[Docs] [txt|pdf] [draft-ietf-tsvwg-...] [Diff1] [Diff2]

Updated by: 5865 INFORMATIONAL

Network Working Group                                           F. Baker
Request for Comments: 4542                                       J. Polk
Category: Informational                                    Cisco Systems
                                                                May 2006


    Implementing an Emergency Telecommunications Service (ETS) for
           Real-Time Services in the Internet Protocol Suite

Status of This Memo

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

Copyright Notice

   Copyright (C) The Internet Society (2006).

Abstract

   RFCs 3689 and 3690 detail requirements for an Emergency
   Telecommunications Service (ETS), of which an Internet Emergency
   Preparedness Service (IEPS) would be a part.  Some of these types of
   services require call preemption; others require call queuing or
   other mechanisms.  IEPS requires a Call Admission Control (CAC)
   procedure and a Per Hop Behavior (PHB) for the data that meet the
   needs of this architecture.  Such a CAC procedure and PHB is
   appropriate to any service that might use H.323 or SIP to set up
   real-time sessions.  The key requirement is to guarantee an elevated
   probability of call completion to an authorized user in time of
   crisis.

   This document primarily discusses supporting ETS in the context of
   the US Government and NATO, because it focuses on the Multi-Level
   Precedence and Preemption (MLPP) and Government Emergency
   Telecommunication Service (GETS) standards.  The architectures
   described here are applicable beyond these organizations.













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Table of Contents

   1. Overview of the Internet Emergency Preference Service
      Problem and Proposed Solutions ..................................3
      1.1. Emergency Telecommunications Services ......................3
           1.1.1. Multi-Level Preemption and Precedence ...............4
           1.1.2. Government Emergency Telecommunications Service .....6
      1.2. Definition of Call Admission ...............................6
      1.3. Assumptions about the Network ..............................7
      1.4. Assumptions about Application Behavior .....................7
      1.5. Desired Characteristics in an Internet Environment .........9
      1.6. The Use of Bandwidth as a Solution for QoS ................10
   2. Solution Proposal ..............................................11
      2.1. Call Admission/Preemption Procedure .......................12
      2.2. Voice Handling Characteristics ............................15
      2.3. Bandwidth Admission Procedure .............................17
           2.3.1. RSVP Admission Using Policy for Both
                  Unicast and Multicast Sessions .....................17
           2.3.2. RSVP Scaling Issues ................................19
           2.3.3. RSVP Operation in Backbones and Virtual
                  Private Networks (VPNs) ............................19
           2.3.4. Interaction with the Differentiated
                  Services Architecture ..............................21
           2.3.5. Admission Policy ...................................21
      2.4. Authentication and Authorization of Calls Placed ..........23
      2.5. Defined User Interface ....................................23
   3. Security Considerations ........................................24
   4. Acknowledgements ...............................................24
   5. References .....................................................25
      5.1. Normative References ......................................25
      5.2. Informative References ....................................27
   Appendix A.  2-Call Preemption Example using RSVP .................29



















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1.  Overview of the Internet Emergency Preference Service Problem and
    Proposed Solutions

   [RFC3689] and [RFC3690] detail requirements for an Emergency
   Telecommunications Service (ETS), of which an Internet Emergency
   Preference Service (IEPS) would be a part.  Some of these types of
   services require call preemption; others require call queuing or
   other mechanisms.  The key requirement is to guarantee an elevated
   probability of call completion to an authorized user in time of
   crisis.

   IEPS requires a Call Admission Control procedure and a Per Hop
   Behavior for the data that meet the needs of this architecture.  Such
   a CAC procedure and PHB is appropriate to any service that might use
   H.323 or SIP to set up real-time sessions.  These obviously include
   but are not limited to Voice and Video applications, although at this
   writing the community is mostly thinking about Voice on IP, and many
   of the examples in the document are taken from that environment.

   In a network where a call permitted initially is not denied or
   rejected at a later time, capacity admission procedures performed
   only at the time of call setup may be sufficient.  However, in a
   network where session status can be reviewed by the network and
   preempted or denied due to changes in routing (when the new routes
   lack capacity to carry calls switched to them) or changes in offered
   load (where higher precedence calls supersede existing calls),
   maintaining a continuing model of the status of the various calls is
   required.

1.1.  Emergency Telecommunications Services

   Before doing so, however, let us discuss the problem that ETS (and
   therefore IEPS) is intended to solve and the architecture of the
   system.  The Emergency Telecommunications Service [ITU.ETS.E106] is a
   successor to and generalization of two services used in the United
   States: Multi-Level Precedence and Preemption (MLPP), and the
   Government Emergency Telecommunication Service (GETS).  Services
   based on these models are also used in a variety of countries
   throughout the world, both Public Switched Telephone Network (PSTN)
   and Global System for Mobile Communications (GSM)-based.  Both of
   these services are designed to enable an authorized user to obtain
   service from the telephone network in times of crisis.  They differ
   primarily in the mechanisms used and number of levels of precedence
   acknowledged.







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1.1.1.  Multi-Level Preemption and Precedence

   The Assured Service is designed as an IP implementation of an
   existing ITU-T/NATO/DoD telephone system architecture known as
   Multi-Level Precedence and Preemption [ITU.MLPP.1990]
   [ANSI.MLPP.Spec] [ANSI.MLPP.Supp], or MLPP.  MLPP is an architecture
   for a prioritized call handling service such that in times of
   emergency in the relevant NATO and DoD commands, the relative
   importance of various kinds of communications is strictly defined,
   allowing higher-precedence communication at the expense of lower-
   precedence communications.  This document describes NATO and US
   Department of Defense uses of MLPP, but the architecture and standard
   are applicable outside of these organizations.

   These precedences, in descending order, are:

   Flash Override Override:  used by the Commander in Chief, Secretary
      of Defense, and Joint Chiefs of Staff, commanders of combatant
      commands when declaring the existence of a state of war.
      Commanders of combatant commands when declaring Defense Condition
      One or Defense Emergency or Air Defense Emergency and other
      national authorities that the President may authorize in
      conjunction with Worldwide Secure Voice Conferencing System
      conferences.  Flash Override Override cannot be preempted.  This
      precedence level is not enabled on all DoD networks.

   Flash Override:  used by the Commander in Chief, Secretary of
      Defense, and Joint Chiefs of Staff, commanders of combatant
      commands when declaring the existence of a state of war.
      Commanders of combatant commands when declaring Defense Condition
      One or Defense Emergency and other national authorities the
      President may authorize.  Flash Override cannot be preempted in
      the DSN.

   Flash:  reserved generally for telephone calls pertaining to command
      and control of military forces essential to defense and
      retaliation, critical intelligence essential to national survival,
      conduct of diplomatic negotiations critical to the arresting or
      limiting of hostilities, dissemination of critical civil alert
      information essential to national survival, continuity of federal
      government functions essential to national survival, fulfillment
      of critical internal security functions essential to national
      survival, or catastrophic events of national or international
      significance.

   Immediate:  reserved generally for telephone calls pertaining to
      situations that gravely affect the security of national and allied
      forces, reconstitution of forces in a post-attack period,



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      intelligence essential to national security, conduct of diplomatic
      negotiations to reduce or limit the threat of war, implementation
      of federal government actions essential to national survival,
      situations that gravely affect the internal security of the
      nation, Civil Defense actions, disasters or events of extensive
      seriousness having an immediate and detrimental effect on the
      welfare of the population, or vital information having an
      immediate effect on aircraft, spacecraft, or missile operations.

   Priority:  reserved generally for telephone calls requiring
      expeditious action by called parties and/or furnishing essential
      information for the conduct of government operations.

   Routine:  designation applied to those official government
      communications that require rapid transmission by telephonic means
      but do not require preferential handling.

   MLPP is intended to deliver a higher probability of call completion
   to the more important calls.  The rule, in MLPP, is that more
   important calls override less important calls when congestion occurs
   within a network.  Station-based preemption is used when a more
   important call needs to be placed to either party in an existing
   call.  Trunk-based preemption is used when trunk bandwidth needs to
   be reallocated to facilitate a higher-precedence call over a given
   path in the network.  In both station- and trunk-based preemption
   scenarios, preempted parties are positively notified, via preemption
   tone, that their call can no longer be supported.  The same
   preemption tone is used, regardless of whether calls are terminated
   for the purposes of station- of trunk-based preemption.  The
   remainder of this discussion focuses on trunk-based preemption
   issues.

   MLPP is built as a proactive system in which callers must assign one
   of the precedence levels listed above at call initiation; this
   precedence level cannot be changed throughout that call.  If an
   elevated status is not assigned by a user at call initiation time,
   the call is assumed to be "routine".  If there is end-to-end capacity
   to place a call, any call may be placed at any time.  However, when
   any trunk group (in the circuit world) or interface (in an IP world)
   reaches a utilization threshold, a choice must be made as to which
   calls to accept or allow to continue.  The system will seize the
   trunk(s) or bandwidth necessary to place the more important calls in
   preference to less important calls by preempting an existing call (or
   calls) of lower precedence to permit a higher-precedence call to be
   placed.






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   More than one call might properly be preempted if more trunks or
   bandwidth is necessary for this higher precedence call.  A video call
   (perhaps of 384 KBPS, or 6 trunks) competing with several lower-
   precedence voice calls is a good example of this situation.

1.1.2.  Government Emergency Telecommunications Service

   A US service similar to MLPP and using MLPP signaling technology, but
   built for use in civilian networks, is the Government Emergency
   Telecommunications Service (GETS).  This differs from MLPP in two
   ways: it does not use preemption, but rather reserves bandwidth or
   queues calls to obtain a high probability of call completion, and it
   has only two levels of service: "Routine" and "Priority".

