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Versions: (draft-baker-tsvwg-mlpp-that-works) 00 01 02 03 04 RFC 4542

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Transport Working Group                                         F. Baker
Internet-Draft                                                   J. Polk
Expires: August 11, 2005                                   Cisco Systems
                                                        February 7, 2005


  Implementing MLPP for Voice and Video in the Internet Protocol Suite
                  draft-ietf-tsvwg-mlpp-that-works-00

Status of this Memo

   This document is an Internet-Draft and is subject to all provisions
   of Section 3 of RFC 3667.  By submitting this Internet-Draft, each
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   This Internet-Draft will expire on August 11, 2005.

Copyright Notice

   Copyright (C) The Internet Society (2005).

Abstract

   The Defense Information Systems Agency of the United States
   Department of Defense, with its contractors, has proposed a service
   architecture for military (NATO and related agencies) telephone
   systems.  This is called the Assured Service, and is defined in two
   documents: "Architecture for Assured Service Capabilities in Voice
   over IP" and "Requirements for Assured Service Capabilities in Voice



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   over IP".  Responding to these are two documents: "Extending the
   Session Initiation Protocol Reason Header to account for Preemption
   Events", "Communications Resource Priority for the Session Initiation
   Protocol".

   What remains to this specification is to provide a Call Admission
   Control procedure and a Per Hop Behavior for the data which 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 that is permitted initially and is not
   denied or rejected at a later time, call and capacity admission
   procedures performed only at the time of call setup may be
   sufficient.  However in a network where sessions‚ÇÖ 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
   supercede existing calls), maintaining a continuing model of the
   status of the various calls is required.




























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Table of Contents

   1.  Overview . . . . . . . . . . . . . . . . . . . . . . . . . . .  4
     1.1   Multi-Level Preemption and Precedence  . . . . . . . . . .  4
     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 . . . .  8
     1.6   The use of bandwidth as a solution for QoS . . . . . . . .  9

   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   Recommended procedure: explicit call admission -
               RSVP Admission using Policy  . . . . . . . . . . . . . 17
       2.3.2   RSVP Scaling Issues  . . . . . . . . . . . . . . . . . 19
       2.3.3   RSVP Operation in backbones and VPNs . . . . . . . . . 19
       2.3.4   Interaction with the Differentiated Services
               Architecture . . . . . . . . . . . . . . . . . . . . . 20
       2.3.5   Admission policy . . . . . . . . . . . . . . . . . . . 20
         2.3.5.1   Admission for variable rate codecs . . . . . . . . 21
         2.3.5.2   Interaction with complex admission policies,
                   AAA, and preemption of bandwidth . . . . . . . . . 22
     2.4   Authentication and authorization of calls placed . . . . . 23
     2.5   Defined User Interface . . . . . . . . . . . . . . . . . . 23

   3.  IANA Considerations  . . . . . . . . . . . . . . . . . . . . . 24

   4.  Security Considerations  . . . . . . . . . . . . . . . . . . . 25

   5.  Acknowledgements . . . . . . . . . . . . . . . . . . . . . . . 26

   6.  References . . . . . . . . . . . . . . . . . . . . . . . . . . 27
     6.1   Normative References . . . . . . . . . . . . . . . . . . . 27
     6.2   Informative References . . . . . . . . . . . . . . . . . . 29

       Authors' Addresses . . . . . . . . . . . . . . . . . . . . . . 30

   A.  2-Call Preemption Example using RSVP . . . . . . . . . . . . . 32

       Intellectual Property and Copyright Statements . . . . . . . . 45









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1.  Overview

   The Defense Information Systems Agency of the United States
   Department of Defense, with is contractors, has proposed a service
   architecture for military (NATO and related agencies) telephone
   systems.  This is called the Assured Service, and is defined in two
   documents: [I-D.pierce-ieprep-assured-service-arch] and
   [I-D.pierce-ieprep-assured-service-req].  Responding to these are two
   documents: [I-D.ietf-sipping-reason-header-for-preemption] and
   [I-D.ietf-sip-resource-priority].

   What remains to this specification is to provide a Call Admission
   Control procedure and a Per Hop Behavior for the data which 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 that is permitted initially and is not
   denied or rejected at a later time, call and capacity admission
   procedures performed only at the time of call setup may be
   sufficient.  However in a network where sessions‚ÇÖ 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
   supercede existing calls), maintaining a continuing model of the
   status of the various calls is required.

