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Network Working Group R. Braden Request for Comments: 1633 ISI Category: Informational D. Clark

                                                            S. Shenker
                                                            Xerox PARC
                                                             June 1994
   Integrated Services in the Internet Architecture: an Overview

Status of this Memo

 This memo provides information for the Internet community.  This memo
 does not specify an Internet standard of any kind.  Distribution of
 this memo is unlimited.


 This memo discusses a proposed extension to the Internet architecture
 and protocols to provide integrated services, i.e., to support real-
 time as well as the current non-real-time service of IP.  This
 extension is necessary to meet the growing need for real-time service
 for a variety of new applications, including teleconferencing, remote
 seminars, telescience, and distributed simulation.
 This memo represents the direct product of recent work by Dave Clark,
 Scott Shenker, Lixia Zhang, Deborah Estrin, Sugih Jamin, John
 Wroclawski, Shai Herzog, and Bob Braden, and indirectly draws upon
 the work of many others.

Table of Contents

 1. Introduction ...................................................2
 2. Elements of the Architecture ...................................3
    2.1 Integrated Services Model ..................................3
    2.2 Reference Implementation Framework .........................6
 3. Integrated Services Model ......................................11
    3.1 Quality of Service Requirements ............................12
    3.2 Resource-Sharing Requirements and Service Models ...........16
    3.3 Packet Dropping ............................................18
    3.4 Usage Feedback .............................................19
    3.5 Reservation Model ..........................................19
 4. Traffic Control Mechanisms .....................................20
    4.1 Basic Functions ............................................20
    4.2 Applying the Mechanisms ....................................23
    4.3 An example .................................................24
 5. Reservation Setup Protocol .....................................25

Braden, Clark & Shenker [Page 1] RFC 1633 Integrated Services Architecture June 1994

    5.1 RSVP Overview ..............................................25
    5.2 Routing and Reservations ...................................28
 6. Acknowledgments ................................................30
 References ........................................................31
 Security Considerations ...........................................32
 Authors' Addresses ................................................33

1. Introduction

 The multicasts of IETF meetings across the Internet have formed a
 large-scale experiment in sending digitized voice and video through a
 packet-switched infrastructure.  These highly-visible experiments
 have depended upon three enabling technologies.  (1) Many modern
 workstations now come equipped with built-in multimedia hardware,
 including audio codecs and video frame-grabbers, and the necessary
 video gear is now inexpensive.  (2) IP multicasting, which is not yet
 generally available in commercial routers, is being provided by the
 MBONE, a temporary "multicast backbone".  (3) Highly-sophisticated
 digital audio and video applications have been developed.
 These experiments also showed that an important technical element is
 still missing: real-time applications often do not work well across
 the Internet because of variable queueing delays and congestion
 losses.  The Internet, as originally conceived, offers only a very
 simple quality of service (QoS), point-to-point best-effort data
 delivery.  Before real-time applications such as remote video,
 multimedia conferencing, visualization, and virtual reality can be
 broadly used, the Internet infrastructure must be modified to support
 real-time QoS, which provides some control over end-to-end packet
 delays.  This extension must be designed from the beginning for
 multicasting; simply generalizing from the unicast (point-to-point)
 case does not work.
 Real-time QoS is not the only issue for a next generation of traffic
 management in the Internet.  Network operators are requesting the
 ability to control the sharing of bandwidth on a particular link
 among different traffic classes.  They want to be able to divide
 traffic into a few administrative classes and assign to each a
 minimum percentage of the link bandwidth under conditions of
 overload, while allowing "unused" bandwidth to be available at other
 times.  These classes may represent different user groups or
 different protocol families, for example.  Such a management facility
 is commonly called controlled link-sharing.  We use the term
 integrated services (IS) for an Internet service model that includes
 best-effort service, real-time service, and controlled link sharing.
 The requirements and mechanisms for integrated services have been the
 subjects of much discussion and research over the past several years

Braden, Clark & Shenker [Page 2] RFC 1633 Integrated Services Architecture June 1994

 (the literature is much too large to list even a representative
 sample here; see the references in [CSZ92, Floyd92, Jacobson91,
 JSCZ93, Partridge92, SCZ93, RSVP93a] for a partial list).  This work
 has led to the unified approach to integrated services support that
 is described in this memo.  We believe that it is now time to begin
 the engineering that must precede deployment of integrated services
 in the Internet.
 Section 2 of this memo introduces the elements of an IS extension of
 the Internet.  Section 3 discusses real-time service models [SCZ93a,
 SCZ93b].  Section 4 discusses traffic control, the forwarding
 algorithms to be used in routers [CSZ92].  Section 5 discusses the
 design of RSVP, a resource setup protocol compatible with the
 assumptions of our IS model [RSVP93a, RSVP93b].

2. Elements of the Architecture

 The fundamental service model of the Internet, as embodied in the
 best-effort delivery service of IP, has been unchanged since the
 beginning of the Internet research project 20 years ago [CerfKahn74].
 We are now proposing to alter that model to encompass integrated
 service.  From an academic viewpoint, changing the service model of
 the Internet is a major undertaking; however, its impact is mitigated
 by the fact that we wish only to extend the original architecture.
 The new components and mechanisms to be added will supplement but not
 replace the basic IP service.
 Abstractly, the proposed architectural extension is comprised of two
 elements: (1) an extended service model, which we call the IS model,
 and (2) a reference implementation framework, which gives us a set of
 vocabulary and a generic program organization to realize the IS
 model.  It is important to separate the service model, which defines
 the externally visible behavior, from the discussion of the
 implementation, which may (and should) change during the life of the
 service model.  However, the two are related; to make the service
 model credible, it is useful to provide an example of how it might be
 2.1 Integrated Services Model
    The IS model we are proposing includes two sorts of service
    targeted towards real-time traffic: guaranteed and predictive
    service.  It integrates these services with controlled link-
    sharing, and it is designed to work well with multicast as well as
    unicast.  Deferring a summary of the IS model to Section 3, we
    first discuss some key assumptions behind the model.

Braden, Clark & Shenker [Page 3] RFC 1633 Integrated Services Architecture June 1994

    The first assumption is that resources (e.g., bandwidth) must be
    explicitly managed in order to meet application requirements.
    This implies that "resource reservation" and "admission control"
    are key building blocks of the service.  An alternative approach,
    which we reject, is to attempt to support real-time traffic
    without any explicit changes to the Internet service model.
    The essence of real-time service is the requirement for some
    service guarantees, and we argue that guarantees cannot be
    achieved without reservations.  The term "guarantee" here is to be
    broadly interpreted; they may be absolute or statistical, strict
    or approximate.  However, the user must be able to get a service
    whose quality is sufficiently predictable that the application can
    operate in an acceptable way over a duration of time determined by
    the user.  Again, "sufficiently" and "acceptable" are vague terms.
    In general, stricter guarantees have a higher cost in resources
    that are made unavailable for sharing with others.
    The following arguments have been raised against resource
    guarantees in the Internet.
    o    "Bandwidth will be infinite."
         The incredibly large carrying capacity of an optical fiber
         leads some to conclude that in the future bandwidth will be
         so abundant, ubiquitous, and cheap that there will be no
         communication delays other than the speed of light, and
         therefore there will be no need to reserve resources.
         However, we believe that this will be impossible in the short
         term and unlikely in the medium term.  While raw bandwidth
         may seem inexpensive, bandwidth provided as a network service
         is not likely to become so cheap that wasting it will be the
         most cost-effective design principle.  Even if low-cost
         bandwidth does eventually become commonly available, we do
         not accept that it will be available "everywhere" in the
         Internet.  Unless we provide for the possibility of dealing
         with congested links, then real-time services will simply be
         precluded in those cases.  We find that restriction
    o    "Simple priority is sufficient."
         It is true that simply giving higher priority to real-time
         traffic would lead to adequate real-time service at some
         times and under some conditions.  But priority is an
         implementation mechanism, not a service model.  If we define
         the service by means of a specific mechanism, we may not get
         the exact features we want.  In the case of simple priority,