   GETS is described here as another example.  Similar architectures are
   applied by other governments and organizations.

1.2.  Definition of Call Admission

   Traditionally, in the PSTN, Call Admission Control (CAC) has had the
   responsibility of implementing bandwidth available thresholds (e.g.,
   to limit resources consumed by some traffic) and determining whether
   a caller has permission (e.g., is an identified subscriber, with
   identify attested to by appropriate credentials) to use an available
   circuit.  IEPS, or any emergency telephone service, has additional
   options that it may employ to improve the probability of call
   completion:

   o  The call may be authorized to use other networks that it would not
      normally use;

   o  The network may preempt other calls to free bandwidth;

   o  The network may hold the call and place it when other calls
      complete; or

   o  The network may use different bandwidth availability thresholds
      than are used for other calls.

   At the completion of CAC, however, the caller either has a circuit
   that he or she is authorized to use or has no circuit.  Since the act
   of preemption or consideration of alternative bandwidth sources is
   part and parcel of the problem of providing bandwidth, the
   authorization step in bandwidth provision also affects the choice of
   networks that may be authorized to be considered.  The three cannot
   be separated.  The CAC procedure finds available bandwidth that the
   caller is authorized to use and preemption may in some networks be
   part of making that happen.



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1.3.  Assumptions about the Network

   IP networks generally fall into two categories: those with
   constrained bandwidth, and those that are massively over-provisioned.
   In a network where over any interval that can be measured (including
   sub-second intervals) capacity exceeds offered load by at least 2:1,
   the jitter and loss incurred in transit are nominal.  This is
   generally a characteristic of properly engineered Ethernet LANs and
   of optical networks (networks that measure their link speeds in
   multiples of 51 MBPS); in the latter, circuit-switched networking
   solutions such as Asynchronous Transfer Mode (ATM), MPLS, and GMPLS
   can be used to explicitly place routes, which improves the odds a
   bit.

   Between those networks, in places commonly called "inter-campus
   links", "access links", or "access networks", for various reasons
   including technology (e.g., satellite links) and cost, it is common
   to find links whose offered load can approximate or exceed the
   available capacity.  Such events may be momentary or may occur for
   extended periods of time.

   In addition, primarily in tactical deployments, it is common to find
   bandwidth constraints in the local infrastructure of networks.  For
   example, the US Navy's network afloat connects approximately 300
   ships, via satellite, to five network operation centers (NOCs), and
   those NOCs are in turn interconnected via the Defense Information
   Systems Agency (DISA) backbone.  A typical ship may have between two
   and six radio systems aboard, often at speeds of 64 KBPS or less.  In
   US Army networks, current radio technology likewise limits tactical
   communications to links below 100 KBPS.

   Over this infrastructure, military communications expect to deploy
   voice communication systems (30-80 KBPS per session) and video
   conferencing using MPEG 2 (3-7 MBPS) and MPEG 4 (80 KBPS to 800
   KBPS), in addition to traditional mail, file transfer, and
   transaction traffic.

1.4.  Assumptions about Application Behavior

   Parekh and Gallagher published a series of papers [Parekh1] [Parekh2]
   analyzing what is necessary to ensure a specified service level for a
   stream of traffic.  In a nutshell, they showed that to predict the
   behavior of a stream of traffic in a network, one must know two
   things:

   o  the rate and arrival distribution with which traffic in a class is
      introduced to the network, and




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   o  what network elements will do, in terms of the departure
      distribution, injected delay jitter, and loss characteristics,
      with the traffic they see.

   For example, TCP tunes its effective window (the amount of data it
   sends per round trip interval) so that the ratio of the window and
   the round trip interval approximate the available capacity in the
   network.  As long as the round trip delay remains roughly stable and
   loss is nominal (which are primarily behaviors of the network), TCP
   is able to maintain a predictable level of throughput.  In an
   environment where loss is random or in which delays wildly vary, TCP
   behaves in a far less predictable manner.

   Voice and video systems, in the main, are designed to deliver a fixed
   level of quality as perceived by the user.  (Exceptions are systems
   that select rate options over a broad range to adapt to ambient loss
   characteristics.  These deliver broadly fluctuating perceived quality
   and have not found significant commercial applicability.)  Rather,
   they send traffic at a rate specified by the codec depending on what
   it perceives is required.  In an MPEG-4 system, for example, if the
   camera is pointed at a wall, the codec determines that an 80 KBPS
   data stream will describe that wall and issues that amount of
   traffic.  If a person walks in front of the wall or the camera is
   pointed an a moving object, the codec may easily send 800 KBPS in its
   effort to accurately describe what it sees.  In commercial broadcast
   sports, which may line up periods in which advertisements are
   displayed, the effect is that traffic rates suddenly jump across all
   channels at certain times because the eye-catching ads require much
   more bandwidth than the camera pointing at the green football field.

   As described in [RFC1633], when dealing with a real-time application,
   there are basically two things one must do to ensure Parekh's first
   requirement.  To ensure that one knows how much offered load the
   application is presenting, one must police (measure load offered and
   discard excess) traffic entering the network.  If that policing
   behavior has a debilitating effect on the application, as non-
   negligible loss has on voice or video, one must admit sessions
   judiciously according to some policy.  A key characteristic of that
   policy must be that the offered load does not exceed the capacity
   dedicated to the application.

   In the network, the other thing one must do is ensure that the
   application's needs are met in terms of loss, variation in delay, and
   end-to-end delay.  One way to do this is to supply sufficient
   bandwidth so that loss and jitter are nominal.  Where that cannot be
   accomplished, one must use queuing technology to deterministically
   apply bandwidth to accomplish the goal.




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1.5.  Desired Characteristics in an Internet Environment

   The key elements of the Internet Emergency Preference Service include
   the following:

   Precedence Level Marking each call:  Call initiators choose the
      appropriate precedence level for each call based on the user-
      perceived importance of the call.  This level is not to be changed
      for the duration of the call.  The call before and the call after
      are independent with regard to this level choice.

   Call Admission/Preemption Policy:  There is likewise a clear policy
      regarding calls that may be in progress at the called instrument.
      During call admission (SIP/H.323), if they are of lower
      precedence, they must make way according to a prescribed
      procedure.  All callers on the preempted call must be informed
      that the call has been preempted, and the call must make way for
      the higher-precedence call.

   Bandwidth Admission Policy:  There is a clear bandwidth admission
      policy: sessions may be placed that assert any of several levels
      of precedence, and in the event that there is demand and
      authorization is granted, other sessions will be preempted to make
      way for a call of higher precedence.

   Authentication and Authorization of calls placed:  Unauthorized
      attempts to place a call at an elevated status are not permitted.
      In the telephone system, this is managed by controlling the policy
      applied to an instrument by its switch plus a code produced by the
      caller identifying himself or herself to the switch.  In the
      Internet, such characteristics must be explicitly signaled.

   Voice handling characteristics:  A call made, in the telephone
      system, gets a circuit and provides the means for the callers to
      conduct their business without significant impact as long as their
      call is not preempted.  In a VoIP system, one would hope for
      essentially the same service.

   Defined User Interface:  If a call is preempted, the caller and the
      callee are notified via a defined signal, so that they know that
      their call has been preempted and that at this instant there is no
      alternative circuit available to them at that precedence level.

   A VoIP implementation of the Internet Emergency Preference Service
   must, by definition, provide those characteristics.






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1.6.  The Use of Bandwidth as a Solution for QoS

   There is a discussion in Internet circles concerning the relationship
   of bandwidth to QoS procedures, which needs to be put to bed before
   this procedure can be adequately analyzed.  The issue is that it is
   possible and common in certain parts of the Internet to solve the
   problem with bandwidth.  In LAN environments, for example, if there
   is significant loss between any two switches or between a switch and
   a server, the simplest and cheapest solution is to buy the next
   faster interface: substitute 100 MBPS for 10 MBPS Ethernet, 1 gigabit
   for 100 MBPS, or, for that matter, upgrade to a 10-gigabit Ethernet.
   Similarly, in optical networking environments, the simplest and
   cheapest solution is often to increase the data rate of the optical
   path either by selecting a faster optical carrier or deploying an
   additional lambda.  In places where the bandwidth can be over-
   provisioned to a point where loss or queuing delay are negligible,
   10:1 over-provisioning is often the cheapest and surest solution and,
   by the way, offers a growth path for future requirements.  However,
   there are many places in communication networks where the provision
   of effectively infinite bandwidth is not feasible, including many
   access networks, satellite communications, fixed wireless, airborne
   and marine communications, island connections, and connections to
   regions in which fiber optic connections are not cost-effective.  It
   is in these places where the question of resource management is
   relevant.  Specifically, we do not recommend the deployment of
   significant QoS procedures on links in excess of 100 MBPS apart from
   the provision of aggregated services that provide specific protection
   to the stability of the network or the continuity of real-time
   traffic as a class, as the mathematics of such circuits do not
   support this as a requirement.

   In short, the fact that we are discussing this class of policy
   control says that such constrictions in the network exist and must be
   dealt with.  However much we might like to, in those places we are
   not solving the problem with bandwidth.
















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2.  Solution Proposal

   A typical voice or video network, including a backbone domain, is
   shown in Figure 1.