1.1  Multi-Level Preemption and Precedence

   Before doing so, however, let us discuss the problem that MLPP is
   intended to solve and the architecture of the system.  The Assured
   Service is designed as an IP implementation of an existing
   ITU-T/NATO/DoD telephone system architecture known as
   [ITU.MLPP.1990][ANSI.MLPP.Spec][ANSI.MLPP.Supplement], 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.  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



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      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,
      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.

   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



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   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 (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
   trunks 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.

   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.

1.2  Definition of Call Admission

   Traditionally, in the PSTN, "Call Admission Control", or CAC, has had
   the responsibility of determining whether a caller has permission (an
   identified subscriber, with identify attested to by appropriate
   credentials, is authorized) to use an available circuit.  MLPP, or
   any emergency telephone service, creates two feedback paths in the
   algorithm: if a caller is authorized to use a higher precedence and
   is asserting that the advanced precedence applies to a given call, he
   may also be authorized to use other networks, or the PSTN may be
   obligated to preempt a call if possible and necessary to create
   appropriate bandwidth, or it may be authorized to use a guard band of
   bandwidth that other callers are not.  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



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   use and preemption may in some networks be part of making that
   happen.

1.3  Assumptions about the Network

   IP networks generally fall into two categories: those with
   constrained bandwidth, and those that are massively overprovisioned.
   In a network wherein 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 ATM, MPLS, and GMPLS can be used to explicitly
   place routes, and so improve the odds a bit.

   Between those networks, in places commonly called "inter-campus
   links", "access links" or "access networks", for various reasons
   including technology 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, and those
   NOCs are in turn interconnected via the 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), 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 do not tune their behavior to that of the
   network.  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 that loss and jitter are nominal.  Where that cannot be
   accomplished, one must use queuing technology to deterministically
   apply bandwidth to accomplish the goal.

1.5  Desired Characteristics in an Internet Environment

   The key elements of the MLPP service include the following:



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   Precedence Level Marking each call: Call initiators choose the
      appropriate precedence level for each call based on 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 which 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 MLPP service must, by definition,
   provide those characteristics.

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



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   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 ten 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
   overprovisioned to a point where loss or queuing delay are
   negligible, 10:1 overprovisioning is often the cheapest and surest
   solution, and by the way offers a growth path for future
   requirements.  However, there are places in communication networks
   where bandwidth is not free and is therefore not effectively
   infinite.  It is in these places, and only these places, where the
   question of resource management is relevant.

   The places where bandwidth constriction takes place is typically
   where one pays a significant amount for bandwidth, such as in access
   paths, or where available technology limits the options.  In military
   networks, Type 1 encryption often presents such a barrier, as do
   satellite links and various kinds of radio systems.

   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 that figure, 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 switches or circuit 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 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.

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




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

   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?


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 specialized policy like call preemption, two capabilities are
   necessary from an administrative perspective:
   [I-D.ietf-sip-resource-priority] provides a way to communicate
   policy-related information regarding the precedence of the call; and
   [I-D.ietf-sipping-reason-header-for-preemption] 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.

   The Communications Resource Priority Header (or RP Header) serves the
   call set-up 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 (phone).  If the receiving phone does not
   support the namespace, it can easily ignore (what
   [I-D.ietf-sip-resource-priority] calls "loose mode") or errors (what
   [I-D.ietf-sip-resource-priority] calls "strict mode") the set-up
   request.  This ability to denote the domain of origin allows SLAs to
   be in place to limit the ability of an unknown requestor to gain
   preferential treatment into an MLPP domain.

   For the DSN infrastructure, this header would look like this:

        Resource Priority : dsn.routine




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

   The DSN has 5 precedence levels of MLPP 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
   is to be IANA registered in [I-D.ietf-sip-resource-priority], has 6
   levels of precedence.  The DRSN simply adds one higher precedence
   level than flash-override:

        drsn.flash-override-override

   to be used by the President and a select few others.  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
   set-up requests.

