Simple Internet Protocol (SIP) Specification
RFC 8507

Document Type RFC - Historic (December 2018; No errata)
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Independent Submission                                        S. Deering
Request for Comments: 8507                                       Retired
Category: Historic                                        R. Hinden, Ed.
ISSN: 2070-1721                                     Check Point Software
                                                           December 2018

              Simple Internet Protocol (SIP) Specification


   This document is published for the historical record.  The Simple
   Internet Protocol was the basis for one of the candidates for the
   IETF's Next Generation (IPng) work that became IPv6.

   The publication date of the original Internet-Draft was November 10,
   1992.  It is presented here substantially unchanged and is neither a
   complete document nor intended to be implementable.

   The paragraph that follows is the Abstract from the original draft.

   This document specifies a new version of IP called SIP, the Simple
   Internet Protocol.  It also describes the changes needed to ICMP,
   IGMP, and transport protocols such as TCP and UDP, in order to work
   with SIP.  A companion document [SIP-ADDR] describes the addressing
   and routing aspects of SIP, including issues of auto-configuration,
   host and subnet mobility, and multicast.

Status of This Memo

   This document is not an Internet Standards Track specification; it is
   published for the historical record.

   This document defines a Historic Document for the Internet community.
   This is a contribution to the RFC Series, independently of any other
   RFC stream.  The RFC Editor has chosen to publish this document at
   its discretion and makes no statement about its value for
   implementation or deployment.  Documents approved for publication by
   the RFC Editor are not candidates for any level of Internet Standard;
   see Section 2 of RFC 7841.

   Information about the current status of this document, any errata,
   and how to provide feedback on it may be obtained at

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Copyright Notice

   Copyright (c) 2018 IETF Trust and the persons identified as the
   document authors.  All rights reserved.

   This document is subject to BCP 78 and the IETF Trust's Legal
   Provisions Relating to IETF Documents
   ( in effect on the date of
   publication of this document.  Please review these documents
   carefully, as they describe your rights and restrictions with respect
   to this document.

Table of Contents

   1. Preface .........................................................3
   2. Introduction ....................................................3
   3. Terminology .....................................................4
   4. SIP Header Format ...............................................5
   5. Addresses .......................................................6
      5.1. Text Representation of Addresses ...........................6
      5.2. Unicast Addresses ..........................................6
      5.3. Multicast Addresses ........................................8
      5.4. Special Addresses ..........................................9
   6. Packet Size Issues .............................................12
   7. Source Routing Header ..........................................13
   8. Fragmentation Header ...........................................14
   9. Changes to Other Protocols .....................................16
      9.1. Changes to ICMP ...........................................16
      9.2. Changes to IGMP ...........................................20
      9.3. Changes to Transport Protocols ............................21
      9.4. Changes to Link-Layer Protocols ...........................22
   10. Security Considerations .......................................22
   11. Acknowledgments ...............................................23
   12. Informative References ........................................23
   Appendix A. SIP Design Rationale ..................................25
   Appendix B. Future Directions .....................................25
   Authors' Addresses ................................................26

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1.  Preface

   This document is published for the historical record.

   Simple IP (SIP) was the basis for one of the candidates for the
   IETF's Next Generation (IPng) work; see "The Recommendation for the
   IP Next Generation Protocol" [RFC1752].  The original 1992
   Internet-Draft describing SIP is published here as part of the record
   of that work.

   SIP evolved into SIP Plus [RFC1710], which was assessed as a
   candidate for IPng [RFC1752] and led eventually to the development of
   IPv6, first published as [RFC1883].  The current specification for
   IPv6 is [RFC8200].

   The original Internet-Draft describing the Simple Internet Protocol
   was written by Steve Deering, and the Internet-Draft was posted on
   November 10, 1992.  The contents of this document are unchanged from
   that Internet-Draft, except for clarifications in the Abstract, the
   addition of this section, modifications to the authors' information,
   the updating of references, removal of the IANA considerations, and
   minor formatting changes.

   It should be noted that the original draft was not complete and that
   no attempt has been made to complete it.  This document is not
   intended to be implementable.

2.  Introduction

   SIP is a new version of IP.  Its salient differences from IP
   version 4 [RFC791], subsequently referred to as "IPv4", are:

       o  expansion of addresses to 64 bits,

       o  simplification of the IP header by eliminating some
          inessential fields, and

       o  relaxation of length restrictions on optional data, such as
          source-routing information.