      ...............             ......................
     .                .          .                      .
    .  H  H  H  H      .        .   H  H  H  H           .
   .  /----------/      .       .  /----------/           .
   .     R     SIP      .       .    R      R              .
   .      \             .       .   /        \              .
   .       R  H  H  H   . .......  /          \             .
   .      /----------/  ..      ../            R     SIP    .
    .              R  ..         /.           /----------/  .
      .....       ..\.    R-----R  .           H  H  H  H   .
            ......  .\   /       \  .                      .
                    . \ /         \  .                    .
                     .  R-----------R  ....................
                     .   \         /   .
                     .    \       /   .
                     .     R-----R   .
                      .             .
                       ............

           SIP   = SIP Proxy
           H     = SIP-enabled Host (Telephone, call gateway or PC)
           R     = Router
           /---/ = Ethernet or Ethernet Switch

              Figure 1: Typical VoIP or Video/IP Network

  Reviewing the figure above, it becomes obvious that Voice/IP and
  Video/IP call flows are very different than call flows in the PSTN.
  In the PSTN, call control traverses a switch, which in turn controls
  data handling services like ATM or Time Division Multiplexing (TDM)
  switches or multiplexers.  While they may not be physically co-
  located, the control plane software and the data plane services are
  closely connected; the switch routes a call using bandwidth that it
  knows is available.  In a voice/video-on-IP network, call control is
  completely divorced from the data plane: It is possible for a
  telephone instrument in the United States to have a Swedish telephone
  number if that is where its SIP proxy happens to be, but on any given
  call for it to use only data paths in the Asia/Pacific region, data
  paths provided by a different company, and, often, data paths provided
  by multiple companies/providers.






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  Call management therefore addresses a variety of questions, all of
  which must be answered:

   o  May I make this call from an administrative policy perspective?
      Am I authorized to make this call?

   o  What IP address correlates with this telephone number or SIP URI?

   o  Is the other instrument "on hook"?  If it is busy, under what
      circumstances may I interrupt?

   o  Is there bandwidth available to support the call?

   o  Does the call actually work, or do other impairments (loss, delay)
      make the call unusable?

2.1.  Call Admission/Preemption Procedure

   Administrative Call Admission is the objective of SIP and H.323.  It
   asks fundamental questions like "What IP address is the callee at?"
   and "Did you pay your bill?".

   For a specialized policy like call preemption, two capabilities are
   necessary from an administrative perspective: [RFC4412] provides a
   way to communicate policy-related information regarding the
   precedence of the call; and [RFC4411] provides a reason code when a
   call fails or is refused, indicating the cause of the event.  If it
   is a failure, it may make sense to redial the call.  If it is a
   policy-driven preemption, even if the call is redialed it may not be
   possible to place the call.  Requirements for this service are
   further discussed in [RFC3689].

   The SIP Communications Resource Priority Header (or RP Header) serves
   the call setup process with the precedence level chosen by the
   initiator of the call.  The syntax is in the form:

        Resource Priority: namespace.priority level

   The "namespace" part of the syntax ensures the domain of significance
   to the originator of the call, and this travels end-to-end to the
   destination (called) device (telephone).  If the receiving phone does
   not support the namespace, it can easily ignore the setup request.
   This ability to denote the domain of origin allows Service Level
   Agreements (SLAs) to be in place to limit the ability of an unknown
   requester to gain preferential treatment into an IEPS domain.






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   For the DSN infrastructure, the header would look like this for a
   routine precedence level call:

        Resource Priority: dsn.routine

   The precedence level chosen in this header would be compared to the
   requester's authorization profile to use that precedence level.  This
   would typically occur in the SIP first-hop Proxy, which can challenge
   many aspects of the call setup request including the requester's
   choice of precedence levels (verifying that they are not using a
   level they are not authorized to use).

   The DSN has 5 precedence levels of IEPS, in descending order:

        dsn.flash-override

        dsn.flash

        dsn.immediate

        dsn.priority

        dsn.routine

   The US Defense Red Switched Network (DRSN), as another example that
   was IANA-registered in [RFC4412], has 6 levels of precedence.  The
   DRSN simply adds one precedence level higher than flash-override to
   be used by the President and a select few others:

        drsn.flash-override-override

   Note that the namespace changed for this level.  The lower 5 levels
   within the DRSN would also have this as their namespace for all
   DRSN-originated call setup requests.

   The Resource-Priority Header (RPH) informs both the use of
   Differentiated Services Code Points (DSCPs) by the callee (who needs
   to use the same DSCP as the caller to obtain the same data path
   service) and to facilitate policy-based preemption of calls in
   progress, when appropriate.

   Once a call is established in an IEPS domain, the Reason Header for
   Preemption, described in [RFC4411], ensures that all SIP nodes are
   synchronized to a preemption event occurring either at the endpoint
   or in a router that experiences congestion.  In SIP, the normal
   indication for the end of a session is for one end system to send a
   BYE Method request as specified in [RFC3261].  This, too, is the
   proper means for signaling a termination of a call due to a



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   preemption event, as it essentially performs a normal termination
   with additional information informing the peer of the reason for the
   abrupt end: it indicates that a preemption occurred.  This will be
   used to inform all relevant SIP entities, and whether this was an
   endpoint-generated preemption event, or that the preemption event
   occurred within a router along the communications path (described in
   Section 2.3.1).

   Figure 2 is a simple example of a SIP call setup that includes the
   layer 7 precedence of a call between Alice and Bob.  After Alice
   successfully sets up a call to Bob at the "Routine" precedence level,
   Carol calls Bob at a higher precedence level (Immediate).  At the SIP
   layer (this has nothing to do with RSVP yet; that example, involving
   SIP and RSVP signaling, is in the appendix), once Bob's user agent
   (phone) receives the INVITE message from Carol, his UA needs to make
   a choice between retaining the call to Alice and sending Carol a
   "busy" indication, or preempting the call to Alice in favor of
   accepting the call from Carol.  That choice in IEPS networks is a
   comparison of Resource Priority headers.  Alice, who controlled the
   precedence level of the call to Bob, sent the precedence level of her
   call to him at "Routine" (the lowest level within the network).
   Carol, who controls the priority of the call signal to Bob, sent her
   priority level to "Immediate" (higher than "Routine").  Bob's UA
   needs to (under IEPS policy) preempt the call from Alice (and provide
   her with a preemption indication in the call termination message).
   Bob needs to successfully answer the call setup from Carol.

























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      UA Alice                     UA Bob                       UA Carol
         |    INVITE (RP: Routine)    |                             |
         |--------------------------->|                             |
         |           200 OK           |                             |
         |<---------------------------|                             |
         |            ACK             |                             |
         |--------------------------->|                             |
         |            RTP             |                             |
         |<==========================>|                             |
         |                            |                             |
         |                            |   INVITE (RP: Immediate)    |
         |                            |<----------------------------|
         |      ************************************************    |
         |      *Resource Priority value comparison by Bob's UA*    |
         |      ************************************************    |
         |                            |                             |
         | BYE (Reason: UA preemption)                              |
         |<---------------------------|                             |
         |                            |           200 OK            |
         |                            |---------------------------->|
         |       200 OK (BYE)         |                             |
         |--------------------------->|                             |
         |                            |            ACK              |
         |                            |<----------------------------|
         |                            |            RTP              |
         |                            |<===========================>|
         |                            |                             |

    Figure 2: Priority Call Establishment and Termination at SIP Layer

   Nothing in this example involved mechanisms other than SIP.  It is
   also assumed each user agent recognized the Resource-Priority header
   namespace value in each message.  Therefore, it is assumed that the
   domain allowed Alice, Bob, and Carol to communicate.  Authentication
   and Authorization are discussed later in this document.

2.2.  Voice Handling Characteristics

   The Quality of Service architecture used in the data path is that of
   [RFC2475].  Differentiated Services uses a flag in the IP header
   called the DSCP [RFC2474] to identify a data stream, and then applies
   a procedure called a Per Hop Behavior, or PHB, to it.  This is
   largely as described in [RFC2998].

   In the data path, the Expedited Forwarding PHB [RFC3246] [RFC3247]
   describes the fundamental needs of voice and video traffic.  This PHB
   entails ensuring that sufficient bandwidth is dedicated to real-time
   traffic to ensure that variation in delay and loss rate are minimal,



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   as codecs are hampered by excessive loss [G711.1] [G711.3].  In parts
   of the network where bandwidth is heavily over-provisioned, there may
   be no remaining concern.  In places in the network where bandwidth is
   more constrained, this may require the use of a priority queue.  If a
   priority queue is used, the potential for abuse exists, meaning that
   it is also necessary to police traffic placed into the queue to
   detect and manage abuse.  A fundamental question is "where does this
   policing need to take place?".  The obvious places would be the
   first-hop routers and any place where converging data streams might
   congest a link.

   Some proposals mark traffic with various code points appropriate to
   the service precedence of the call.  In normal service, if the
   traffic is all in the same queue and EF service requirements are met
   (applied capacity exceeds offered load, variation in delay is
   minimal, and loss is negligible), details of traffic marking should
   be irrelevant, as long as packets get into the right service class.
   Then, the major issues are appropriate policing of traffic,
   especially around route changes, and ensuring that the path has
   sufficient capacity.

   The real-time voice/video application should be generating traffic at
   a rate appropriate to its content and codec, which is either a
   constant bit rate stream or a stream whose rate is variable within a
   specified range.  The first-hop router should be policing traffic
   originated by the application, as is performed in traditional virtual
   circuit networks like Frame Relay and ATM.  Between these two checks
   (at what some networks call the Data Terminal Equipment (DTE) and
   Data Communications Equipment (DCE)), the application traffic should
   be guaranteed to be within acceptable limits.  As such, given
   bandwidth-aware call admission control, there should be minimal
   actual loss.  The cases where loss would occur include cases where
   routing has recently changed and CAC has not caught up, or cases
   where statistical thresholds are in use in CAC and the data streams
   happen to coincide at their peak rates.