   This informs both the use of 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 MLPP domain, the Reason Header for
   Preemption, described in
   [I-D.ietf-sipping-reason-header-for-preemption], 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
   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 a



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   endpoint generated preemption event, or that the preemption event
   occurred within a router along the communications path (described in
   Section 2.3.1 ).

   Figure X is a simple example of a SIP call set-up 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 will be 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 MLPP 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 MLPP 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 set-up 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's namespace value.  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 the [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



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   traffic to ensure minimal variation in delay and a minimal loss rate,
   as codecs are hampered by excessive loss [G711.1][G711.2][G711.3] .
   In parts of the network where bandwidth is heavily overprovisioned,
   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.

   For policy reasons, DISA would like to 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 they get the packets
   into the right service class.  The major issue, then is primarily one
   of appropriate policing of traffic, especially around route changes.

   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, 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
   bandwidth, the network will remark and drop the excess traffic.  The
   key questions become "How does one limit offered load to a rate less



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   than or equal to available bandwidth?" and "how much traffic does one
   admit with each appropriate marking?"

2.3  Bandwidth admission procedure

   Since the 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 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 requires a single ingress/egress - a
   single point of failure.  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.

   The [RFC1633] provides for signalled admission for the use of
   capacity.  This is currently implemented using the Resource
   Reservation Protocol [RFC2205][RFC2209] (RSVP).  The use of Capacity
   Admission with SIP is described in [RFC3312] ; at this writing,
   Capacity Admission is not integrated with H.323.

2.3.1  Recommended procedure: explicit call admission - RSVP Admission
      using Policy

   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
   setup 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 appropriate.

   RSVP is a locally-oriented process, not a globally- or
   domain-oriented one like a routing protocol or like H.323 Call
   Counting.  Although it uses the local routing databases to determine



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   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 enabled in a network without
   the need to have every node participate.

                 HOST                              ROUTER
    _____________________________       ____________________________
   |  _______                    |     |                            |
   | |       |   _______         |     |            _______         |
   | |Appli- |  |       |        |RSVP |           |       |        |
   | | cation|  | RSVP <---------------------------> RSVP  <---------->
   | |       <-->       |        |     | _______   |       |        |
   | |       |  |process|  _____ |     ||Routing|  |process|  _____ |
   | |_._____|  |       -->Polcy||     ||       <-->       -->Polcy||
   |   |        |__.__._| |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: which resolves the QoS class for each
      packet; this can determine the route as well.

   An admission control mechanism: this consists of two decision
      modules: the admission control module and the policy control
      module.  Determining whether there is satisfactory resources for
      the requested QoS is the function of admission control.
      Determining if the user has the authorization to request such



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      resources is the function of policy control.  If the parameters
      carried within this flow fail either of these two modules, RSVP
      errors the request.

   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, RSVP had scaling limitations due to its data
   plane behavior.  This has, in time, largely been corrected.  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, and may also include any type 1 encrypted
   interface, as these are generally limited in bandwidth by the
   encryption.

2.3.3  RSVP Operation in backbones and 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 and of networks that carry traffic from an encryptor to a
   decryptor, and also of VPNs - applicable technologies include
   [RFC2207], [RFC2746], and [RFC2983].  An IP tunnel is simplistically
   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 enabled, there must be a 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.




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   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, similar tunnels between HAIPE
   encryptors and decryptors, IP/IP tunnels, and 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 MLPP 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.

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



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   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 overprovisioning, and
   approximation to ambient load.

   Simple overprovisioning 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 which
   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
   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.  But they offer the
   opportunity to take advantage of statistical multiplexing gains that
   might otherwise be missed.





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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 Int-Serv model.  This model was created to provide
   ability to monitor and control reservation flows based on user
   identify, specific traffic and security requirements and conditions
   which might change for various reasons, including as 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
   Decision Point.  The easier the instruction set of rules, the more
   likely this set can reside in the LDPD 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 which 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.



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   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 to disconnect 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; it is the capabilities assigned to an
   authenticated user that may be considered to have been 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 MLPP 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
   which don't necessarily apply.  For example, in an end-to-end IP
   call, there is no known benefit to informing the user which Ethernet
   switch or router caused the call to fail - as is the equivalent case
   if a TDM Switch were the cause.  All indications to the user, of
   course, depend on positive signals, not unreliable measures based on
   changing measurements.






