   SIP retains the IP model of globally-unique addresses,
   hierarchically-structured for efficient routing.  Increasing the
   address size from 32 to 64 bits allows more levels of hierarchy to be
   encoded in the addresses, enough to enable efficient routing in an
   internet with tens of thousands of addressable devices in every
   office, every residence, and every vehicle in the world.  Keeping the

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   addresses fixed-length and relatively compact facilitates
   high-performance router and host implementation, and keeps the
   bandwidth overhead of the SIP headers almost as low as IPv4.

   The elimination of inessential fields also contributes to
   high-performance implementation, and to the likelihood of correct
   implementation.  A change in the way that optional data, such as
   source-routing information, is encoded allows for more efficient
   forwarding and less stringent limits on the length of such data.

   Despite these changes, SIP remains very similar to IPv4.  This
   similarity makes it easy to understand SIP (for those who already
   understand IPv4), makes it possible to reuse much of the code and
   data structures from IPv4 in an implementation of SIP (including
   almost all of ICMP and IGMP), and makes it straightforward to
   translate between SIP packets and IPv4 packets for transition
   purposes [IPAE].

   The subsequent sections of this document specify SIP and its
   associated protocols without much explanation of why the design
   choices were made the way they were.  Appendix A presents the
   rationale for those aspects of SIP that differ from IPv4.

3.  Terminology

    system      - a device that implements SIP.

    router      - a system that forwards SIP packets.

    host        - any system that is not a router.

    link        - a communication facility or medium over which systems
                  can communicate at the link layer, i.e., the layer
                  immediately below SIP.

    interface   - a system's attachment point to a link.

    address     - a SIP-layer identifier for an interface or a group of

    subnet      - in the SIP unicast addressing hierarchy, a
                  lowest-level (finest-grain) cluster of addresses,
                  sharing a common address prefix (i.e., high-order
                  address bits).  Typically, all interfaces attached to
                  the same link have addresses in the same subnet;
                  however, in some cases, a single link may support more
                  than one subnet, or a single subnet may span more than
                  one link.

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    link MTU    - the maximum transmission unit, i.e., maximum packet
                  size in octets, that can be conveyed in one piece over
                  a link (where a packet is a SIP header plus payload).

    path MTU    - the minimum link MTU of all the links in a path
                  between a source system and a destination system.

    layer       - any protocol layer above SIP that is responsible for
                  packetizing data to fit within outgoing SIP packets.
                  Typically, transport-layer protocols, such as TCP, are
                  packetization protocols, but there may also be
                  higher-layer packetization protocols, such as
                  protocols implemented on top of UDP.

4.  SIP Header Format

   |Version|                        Reserved                       |
   |         Payload Length        |  Payload Type |   Hop Limit   |
   |                                                               |
   +                         Source Address                        +
   |                                                               |
   |                                                               |
   +                      Destination Address                      +
   |                                                               |

   Version              4-bit IP version number = decimal 6.
                        <to be confirmed>

   Reserved             28-bit reserved field.  Initialized to zero
                        for transmission; ignored on reception.

   Payload Length       16-bit unsigned integer.  Length of payload,
                        i.e., the rest of the packet following the
                        SIP header, in octets.

   Payload Type         8-bit selector.  Identifies the type of
                        payload, e.g., TCP.

   Hop Limit            8-bit unsigned integer.  Decremented by 1
                        by each system that forwards the packet.
                        The packet is discarded if Hop Limit is
                        decremented to zero.

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   Source Address       64 bits.  See "Addresses" section, following.

   Destination Address  64 bits.  See "Addresses" section, following.

5.  Addresses

5.1.  Text Representation of Addresses

   SIP addresses are 64 bits (8 octets) long.  The text representation
   of a SIP addresses is 16 hexadecimal digits, with a colon between the
   4th and 5th digits, and optional colons between any subsequent pair
   of digits.  Leading zeros must not be dropped.  Examples:




   Programs that read the text representation of SIP addresses must be
   insensitive to the presence or absence of optional colons.  Programs
   that write the text representation of a SIP address should use the
   first format above (i.e., colons after the 4th, 8th, and 12th
   digits), in the absence of any knowledge of the structure or
   preferred format of the address, such as knowledge of the format in
   which it was originally read.

   The presence of at least one colon in the text representation allows
   SIP addresses to be easily distinguished from both domain names and
   the text representation of IPv4 addresses.

5.2.  Unicast Addresses

   A SIP unicast address is a globally-unique identifier for a single
   interface, i.e., no two interfaces in a SIP internet may have the
   same unicast address.  A single interface may, however, have more
   than one unicast address.

   A system considers its own unicast address(es) to have the following
   structure, where different addresses may have different values for n:

    |                         n bits                     |  64-n bits |
    |                     subnet prefix                  |interface ID|

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   To know the length of the subnet prefix, the system is required to
   associate with each of its addresses a 'subnet mask' of the following

    |                         n bits                     |  64-n bits |

   A system may have a subnet mask of all-ones, which means that the
   system belongs to a subnet containing exactly one system -- itself.