   If it is demonstrated that routing transients and variable rate beat
   frequencies present a sufficient problem, it is possible to provide a
   policing mechanism that isolates intentional loss among an ordered
   set of classes.  While the ability to do so, by various algorithms,
   has been demonstrated, the technical requirement has not.  If
   dropping random packets from all calls is not appropriate,
   concentrating random loss in a subset of the calls makes the problem
   for those calls worse; a superior approach would reject or preempt an
   entire call.

   Parekh's second condition has been met: we must know what the network
   will do with the traffic.  If the offered load exceeds the available



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   bandwidth, the network will remark and drop the excess traffic.  The
   key questions become "How does one limit offered load to a rate less
   than or equal to available bandwidth?" and "How much traffic does one
   admit with each appropriate marking?"

2.3.  Bandwidth Admission Procedure

   Since many available voice and video codecs require a nominal loss
   rate to deliver acceptable performance, Parekh's first requirement is
   that offered load be within the available capacity.  There are
   several possible approaches.

   An approach that is commonly used in H.323 networks is to limit the
   number of calls simultaneously accepted by the gatekeeper.  SIP
   networks do something similar when they place a stateful SIP proxy
   near a single ingress/egress to the network.  This is able to impose
   an upper bound on the total number of calls in the network or the
   total number of calls crossing the significant link.  However, the
   gatekeeper has no knowledge of routing, so the engineering must be
   very conservative and usually presumes a single ingress/egress or the
   failure of one of its data paths.  While this may serve as a short-
   term work-around, it is not a general solution that is readily
   deployed.  This limits the options in network design.

   [RFC1633] provides for signaled admission for the use of capacity.
   The recommended approach is explicit capacity admission, supporting
   the concepts of preemption.  An example of such a procedure uses the
   Resource Reservation Protocol [RFC2205] [RFC2209] (RSVP).  The use of
   Capacity Admission using RSVP with SIP is described in [RFC3312].
   While call counting is specified in H.323, network capacity admission
   is not integrated with H.323 at this time.

2.3.1.  RSVP Admission Using Policy for Both Unicast and Multicast
        Sessions

   RSVP is a resource reservation setup protocol providing the one-way
   (at a time) setup of resource reservations for multicast and unicast
   flows.  Each reservation is set up in one direction (meaning one
   reservation from each end system; in a multicast environment, N
   senders set up N reservations).  These reservations complete a
   communication path with a deterministic bandwidth allocation through
   each router along that path between end systems.  These reservations
   set up a known quality of service for end-to-end communications and
   maintain a "soft-state" within a node.  The meaning of the term "soft
   state" is that in the event of a network outage or change of routing,
   these reservations are cleared without manual intervention, but must
   be periodically refreshed.  In RSVP, the refresh period is by default
   30 seconds, but may be as long as is appropriate.



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   RSVP is a locally-oriented process, not a globally- or domain-
   oriented one like a routing protocol or H.323 Call Counting.
   Although it uses the local routing databases to determine the routing
   path, it is only concerned with the quality of service for a
   particular or aggregate flow through a device.  RSVP is not aware of
   anything other than the local goal of QoS and its RSVP-enabled
   adjacencies, operating below the network layer.  The process by
   itself neither requires nor has any end-to-end network knowledge or
   state.  Thus, RSVP can be effective when it is enabled at some nodes
   in a network without the need to have every node participate.

                 HOST                              ROUTER
    _____________________________       ____________________________
   |  _______                    |     |                            |
   | |       |   _______         |     |            _______         |
   | |Appli- |  |       |        |RSVP |           |       |        |
   | | cation|  | RSVP <---------------------------> RSVP  <---------->
   | |       <-->       |        |     | _______   |       |        |
   | |       |  |process|  _____ |     ||Routing|  |process|  _____ |
   | |_._____|  |       -->Policy|     ||       <-->       -->Policy||
   |   |        |__.__._| |Cntrl||     ||process|  |__.__._| |Cntrl||
   |   |data       |  |   |_____||     ||__.____|     |  |   |_____||
   |===|===========|==|==========|     |===|==========|==|==========|
   |   |   --------|  |    _____ |     |   |  --------|  |    _____ |
   |   |  |        |  ---->Admis||     |   |  |       |  ---->Admis||
   |  _V__V_    ___V____  |Cntrl||     |  _V__V_    __V_____ |Cntrl||
   | |      |  |        | |_____||     | |      |  |        ||_____||
   | |Class-|  | Packet |        |     | |Class-|  | Packet |       |
   | | ifier|==>Schedulr|================> ifier|==>Schedulr|=========>
   | |______|  |________|        |data | |______|  |________|      data
   |                             |     |                            |
   |_____________________________|     |____________________________|

                    Figure 3: RSVP in Hosts and Routers

   Figure 3 shows the internal process of RSVP in both hosts (end
   systems) and routers, as shown in [RFC2209].

   RSVP uses the phrase "traffic control" to describe the mechanisms of
   how a data flow receives quality of service.  There are 3 different
   mechanisms to traffic control (shown in Figure 2 in both hosts and
   routers).  They are:

   A packet classifier mechanism: This resolves the QoS class for each
      packet; this can determine the route as well.






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   An admission control mechanism: This consists of two decision
      modules: admission control and policy control.  Determining
      whether there are satisfactory resources for the requested QoS is
      the function of admission control.  Determining whether the user
      has the authorization to request such resources is the function of
      policy control.  If the parameters carried within this flow fail,
      either of these two modules errors the request using RSVP.

   A packet scheduler mechanism:  At each outbound interface, the
      scheduler attains the guaranteed QoS for that flow.

2.3.2.  RSVP Scaling Issues

   As originally written, there was concern that RSVP had scaling
   limitations due to its data plane behavior [RFC2208].  This either
   has not proven to be the case or has in time largely been corrected.
   Telephony services generally require peak call admission rates on the
   order of thousands of calls per minute and peak call levels
   comparable to the capacities of the lines in question, which is
   generally on the order of thousands to tens of thousands of calls.
   Current RSVP implementations admit calls at the rate of hundreds of
   calls per second and maintain as many calls in progress as memory
   configurations allow.

   In edge networks, RSVP is used to signal for individual microflows,
   admitting the bandwidth.  However, Differentiated Services is used
   for the data plane behavior.  Admission and policing may be performed
   anywhere, but need only be performed in the first-hop router (which,
   if the end system sending the traffic is a DTE, constitutes a DCE for
   the remaining network) and in routers that have interfaces threatened
   by congestion.  In Figure 1, these would normally be the links that
   cross network boundaries.

2.3.3.  RSVP Operation in Backbones and Virtual Private Networks (VPNs)

   In backbone networks, networks that are normally awash in bandwidth,
   RSVP and its affected data flows may be carried in a variety of ways.
   If the backbone is a maze of tunnels between its edges (true of MPLS
   networks, networks that carry traffic from an encryptor to a
   decryptor, and also VPNs), applicable technologies include [RFC2207],
   [RFC2746], and [RFC2983].  An IP tunnel is, simplistically put, a IP
   packet enveloped inside another IP packet as a payload.  When IPv6 is
   transported over an IPv4 network, encapsulating the entire v6 packet
   inside a v4 packet is an effective means to accomplish this task.  In
   this type of tunnel, the IPv6 packet is not read by any of the
   routers while inside the IPv4 envelope.  If the inner packet is RSVP





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   enabled, there must be an active configuration to ensure that all
   relevant backbone nodes read the RSVP fields; [RFC2746] describes
   this.

   This is similar to how IPsec tunnels work.  Encapsulating an RSVP
   packet inside an encrypted packet for security purposes without
   copying or conveying the RSVP indicators in the outside IP packet
   header would make RSVP inoperable while in this form of a tunnel.
   [RFC2207] describes how to modify an IPsec packet header to allow for
   RSVP awareness by nodes that need to provide QoS for the flow or
   flows inside a tunnel.

   Other networks may simply choose to aggregate the reservations across
   themselves as described in [RFC3175].  The problem with an individual
   reservation architecture is that each flow requires a non-trivial
   amount of message exchange, computation, and memory resources in each
   router between each endpoint.  Aggregation of flows reduces the
   number of completely individual reservations into groups of
   individual flows that can act as one for part or all of the journey
   between end systems.  Aggregates are not intended to be from the
   first router to the last router within a flow, but to cover common
   paths of a large number of individual flows.

   Examples of aggregated data flows include streams of IP data that
   traverse common ingress and egress points in a network and also
   include tunnels of various kinds.  MPLS LSPs, IPsec Security
   Associations between VPN edge routers, IP/IP tunnels, and Generic
   Routing Encapsulation (GRE) tunnels all fall into this general
   category.  The distinguishing factor is that the system injecting an
   aggregate into the aggregated network sums the PATH and RESV
   statistical information on the un-aggregated side and produces a
   reservation for the tunnel on the aggregated side.  If the bandwidth
   for the tunnel cannot be expanded, RSVP leaves the existing
   reservation in place and returns an error to the aggregator, which
   can then apply a policy such as IEPS to determine which session to
   refuse.  In the data plane, the DSCP for the traffic must be copied
   from the inner to the outer header, to preserve the PHB's effect.

   One concern with this approach is that this leaks information into
   the aggregated zone concerning the number of active calls or the
   bandwidth they consume.  In fact, it does not, as the data itself is
   identifiable by aggregator address, deaggregator address, and DSCP.
   As such, even if it is not advertised, such information is
   measurable.







