cache.
            Hosts wishing to receive trailer encapsulations send
            trailer ARP replies whenever they complete exchanges of
            normal ARP messages for IP.  Thus, a host that received an
            ARP request for its IP protocol address would send a
            trailer ARP reply in addition to the normal IP ARP reply;
            a host that sent the IP ARP request would send a trailer
            ARP reply when it received the corresponding IP ARP reply.
            In this way, either the requesting or responding host in
            an IP ARP exchange may request that it receive trailer
            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

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               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

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                    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-

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       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.

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         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

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RFC1122 INTERNET LAYER October 1989

    For normal datagrams, the processing is straightforward.  For
    incoming datagrams, the IP layer:
    (1)  verifies that the datagram is correctly formatted;
    (2)  verifies that it is destined to the local host;
    (3)  processes options;
    (4)  reassembles the datagram if necessary; and
    (5)  passes the encapsulated message to the appropriate
         transport-layer protocol module.
    For outgoing datagrams, the IP layer:
    (1)  sets any fields not set by the transport layer;
    (2)  selects the correct first hop on the connected network (a
         process called "routing");
    (3)  fragments the datagram if necessary and if intentional
         fragmentation is implemented (see Section 3.3.3); and
    (4)  passes the packet(s) to the appropriate link-layer driver.
    A host is said to be multihomed if it has multiple IP addresses.
    Multihoming introduces considerable confusion and complexity into
    the protocol suite, and it is an area in which the Internet
    architecture falls seriously short of solving all problems.  There
    are two distinct problem areas in multihoming:
    (1)  Local multihoming --  the host itself is multihomed; or
    (2)  Remote multihoming -- the local host needs to communicate
         with a remote multihomed host.
    At present, remote multihoming MUST be handled at the application
    layer, as discussed in the companion RFC [INTRO:1].  A host MAY
    support local multihoming, which is discussed in this document,
    and in particular in Section 3.3.4.
    Any host that forwards datagrams generated by another host is
    acting as a gateway and MUST also meet the specifications laid out
    in the gateway requirements RFC [INTRO:2].  An Internet host that
    includes embedded gateway code MUST have a configuration switch to
    disable the gateway function, and this switch MUST default to the

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    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
          discarded.  Checksum: RFC-791 Section 3.1
          A host MUST verify the IP header checksum on every received
          datagram and silently discard every datagram that has a bad
          checksum.  Addressing: RFC-791 Section 3.2
          There are now five classes of IP addresses: Class A through
          Class E.  Class D addresses are used for IP multicasting
          [IP:4], while Class E addresses are reserved for
          experimental use.
          A multicast (Class D) address is a 28-bit logical address
          that stands for a group of hosts, and may be either
          permanent or transient.  Permanent multicast addresses are
          allocated by the Internet Assigned Number Authority
          [INTRO:6], while transient addresses may be allocated

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          dynamically to transient groups.  Group membership is
          determined dynamically using IGMP [IP:4].
          We now summarize the important special cases for Class A, B,
          and C IP addresses, using the following notation for an IP
              { <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.

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          (f)  { <Network-number>, -1, -1 }
               Directed broadcast to all subnets of the specified
               subnetted network.  It MUST NOT be used as a source
          (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
               recommended.  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.

Internet Engineering Task Force [Page 44]

RFC1122 INTERNET LAYER October 1989  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
               cache.  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.  Initialization
          The following information MUST be configurable:
          (1)  IP address(es).
          (2)  Address mask(s).
          (3)  A list of default gateways, with a preference level.
          A manual method of entering this configuration data MUST be
          provided.  In addition, a variety of methods can be used to
          determine this information dynamically; see the section on
          "Host Initialization" in [INTRO:1].
               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  Introduction
          A multihomed host has multiple IP addresses, which we may
          think of as "logical interfaces".  These logical interfaces
          may be associated with one or more physical interfaces, and
          these physical interfaces may be connected to the same or
          different networks.
          Here are some important cases of multihoming:
          (a)  Multiple Logical Networks
               The Internet architects envisioned that each physical
               network would have a single unique IP network (or
               subnet) number.  However, LAN administrators have
               sometimes found it useful to violate this assumption,
               operating a LAN with multiple logical networks per
               physical connected network.
               If a host connected to such a physical network is
               configured to handle traffic for each of N different
               logical networks, then the host will have N logical
               interfaces.  These could share a single physical
               interface, or might use N physical interfaces to the
               same network.
          (b)  Multiple Logical Hosts
               When a host has multiple IP addresses that all have the
               same <Network-number> part (and the same <Subnet-
               number> part, if any), the logical interfaces are known
               as "logical hosts".  These logical interfaces might
               share a single physical interface or might use separate