   A system acquires its subnet mask(s) at the same time, and by the
   same mechanism, as it acquires its address(es), for example, by
   manual configuration or by a dynamic configuration protocol such as
   BOOTP [RFC951].

   Hosts are ignorant of any further structure in a unicast address.

   Routers may acquire, through manual configuration or the operation of
   routing protocols, additional masks that identify higher-level
   clusters in a hierarchical addressing plan.  For example, the routers
   within a single site would typically have a 'site mask', such as the

    |                  m bits                 |       64-m bits       |

   by which they could deduce the following structure in the site's

    |                  m bits                 |  p bits  | 64-m-p bits|
    |                site prefix              |subnet  ID|interface ID|

   All knowledge by SIP systems of the structure of unicast addresses is
   based on possession of such masks -- there is no "wired-in" knowledge
   of unicast address formats.

   The SIP Addressing and Routing document [SIP-ADDR] proposes two
   hierarchical addressing plans, one based on a hierarchy of SIP
   service providers, and one based on a geographic hierarchy.

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5.3.  Multicast Addresses

   A SIP multicast address is an identifier for a group of interfaces.
   An interface may belong to any number of multicast groups.  Multicast
   addresses have the following format:

    |1|   7   |  4 |  4 |                  48 bits                    |
    |C|1111111|flgs|scop|                  group ID                   |


     C = IPv4 compatibility flag; see [IPAE].

     1111111 in the rest of the first octet identifies the address as
             being a multicast address.

     flgs is a set of 4 flags:   |0|0|0|T|

       the high-order 3 flags are reserved, and must be initialized
       to 0.

       T = 0 indicates a permanently-assigned ("well-known") multicast
             address, assigned by the global internet numbering

       T = 1 indicates a non-permanently-assigned ("transient")
             multicast address.

     scop is a 4-bit multicast scope value:

       0 reserved
       1 intra-system scope
       2 intra-link scope
       3 (unassigned)
       4 (unassigned)
       5 intra-site scope
       6 (unassigned)
       7 (unassigned)
       8 intra-metro scope
       9 (unassigned)
       A (unassigned)
       B intra-country scope
       C (unassigned)

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       D (unassigned)
       E global scope
       F reserved

     group ID identifies the multicast group, either permanent or
     transient, within the given scope.

   The "meaning" of a permanently-assigned multicast address is
   independent of the scope value.  For example, if the "NTP servers
   group" is assigned a permanent multicast address with a group ID of
   43 (hex), then:

     7F01:000000000043 means all NTP servers on the same system as the

     7F02:000000000043 means all NTP servers on the same link as the

     7F05:000000000043 means all NTP servers at the same site as the

     7F0E:000000000043 means all NTP servers in the internet.

   Non-permanently-assigned multicast addresses are meaningful only
   within a given scope.  For example, a group identified by the
   non-permanent, intra-site multicast address 7F15:000000000043 at one
   site bears no relationship to a group using the same address at a
   different site, nor to a non-permanent group using the same group ID
   with different scope, nor to a permanent group with the same
   group ID.

5.4.  Special Addresses

   There are a number of "special purpose" SIP addresses:

     The Unspecified Address: 0000:0000:0000:0000

       This address shall never be assigned to any system.  It may be
       used wherever an address appears, to indicate the absence of an
       address.  One example of its use is in the Source Address field
       of a SIP packet sent by an initializing host, before it has
       learned its own address.

     The Loopback Address: 0000:0000:0000:0001

       This address may be used by a system to send a SIP packet to

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     Anyone Addresses: <prefix><zero>

       Addresses of this form may be used to send to the "nearest"
       system (according the routing protocols' measure of distance)
       that "knows" it has a unicast address prefix of <prefix>.

       Since hosts know only their subnet prefix(es), and no
       higher-level prefixes, a host with the following address:

       |               subnet prefix = A              |interface ID = B|

       shall recognize only the following Anyone address as identifying

       |               subnet prefix = A              |0000000000000000|

       An intra-site router that knows that one of its addresses has the

       |         site prefix = X       |subnet  ID = Y|interface ID = Z|

       shall accept packets sent to either of the following two Anyone
       addresses as if they had been sent to the router's own address:

       |         site prefix = X       |0000000000000000000000000000000|

       |         site prefix = X       |subnet  ID = Y|0000000000000000|

       Anyone Addresses work as follows:

         If any system belonging to subnet A sends a packet to
         subnet A's Anyone address, the packet shall be looped-back
         within the sending system itself, since it is the nearest
         system to itself with the subnet A prefix.  If a system outside
         of subnet A sends a packet to subnet A's Anyone address, the
         packet shall be accepted by the first router on subnet A that
         the packet reaches.