Braden, Clark & Shenker [Page 4] RFC 1633 Integrated Services Architecture June 1994

         the issue is that as soon as there are too many real-time
         streams competing for the higher priority, every stream is
         degraded.  Restricting our service to this single failure
         mode is unacceptable.  In some cases, users will demand that
         some streams succeed while some new requests receive a "busy
    o    "Applications can adapt."
         The development of adaptive real-time applications, such as
         Jacobson's audio program VAT, does not eliminate the need to
         bound packet delivery time.  Human requirements for
         interaction and intelligibility limit the possible range of
         adaptation to network delays.  We have seen in real
         experiments that, while VAT can adapt to network delays of
         many seconds, the users find that interaction is impossible
         in these cases.
    We conclude that there is an inescapable requirement for routers
    to be able to reserve resources, in order to provide special QoS
    for specific user packet streams, or "flows".  This in turn
    requires flow-specific state in the routers, which represents an
    important and fundamental change to the Internet model.  The
    Internet architecture was been founded on the concept that all
    flow-related state should be in the end systems [Clark88].
    Designing the TCP/IP protocol suite on this concept led to a
    robustness that is one of the keys to its success.  In section 5
    we discuss how the flow state added to the routers for resource
    reservation can be made "soft", to preserve the robustness of the
    Internet protocol suite.
    There is a real-world side effect of resource reservation in
    routers.  Since it implies that some users are getting privileged
    service, resource reservation will need enforcement of policy and
    administrative controls.  This in turn will lead to two kinds of
    authentication requirements:  authentication of users who make
    reservation requests, and authentication of packets that use the
    reserved resources.  However, these issues are not unique to "IS";
    other aspects of the evolution of the Internet, including
    commercialization and commercial security, are leading to the same
    requirements.  We do not discuss the issues of policy or security
    further in this memo, but they will require attention.
    We make another fundamental assumption, that it is desirable to
    use the Internet as a common infrastructure to support both non-
    real-time and real-time communication.  One could alternatively
    build an entirely new, parallel infrastructure for real-time
    services, leaving the Internet unchanged.  We reject this

Braden, Clark & Shenker [Page 5] RFC 1633 Integrated Services Architecture June 1994

    approach, as it would lose the significant advantages of
    statistical sharing between real-time and non-real-time traffic,
    and it would be much more complex to build and administer than a
    common infrastructure.
    In addition to this assumption of common infrastructure, we adopt
    a unified protocol stack model, employing a single internet-layer
    protocol for both real-time and non-real-time service.  Thus, we
    propose to use the existing internet-layer protocol (e.g., IP or
    CLNP) for real-time data.  Another approach would be to add a new
    real-time protocol in the internet layer [ST2-90].  Our unified
    stack approach provides economy of mechanism, and it allows us to
    fold controlled link-sharing in easily.  It also handles the
    problem of partial coverage, i.e., allowing interoperation between
    IS-capable Internet systems and systems that have not been
    extended, without the complexity of tunneling.
    We take the view that there should be a single service model for
    the Internet.  If there were different service models in different
    parts of the Internet, it is very difficult to see how any end-
    to-end service quality statements could be made.  However, a
    single service model does not necessarily imply a single
    implementation for packet scheduling or admission control.
    Although specific packet scheduling and admission control
    mechanisms that satisfy our service model have been developed, it
    is quite possible that other mechanisms will also satisfy the
    service model.  The reference implementation framework, introduced
    below, is intended to allow discussion of implementation issues
    without mandating a single design.
    Based upon these considerations, we believe that an IS extension
    that includes additional flow state in routers and an explicit
    setup mechanism is necessary to provide the needed service.  A
    partial solution short of this point would not be a wise
    investment.  We believe that the extensions we propose preserve
    the essential robustness and efficiency of the Internet
    architecture, and they allow efficient management of the network
    resources; these will be important goals even if bandwidth becomes
    very inexpensive.
 2.2 Reference Implementation Framework
    We propose a reference implementation framework to realize the IS
    model.  This framework includes four components: the packet
    scheduler, the admission control routine, the classifier, and the
    reservation setup protocol.  These are discussed briefly below and
    more fully in Sections 4 and 5.

Braden, Clark & Shenker [Page 6] RFC 1633 Integrated Services Architecture June 1994

    In the ensuing discussion, we define the "flow" abstraction as a
    distinguishable stream of related datagrams that results from a
    single user activity and requires the same QoS.  For example, a
    flow might consist of one transport connection or one video stream
    between a given host pair.  It is the finest granularity of packet
    stream distinguishable by the IS.  We define a flow to be simplex,
    i.e., to have a single source but N destinations.  Thus, an N-way
    teleconference will generally require N flows, one originating at
    each site.
    In today's Internet, IP forwarding is completely egalitarian; all
    packets receive the same quality of service, and packets are
    typically forwarded using a strict FIFO queueing discipline.  For
    integrated services, a router must implement an appropriate QoS
    for each flow, in accordance with the service model.  The router
    function that creates different qualities of service is called
    "traffic control".  Traffic control in turn is implemented by
    three components: the packet scheduler, the classifier, and
    admission control.
    o    Packet Scheduler
         The packet scheduler manages the forwarding of different
         packet streams using a set of queues and perhaps other
         mechanisms like timers.  The packet scheduler must be
         implemented at the point where packets are queued; this is
         the output driver level of a typical operating system, and
         corresponds to the link layer protocol.  The details of the
         scheduling algorithm may be specific to the particular output
         medium.  For example, the output driver will need to invoke
         the appropriate link-layer controls when interfacing to a
         network technology that has an internal bandwidth allocation
         An experimental packet scheduler has been built that
         implements the IS model described in Section 3 and [SCZ93];
         this is known as the CSZ scheduler and is discussed further
         in Section 4.  We note that the CSZ scheme is not mandatory
         to accomplish our service model; indeed for parts of the
         network that are known always to be underloaded, FIFO will
         deliver satisfactory service.
         There is another component that could be considered part of
         the packet scheduler or separate: the estimator [Jacobson91].
         This algorithm is used to measure properties of the outgoing
         traffic stream, to develop statistics that control packet
         scheduling and admission control.  This memo will consider
         the estimator to be a part of the packet scheduler.

Braden, Clark & Shenker [Page 7] RFC 1633 Integrated Services Architecture June 1994

    o    Classifier
         For the purpose of traffic control (and accounting), each
         incoming packet must be mapped into some class; all packets
         in the same class get the same treatment from the packet
         scheduler.  This mapping is performed by the classifier.
         Choice of a class may be based upon the contents of the
         existing packet header(s) and/or some additional
         classification number added to each packet.
         A class might correspond to a broad category of flows, e.g.,
         all video flows or all flows attributable to a particular
         organization.  On the other hand, a class might hold only a
         single flow.  A class is an abstraction that may be local to
         a particular router; the same packet may be classified
         differently by different routers along the path.  For
         example, backbone routers may choose to map many flows into a
         few aggregated classes, while routers nearer the periphery,
         where there is much less aggregation, may use a separate
         class for each flow.
    o    Admission Control
         Admission control implements the decision algorithm that a
         router or host uses to determine whether a new flow can be
         granted the requested QoS without impacting earlier
         guarantees.  Admission control is invoked at each node to
         make a local accept/reject decision, at the time a host
         requests a real-time service along some path through the
         Internet.  The admission control algorithm must be consistent
         with the service model, and it is logically part of traffic
         control.  Although there are still open research issues in
         admission control, a first cut exists [JCSZ92].
         Admission control is sometimes confused with policing or
         enforcement, which is a packet-by-packet function at the
         "edge" of the network to ensure that a host does not violate
         its promised traffic characteristics.  We consider policing
         to be one of the functions of the packet scheduler.
         In addition to ensuring that QoS guarantees are met,
         admission control will be concerned with enforcing
         administrative policies on resource reservations.  Some
         policies will demand authentication of those requesting
         reservations.  Finally, admission control will play an