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2.3.4.  Interaction with the Differentiated Services Architecture

   In the PATH message, the DCLASS object described in [RFC2996] is used
   to carry the determined DSCP for the precedence level of that call in
   the stream.  This is reflected back in the RESV message.  The DSCP
   will be determined from the authorized SIP message exchange between
   end systems by using the R-P header.  The DCLASS object permits both
   bandwidth admission within a class and the building up of the various
   rates or token buckets.

2.3.5.  Admission Policy

   RSVP's basic admission policy, as defined, is to grant any user
   bandwidth if there is bandwidth available within the current
   configuration.  In other words, if a new request arrives and the
   difference between the configured upper bound and the currently
   reserved bandwidth is sufficiently large, RSVP grants use of that
   bandwidth.  This basic policy may be augmented in various ways, such
   as using a local or remote policy engine to apply AAA procedures and
   further qualify the reservation.

2.3.5.1.  Admission for Variable Rate Codecs

   For certain applications, such as broadcast video using MPEG-1 or
   voice without activity detection and using a constant bit rate codec
   such as G.711, this basic policy is adequate apart from AAA.  For
   variable rate codecs, such as MPEG-4 or a voice codec with Voice
   Activity Detection, however, this may be deemed too conservative.  In
   such cases, two basic types of statistical policy have been studied
   and reported on in the literature: simple over-provisioning, and
   approximation to ambient load.

   Simple over-provisioning sets the bandwidth admission limit higher
   than the desired load, on the assumption that a session that admits a
   certain bandwidth will in fact use a fraction of the bandwidth.  For
   example, if MPEG-4 data streams are known to use data rates between
   80 and 800 KBPS and there is no obvious reason that sessions would
   synchronize (such as having commercial breaks on 15 minute
   boundaries), one could imagine estimating that the average session
   consumes 400 KBPS and treating an admission of 800 KBPS as actually
   consuming half the amount.

   One can also approximate to average load, which is perhaps a more
   reliable procedure.  In this case, one maintains a variable that
   measures actual traffic through the admitted data's queue,
   approximating it using an exponentially weighted moving average.
   When a new reservation request arrives, if the requested rate is less
   than the difference between the configured upper bound and the



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   current value of the moving average, the reservation is accepted, and
   the moving average is immediately increased by the amount of the
   reservation to ensure that the bandwidth is not promised out to
   several users simultaneously.  In time, the moving average will decay
   from this guard position to an estimate of true load, which may offer
   a chance to another session to be reserved that would otherwise have
   been refused.

   Statistical reservation schemes such as these are overwhelmingly
   dependent on the correctness of their configuration and its
   appropriateness for the codecs in use.  However, they offer the
   opportunity to take advantage of statistical multiplexing gains that
   might otherwise be missed.

2.3.5.2.  Interaction with Complex Admission Policies, AAA, and
          Preemption of Bandwidth

   Policy is carried and applied as described in [RFC2753].  Figure 4,
   below, is the basic conceptual model for policy decisions and
   enforcement in an Integrated Services model.  This model was created
   to provide the ability to monitor and control reservation flows based
   on user identify, specific traffic and security requirements, and
   conditions that might change for various reasons, including a
   reaction to a disaster or emergency event involving the network or
   its users.

     Network Node       Policy server
    ______________
   |   ______     |
   |  |      |    |      _____
   |  | PEP  |    |     |     |------------->
   |  |______|<---|---->| PDP |May use LDAP,SNMP,COPS...for accessing
   |     ^        |     |     | policy database, authentication, etc.
   |     |        |     |_____|------------->
   |   __v___     |
   |  |      |    |     PDP = Policy Decision Point
   |  | LPDP |    |     PEP = Policy Enforcement Point
   |  |______|    |    LPDP = Local Policy Decision Point
   |______________|

         Figure 4: Conceptual Model for Policy Control of Routers

   The Network Node represents a router in the network.  The Policy
   Server represents the point of admission and policy control by the
   network operator.  Policy Enforcement Point (PEP) (the router) is
   where the policy action is carried out.  Policy decisions can be
   either locally present in the form of a Local Policy Decision Point
   (LPDP), or in a separate server on the network called the Policy



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   Decision Point.  The easier the instruction set of rules, the more
   likely this set can reside in the LPDP for speed of access reasons.
   The more complex the rule set, the more likely this is active on a
   remote server.  The PDP will use other protocols (LDAP, SNMP, etc.)
   to request information (e.g., user authentication and authorization
   for precedence level usage) to be used in creating the rule sets of
   network components.  This remote PDP should also be considered where
   non-reactive policies are distributed out to the LPDPs.

   Taking the above model as a framework, [RFC2750] extends RSVP's
   concept of a simple reservation to include policy controls, including
   the concepts of Preemption [RFC3181] and Identity [RFC3182],
   specifically speaking to the usage of policies that preempt calls
   under the control of either a local or remote policy manager.  The
   policy manager assigns a precedence level to the admitted data flow.
   If it admits a data flow that exceeds the available capacity of a
   system, the expectation is that the RSVP-affected RSVP process will
   tear down a session among the lowest precedence sessions it has
   admitted.  The RESV Error resulting from that will go to the receiver
   of the data flow and be reported to the application (SIP or H.323).
   That application is responsible for disconnecting its call, with a
   reason code of "bandwidth preemption".

2.4.  Authentication and Authorization of Calls Placed

   It will be necessary, of course, to ensure that any policy is applied
   to an authenticated user; the capabilities assigned to an
   authenticated user may be considered authorized for use in the
   network.  For bandwidth admission, this will require the utilization
   of [RFC2747] [RFC3097].  In SIP and H.323, AAA procedures will also
   be needed.

2.5.  Defined User Interface

   The user interface -- the chimes and tones heard by the user --
   should ideally remain the same as in the PSTN for those indications
   that are still applicable to an IP network.  There should be some new
   effort generated to update the list of announcements sent to the user
   that don't necessarily apply.  All indications to the user, of
   course, depend on positive signals, not unreliable measures based on
   changing measurements.










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3.  Security Considerations

   This document outlines a networking capability composed entirely of
   existing specifications.  It has significant security issues, in the
   sense that a failure of the various authentication or authorization
   procedures can cause a fundamental breakdown in communications.
   However, the issues are internal to the various component protocols
   and are covered by their various security procedures.

4.  Acknowledgements

   This document was developed with the knowledge and input of many
   people, far too numerous to be mentioned by name.  However, key
   contributors of thoughts include Francois Le Faucheur, Haluk
   Keskiner, Rohan Mahy, Scott Bradner, Scott Morrison, Subha Dhesikan,
   and Tony De Simone.  Pete Babendreier, Ken Carlberg, and Mike Pierce
   provided useful reviews.


































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5.  References

5.1.  Normative References

   [RFC3689]         Carlberg, K. and R. Atkinson, "General Requirements
                     for Emergency Telecommunication Service (ETS)", RFC
                     3689, February 2004.

   [RFC3690]         Carlberg, K. and R. Atkinson, "IP Telephony
                     Requirements for Emergency Telecommunication
                     Service (ETS)", RFC 3690, February 2004.

   Integrated Services Architecture References

   [RFC1633]         Braden, B., Clark, D., and S. Shenker, "Integrated
                     Services in the Internet Architecture: an
                     Overview", RFC 1633, June 1994.

   [RFC2205]         Braden, B., Zhang, L., Berson, S., Herzog, S., and
                     S.  Jamin, "Resource ReSerVation Protocol (RSVP) --
                     Version 1 Functional Specification", RFC 2205,
                     September 1997.

   [RFC2207]         Berger, L. and T. O'Malley, "RSVP Extensions for
                     IPSEC Data Flows", RFC 2207, September 1997.

   [RFC2208]         Mankin, A., Baker, F., Braden, B., Bradner, S.,
                     O'Dell, M., Romanow, A., Weinrib, A., and L. Zhang,
                     "Resource ReSerVation Protocol (RSVP) Version 1
                     Applicability Statement Some Guidelines on
                     Deployment", RFC 2208, September 1997.

   [RFC2209]         Braden, B. and L. Zhang, "Resource ReSerVation
                     Protocol (RSVP) -- Version 1 Message Processing
                     Rules", RFC 2209, September 1997.

   [RFC2746]         Terzis, A., Krawczyk, J., Wroclawski, J., and L.
                     Zhang, "RSVP Operation Over IP Tunnels", RFC 2746,
                     January 2000.

   [RFC2747]         Baker, F., Lindell, B., and M. Talwar, "RSVP
                     Cryptographic Authentication", RFC 2747, January
                     2000.

   [RFC2750]         Herzog, S., "RSVP Extensions for Policy Control",
                     RFC 2750, January 2000.





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   [RFC2753]         Yavatkar, R., Pendarakis, D., and R. Guerin, "A
                     Framework for Policy-based Admission Control", RFC
                     2753, January 2000.

   [RFC2996]         Bernet, Y., "Format of the RSVP DCLASS Object", RFC
                     2996, November 2000.

   [RFC2998]         Bernet, Y., Ford, P., Yavatkar, R., Baker, F.,
                     Zhang, L., Speer, M., Braden, R., Davie, B.,
                     Wroclawski, J., and E.  Felstaine, "A Framework for
                     Integrated Services Operation over Diffserv
                     Networks", RFC 2998, November 2000.

   [RFC3097]         Braden, R. and L. Zhang, "RSVP Cryptographic
                     Authentication -- Updated Message Type Value", RFC
                     3097, April 2001.

   [RFC3175]         Baker, F., Iturralde, C., Le Faucheur, F., and B.
                     Davie, "Aggregation of RSVP for IPv4 and IPv6
                     Reservations", RFC 3175, September 2001.

   [RFC3181]         Herzog, S., "Signaled Preemption Priority Policy
                     Element", RFC 3181, October 2001.