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         Similarly, a packet sent to site X's Anyone address from
         outside of site X shall be accepted by the first encountered
         router belonging to site X, i.e., one of site X's boundary
         routers.  If a higher-level prefix P identifies, say, a
         particular service provider, then a packet sent to <P> <zero>
         from outside of provider P's facilities shall be delivered to
         the nearest entry router into P's facilities.

       Anyone addresses are most commonly used in conjunction with the
       SIP source routing header, to cause a packet to be routed via one
       or more specified "transit domains", without the need to identify
       individual routers in those domains.

       The value zero is reserved at each level of every unicast address
       hierarchy, to serve as an Anyone address for that level.

     The Reserved Multicast Address:   7F0s:0000:0000:0000

       This multicast address (with any scope value, s) is reserved, and
       shall never be assigned to any multicast group.

     The All Systems Addresses:   7F01:0000:0000:0001

       These multicast addresses identify the group of all SIP systems,
       within scope 1 (intra-system) or 2 (intra-link).

     The All Hosts Addresses:   7F01:0000:0000:0002

       These multicast addresses identify the group of all SIP hosts,
       within scope 1 (intra-system) or 2 (intra-link).

     The All Routers Addresses:   7F01:0000:0000:0003

       These multicast addresses identify the group of all SIP routers,
       within scope 1 (intra-system) or 2 (intra-link).

   A host is required to recognize the following addresses as
   identifying itself: its own unicast addresses, the Anyone addresses
   with the same subnet prefixes as its unicast addresses, the Loopback
   address, the All Systems and All Hosts addresses, and any other
   multicast addresses to which the host belongs.

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   A router is required to recognize the following addresses as
   identifying itself: its own unicast addresses, the Anyone addresses
   with the same subnet or higher-level prefixes as its unicast
   addresses, the Loopback address, the All Systems and All Routers
   addresses, and any other multicast addresses to which the host

6.  Packet Size Issues

   SIP requires that every link in the internet have an MTU of 576
   octets or greater.  On any link that cannot convey a 576-octet packet
   in one piece, link-specific fragmentation and reassembly must be
   provided at a layer below SIP.

       (Note: this minimum link MTU is NOT the same as the one in IPv4.
       In IPv4, the minimum link MTU is 68 octets [ [RFC791], page 25 ];
       576 octets is the minimum reassembly buffer size required in an
       IPv4 system, which has nothing to do with link MTUs.)

   From each link to which a system is directly attached, the system
   must be able to accept packets as large as that link's MTU.  Links
   that have a configurable MTU, such as PPP links [RFC1661], should be
   configured with an MTU of 600 octets or greater.

   SIP systems are expected to implement Path MTU Discovery [RFC1191],
   in order to discover and take advantage of paths with MTU greater
   than 576 octets.  However, a minimal SIP implementation (e.g., in a
   boot ROM) may simply restrict itself to sending packets no larger
   than 576 octets, and omit implementation of Path MTU Discovery.

   Path MTU Discovery requires support both in the SIP layer and in the
   packetization layers.  A system that supports Path MTU Discovery at
   the SIP layer may serve packetization layers that are unable to adapt
   to changes of the path MTU.  Such packetization layers must limit
   themselves to sending packets no longer than 576 octets, even when
   sending to destinations that belong to the same subnet.

       (Note: Unlike IPv4, it is unnecessary in SIP to set a "Don't
       Fragment" flag in the packet header in order to perform Path MTU
       Discovery; that is an implicit attribute of every SIP packet.
       Also, those parts of the RFC-1191 procedures that involve use of
       a table of MTU "plateaus" do not apply to SIP, because the SIP
       version of the "Datagram Too Big" message always identifies the
       exact MTU to be used.)

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7.  Source Routing Header

   A Payload Type of <TBD> in the immediately preceding header indicates
   the presence of this Source Routing header:

      |                            Reserved                           |
      |   Num Addrs   |   Next Addr   |  Payload Type |    Reserved   |
      |                                                               |
      +                           Address[0]                          +
      |                                                               |
      |                                                               |
      +                           Address[1]                          +
      |                                                               |
      .                               .                               .
      .                               .                               .
      .                               .                               .
      |                                                               |
      +                     Address[Num Addrs - 1]                    +
      |                                                               |

      Reserved             Initialized to zero for transmission; ignored
                           on reception.