Braden, Clark & Shenker [Page 8] RFC 1633 Integrated Services Architecture June 1994

         important role in accounting and administrative reporting.
    The fourth and final component of our implementation framework is
    a reservation setup protocol, which is necessary to create and
    maintain flow-specific state in the endpoint hosts and in routers
    along the path of a flow.  Section  discusses a reservation setup
    protocol called RSVP (for "ReSerVation Protocol") [RSVP93a,
    RSVP93b].  It may not be possible to insist that there be only one
    reservation protocol in the Internet, but we will argue that
    multiple choices for reservation protocols will cause confusion.
    We believe that multiple protocols should exist only if they
    support different modes of reservation.
    The setup requirements for the link-sharing portion of the service
    model are far less clear than those for resource reservations.
    While we expect that much of this can be done through network
    management interfaces, and thus need not be part of the overall
    architecture, we may also need RSVP to play a role in providing
    the required state.
    In order to state its resource requirements, an application must
    specify the desired QoS using a list of parameters that is called
    a "flowspec" [Partridge92].  The flowspec is carried by the
    reservation setup protocol, passed to admission control for to
    test for acceptability, and ultimately used to parametrize the
    packet scheduling mechanism.
    Figure  shows how these components might fit into an IP router
    that has been extended to provide integrated services.  The router
    has two broad functional divisions:  the forwarding path below the
    double horizontal line, and the background code above the line.
    The forwarding path of the router is executed for every packet and
    must therefore be highly optimized.  Indeed, in most commercial
    routers, its implementation involves a hardware assist.  The
    forwarding path is divided into three sections: input driver,
    internet forwarder, and output driver.  The internet forwarder
    interprets the internetworking protocol header appropriate to the
    protocol suite, e.g., the IP header for TCP/IP, or the CLNP header
    for OSI.  For each packet, an internet forwarder executes a
    suite-dependent classifier and then passes the packet and its
    class to the appropriate output driver.  A classifier must be both
    general and efficient.  For efficiency, a common mechanism should
    be used for both resource classification and route lookup.
    The output driver implements the packet scheduler.  (Layerists
    will observe that the output driver now has two distinct sections:
    the packet scheduler that is largely independent of the detailed

Braden, Clark & Shenker [Page 9] RFC 1633 Integrated Services Architecture June 1994

    mechanics of the interface, and the actual I/O driver that is only
    concerned with the grittiness of the hardware.  The estimator
    lives somewhere in between.  We only note this fact, without
    suggesting that it be elevated to a principle.).
     |         ____________     ____________     ___________       |
     |        |            |   | Reservation|   |           |      |
     |        |   Routing  |   |    Setup   |   | Management|      |
     |        |    Agent   |   |    Agent   |   |  Agent    |      |
     |        |______._____|   |______._____|   |_____._____|      |
     |               .                .    |          .            |
     |               .                .   _V________  .            |
     |               .                .  | Admission| .            |
     |               .                .  |  Control | .            |
     |               V                .  |__________| .            |
     |           [Routing ]           V               V            |
     |           [Database]     [Traffic Control Database]         |
     |        |                  |     _______                     |
     |        |   __________     |    |_|_|_|_| => o               |
     |        |  |          |    |      Packet     |     _____     |
     |     ====> |Classifier| =====>   Scheduler   |===>|_|_|_| ===>
     |        |  |__________|    |     _______     |               |
     |        |                  |    |_|_|_|_| => o               |
     | Input  |   Internet       |                                 |
     | Driver |   Forwarder      |     O u t p u t   D r i v e r   |
           Figure 1: Implementation Reference Model for Routers
    The background code is simply loaded into router memory and
    executed by a general-purpose CPU.  These background routines
    create data structures that control the forwarding path.  The
    routing agent implements a particular routing protocol and builds
    a routing database.  The reservation setup agent implements the
    protocol used to set up resource reservations; see Section .  If
    admission control gives the "OK" for a new request, the
    appropriate changes are made to the classifier and packet
    scheduler database to implement the desired QoS.  Finally, every
    router supports an agent for network management.  This agent must
    be able to modify the classifier and packet scheduler databases to
    set up controlled link-sharing and to set admission control

Braden, Clark & Shenker [Page 10] RFC 1633 Integrated Services Architecture June 1994

    The implementation framework for a host is generally similar to
    that for a router, with the addition of applications.  Rather than
    being forwarded, host data originates and terminates in an
    application.  An application needing a real-time QoS for a flow
    must somehow invoke a local reservation setup agent.  The best way
    to interface to applications is still to be determined.  For
    example, there might be an explicit API for network resource
    setup, or the setup might be invoked implicitly as part of the
    operating system scheduling function.  The IP output routine of a
    host may need no classifier, since the class assignment for a
    packet can be specified in the local I/O control structure
    corresponding to the flow.
    In routers, integrated service will require changes to both the
    forwarding path and the background functions.  The forwarding
    path, which may depend upon hardware acceleration for performance,
    will be the more difficult and costly to change.  It will be vital
    to choose a set of traffic control mechanisms that is general and
    adaptable to a wide variety of policy requirements and future
    circumstances, and that can be implemented efficiently.

3. Integrated Services Model

 A service model is embedded within the network service interface
 invoked by applications to define the set of services they can
 request.  While both the underlying network technology and the
 overlying suite of applications will evolve, the need for
 compatibility requires that this service interface remain relatively
 stable (or, more properly, extensible; we do expect to add new
 services in the future but we also expect that it will be hard to
 change existing services).  Because of its enduring impact, the
 service model should not be designed in reference to any specific
 network artifact but rather should be based on fundamental service
 We now briefly describe a proposal for a core set of services for the
 Internet; this proposed core service model is more fully described in
 [SCZ93a, SCZ93b].  This core service model addresses those services
 which relate most directly to the time-of-delivery of packets.  We
 leave the remaining services (such as routing, security, or stream
 synchronization) for other standardization venues.  A service model
 consists of a set of service commitments; in response to a service
 request the network commits to deliver some service.  These service
 commitments can be categorized by the entity to whom they are made:
 they can be made to either individual flows or to collective entities
 (classes of flows).  The service commitments made to individual flows
 are intended to provide reasonable application performance, and thus
 are driven by the ergonomic requirements of the applications; these

Braden, Clark & Shenker [Page 11] RFC 1633 Integrated Services Architecture June 1994

 service commitments relate to the quality of service delivered to an
 individual flow.  The service commitments made to collective entities
 are driven by resource-sharing, or economic, requirements; these
 service commitments relate to the aggregate resources made available
 to the various entities.
 In this section we start by exploring the service requirements of
 individual flows and propose a corresponding set of services.  We
 then discuss the service requirements and services for resource
 sharing.  Finally, we conclude with some remarks about packet
 3.1 Quality of Service Requirements
    The core service model is concerned almost exclusively with the
    time-of-delivery of packets.  Thus, per-packet delay is the
    central quantity about which the network makes quality of service
    commitments.  We make the even more restrictive assumption that
    the only quantity about which we make quantitative service
    commitments are bounds on the maximum and minimum delays.
    The degree to which application performance depends on low delay
    service varies widely, and we can make several qualitative
    distinctions between applications based on the degree of their
    dependence.  One class of applications needs the data in each
    packet by a certain time and, if the data has not arrived by then,
    the data is essentially worthless; we call these real-time
    applications.  Another class of applications will always wait for
    data to arrive; we call these " elastic" applications.  We now
    consider the delay requirements of these two classes separately.
    3.1.1 Real-Time Applications
       An important class of such real-time applications, which are
       the only real-time applications we explicitly consider in the
       arguments that follow, are "playback" applications.  In a
       playback application, the source takes some signal, packetizes
       it, and then transmits the packets over the network.  The
       network inevitably introduces some variation in the delay of
       the delivered packets.  The receiver depacketizes the data and
       then attempts to faithfully play back the signal.  This is done
       by buffering the incoming data and then replaying the signal at
       some fixed offset delay from the original departure time; the
       term "playback point" refers to the point in time which is
       offset from the original departure time by this fixed delay.
       Any data that arrives before its associated playback point can
       be used to reconstruct the signal; data arriving after the
       playback point is essentially useless in reconstructing the