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               physical interfaces to the same physical network.
          (c)  Simple Multihoming
               In this case, each logical interface is mapped into a
               separate physical interface and each physical interface
               is connected to a different physical network.  The term
               "multihoming" was originally applied only to this case,
               but it is now applied more generally.
               A host with embedded gateway functionality will
               typically fall into the simple multihoming case.  Note,
               however, that a host may be simply multihomed without
               containing an embedded gateway, i.e., without
               forwarding datagrams from one connected network to
               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

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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
                                               |        | | | | | |


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


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
                                               |        | | | | | |


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


       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  Ports
          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.

Internet Engineering Task Force [Page 77]

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
               port(s).  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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               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| | |

  1. Default is to checksum | |x| | | | |

- Receiver Option to require checksum | | | |x| | |

                                               |        | | | | | |

UDP Multihoming | | | | | | | - Pass spec-dest addr to application | |x| | | | |

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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

       The Transmission Control Protocol TCP [TCP:1] is the primary
       virtual-circuit transport protocol for the Internet suite.  TCP
       provides reliable, in-sequence delivery of a full-duplex stream
       of octets (8-bit bytes).  TCP is used by those applications
       needing reliable, connection-oriented transport service, e.g.,
       mail (SMTP), file transfer (FTP), and virtual terminal service
       (Telnet); requirements for these application-layer protocols
       are described in [INTRO:1].
    4.2.2  PROTOCOL WALK-THROUGH  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).

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

Internet Engineering Task Force [Page 91]

RFC1122 TRANSPORT LAYER – TCP October 1989  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
               future.  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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RFC1122 TRANSPORT LAYER – TCP October 1989  TCP Multihoming
          If an application on a multihomed host does not specify the
          local IP address when actively opening a TCP connection,
          then the TCP MUST ask the IP layer to select a local IP
          address before sending the (first) SYN.  See the function
          GET_SRCADDR() in Section 3.4.
          At all other times, a previous segment has either been sent
          or received on this connection, and TCP MUST use the same
          local address is used that was used in those previous
          segments.  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
          address.  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.  Efficiency
               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.
    4.2.4  TCP/APPLICATION LAYER INTERFACE  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.  Type-of-Service
          The application layer MUST be able to specify the Type-of-
          Service (TOS) for segments that are sent on a connection.
          It not required, but the application SHOULD be able to
          change the TOS during the connection lifetime.  TCP SHOULD
          pass the current TOS value without change to the IP layer,
          when it sends segments on the connection.
          The TOS will be specified independently in each direction on
          the connection, so that the receiver application will
          specify the TOS used for ACK segments.
          TCP MAY pass the most recently received TOS up to the
               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.

Internet Engineering Task Force [Page 107]

RFC1122 TRANSPORT LAYER – TCP October 1989  Multihoming
          The user interface outlined in sections 2.7 and 3.8 of RFC-
          793 needs to be extended for multihoming.  The OPEN call
          MUST have an optional parameter:
              OPEN( ... [local IP address,] ... )
          to allow the specification of the local IP address.
               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| | |

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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| | | | |

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                                               |        | | | | | |

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| | |

  1. Application can request | |x| | | | |
  2. Default is "off" | |x| | | | |
  3. Only send if idle for interval | |x| | | | |
  4. Interval configurable | |x| | | | |

Internet Engineering Task Force [Page 110]

RFC1122 TRANSPORT LAYER – TCP October 1989

  1. Default at least 2 hrs. | |x| | | | |
  2. 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]

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[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]

/data/webs/external/dokuwiki/data/pages/rfc/rfc1122.txt · Last modified: 1989/10/02 22:48 (external edit)