      Num Addrs            Number of addresses in the Source Routing

      Next Addr            Index of next address to be processed;
                           initialized to 0 by the originating system.

      Payload Type         Identifies the type of payload following the
                           Source Routing header.

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   A Source Routing header is not examined or processed until it reaches
   the system identified in the Destination Address field of the SIP
   header.  In that system, dispatching on the Payload Type of the SIP
   (or subsequent) header causes the Source Routing module to be
   invoked, which performs the following algorithm:

       o  If Next Addr < Num Addrs, swap the SIP Destination Address and
          Address[Next Addr], increment Next Addr by one, and re-submit
          the packet to the SIP module for forwarding to the next

       o  If Next Addr = Num Addrs, dispatch to the local protocol
          module identified by the Payload Type field in the Source
          Routing header.

       o  If Next Addr > Num Addrs, send an ICMP Parameter Problem
          message to the Source Address, pointing to the Num Addrs

8.  Fragmentation Header

   A Payload Type of <TBD> in the immediately preceding header indicates
   the presence of this Fragmentation header:

      |                         Identification                        |
      |0 0 M|      Fragment Offset    |  Payload Type |    Reserved   |

      Identification       A value that changes on each packet sent with
                           the same Source Address, Destination Address,
                           and preceding Payload Type.

      M flag               1 = more fragments; 0 = last fragment.

      Fragment Offset      The offset, in 8-octet chunks, of the
                           following payload, relative to the original,
                           unfragmented payload.

      Payload Type         Identifies the type of payload following the
                           Fragmentation header.

      Reserved             Initialized to zero for transmission; ignored
                           on reception.

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   The Fragmentation header is NOT intended to support general,
   SIP-layer fragmentation.  In particular, SIP routers shall not
   fragment a SIP packet that is too big for the MTU of its next hop,
   except in the special cases described below; in the normal case, such
   a packet results in an ICMP Packet Too Big message being sent back to
   its source, for use by the source system's Path MTU Discovery

   The special cases for which the Fragmentation header is intended are
   the following:

       o  A SIP packet that is "tunneled", either by encapsulation
          within another SIP packet or by insertion of a Source Routing
          header en-route, may, after the addition of the extra header
          fields, exceed the MTU of the tunnel's path; if the original
          packet is 576 octets or less in length, the tunnel entry
          system cannot respond to the source with a Packet Too Big
          message, and therefore must insert a Fragmentation header and
          fragment the packet to fit within the tunnel's MTU.

       o  An IPv4 fragment that is translated into a SIP packet, or an
          unfragmented IPv4 packet that is translated into too long a
          SIP packet to fit in the remaining path MTU, must include the
          SIP Fragmentation header, so that it may be properly
          reassembled at the destination SIP system.

   Every SIP system must support SIP fragmentation and reassembly, since
   any system may be configured to serve as a tunnel entry or exit
   point, and any SIP system may be destination of IPv4 fragments.  All
   SIP systems must be capable of reassembling, from fragments, a SIP
   packet of up to 1024 octets in length, including the SIP header; a
   system may be capable of assembling packets longer than 1024 octets.

   Routers do not examine or process Fragmentation headers of packets
   that they forward; only at the destination system is the
   Fragmentation header acted upon (i.e., reassembly performed), as a
   result of dispatching on the Payload Type of the preceding header.

   Fragmentation and reassembly employ the same algorithm as IPv4, with
   the following exceptions:

       o  All headers up to and including the Fragmentation header are
          repeated in each fragment; no headers or data following the
          Fragmentation header are repeated in each fragment.

       o  the Identification field is increased to 32 bits, to decrease
          the risk of wraparound of that field within the maximum packet
          lifetime over very high-throughput paths.

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   The similarity of the algorithm and the field layout to that of IPv4
   enables existing IPv4 fragmentation and reassembly code and data
   structures to be re-used with little modification.

9.  Changes to Other Protocols

   Upgrading IPv4 to SIP entails changes to the associated control
   protocols, ICMP and IGMP, as well as to the transport layer, above,
   and possibly to the link-layer, below.  This section identifies those

9.1.  Changes to ICMP

   SIP uses a subset of ICMP [[RFC792], [RFC950], [RFC1122], [RFC1191],
   [RFC1256]], with a few minor changes and some additions.  The
   presence of an ICMP header is indicated by a Payload Type of 1.

   One change to all ICMP messages is that, when used with SIP, the ICMP
   checksum includes a pseudo-header, like TCP and UDP, consisting of
   the SIP Source Address, Destination Address, Payload Length, and
   Payload Type (see section 8.3).