Braden, Clark & Shenker [Page 12] RFC 1633 Integrated Services Architecture June 1994

       real-time signal.
       In order to choose a reasonable value for the offset delay, an
       application needs some "a priori" characterization of the
       maximum delay its packets will experience.  This "a priori"
       characterization could either be provided by the network in a
       quantitative service commitment to a delay bound, or through
       the observation of the delays experienced by the previously
       arrived packets; the application needs to know what delays to
       expect, but this expectation need not be constant for the
       entire duration of the flow.
       The performance of a playback application is measured along two
       dimensions:  latency and fidelity.  Some playback applications,
       in particular those that involve interaction between the two
       ends of a connection such as a phone call, are rather sensitive
       to the latency; other playback applications, such as
       transmitting a movie or lecture, are not.  Similarly,
       applications exhibit a wide range of sensitivity to loss of
       fidelity.  We will consider two somewhat artificially
       dichotomous classes: intolerant applications, which require an
       absolutely faithful playback, and tolerant applications, which
       can tolerate some loss of fidelity.  We expect that the vast
       bulk of audio and video applications will be tolerant, but we
       also suspect that there will be other applications, such as
       circuit emulation, that are intolerant.
       Delay can affect the performance of playback applications in
       two ways.  First, the value of the offset delay, which is
       determined by predictions about the future packet delays,
       determines the latency of the application.  Second, the delays
       of individual packets can decrease the fidelity of the playback
       by exceeding the offset delay; the application then can either
       change the offset delay in order to play back late packets
       (which introduces distortion) or merely discard late packets
       (which creates an incomplete signal).  The two different ways
       of coping with late packets offer a choice between an
       incomplete signal and a distorted one, and the optimal choice
       will depend on the details of the application, but the
       important point is that late packets necessarily decrease
       Intolerant applications must use a fixed offset delay, since
       any variation in the offset delay will introduce some
       distortion in the playback.  For a given distribution of packet
       delays, this fixed offset delay must be larger than the
       absolute maximum delay, to avoid the possibility of late
       packets.   Such an application can only set its offset delay

Braden, Clark & Shenker [Page 13] RFC 1633 Integrated Services Architecture June 1994

       appropriately if it is given a perfectly reliable upper bound
       on the maximum delay of each packet.  We call a service
       characterized by a perfectly reliable upper bound on delay "
       guaranteed service", and propose this as the appropriate
       service model for intolerant playback applications.
       In contrast, tolerant applications need not set their offset
       delay greater than the absolute maximum delay, since they can
       tolerate some late packets.  Moreover, instead of using a
       single fixed value for the offset delay, they can attempt to
       reduce their latency by varying their offset delays in response
       to the actual packet delays experienced in the recent past.  We
       call applications which vary their offset delays in this manner
       "adaptive" playback applications.
       For tolerant applications we propose a service model called "
       predictive service" which supplies a fairly reliable, but not
       perfectly reliable, delay bound.  This bound, in contrast to
       the bound in the guaranteed service, is not based on worst case
       assumptions on the behavior of other flows.  Instead, this
       bound might be computed with properly conservative predictions
       about the behavior of other flows.  If the network turns out to
       be wrong and the bound is violated, the application's
       performance will perhaps suffer, but the users are willing to
       tolerate such interruptions in service in return for the
       presumed lower cost of the service.  Furthermore, because many
       of the tolerant applications are adaptive, we augment the
       predictive service to also give "minimax" service, which is to
       attempt to minimize the ex post maximum delay.  This service is
       not trying to minimize the delay of every packet, but rather is
       trying to pull in the tail of the delay distribution.
       It is clear that given a choice, with all other things being
       equal, an application would perform no worse with absolutely
       reliable bounds than with fairly reliable bounds.  Why, then,
       do we offer predictive service?  The key consideration here is
       efficiency; when one relaxes the service requirements from
       perfectly to fairly reliable bounds, this increases the level
       of network utilization that can be sustained, and thus the
       price of the predictive service will presumably be lower than
       that of guaranteed service.  The predictive service class is
       motivated by the conjecture that the performance penalty will
       be small for tolerant applications but the overall efficiency
       gain will be quite large.
       In order to provide a delay bound, the nature of the traffic
       from the source must be characterized, and there must be some
       admission control algorithm which insures that a requested flow

Braden, Clark & Shenker [Page 14] RFC 1633 Integrated Services Architecture June 1994

       can actually be accommodated. A fundamental point of our
       overall architecture is that traffic characterization and
       admission control are necessary for these real-time delay bound
       services.  So far we have assumed that an application's data
       generation process is an intrinsic property unaffected by the
       network.  However, there are likely to be many audio and video
       applications which can adjust their coding scheme and thus can
       alter the resulting data generation process depending on the
       network service available.  This alteration of the coding
       scheme will present a tradeoff between fidelity (of the coding
       scheme itself, not of the playback process) and the bandwidth
       requirements of the flow.  Such "rate-adaptive" playback
       applications have the advantage that they can adjust to the
       current network conditions not just by resetting their playback
       point but also by adjusting the traffic pattern itself.  For
       rate-adaptive applications, the traffic characterizations used
       in the service commitment are not immutable.  We can thus
       augment the service model by allowing the network to notify
       (either implicitly through packet drops or explicitly through
       control packets) rate-adaptive applications to change their
       traffic characterization.
    3.1.2 Elastic Applications
       While real-time applications do not wait for late data to
       arrive, elastic applications will always wait for data to
       arrive.  It is not that these applications are insensitive to
       delay; to the contrary, significantly increasing the delay of a
       packet will often harm the application's performance.  Rather,
       the key point is that the application typically uses the
       arriving data immediately, rather than buffering it for some
       later time, and will always choose to wait for the incoming
       data rather than proceed without it.  Because arriving data can
       be used immediately, these applications do not require any a
       priori characterization of the service in order for the
       application to function.  Generally speaking, it is likely that
       for a given distribution of packet delays, the perceived
       performance of elastic applications will depend more on the
       average delay than on the tail of the delay distribution.  One
       can think of several categories of such elastic applications:
       interactive burst (Telnet, X, NFS), interactive bulk transfer
       (FTP), and asynchronous bulk transfer (electronic mail, FAX).
       The delay requirements of these elastic applications vary from
       rather demanding for interactive burst applications to rather
       lax for asynchronous bulk transfer, with interactive bulk
       transfer being intermediate between them.

Braden, Clark & Shenker [Page 15] RFC 1633 Integrated Services Architecture June 1994

       An appropriate service model for elastic applications is to
       provide "as-soon-as-possible", or ASAP service. (For
       compatibility with historical usage, we will use the term
       best-effort service when referring to ASAP service.).  We
       furthermore propose to offer several classes of best-effort
       service to reflect the relative delay sensitivities of
       different elastic applications.  This service model allows
       interactive burst applications to have lower delays than
       interactive bulk applications, which in turn would have lower
       delays than asynchronous bulk applications.  In contrast to the
       real-time service models, applications using this service are
       not subject to admission control.
       The taxonomy of applications into tolerant playback, intolerant
       playback, and elastic is neither exact nor complete, but was
       only used to guide the development of the core service model.
       The resulting core service model should be judged not on the
       validity of the underlying taxonomy but rather on its ability
       to adequately meet the needs of the entire spectrum of
       applications.  In particular, not all real-time applications
       are playback applications; for example, one might imagine a
       visualization application which merely displayed the image
       encoded in each packet whenever it arrived.  However, non-
       playback applications can still use either the guaranteed or
       predictive real-time service model, although these services are
       not specifically tailored to their needs.  Similarly, playback
       applications cannot be neatly classified as either tolerant or
       intolerant, but rather fall along a continuum; offering both
       guaranteed and predictive service allows applications to make
       their own tradeoff between fidelity, latency, and cost.
       Despite these obvious deficiencies in the taxonomy, we expect
       that it describes the service requirements of current and
       future applications well enough so that our core service model
       can adequately meet all application needs.
 3.2 Resource-Sharing Requirements and Service Models
    The last section considered quality of service commitments; these
    commitments dictate how the network must allocate its resources
    among the individual flows.  This allocation of resources is
    typically negotiated on a flow-by-flow basis as each flow requests
    admission to the network, and does not address any of the policy
    issues that arise when one looks at collections of flows.  To
    address these collective policy issues, we now discuss resource-
    sharing service commitments.  Recall that for individual quality
    of service commitments we focused on delay as the only quantity of
    interest.  Here, we postulate that the quantity of primary
    interest in resource-sharing is aggregate bandwidth on individual