   [RFC3182]         Yadav, S., Yavatkar, R., Pabbati, R., Ford, P.,
                     Moore, T., Herzog, S., and R. Hess, "Identity
                     Representation for RSVP", RFC 3182, October 2001.

   [RFC3312]         Camarillo, G., Marshall, W., and J. Rosenberg,
                     "Integration of Resource Management and Session
                     Initiation Protocol (SIP)", RFC 3312, October 2002.

   Differentiated Services Architecture References

   [RFC2474]         Nichols, K., Blake, S., Baker, F., and D. Black,
                     "Definition of the Differentiated Services Field
                     (DS Field) in the IPv4 and IPv6 Headers", RFC 2474,
                     December 1998.

   [RFC2475]         Blake, S., Black, D., Carlson, M., Davies, E.,
                     Wang, Z., and W. Weiss, "An Architecture for
                     Differentiated Services", RFC 2475, December 1998.

   [RFC2983]         Black, D., "Differentiated Services and Tunnels",
                     RFC 2983, October 2000.






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   [RFC3246]         Davie, B., Charny, A., Bennet, J., Benson, K., Le
                     Boudec, J., Courtney, W., Davari, S., Firoiu, V.,
                     and D.  Stiliadis, "An Expedited Forwarding PHB
                     (Per-Hop Behavior)", RFC 3246, March 2002.

   [RFC3247]         Charny, A., Bennet, J., Benson, K., Boudec, J.,
                     Chiu, A., Courtney, W., Davari, S., Firoiu, V.,
                     Kalmanek, C., and K.  Ramakrishnan, "Supplemental
                     Information for the New Definition of the EF PHB
                     (Expedited Forwarding Per-Hop Behavior)", RFC 3247,
                     March 2002.

   Session Initiation Protocol and Related References

   [RFC2327]         Handley, M. and V. Jacobson, "SDP: Session
                     Description Protocol", RFC 2327, April 1998.

   [RFC3261]         Rosenberg, J., Schulzrinne, H., Camarillo, G.,
                     Johnston, A., Peterson, J., Sparks, R., Handley,
                     M., and E.  Schooler, "SIP: Session Initiation
                     Protocol", RFC 3261, June 2002.

   [RFC4411]         Polk, J., "Extending the Session Initiation
                     Protocol (SIP) Reason Header for Preemption
                     Events", RFC 4411, February 2006.

   [RFC4412]         Schulzrinne, H. and J. Polk, "Communications
                     Resource Priority for the Session Initiation
                     Protocol (SIP)", RFC 4412, February 2006.

5.2.  Informative References

   [ANSI.MLPP.Spec]  American National Standards Institute,
                     "Telecommunications - Integrated Services Digital
                     Network (ISDN) - Multi-Level Precedence and
                     Preemption (MLPP) Service Capability", ANSI
                     T1.619-1992 (R1999), 1992.

   [ANSI.MLPP.Supp]  American National Standards Institute, "MLPP
                     Service Domain Cause Value Changes", ANSI ANSI
                     T1.619a-1994 (R1999), 1990.

   [G711.1]          Viola Networks, "Netally VoIP Evaluator", January
                     2003, <http://www.brainworks.de/Site/hersteller/
                     viola_networks/Dokumente/Compr_Report_Sample.pdf>.






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   [G711.3]          Nortel Networks, "Packet Loss and Packet Loss
                     Concealment", 2000, <http://www.nortelnetworks.com/
                     products/01/succession/es/collateral/
                     tb_pktloss.pdf>.

   [ITU.ETS.E106]    International Telecommunications Union,
                     "International Emergency Preference Scheme for
                     disaster relief operations (IEPS)", ITU-T
                     Recommendation E.106, October 2003.

   [ITU.MLPP.1990]   International Telecommunications Union, "Multilevel
                     Precedence and Preemption Service (MLPP)", ITU-T
                     Recommendation I.255.3, 1990.

   [Parekh1]         Parekh, A. and R. Gallager, "A Generalized
                     Processor Sharing Approach to Flow Control in
                     Integrated Services Networks: The Multiple Node
                     Case", INFOCOM 1993: 521-530, 1993.

   [Parekh2]         Parekh, A. and R. Gallager, "A Generalized
                     Processor Sharing Approach to Flow Control in
                     Integrated Services Networks: The Single Node
                     Case", INFOCOM 1992: 915-924, 1992.




























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Appendix A.  2-Call Preemption Example Using RSVP

   This appendix will present a more complete view of the interaction
   among SIP, SDP, and RSVP.  The bulk of the material is referenced
   from [RFC2327], [RFC3312], [RFC4411], and [RFC4412].  There will be
   some discussion on basic RSVP operations regarding reservation paths;
   this will be mostly from [RFC2205].

   SIP signaling occurs at the Application Layer, riding on a UDP/IP or
   TCP/IP (including TLS/TCP/IP) transport that is bound by routing
   protocols such as BGP and OSPF to determine the route the packets
   traverse through a network between source and destination devices.
   RSVP is riding on top of IP as well, which means RSVP is at the mercy
   of the IP routing protocols to determine a path through the network
   between endpoints.  RSVP is not a routing protocol.  In this
   appendix, there will be an escalation of building blocks getting to
   how the many layers are involved in SIP.  QoS Preconditions require
   successful RSVP signaling between endpoints prior to SIP successfully
   acknowledging the setup of the session (for voice, video, or both).
   Then we will present what occurs when a network overload occurs
   (congestion), causing a SIP session to be preempted.

   Three diagrams in this appendix show multiple views of the same
   example of connectivity for discussion throughout this appendix.  The
   first diagram (Figure 5) is of many routers between many endpoints
   (SIP user agents, or UAs).  There are 4 UAs of interest; those are
   for users Alice, Bob, Carol, and Dave.  When a user (the human) of a
   UA gets involved and must do something to a UA to progress a SIP
   process, this will be explicitly mentioned to avoid confusion;
   otherwise, when Alice is referred to, it means Alice's UA (her
   phone).

   RSVP reserves bandwidth in one direction only (the direction of the
   RESV message), as has been discussed, IP forwarding of packets are
   dictated by the routing protocol for that portion of the
   infrastructure from the point of view of where the packet is to go
   next.

   The RESV message traverses the routers in the reverse path taken by
   the PATH message.  The PATH message establishes a record of the route
   taken through a network portion to the destination endpoint, but it
   does not reserve resources (bandwidth).  The RESV message back to the
   original requester of the RSVP flow requests for the bandwidth
   resources.  This means the endpoint that initiates the RESV message
   controls the parameters of the reservation.  This document specifies
   in the body text that the SIP initiator (the UAC) establishes the
   parameters of the session in an INVITE message, and that the INVITE
   recipient (the UAS) must follow the parameters established in that



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   INVITE message.  One exception to this is which codec to use if the
   UAC offered more than one to the UAS.  This exception will be shown
   when the INVITE message is discussed in detail later in the appendix.
   If there was only one codec in the SDP of the INVITE message, the
   parameters of the reservation will follow what the UAC requested
   (specifically to include the Resource-Priority header namespace and
   priority value).

   Here is the first figure with the 4 UAs and a meshed routed
   infrastructure between each.  For simplicity of this explanation,
   this appendix will only discuss the reservations from Alice to Bob
   (one direction) and from Carol to Dave (one direction).  An
   interactive voice service will require two one-way reservations that
   end in each UA.  This gives the appearance of a two-way reservation,
   when indeed it is not.

           Alice -----R1----R2----R3----R4------ Bob
                      | \  /  \  /  \  / |
                      |  \/    \/    \/  |
                      |  /\    /\    /\  |
                      | /  \  /  \  /  \ |
           Carol -----R5----R6----R7----R8------ Dave

            Figure 5: Complex Routing and Reservation Topology

   The PATH message from Alice to Bob (establishing the route for the
   RESV message) will be through routers:

      Alice -> R1 -> R2 -> R3 -> R4 -> Bob

   The RESV message (and therefore the reservation of resources) from
   Bob to Alice will be through routers:

      Bob -> R4 -> R3 -> R2 -> R1 -> Alice

   The PATH message from Carol to Dave (establishing the route for the
   RESV message) will be through routers:

      Carol -> R5 -> R2 -> R3 -> R8 -> Dave

   The RESV message (and therefore the reservation of resources) from
   Dave to Carol will be through routers:

      Dave -> R8 -> R3 -> R2 -> R5 -> Carol

   The reservations from Alice to Bob traverse a common router link:
   between R3 and R2 and thus a common interface at R2.  Here is where
   there will be congestion in this example, on the link between R2 and



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   R3.  Since the flow of data (in this case voice media packets)
   travels the direction of the PATH message, and RSVP establishes
   reservation of resources at the egress interface of a router, the
   interface in Figure 6 shows that Int7 will be what first knows about
   a congestion condition.

             Alice                               Bob
                \                                /
                 \                              /
                  +--------+          +--------+
                  |        |          |        |
                  |   R2   |          |   R3   |
                  |       Int7-------Int5      |
                  |        |          |        |
                  +--------+          +--------+
                 /                              \
                /                                \
            Carol                                Dave

                  Figure 6: Reduced Reservation Topology

   Figure 6 illustrates how the messaging between the UAs and the RSVP
   messages between the relevant routers can be shown to understand the
   binding that was established in [RFC3312] (more suitably titled "SIP
   Preconditions for QoS" from this document's point of view).

   We will assume all devices have powered up and received whatever
   registration or remote policy downloads were necessary for proper
   operation.  The routing protocol of choice has performed its routing
   table update throughout this part of the network.  Now we are left to
   focus only on end-to-end communications and how that affects the
   infrastructure between endpoints.