   There are a set of ICMP messages called "error messages", each of
   which, for IPv4, carries the IPv4 header plus 64 bits or more of data
   from the packet that invoked the error message.  When used with SIP,
   ICMP error messages carry the first 256 octets of the invoking SIP
   packet, or the entire invoking packet if it is shorter than
   256 octets.

   For most of the ICMP message types, the packets retain the same
   format and semantics as with IPv4; however, some of the fields are
   given new names to match SIP terminology.

   The changes to specific message types are as follows:

     Destination Unreachable

       The following Codes have different names when used with SIP:

         1 - destination address unreachable (IPv4 "host unreachable")
         7 - destination address unknown (IPv4 "dest. host unknown")
         2 - payload type unknown (IPv4 "protocol unreachable")
         4 - packet too big (IPv4 "fragmentation needed and DF set")

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       The following Codes retain the same names when used with SIP:

         3 - port unreachable
         5 - source route failed
         8 - source host isolated
        13 - communication administratively prohibited

       The following Codes are not used with SIP:

         0 - net unreachable
         6 - destination network unknown
         9 - comm. with dest. network administratively prohibited
        10 - comm. with dest. host administratively prohibited
        11 - network unreachable for type of service
        12 - host unreachable for type of service

       For "packet too big" messages (Code 4), the minimum legal value
       in the Next-Hop MTU field [RFC1191] is 576.

     Time Exceeded

       The name of Code 0 is changed to "hop limit exceeded in transit".

     Parameter Problem

       The Pointer field is extended to 16 bits and moved to the
       low-order end of the second 32-bit word, as follows:

       |    Type     |      Code     |            Checksum         |
       |          Reserved           |            Pointer          |
       |                                                           |
       |           first 256 octets of the invoking packet         |
       |                                                           |

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       Only Code 1 is used for SIP, meaning "redirect packets for the
       destination address".

       The Redirect header is modified for SIP, to accommodate the
       64-bit address of the "preferred router" and to retain 64-bit
       alignment, as follows:

       |      Type     |      Code     |            Checksum         |
       |                            Reserved                         |
       |                                                             |
       +                        Preferred Router                     +
       |                                                             |
       |                                                             |
       |             first 256 octets of the invoking packet         |
       |                                                             |

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     Router Advertisement

       The format of the Router Advertisement message is changed to:

       |     Type      |     Code      |           Checksum          |
       |   Num Addrs   |Addr Entry Size|           Lifetime          |
       |                                                             |
       +                       Router Address[0]                     +
       |                                                             |
       |                      Preference Level[0]                    |
       |                          Reserved[0]                        |
       |                                                             |
       +                       Router Address[1]                     +
       |                                                             |
       |                      Preference Level[1]                    |
       |                          Reserved[1]                        |
       |                               .                             |
       |                               .                             |
       |                               .                             |

       The value in the Addr Entry Size field is 4, and all of the
       Reserved fields are initialized to zero by senders and ignored by

     Router Solicitation

       No changes.

     Echo and Echo Reply

       No changes.

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     The following ICMP message types are not used with SIP:

       Source Quench
       Timestamp Reply
       Information Request
       Information Reply
       Address Mask Request
       Address Mask Reply

9.2.  Changes to IGMP

   SIP uses the Internet Group Management Protocol, IGMP [RFC1112].  The
   presence of an IGMP header is indicated by a Payload Type of 2.

   When used with SIP, the IGMP checksum includes a pseudo-header, like
   TCP and UDP, consisting of the SIP Source Address, Destination
   Address, Payload Length, and Payload Type (see section 8.3).

   The format of an IGMP Host Membership Query message becomes:

       |Version| Type  |    Reserved   |           Checksum            |
       |                            Reserved                           |

   The format of an IGMP Host Membership Report message becomes:

       |Version| Type  |    Reserved   |           Checksum            |
       |                            Reserved                           |
       |                                                               |
       +                       Multicast Address                       +
       |                                                               |

   For both message types, the Version number remains 1, and the
   Reserved fields are set to zero by senders and ignored by receivers.

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9.3.  Changes to Transport Protocols

   The service interface to SIP has some differences from IPv4's service
   interface.  Existing transport protocols that use IPv4 must be
   changed to operate over SIP's service interface.  The differences
   from IPv4 are:

       o  Any addresses passed across the interface are 64 bits long,
          rather than 32 bits.

       o  The following IPv4 variables are not passed across the
          interface: Precedence, Type-of-Service, Identifier,
          Don't Fragment Flag

       o  SIP options have a different format than IPv4 options.  (For
          SIP, "options" are all headers between, and not including, the
          SIP header and the transport header.  The only IPv4 option
          currently specified for SIP is Loose Source Routing.

       o  ICMP error messages for SIP that are passed up to the
          transport layer carry the first 256 octets of the invoking SIP

   Transport protocols that use IPv4 addresses for their own purposes,
   such as identifying connection state or inclusion in a pseudo-header
   checksum, must be changed to use 64-bit SIP addresses for those
   purposes instead.