Braden, Clark & Shenker [Page 16] RFC 1633 Integrated Services Architecture June 1994

    links.  Thus, this component of the service model, called "link-
    sharing", addresses the question of how to share the aggregate
    bandwidth of a link among various collective entities according to
    some set of specified shares.  There are several examples that are
    commonly used to explain the requirement of link-sharing among
    collective entities.
    Multi-entity link-sharing. -- A link may be purchased and used
    jointly by several organizations, government agencies or the like.
    They may wish to insure that under overload the link is shared in
    a controlled way, perhaps in proportion to the capital investment
    of each entity.  At the same time, they might wish that when the
    link is underloaded, any one of the entities could utilize all the
    idle bandwidth.
    Multi-protocol link-sharing -- In a multi-protocol Internet, it
    may be desired to prevent one protocol family (DECnet, IP, IPX,
    OSI, SNA, etc.) from overloading the link and excluding the other
    families. This is important because different families may have
    different methods of detecting and responding to congestion, and
    some methods may be more "aggressive" than others. This could lead
    to a situation in which one protocol backs off more rapidly than
    another under congestion, and ends up getting no bandwidth.
    Explicit control in the router may be required to correct this.
    Again, one might expect that this control should apply only under
    overload, while permitting an idle link to be used in any
    Multi-service sharing -- Within a protocol family such as IP, an
    administrator might wish to limit the fraction of bandwidth
    allocated to various service classes.  For example, an
    administrator might wish to limit the amount of real-time traffic
    to some fraction of the link, to avoid preempting elastic traffic
    such as FTP.
    In general terms, the link-sharing service model is to share the
    aggregate bandwidth according to some specified shares.  We can
    extend this link-sharing service model to a hierarchical version.
    For instance, a link could be divided between a number of
    organizations, each of which would divide the resulting allocation
    among a number of protocols, each of which would be divided among
    a number of services.  Here, the sharing is defined by a tree with
    shares assigned to each leaf node.
    An idealized fluid model of instantaneous link-sharing with
    proportional sharing of excess is the fluid processor sharing
    model (introduced in [DKS89] and further explored in [Parekh92]
    and generalized to the hierarchical case) where at every instant

Braden, Clark & Shenker [Page 17] RFC 1633 Integrated Services Architecture June 1994

    the available bandwidth is shared between the active entities
    (i.e., those having packets in the queue) in proportion to the
    assigned shares of the resource.  This fluid model exhibits the
    desired policy behavior but is, of course, an unrealistic
    idealization.  We then propose that the actual service model
    should be to approximate, as closely as possible, the bandwidth
    shares produced by this ideal fluid model.  It is not necessary to
    require that the specific order of packet departures match those
    of the fluid model since we presume that all detailed per-packet
    delay requirements of individual flows are addressed through
    quality of service commitments and, furthermore, the satisfaction
    with the link-sharing service delivered will probably not depend
    very sensitively on small deviations from the scheduling implied
    by the fluid link-sharing model.
    We previously observed that admission control was necessary to
    ensure that the real-time service commitments could be met.
    Similarly, admission control will again be necessary to ensure
    that the link-sharing commitments can be met.  For each entity,
    admission control must keep the cumulative guaranteed and
    predictive traffic from exceeding the assigned link-share.
 3.3 Packet Dropping
    So far, we have implicitly assumed that all packets within a flow
    were equally important.  However, in many audio and video streams,
    some packets are more valuable than others.  We therefore propose
    augmenting the service model with a "preemptable" packet service,
    whereby some of the packets within a flow could be marked as
    preemptable.  When the network was in danger of not meeting some
    of its quantitative service commitments, it could exercise a
    certain packet's "preemptability option" and discard the packet
    (not merely delay it, since that would introduce out-of-order
    problems).  By discarding these preemptable packets, a router can
    reduce the delays of the not-preempted packets.
    Furthermore, one can define a class of packets that is not subject
    to admission control.  In the scenario described above where
    preemptable packets are dropped only when quantitative service
    commitments are in danger of being violated, the expectation is
    that preemptable packets will almost always be delivered and thus
    they must included in the traffic description used in admission
    control.  However, we can extend preemptability to the extreme
    case of "expendable" packets (the term expendable is used to
    connote an extreme degree of preemptability), where the
    expectation is that many of these expendable packets may not be
    delivered.  One can then exclude expendable packets from the
    traffic description used in admission control; i.e., the packets

Braden, Clark & Shenker [Page 18] RFC 1633 Integrated Services Architecture June 1994

    are not considered part of the flow from the perspective of
    admission control, since there is no commitment that they will be
 3.4 Usage Feedback
    Another important issue in the service is the model for usage
    feedback, also known as "accounting", to prevent abuse of network
    resources.   The link-sharing service described earlier can be
    used to provide administratively-imposed limits on usage.
    However, a more free-market model of network access will require
    back-pressure on users for the network resources they reserve.
    This is a highly contentious issue, and we are not prepared to say
    more about it at this time.
 3.5 Reservation Model
    The "reservation model" describes how an application negotiates
    for a QoS level.  The simplest model is that the application asks
    for a particular QoS and the network either grants it or refuses.
    Often the situation will be more complex.  Many applications will
    be able to get acceptable service from a range of QoS levels, or
    more generally, from anywhere within some region of the multi-
    dimensional space of a flowspec.
    For example, rather than simply refusing the request, the network
    might grant a lower resource level and inform the application of
    what QoS has been actually granted.  A more complex example is the
    "two-pass" reservation model, In this scheme, an "offered"
    flowspec is propagated along the multicast distribution tree from
    each sender Si to all receivers Rj.  Each router along the path
    records these values and perhaps adjusts them to reflect available
    capacity.  The receivers get these offers, generate corresponding
    "requested" flowspecs, and propagate them back along the same
    routes to the senders.  At each node, a local reconciliation must
    be performed between the offered and the requested flowspec to
    create a reservation, and an appropriately modified requested
    flowspec is passed on.  This two-pass scheme allows extensive
    properties like allowed delay to be distributed across hops in the
    path [Tenet90, ST2-90].  Further work is needed to define the
    amount of generality, with a corresponding level of complexity,
    that is required in the reservation model.

Braden, Clark & Shenker [Page 19] RFC 1633 Integrated Services Architecture June 1994

4. Traffic Control Mechanisms

 We first survey very briefly the possible traffic control mechanisms.
 Then in Section 4.2 we apply a subset of these mechanisms to support
 the various services that we have proposed.
 4.1 Basic Functions
    In the packet forwarding path, there is actually a very limited
    set of actions that a router can take.  Given a particular packet,
    a router must select a route for it; in addition the router can
    either forward it or drop it, and the router may reorder it with
    respect to other packets waiting to depart.  The router can also
    hold the packet, even though the link is idle.  These are the
    building blocks from which we must fashion the desired behavior.
    4.1.1 Packet Scheduling
       The basic function of packet scheduling is to reorder the
       output queue.  There are many papers that have been written on
       possible ways to manage the output queue, and the resulting
       behavior.  Perhaps the simplest approach is a priority scheme,
       in which packets are ordered by priority, and highest priority
       packets always leave first.  This has the effect of giving some
       packets absolute preference over others; if there are enough of
       the higher priority packets, the lower priority class can be
       completely prevented from being sent.
       An alternative scheduling scheme is round-robin or some
       variant, which gives different classes of packets access to a
       share of the link. A variant called Weighted Fair Queueing, or
       WFQ, has been demonstrated to allocate the total bandwidth of a
       link into specified shares.
       There are more complex schemes for queue management, most of
       which involve observing the service objectives of individual
       packets, such as delivery deadline, and ordering packets based
       on these criteria.
    4.1.2 Packet Dropping
       The controlled dropping of packets is as important as their
       Most obviously, a router must drop packets when its buffers are
       all full.  This fact, however, does not determine which packet
       should be dropped.  Dropping the arriving packet, while simple,
       may cause undesired behavior.

