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