   For SIP, the pseudo-header checksums of TCP, UDP, ICMP, and IGMP
   include the SIP Source Address, Destination Address, Payload Length,
   and Payload Type, with the following caveats:

       o  If the packet contains a Source Routing header, the
          destination address used in the pseudo-header checksum is that
          of the final destination.

       o  The Payload Length used in the pseudo-header checksum is the
          length of the transport-layer packet, including the transport

       o  The Payload Type used in the pseudo-header checksum is the
          Payload Type from the header immediately preceding the
          transport header.

       o  When added to the pseudo-header checksum, the Payload Type is
          treated as the left octet of a 16-bit word, with zeros in the
          the right octet, when viewed in IP standard octet order.

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       o  If either of the two addresses used in the pseudo-header
          checksum has its high-order bit set to 1, only the low-order
          32-bits of that address shall be used in the sum.  The
          high-order bit is used to indicate that the addressed system
          is an IPv4 system, and that the low-order 32-bits of the
          address contain that system's IPv4 address [IPAE].

   The semantics of SIP service differ from IPv4 service in three ways
   that may affect some transport protocols:

     (1)  SIP does not enforce maximum packet lifetime.  Any transport
          protocol that relies on IPv4 to limit packet lifetime must
          take this change into account, for example, by providing its
          own mechanisms for detecting and discarding obsolete packets.

     (2)  SIP does not checksum its own header fields.  Any transport
          protocol that relies on IPv4 to assure the integrity of the
          source and destinations addresses, packet length, and
          transport protocol identifier must take this change into
          account.  In particular, when used with SIP, the UDP checksum
          is mandatory, and ICMP and IGMP are changed to use a
          pseudo-header checksum.

     (3)  SIP does not (except in special cases) fragment packets that
          exceed the MTU of their delivery paths.  Therefore, a
          transport protocol must not send packets longer than
          576 octets unless it implements Path MTU Discovery [RFC1191]
          and is capable of adapting its transmitted packet size in
          response to changes of the path MTU.

9.4.  Changes to Link-Layer Protocols

   Link-layer media that have an MTU less than 576 must be enhanced
   with a link-specific fragmentation and reassembly mechanism, to
   support SIP.

   For links on which ARP is used by IPv4, the identical ARP protocol is
   used for SIP.  The low-order 32-bits of SIP addresses are used
   wherever IPv4 addresses would appear; since ARP is used only among
   systems on the same subnet, the high-order 32-bits of the SIP
   addresses may be inferred from the subnet prefix (assuming the subnet
   prefix is at least 32 bits long).  [This is subject to change -- see
   Appendix B.]

10.  Security Considerations

   <to be done>

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11.  Acknowledgments

   The author acknowledges the many helpful suggestions and the words of
   encouragement from Dave Clark, Dave Crocker, Deborah Estrin, Bob
   Hinden, Christian Huitema, Van Jacobson, Jeff Mogul, Dave Nichols,
   Erik Nordmark, Dave Oran, Craig Partridge, Scott Shenker, Paul
   Tsuchiya, Lixia Zhang, the members of End-to-End Research Group and
   the IPAE Working Group, and the participants in the big-internet and
   sip mailing lists.  He apologizes to those whose names he has not
   explicitly listed.  [If you want to be on the list in the next draft,
   just let him know!]

   Editor's note: Steve Deering was employed by the Xerox Palo Alto
   Research Center in Palo Alto, CA USA when this work was done.

12.  Informative References

   [IPAE]     Crocker, D. and R. Hinden, "IP Address Encapsulation
              (IPAE): A Mechanism for Introducing a New IP", Work in
              Progress, draft-crocker-ip-encaps-01, November 1992.

   [RFC791]   Postel, J., "Internet Protocol", STD 5, RFC 791,
              DOI 10.17487/RFC0791, September 1981,

   [RFC792]  Postel, J., "Internet Control Message Protocol", STD 5,
              RFC 792, DOI 10.17487/RFC0792, September 1981,

   [RFC950]  Mogul, J. and J. Postel, "Internet Standard Subnetting
              Procedure", STD 5, RFC 950, DOI 10.17487/RFC0950,
              August 1985, <>.