Braden, Clark & Shenker [Page 20] RFC 1633 Integrated Services Architecture June 1994

       In the context of today's Internet, with TCP operating over
       best effort IP service, dropping a packet is taken by TCP as a
       signal of congestion and causes it to reduce its load on the
       network.  Thus, picking a packet to drop is the same as picking
       a source to throttle.  Without going into any particular
       algorithm, this simple relation suggests that some specific
       dropping controls should be implemented in routers to improve
       congestion control.
       In the context of real-time services, dropping more directly
       relates to achieving the desired quality of service.  If a
       queue builds up, dropping one packet reduces the delay of all
       the packets behind it in the queue.  The loss of one can
       contribute to the success of many.  The problem for the
       implementor is to determine when the service objective (the
       delay bound) is in danger of being violated.  One cannot look
       at queue length as an indication of how long packets have sat
       in a queue.  If there is a priority scheme in place, packets of
       lower priority can be pre-empted indefinitely, so even a short
       queue may have very old packets in it.  While actual time
       stamps could be used to measure holding time, the complexity
       may be unacceptable.
       Some simple dropping schemes, such as combining all the buffers
       in a single global pool, and dropping the arriving packet if
       the pool is full, can defeat the service objective of a WFQ
       scheduling scheme.  Thus, dropping and scheduling must be
    4.1.3 Packet Classification
       The above discussion of scheduling and dropping presumed that
       the packet had been classified into some flow or sequence of
       packets that should be treated in a specified way.  A
       preliminary to this sort of processing is the classification
       itself.  Today a router looks at the destination address and
       selects a route.  The destination address is not sufficient to
       select the class of service a packet must receive; more
       information is needed.
       One approach would be to abandon the IP datagram model for a
       virtual circuit model, in which a circuit is set up with
       specific service attributes, and the packet carries a circuit
       identifier.  This is the approach of ATM as well as protocols
       such as ST-II [ST2-90].  Another model, less hostile to IP, is
       to allow the classifier to look at more fields in the packet,
       such as the source address, the protocol number and the port
       fields.  Thus, video streams might be recognized by a

Braden, Clark & Shenker [Page 21] RFC 1633 Integrated Services Architecture June 1994

       particular well-known port field in the UDP header, or a
       particular flow might be recognized by looking at both the
       source and destination port numbers.  It would be possible to
       look even deeper into the packets, for example testing a field
       in the application layer to select a subset of a
       hierarchically-encoded video stream.
       The classifier implementation issues are complexity and
       processing overhead.  Current experience suggests that careful
       implementation of efficient algorithms can lead to efficient
       classification of IP packets.  This result is very important,
       since it allows us to add QoS support to existing applications,
       such as Telnet, which are based on existing IP headers.
       One approach to reducing the overhead of classification would
       be to provide a "flow-id" field in the Internet-layer packet
       header.  This flow-id would be a handle that could be cached
       and used to short-cut classification of the packet.  There are
       a number of variations of this concept, and engineering is
       required to choose the best design.
    4.1.4 Admission Control
       As we stated in the introduction, real-time service depends on
       setting up state in the router and making commitments to
       certain classes of packets.  In order to insure that these
       commitments can be met, it is necessary that resources be
       explicitly requested, so that the request can be refused if the
       resources are not available.  The decision about resource
       availability is called admission control.
       Admission control requires that the router understand the
       demands that are currently being made on its assets.  The
       approach traditionally proposed is to remember the service
       parameters of past requests, and make a computation based on
       the worst-case bounds on each service.  A recent proposal,
       which is likely to provide better link utilization, is to
       program the router to measure the actual usage by existing
       packet flows, and to use this measured information as a basis
       of admitting new flows [JCSZ92]. This approach is subject to
       higher risk of overload, but may prove much more effective in
       using bandwidth.
       Note that while the need for admission control is part of the
       global service model, the details of the algorithm run in each
       router is a local matter.  Thus, vendors can compete by
       developing and marketing better admission control algorithms,
       which lead to higher link loadings with fewer service

Braden, Clark & Shenker [Page 22] RFC 1633 Integrated Services Architecture June 1994

 4.2 Applying the Mechanisms
    The various tools described above can be combined to support the
    services which were discussed in section 3.
    o    Guaranteed delay bounds
         A theoretical result by Parekh [Parekh92] shows that if the
         router implements a WFQ scheduling discipline, and if the
         nature of the traffic source can be characterized (e.g. if it
         fits within some bound such as a token bucket) then there
         will be an absolute upper bound on the network delay of the
         traffic in question.  This simple and very powerful result
         applies not just to one switch, but to general networks of
         routers.  The result is a constructive one; that is, Parekh
         displays a source behavior which leads to the bound, and then
         shows that this behavior is the worst possible.  This means
         that the bound he computes is the best there can be, under
         these assumptions.
    o    Link sharing
         The same WFQ scheme can provide controlled link sharing.  The
         service objective here is not to bound delay, but to limit
         overload shares on a link, while allowing any mix of traffic
         to proceed if there is spare capacity.  This use of WFQ is
         available in commercial routers today, and is used to
         segregate traffic into classes based on such things as
         protocol type or application.  For example, one can allocate
         separate shares to TCP, IPX and SNA, and one can assure that
         network control traffic gets a guaranteed share of the link.
    o    Predictive real-time service
         This service is actually more subtle than guaranteed service.
         Its objective is to give a delay bound which is, on the one
         hand, as low as possible, and on the other hand, stable
         enough that the receiver can estimate it.  The WFQ mechanism
         leads to a guaranteed bound, but not necessarily a low bound.
         In fact, mixing traffic into one queue, rather than
         separating it as in WFQ, leads to lower bounds, so long as
         the mixed traffic is generally similar (e.g., mixing traffic
         from multiple video coders makes sense, mixing video and FTP
         does not).

Braden, Clark & Shenker [Page 23] RFC 1633 Integrated Services Architecture June 1994

         This suggests that we need a two-tier mechanism, in which the
         first tier separates traffic which has different service
         objectives, and the second tier schedules traffic within each
         first tier class in order to meet its service objective.
 4.3 An example: The CSZ scheme
    As a proof of concept, a code package has been implemented which
    realizes the services discussed above.  It actually uses a number
    of the basic tools, combined in a way specific to the service
    needs.  We describe in general terms how it works, to suggest how
    services can be realized.  We stress that there are other ways of
    building a router to meet the same service needs, and there are in
    fact other implementations being used today.
    At the top level, the CSZ code uses WFQ as an isolation mechanism
    to separate guaranteed flows from each other, as well as from the
    rest of the traffic.  Guaranteed service gets the highest priority
    when and only when it needs the access to meets its deadline.  WFQ
    provides a separate guarantee for each and every guaranteed flow.
    Predictive service and best effort service are separated by
    priority.  Within the predictive service class, a further priority
    is used to provide sub-classes with different delay bounds.
    Inside each predictive sub-class, simple FIFO queueing is used to
    mix the traffic, which seems to produce good overall delay
    behavior.  This works because the top-tier algorithm has separated
    out the best effort traffic such as FTP.
    Within the best-effort class, WFQ is used to provide link sharing.
    Since there is a possible requirement for nested shares, this WFQ
    code can be used recursively.  There are thus two different uses
    of WFQ in this code, one to segregate the guaranteed classes, and
    one to segregate the link shares.  They are similar, but differ in
    Within each link share of the best effort class, priority is used
    to permit more time-sensitive elastic traffic to precede other
    elastic traffic, e.g., to allow interactive traffic to precede
    asynchronous bulk transfers.
    The CSZ code thus uses both WFQ and priority in an alternating
    manner to build a mechanism to support a range of rather
    sophisticated service offerings.  This discussion is very brief,
    and does not touch on a number of significant issues, such as how
    the CSZ code fits real time traffic into the link sharing
    objectives.  But the basic building blocks are very simple, and

Braden, Clark & Shenker [Page 24] RFC 1633 Integrated Services Architecture June 1994

    very powerful.  In particular, while priority has been proposed as
    a key to real-time services, WFQ may be the more general and
    powerful of the two schemes.  It, rather than priority, supports
    guaranteed service and link sharing.