   The next diagram (Figure 7) (nearly identical to Figure 1 from
   [RFC3312]) shows the minimum SIP messaging (at layer 7) between Alice
   and Bob for a good-quality voice call.  The SIP messages are numbered
   to identify special qualities of each.  During the SIP signaling,
   RSVP will be initiated.  That messaging will also be discussed below.













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      UA Alice                                      UA Bob
          |                                            |
          |                                            |
          |-------------(1) INVITE SDP1--------------->|
          |                                            |   Note 1
          |<------(2) 183 Session Progress SDP2--------|     |
       ***|********************************************|***<-+
       *  |----------------(3) PRACK------------------>|  *
       *  |                                            |  * Where
       *  |<-----------(4) 200 OK (PRACK)--------------|  * RSVP
       *  |                                            |  * is
       *  |                                            |  * signaled
       ***|********************************************|***
          |-------------(5) UPDATE SDP3--------------->|
          |                                            |
          |<--------(6) 200 OK (UPDATE) SDP4-----------|
          |                                            |
          |<-------------(7) 180 Ringing---------------|
          |                                            |
          |-----------------(8) PRACK----------------->|
          |                                            |
          |<------------(9) 200 OK (PRACK)-------------|
          |                                            |
          |                                            |
          |<-----------(10) 200 OK (INVITE)------------|
          |                                            |
          |------------------(11) ACK----------------->|
          |                                            |
          |         RTP (within the reservation)       |
          |<==========================================>|
          |                                            |

        Figure 7: SIP Reservation Establishment Using Preconditions

   The session initiation starts with Alice wanting to communicate with
   Bob.  Alice decides on an IEPS precedence level for their call (the
   default is the "routine" level, which is for normal everyday calls,
   but a priority level has to be chosen for each call).  Alice puts
   into her UA Bob's address and precedence level and (effectively) hits
   the send button.  This is reflected in SIP with an INVITE Method
   Request message [M1].  Below is what SIP folks call a well-formed SIP
   message (meaning it has all the headers that are mandatory to
   function properly).  We will pick on the US Marine Corps (USMC) for
   the addressing of this message exchange.







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      [M1 - INVITE from Alice to Bob, RP=Routine, QOS=e2e and mandatory]
      INVITE sip:bob@usmc.example.mil SIP/2.0
      Via: SIP/2.0/TCP pc33.usmc.example.mil:5060
        ;branch=z9hG4bK74bf9
      Max-Forwards: 70
      From: Alice <sip:alice@usmc.example.mil>;tag=9fxced76sl
      To: Bob <sip:bob@usmc.example.mil>
      Call-ID: 3848276298220188511@pc33.usmc.example.mil
      CSeq: 31862 INVITE
      Require: 100rel, preconditions, resource-priority
      Resource-Priority: dsn.routine
      Contact: <sip:alice@usmc.example.mil>
      Content-Type: application/sdp
      Content-Length: 191

      v=0
      o=alice 2890844526 2890844526 IN IP4 usmc.example.mil
      c=IN IP4 10.1.3.33
      t=0 0
      m=audio 49172 RTP/AVP 0 4 8
      a=rtpmap:0 PCMU/8000
      a=curr:qos e2e none
      a=des:qos mandatory e2e sendrecv

   From the INVITE above, Alice is inviting Bob to a session.  The upper
   half of the lines (above the line "v=0") is SIP headers and header
   values, and the lower half is Session Description Protocol (SDP)
   lines.  SIP headers (after the first line, called the Status line)
   are not mandated in any particular order, with one exception: the Via
   header.  It is a SIP hop (through a SIP Proxy) route path that has a
   new Via header line added by each SIP element this message traverses
   towards the destination UA.  This is similar in function to an RSVP
   PATH message (building a reverse path back to the originator of the
   message).  At any point in the message's path, a SIP element knows
   the path to the originator of the message.  There will be no SIP
   Proxies in this example, because for Preconditions, Proxies only make
   more messages that look identical (with the exception of the Via and
   Max-Forwards headers), and it is not worth the space here to
   replicate what has been done in SIP RFCs already.












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   SIP headers that are used for Preconditions are as follows:

   o  Require header, which contains 3 option tags: "100rel" mandates a
      reliable provisional response message to the conditions requesting
      in this INVITE (knowing they are special), "preconditions"
      mandates that preconditions are attempted, and "resource-priority"
      mandates support for the Resource-Priority header.  Each of these
      option tags can be explicitly identified in a message failure
      indication from the called UA to tell the calling UA exactly what
      was not supported.

      Provided that this INVITE message is received as acceptable, this
      will result in the 183 "Session Progress" message from Bob's UA, a
      reliable confirmation that preconditions are required for this
      call.

   o  Resource-Priority header, which denotes the domain namespace and
      precedence level of the call on an end-to-end basis.

   This completes SIP's functions in session initiation.  Preconditions
   are requested, required, and signaled for in the SDP portion of the
   message.  SDP is carried in what's called a SIP message body (much
   like the text in an email message is carried).  SDP has special
   properties (see [RFC2327] for more on SDP, or the MMUSIC WG for
   ongoing efforts regarding SDP).  SDP lines are in a specific order
   for parsing by end systems.  Dialog-generating (or call-generating)
   SDP message bodies all must have an "m=" line (or media description
   line).  Following the "m=" line are zero or more "a=" lines (or
   Attribute lines).  The "m=" line in Alice's INVITE calls for a voice
   session (this is where video is identified also) using one of 3
   different codecs that Alice supports (0 = G.711, 4 = G.723, and 18 =
   G.729) that Bob gets to choose from for this session.  Bob can choose
   any of the 3.  The first a=rtpmap line is specific to the type of
   codec these 3 are (PCMU).  The next two "a=" lines are the only
   identifiers that RSVP is to be used for this call.  The second "a="
   line:

      a=curr:qos e2e none

   identifies the "current" status of qos at Alice's UA.  Note:
   everything in SDP is with respect to the sender of the SDP message
   body (Alice will never tell Bob how his SDP is; she will only tell
   Bob about her SDP).

      "e2e" means that capacity assurance is required from Alice's UA to
      Bob's UA; thus, a lack of available capacity assurance in either
      direction will fail the call attempt.




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      "none" means there is no reservation at Alice's UA (to Bob) at
      this time.

   The final "a=" line (a=des) identifies the "desired" level of qos:

      a=des:qos mandatory e2e sendrecv

      "mandatory" means this request for qos MUST be successful, or the
      call fails.

      "e2e" means RSVP is required from Alice's UA to Bob's UA.

      "sendrecv" means the reservation is in both directions.

   As discussed, RSVP does not reserve bandwidth in both directions, and
   it is up to the endpoints to have 2 one-way reservations if that
   particular application (here, voice) requires it.  Voice between
   Alice and Bob requires 2 one-way reservations.  The UAs will be the
   focal points for both reservations in both directions.

   Message 2 is the 183 "Session Progress" message sent by Bob to Alice,
   which indicates to Alice that Bob understands that preconditions are
   required for this call.

      [M2 - 183 "Session Progress"]
      SIP/2.0 183 Session Progress
      Via: SIP/2.0/TCP pc33.usmc.example.mil:5060
        ;branch=z9hG4bK74bf9 ;received=10.1.3.33
      From: Alice <sip:alice@usmc.example.mil>;tag=9fxced76sl
      To: Bob <sip:bob@usmc.example.mil>;tag=8321234356
      Call-ID: 3848276298220188511@pc33.usmc.example.mil
      CSeq: 31862 INVITE
      RSeq: 813520
      Resource-Priority: dsn.routine
      Contact: <sip:bob@usmc.example.mil>
      Content-Type: application/sdp
      Content-Length: 210

      v=0
      o=bob 2890844527 2890844527 IN IP4 usmc.example.mil
      c=IN IP4 10.100.50.51
      t=0 0
      m=audio 3456 RTP/AVP 0
      a=rtpmap:0 PCMU/8000
      a=curr:qos e2e none
      a=des:qos mandatory e2e sendrecv
      a=conf:qos e2e recv




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   The only interesting header in the SIP portion of this message is the
   RSeq header, which is the "Reliable Sequence" header.  The value is
   incremented for every Reliable message that's sent in this call setup
   (to make sure none are lost or to ignore duplicates).

   Bob's SDP indicates several "a=" line statuses and picks a codec for
   the call.  The codec picked is in the m=audio line (the "0" at the
   end of this line means G.711 will be the codec).

   The a=curr line gives Alice Bob's status with regard to RSVP
   (currently "none").

   The a=des line also states the desire for mandatory qos e2e in both
   directions.

   The a=conf line is new.  This line means Bob wants confirmation that
   Alice has 2 one-way reservations before Bob's UA proceeds with the
   SIP session setup.

   This is where "Note-1" applies in Figure 7.  At the point that Bob's
   UA transmits this 183 message, Bob's UA (the one that picked the
   codec, so it knows the amount of bandwidth to reserve) transmits an
   RSVP PATH message to Alice's UA.  This PATH message will take the
   route previously discussed in Figure 5:

      Bob -> R4 -> R3 -> R2 -> R1 -> Alice

   This is the path of the PATH message, and the reverse will be the
   path of the reservation setup RESV message, or:

      Alice -> R1 -> R2 -> R3 -> R4 -> Bob

   Immediately after Alice transmits the RESV message towards Bob, Alice
   sends her own PATH message to initiate the other one-way reservation.
   Bob, receiving that PATH message, will reply with a RESV.