   [RFC951]  Croft, W. and J. Gilmore, "Bootstrap Protocol", RFC 951,
              DOI 10.17487/RFC0951, September 1985,

   [RFC1112]  Deering, S., "Host extensions for IP multicasting", STD 5,
              RFC 1112, DOI 10.17487/RFC1112, August 1989,

   [RFC1122]  Braden, R., Ed., "Requirements for Internet Hosts -
              Communication Layers", STD 3, RFC 1122,
              DOI 10.17487/RFC1122, October 1989,

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   [RFC1191]  Mogul, J. and S. Deering, "Path MTU discovery", RFC 1191,
              DOI 10.17487/RFC1191, November 1990,

   [RFC1256]  Deering, S., Ed., "ICMP Router Discovery Messages",
              RFC 1256, DOI 10.17487/RFC1256, September 1991,

   [RFC1661]  Simpson, W., Ed., "The Point-to-Point Protocol (PPP)",
              STD 51, RFC 1661, DOI 10.17487/RFC1661, July 1994,

   [RFC1710]  Hinden, R., "Simple Internet Protocol Plus White Paper",
              RFC 1710, DOI 10.17487/RFC1710, October 1994,

   [RFC1752]  Bradner, S. and A. Mankin, "The Recommendation for the IP
              Next Generation Protocol", RFC 1752, DOI 10.17487/RFC1752,
              January 1995, <>.

   [RFC1883]  Deering, S. and R. Hinden, "Internet Protocol, Version 6
              (IPv6) Specification", RFC 1883, DOI 10.17487/RFC1883,
              December 1995, <>.

   [RFC8200]  Deering, S. and R. Hinden, "Internet Protocol, Version 6
              (IPv6) Specification", STD 86, RFC 8200,
              DOI 10.17487/RFC8200, July 2017,

   [SIP-ADDR] Deering, S., "Simple Internet Protocol (SIP) Addressing
              and Routing", Work in Progress, November 1992.

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Appendix A.  SIP Design Rationale

   <this section still to be done>

   Fields present in IPv4, but absent in SIP:

     Header Length    Not needed; SIP header length is fixed.

     Precedence &
     Type of Service  Not used; transport-layer Port fields (or perhaps
                      a to-be-defined value in the Reserved field of the
                      SIP header) may be used for classifying packets at
                      a granularity finer than host-to-host, as required
                      for special handling.

     Header Checksum  Not used; transport pseudo-header checksum
                      protects destinations from accepting corrupted

   Need to justify:

     change of Total Length -> Payload Length, excluding header
     change of Protocol -> Payload Type
     change of Time to Live -> Hop Limit
     movement of fragmentation fields out of fixed header
     bigger minimum MTU, and reliance on PMTU Discovery

Appendix B.  Future Directions

   SIP as specified above is a fully functional replacement for IPv4,
   with a number of improvements, particularly in the areas of
   scalability of routing and addressing, and performance.  Some
   additional improvements are still under consideration:

       o  ARP may be modified to carry full 64-bit addresses, and to use
          link-layer multicast addresses, rather than broadcast

       o  The 28-bit Reserved field in the SIP header may be defined as
          a "Flow ID", or partitioned into a Type of Service field and a
          Flow ID field, for classifying packets deserving of special
          handling, e.g., non-default quality of service or real-time
          service.  On the other hand, the transport-layer port fields
          may be adequate for performing any such classification.  (One
          possibility would be simply to remove the port fields from TCP
          & UDP and append them to the SIP header, as in XNS.)

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       o  A new ICMP "destination has moved" message may defined, for
          re-routing to mobile hosts or subnets, and to domains that
          have changed their address prefixes.

       o  An explicit Trace Route message or option may be defined; the
          current IPv4 traceroute scheme will work fine with SIP, but it
          does not work for multicast, for which it has become very
          apparent that management and debugging tools are needed.

       o  A new Host-to-Router protocol may be specified, encompassing
          the requirements of router discovery, black-hole detection,
          auto- configuration of subnet prefixes, "beaconing" for mobile
          hosts, and, possibly, address resolution.  The OSI End System
          To Intermediate System Protocol may serve as a good model for
          such a protocol.

       o  The requirement that SIP addresses be strictly bound to
          interfaces may be relaxed, so that, for example, a system
          might have fewer addresses than interfaces.  There is some
          experience with this approach in the current Internet, with
          the use of "unnumbered links" in routing protocols such as

       o  Authentication and integrity-assurance mechanisms for all
          clients of SIP, including ICMP and IGMP, may be specified,
          possibly based on the Secure Data Network System (SNDS) SP-3
          or SP-4 protocol.

Authors' Addresses

   Stephen E. Deering
   Vancouver, British Columbia

   Robert M. Hinden (editor)
   Check Point Software
   959 Skyway Road
   San Carlos, CA  94070


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