5. Reservation Setup Protocol

 There are a number of requirements to be met by the design of a
 reservation setuop protocol.  It should be fundamentally designed for
 a multicast environment, and it must accommodate heterogeneous
 service needs.  It must give flexible control over the manner in
 which reservations can be shared along branches of the multicast
 delivery trees.  It should be designed around the elementary action
 of adding one sender and/or receiver to an existing set, or deleting
 one.  It must be robust and scale well to large multicast groups.
 Finally, it must provide for advance reservation of resources, and
 for the preemption that this implies.  The reservation setup protocol
 RSVP has been designed to meet these requirements [RSVP93a, RSVP93b].
 This section gives an overview of the design of RSVP.
 5.1 RSVP Overview
    Figure  shows multi-source, multi-destination data delivery for a
    particular shared, distributed application.  The arrows indicate
    data flow from senders S1 and S2 to receivers R1, R2, and R3, and
    the cloud represents the distribution mesh created by the
    multicast routing protocol.  Multicasting distribution replicates
    each data packet from a sender Si, for delivery to every receiver
    Rj.  We treat uncast delivery from S1 to R1 as a special case, and
    we call this multicast distribution mesh a session.  A session is
    defined by the common IP (multicast) destination address of the
               Senders                              Receivers
                          (                     ) ===> R1
                  S1 ===> (    Multicast        )
                          (                     ) ===> R2
                          (    distribution     )
                  S2 ===> (                     )
                          (                     ) ===> R3
                 Figure 2: Multicast Distribution Session

Braden, Clark & Shenker [Page 25] RFC 1633 Integrated Services Architecture June 1994

    5.1.1 Flowspecs and Filter Specs
       In general, an RSVP reservation request specifies the amount of
       resources to be reserved for all, or some subset of, the
       packets in a particular session.  The resource quantity is
       specified by a flowspec, while the packet subset to receive
       those resources is specified by a filter spec.  Assuming
       admission control succeeds, the flowspec will be used to
       parametrize a resource class in the packet scheduler, and the
       filter spec will be instantiated in the packet classifier to
       map the appropriate packets into this class.  The subset of the
       classifier state that selects a particular class is referred to
       in RSVP documentation as a (packet) "filter".
       The RSVP protocol mechanisms provide a very general facility
       for creating and maintaining distributed reservation state
       across the mesh of multicast delivery paths.  These mechanisms
       treat flowspecs and filter specs as mostly opaque binary data,
       handing them to the local traffic control machinery for
       interpretation.  Of course, the service model presented to an
       application must specify how to encode flowspecs and filter
    5.1.2 Reservation Styles
       RSVP offers several different reservation "styles", which
       determine the manner in which the resource requirements of
       multiple receivers are aggregated in the routers.  These styles
       allow the reserved resources to more efficiently meet
       application requirements.  Currently there are three
       reservation styles, "wildcard", "fixed-filter", and " dynamic-
       filter".  A wildcard reservation uses a filter spec that is not
       source-specific, so all packets destined for the associated
       destination (session) may use a common pool of reserved
       resources.  This allows a single resource allocation to be made
       across all distribution paths for the group.  The wildcard
       reservation style is useful in support of an audio conference,
       where at most a small number of sources are active
       simultaneously and may share the resource allocation.
       The other two styles use filter specs that select particular
       sources.  A receiver may desire to receive from a fixed set of
       sources, or instead it may desire the network to switch between
       different source, by changing its filter spec(s) dymamically.
       A fixed-filter style reservation cannot be changed during its
       lifetime without re-invoking admission control.  Dynamic-filter
       reservations do allow a receiver to modify its choice of
       source(s) over time without additional admission control;

Braden, Clark & Shenker [Page 26] RFC 1633 Integrated Services Architecture June 1994

       however, this requires that sufficient resources be allocated
       to handle the worst case when all downstream receivers take
       input from different sources.
    5.1.3 Receiver Initiation
       An important design question is whether senders or receivers
       should have responsibility for initiating reservations.  A
       sender knows the qualities of the traffic stream it can send,
       while a receiver knows what it wants to (or can) receive.
       Perhaps the most obvious choice is to let the sender initiate
       the reservation.  However, this scales poorly for large,
       dynamic multicast delivery trees and for heterogeneous
       Both of these scaling problems are solved by making the
       receiver responsible for initiating a reservation.  Receiver
       initiation  handles heterogeneous receivers easily; each
       receiver simply asks for a reservation appropriate to itself,
       and any differences among reservations from different receivers
       are resolved ("merged") within the network by RSVP.  Receiver
       initiation is also consisent with IP multicast, in which a
       multicast group is created implicitly by receivers joining it.
       Although receiver-initiated reservation is the natural choice
       for multicast sessions, the justification for receiver
       initiateion may appear weaker for unicast sessions, where the
       sender may be the logical session initiator.  However, we
       expect that every realtime application will have its higher-
       level signalling and control protocol, and this protocol can be
       used to signal the receiver to initiate a reservation (and
       perhaps indicate the flowspec to be used).  For simplicity and
       economy, a setup protocol should support only one direction of
       initiation, and, and receiver initiation appears to us to be
       the clear winner.
       RSVP uses receiver-initiation of rservations [RSVP93b].  A
       receiver is assumed to learn the senders' offered flowspecs by
       a higher-level mechanism ("out of band"), it then generates its
       own desired flowspec and propagates it towards the senders,
       making reservations in each router along the way.
    5.1.4 Soft State
       There are two different possible styles for reservation setup
       protocols, the "hard state" (HS) approach (also called
       "connection-oriented"), and the "soft state" (SS) approach
       (also called "connectionless").  In both approaches, multicast

Braden, Clark & Shenker [Page 27] RFC 1633 Integrated Services Architecture June 1994

       distribution is performed using flow-specific state in each
       router along the path.  Under the HS approach, this state is
       created and deleted in a fully deterministic manner by
       cooperation among the routers.  Once a host requests a session,
       the "network" takes responsibility for creating and later
       destroying the necessary state.  ST-II is an example of the HS
       approach [ST2-90].  Since management of HS session state is
       completely deterministic, the HS setup protocol must be
       reliable, with acknowledgments and retransmissions.  In order
       to achieve deterministic cleanup of state after a failure,
       there must be some mechanism to detect failures, i.e., an
       "up/down" protocol.  The router upstream (towards the source)
       from a failure takes responsibility for rebuilding the
       necessary state on the router(s) along an alternate route.
       RSVP takes the SS approach, which regards the reservation state
       as cached information that is installed and periodically
       refreshed by the end hosts.  Unused state is timed out by the
       routers.  If the route changes, the refresh messages
       automatically install the necessary state along the new route.
       The SS approach was chosen to obtain the simplicity and
       robustness that have been demonstrated by connectionless
       protocols such as IP [Clark88].
 5.2 Routing and Reservations
    There is a fundamental interaction between resource reservation
    set up and routing, since reservation requires the installation of
    flow state along the route of data packets.  If and when a route
    changes, there must be some mechanism to set up a reservation
    along the new route.
    Some have suggested that reservation setup necessarily requires
    route set up, i.e., the imposition of a virtual-circuit internet
    layer.  However, our goal is to simply extend the Internet
    architecture, not replace it.  The fundamental connectionless
    internet layer [Clark88] has been highly successful, and we wish
    to retain it as an architectural foundation.  We propose instead
    to modify somewhat the pure datagram forwarding mechanism of the
    present Internet to accomodate "IS".