   All this is independent of SIP.  However, during this time of
   reservation establishment, a Provisional Acknowledgement (PRACK) [M3]
   is sent from Alice to Bob to confirm the request for confirmation of
   2 one-way reservations at Alice's UA.  This message is acknowledged
   with a normal 200 OK message [M4].  This is shown in Figure 7.

   As soon as the RSVP is successfully completed at Alice's UA (knowing
   that it was the last in the two-way cycle or reservation
   establishment), at the SIP layer an UPDATE message [M5] is sent to
   Bob's UA to inform his UA that the current status of RSVP (or qos) is
   "e2e" and "sendrecv".




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      [M5 - UPDATE to Bob that Alice has qos e2e and sendrecv]
      UPDATE sip:bob@usmc.example.mil SIP/2.0
      Via: SIP/2.0/TCP pc33.usmc.example.mil:5060
        ;branch=z9hG4bK74bfa
      From: Alice <sip:alice@usmc.example.mil>;tag=9fxced76sl
      To: Bob <sip:bob@usmc.example.mil>
      Call-ID: 3848276298220188511@pc33.usmc.example.mil
      Resource-Priority: dsn.routine
      Contact: <sip:alice@usmc.example.mil>
      CSeq: 10197 UPDATE
      Content-Type: application/sdp
      Content-Length: 191

      v=0
      o=alice 2890844528 2890844528 IN IP4 usmc.example.mil
      c=IN IP4 10.1.3.33
      t=0 0
      m=audio 49172 RTP/AVP 0
      a=rtpmap:0 PCMU/8000
      a=curr:qos e2e send
      a=des:qos mandatory e2e sendrecv

   This is shown by the matching table that can be built from the a=curr
   line and a=des line.  If the two lines match, then no further
   signaling needs take place with regard to "qos".  [M6] is the 200 OK
   acknowledgement of this synchronization between the two UAs.

      [M6 - 200 OK to the UPDATE from Bob indicating synchronization]
      SIP/2.0 200 OK sip:bob@usmc.example.mil
      Via: SIP/2.0/TCP pc33.usmc.example.mil:5060
        ;branch=z9hG4bK74bfa
      From: Alice <sip:alice@usmc.example.mil>;tag=9fxced76sl
      To: Bob <sip:bob@usmc.example.mil>
      Call-ID: 3848276298220188511@pc33.usmc.example.mil
      Resource-Priority: dsn.routine
      Contact: < sip:alice@usmc.example.mil >
      CSeq: 10197 UPDATE
      Content-Type: application/sdp
      Content-Length: 195

      v=0
      o=alice 2890844529 2890844529 IN IP4 usmc.example.mil
      c=IN IP4 10.1.3.33
      t=0 0
      m=audio 49172 RTP/AVP 0
      a=rtpmap:0 PCMU/8000
      a=curr:qos e2e sendrecv
      a=des:qos mandatory e2e sendrecv



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   At this point, the reservation is operational and both UAs know it.
   Bob's UA now rings, telling Bob the user that Alice is calling him.
   ([M7] is the SIP indication to Alice that this is taking place).
   Nothing up until now has involved Bob the user.  Bob picks up the
   phone (generating [M10], from which Alice's UA responds with the
   final ACK), and RTP is now operating within the reservations between
   the two UAs.

   Now we get to Carol calling Dave.  Figure 6 shows a common router
   interface for the reservation between Alice to Bob, and one that will
   also be the route for one of the reservations between Carol to Dave.
   This interface will experience congestion in our example.

   Carol is now calling Dave at a Resource-Priority level of
   "Immediate", which is higher in priority than Alice to Bob's
   "routine".  In this continuing example, Router 2's Interface-7 is
   congested and cannot accept any more RSVP traffic.  Perhaps the
   offered load is at interface capacity.  Perhaps Interface-7 is
   configured with a fixed amount of bandwidth it can allocate for RSVP
   traffic, and it has reached its maximum without one of the
   reservations going away through normal termination or forced
   termination (preemption).

   Interface-7 is not so full of offered load that it cannot transmit
   signaling packets, such as Carol's SIP messaging to set up a call to
   Dave.  This should be by design (that not all RSVP traffic can starve
   an interface from signaling packets).  Carol sends her own INVITE
   with the following important characteristics:

   [M1 - INVITE from Carol to Dave, RP=Immediate, QOS=e2e and mandatory]

   This packet does *not* affect the reservations between Alice and Bob
   (SIP and RSVP are at different layers, and all routers are passing
   signaling packets without problems).  Dave sends his M2:

   [M2 - 183 "Session Progress"]

   with the SDP chart of:

      a=curr:qos e2e none

      a=des:qos mandatory e2e sendrecv

      a=conf:qos e2e recv







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   indicating he understands RSVP reservations are required e2e for this
   call to be considered successful.  Dave sends his PATH message.  The
   PATH message does *not* affect Alice's reservation; it merely
   establishes a path for the RESV reservation setup message to take.

   To keep this example simple, the PATH message from Dave to Carol took
   this route (which we make different from the route in the reverse
   direction):

      Dave -> R8 -> R7 -> R6 -> R5 -> Carol

   causing the reservation to be this route:

      Carol -> R5 -> R6 -> R7 -> R8 -> Dave

   The Carol-to-Dave reservation above will not traverse any of the same
   routers as the Alice-to-Bob reservation.  When Carol transmits her
   RESV message towards Dave, she immediately transmits her PATH message
   to set up the complementary reservation.

   The PATH message from Carol to Dave be through routers:

      Carol -> R5 -> R2 -> R3 -> R8 -> Dave

   Thus, the RESV message will be through routers:

      Dave -> R8 -> R3 -> R2 -> R5 -> Carol

   This RESV message will traverse the same routers, R3 and R2, as the
   Alice-to-Bob reservation.  This RESV message, when received at
   Interface-7 of R2, will create a congestion situation such that R2
   will need to make a decision on whether:

   o  to keep the Alice-to-Bob reservation and error the new RESV from
      Dave, or

   o  to error the reservation from Alice to Bob in order to make room
      for the Carol-to-Dave reservation.

   Alice's reservation was set up in SIP at the "routine" precedence
   level.  This will equate to a comparable RSVP priority number (RSVP
   has 65,535 priority values, or 2*32 bits per [RFC3181]).  Dave's RESV
   equates to a precedence value of "immediate", which is a higher
   priority.  Thus, R2 will preempt the reservation from Alice to Bob
   and allow the reservation request from Dave to Carol.  The proper
   RSVP error is the ResvErr that indicates preemption.  This message
   travels downstream towards the originator of the RESV message (Bob).
   This clears the reservation in all routers downstream of R2 (meaning



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   R3 and R4).  Once Bob receives the ResvErr message indicating
   preemption has occurred on this reservation, Bob's UA transmits a SIP
   preemption indication back towards Alice's UA.  This accomplishes two
   things: first, it informs all SIP Servers that were in the session
   setup path that wanted to remain "dialog stateful" per [RFC3261], and
   second, it informs Alice's UA that this was a purposeful termination,
   and to play a preemption tone.  The proper indication in SIP of this
   termination due to preemption is a BYE Method message that includes a
   Reason Header indicating why this occurred (in this case, "Reserved
   Resources Preempted").  Here is the message from Bob to Alice that
   terminates the call in SIP.

      BYE sip:alice@usmc.example.mil SIP/2.0
      Via: SIP/2.0/TCP swp34.usmc.example.mil
        ;branch=z9hG4bK776asegma
      To: Alice <sip:alice@usmc.example.mil>
      From: Bob <sip:bob@usmc.example.mil>;tag=192820774
      Reason: preemption ;cause=2 ;text=reserved resourced preempted
      Call-ID: 3848276298220188511@pc33.usmc.example.mil
      CSeq: 6187 BYE
      Contact: <sip:bob@usmc.example.mil>

   When Alice's UA receives this message, her UA terminates the call,
   sends a 200 OK to Bob to confirm reception of the BYE message, and
   plays a preemption tone to Alice the user.

   The RESV message from Dave successfully traverses R2, and Carol's UA
   receives it.  Just as with the Alice-to-Bob call setup, Carol sends
   an UPDATE message to Dave, confirming she has QoS "e2e" in "sendrecv"
   directions.  Bob acknowledges this with a 200 OK that gives his
   current status (QoS "e2e" and "sendrecv"), and the call setup in SIP
   continues to completion.

   In summary, Alice set up a call to Bob with RSVP at a priority level
   of Routine.  When Carol called Dave at a high priority, their call
   would have preempted any lower priority calls if there were a
   contention for resources.  In this case, it occurred and affected the
   call between Alice and Bob.  A router at this congestion point
   preempted Alice's call to Bob in order to place the higher-priority
   call between Carol and Dave.  Alice and Bob were both informed of the
   preemption event.  Both Alice and Bob's UAs played preemption
   indications.  What was not mentioned in this appendix was that this
   document RECOMMENDS that router R2 (in this example) generate a
   syslog message to the domain administrator to properly manage and
   track such events within this domain.  This will ensure that the
   domain administrators have recorded knowledge of where such events
   occur, and what the conditions were that caused them.




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RFC 4542                  ETS in an IP Network                  May 2006


Authors' Addresses

   Fred Baker
   Cisco Systems
   1121 Via Del Rey
   Santa Barbara, California  93117
   USA

   Phone: +1-408-526-4257
   Fax:   +1-413-473-2403
   EMail: fred@cisco.com


   James Polk
   Cisco Systems
   2200 East President George Bush Turnpike
   Richardson, Texas  75082
   USA

   Phone: +1-817-271-3552
   EMail: jmpolk@cisco.com






























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RFC 4542                  ETS in an IP Network                  May 2006


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