Braden, Clark & Shenker [Page 28] RFC 1633 Integrated Services Architecture June 1994

    There are four routing issues faced by a reservation setup
    protocol such as RSVP.
    1.   Find a route that supports resource reservation.
         This is simply "type-of-service" routing, a facility that is
         already available in some modern routing protocols.
    2.   Find a route that has sufficient unreserved capacity for a
         new flow.
         Early experiments on the ARPANET showed that it is difficult
         to do load-dependent dynamic routing on a packet-by-packet
         basis without instability problems.  However, instability
         should not be a problem if load-dependent routing is
         performed only at reservation setup time.
         Two different approaches might be taken to finding a route
         with enough capacity.  One could modify the routing
         protocol(s) and interface them to the traffic control
         mechanism, so the route computation can consider the average
         recent load.  Alternatively, the routing protocol could be
         (re-)designed to provide multiple alternative routes, and
         reservation setup could be attempted along each in turn.
    3.   Adapt to a route failure
         When some node or link fails, adaptive routing finds an
         alternate path.  The periodic refresh messages of RSVP will
         automatically request a reservation along the new path.  Of
         course, this reservation may fail because there is
         insufficienct available capacity on the new path.  This is a
         problem of provisioning and network engineering, which cannot
         be solved by the routing or setup protocols.
         There is a problem of timeliness of establishing reservation
         state on the new path.  The end-to-end robustness mechanism
         of refreshes is limited in frequency by overhead, which may
         cause a gap in realtime service when an old route breaks and
         a new one is chosen.  It should be possible to engineer RSVP
         to sypplement the global refresh mechanism with a local
         repair mechanism, using hints about route changes from the
         routing mechanism.
    4.   Adapt to a route change (without failure)
         Route changes may occur even without failure in the affected
         path.  Although RSVP could use the same repair techniques as

Braden, Clark & Shenker [Page 29] RFC 1633 Integrated Services Architecture June 1994

         those described in (3), this case raises a problem with the
         robustness of the QoS guarantees.  If it should happen that
         admission control fails on the new route, the user will see
         service degradation unnecessarily and capriciously, since the
         orginal route is still functional.
         To avoid this problem, a mechanism called "route pinning" has
         been suggested.  This would modify the routing protocol
         implementation and the interface to the classifier, so that
         routes associated with resource reservations would be
         "pinned".  The routing prootocol would not change a pinned
         route if it was still viable.
    It may eventually be possible to fold together the routing and
    reservation setup problems, but we do not yet understand enough to
    do that.  Furthermore, the reservation protocol needs to coexist
    with a number of different routing protocols in use in the
    Internet.  Therefore, RSVP is currently designed to work with any
    current-generation routing protocol without modification.  This is
    a short-term compromise, which may result in an occasional failure
    to create the best, or even any, real-time session, or an
    occasional service degradation due to a route change.  We expect
    that future generations of routing protocols will remove this
    compromise, by including hooks and mechanisms that, in conjunction
    with RSVP, will solve the problems (1) through (4) just listed.
    They will support route pinning, notification of RSVP to trigger
    local repair, and selection of routes with "IS" support and
    adequate capacity.
    The last routing-related issue is provided by mobile hosts.  Our
    conjecture is that mobility is not essentially different from
    other route changes, so that the mechanism suggested in (3) and
    (4) will suffice.  More study and experimentation is needed to
    prove or disprove this conjecture.


 Many Internet researchers have contributed to the work described in
 this memo.  We want to especially acknowledge, Steve Casner, Steve
 Deering, Deborah Estrin, Sally Floyd, Shai Herzog, Van Jacobson,
 Sugih Jamin, Craig Partridge, John Wroclawski, and Lixia Zhang.  This
 approach to Internet integrated services was initially discussed and
 organized in the End-to-End Research Group of the Internet Research
 Taskforce, and we are grateful to all members of that group for their
 interesting (and sometimes heated) discussions.

Braden, Clark & Shenker [Page 30] RFC 1633 Integrated Services Architecture June 1994


[CerfKahn74] Cerf, V., and R. Kahn, "A Protocol for Packet Network

  Intercommunication", IEEE Trans on Comm., Vol. Com-22, No. 5, May

[Clark88] Clark, D., "The Design Philosophy of the DARPA Internet

  Protocols", ACM SIGCOMM '88, August 1988.

[CSZ92] Clark, D., Shenker, S., and L. Zhang, "Supporting Real-Time

  Applications in an Integrated Services Packet Network: Architecture
  and Mechanisms", Proc. SIGCOMM '92, Baltimore, MD, August 1992.

[DKS89] Demers, A., Keshav, S., and S. Shenker. "Analysis and

  Simulation of a Fair Queueing Algorithm", Journal of
  Internetworking: Research and Experience, 1, pp. 3-26, 1990.  Also
  in Proc. ACM SIGCOMM '89, pp 3-12.

[SCZ93a] Shenker, S., Clark, D., and L. Zhang, "A Scheduling Service

  Model and a Scheduling Architecture for an Integrated Services
  Packet Network", submitted to ACM/IEEE Trans. on Networking.

[SCZ93b] Shenker, S., Clark, D., and L. Zhang, "A Service Model for the

  Integrated Services Internet", Work in Progress, October 1993.

[Floyd92] Floyd, S., "Issues in Flexible Resource Management for

  Datagram Networks", Proceedings of the 3rd Workshop on Very High
  Speed Networks, March 1992.

[Jacobson91] Jacobson, V., "Private Communication", 1991.

[JCSZ92] Jamin, S., Shenker, S., Zhang, L., and D. Clark, "An Admission

  Control Algorithm for Predictive Real-Time Service", Extended
  abstract, in Proc. Third International Workshop on Network and
  Operating System Support for Digital Audio and Video, San Diego, CA,
  Nov. 1992, pp.  73-91.

[Parekh92] Parekh, A., "A Generalized Processor Sharing Approach to

  Flow Control in Integrated Services Networks", Technical Report
  LIDS-TR-2089, Laboratory for Information and Decision Systems,
  Massachusetts Institute of Technology, 1992.

[Partridge92] Partridge, C., "A Proposed Flow Specification", RFC 1363,

  BBN, July 1992.

[RSVP93a] Zhang, L., Deering, S., Estrin, D., Shenker, S., and D.

  Zappala, "RSVP: A New Resource ReSerVation Protocol", Accepted for
  publication in IEEE Network, 1993.

Braden, Clark & Shenker [Page 31] RFC 1633 Integrated Services Architecture June 1994

[RSVP93b] Zhang, L., Braden, R., Estrin, D., Herzog, S., and S. Jamin,

  "Resource ReSerVation Protocol (RSVP) - Version 1 Functional
  Specification", Work in Progress, 1993.

[ST2-90] Topolcic, C., "Experimental Internet Stream Protocol: Version

  2 (ST-II)", RFC 1190, BBN, October 1990.

[Tenet90] Ferrari, D., and D. Verma, "A Scheme for Real-Time Channel

  Establishment in Wide-Area Networks", IEEE JSAC, Vol. 8, No. 3, pp
  368-379, April 1990.

Security Considerations

 As noted in Section 2.1, the ability to reserve resources will create
 a requirement for authentication, both of users requesting resource
 guarantees and of packets that claim to have the right to use those
 guarantees.  These authentication issues are not otherwise addressed
 in this memo, but are for further study.

Braden, Clark & Shenker [Page 32] RFC 1633 Integrated Services Architecture June 1994

Authors' Addresses

 Bob Braden
 USC Information Sciences Institute
 4676 Admiralty Way
 Marina del Rey, CA 90292
 Phone: (310) 822-1511
 EMail: Braden@ISI.EDU
 David Clark
 MIT Laboratory for Computer Science
 545 Technology Square
 Cambridge, MA 02139-1986
 Phone: (617) 253-6003
 EMail: ddc@LCS.MIT.EDU
 Scott Shenker
 Xerox Palo Alto Research Center
 3333 Coyote Hill Road
 Palo Alto, CA 94304
 Phone: (415) 812-4840
 EMail: Shenker@PARC.XEROX.COM

Braden, Clark & Shenker [Page 33]

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