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3.  IANA Considerations

   This document makes no request of IANA.

   Note to RFC Editor: this section may be removed on publication as an
   RFC.













































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4.  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.











































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5.  Acknowledgements

   This document was developed with the knowledge and input of many
   people, far too numerous to be mentioned by name.  Key contributors
   of thoughts include, however, Francois Le Faucheur, Haluk Keskiner,
   Rohan Mahy, Scott Bradner, Scott Morrison, and Subha Dhesikan.  Pete
   Babendreier's review was especially useful.












































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6.  References

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

   [I-D.ietf-sip-resource-priority]
              Schulzrinne, H. and J. Polk, "Communications Resource
              Priority for the Session Initiation Protocol (SIP)",
              Internet-Draft draft-ietf-sip-resource-priority-05,
              October 2004.

   [I-D.ietf-sipping-reason-header-for-preemption]
              Polk, J., "Extending the Session Initiation Protocol
              Reason Header for Preemption  Events",
              Internet-Draft draft-ietf-sipping-reason-header-for-preemption-02
              , August 2004.

   [I-D.pierce-ieprep-assured-service-arch]
              Pierce, M. and D. Choi, "Architecture for Assured Service
              Capabilities in Voice over IP",
              Internet-Draft draft-pierce-ieprep-assured-service-arch-02, January 2004
              .

   [I-D.pierce-ieprep-assured-service-req]
              Pierce, M. and D. Choi, "Requirements for Assured Service
              Capabilities in Voice over IP",
              Internet-Draft draft-pierce-ieprep-assured-service-req-02,
              January 2004.

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

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




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   [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.

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

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

   [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.

   [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.

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

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



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              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.

   [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.

   [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.

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

   [RFC3326]  Schulzrinne, H., Oran, D. and G. Camarillo, "The Reason
              Header Field for the Session Initiation Protocol (SIP)",
              RFC 3326, December 2002.

6.2  Informative References

   [G711.1]   Viola Networks, "Netally VoIP Evaluator", January 2003,
              <http://www.sygnusdata.co.uk/white_papers/viola/netally_vo
              ip_sample_report_preliminary.pdf>.

   [G711.2]   ETSI Tiphon, "ETSI Tiphon Temporary Document 64", July
              1999,
              <http://docbox.etsi.org/tiphon/tiphon/archives/1999/05-990
              7-Amsterdam/14TD113.pdf>.




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

   [G711.4]   Clark, A., "Modeling the Effects of Burt Packet Loss and
              recency on Subjective Voice Quality", 2000,
              <http://www.telchemy.com/references/tech_papers/iptel2001.
              pdf>.

   [G711.5]   Cisco Systems, "Understanding Codecs: Complexity, Hardware
              Support, MOS, and Negotiation", 2003,
              <http://www.cisco.com/en/US/tech/tk652/tk701/technologies_
              tech_note09186a00800b6710.shtml#mos>.

   [ILBC]     Chen, M. and M. Murthi, "On The Performance Of ILBC Over
              Networks With Bursty Packet Loss", July 2003.

   [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.

   [RFC3951]  Andersen, S., Duric, A., Astrom, H., Hagen, R., Kleijn, W.
              and J. Linden, "Internet Low Bit Rate Codec (iLBC)",
              RFC 3951, December 2004.


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







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

   Phone: +1-469-255-5208
   Email: jmpolk@cisco.com











































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Appendix A.  2-Call Preemption Example using RSVP

   This appendix will present a more complete view of the interaction
   between SIP, SDP and RSVP.  The bulk of the material is referenced
   from [RFC2327], [RFC3312],
   [I-D.ietf-sipping-reason-header-for-preemption],
   [I-D.ietf-sip-resource-priority].  There will be some discussion on
   basic RSVP operations regarding reservation paths, this will be
   mostly from [RFC2205].

   SIP signaling occurs at layer 7, 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 a escalation of building blocks getting to how the many
   layers are involved in SIP with QoS Preconditions requiring
   successful RSVP signaling between endpoints prior to SIP successfully
   acknowledging the set-up of the session (for voice or video or both).
   Then we will present what occurs when a network overload occurs
   (congestion), causing a SIP session to be preempted.

   There are 3 diagrams in this appendix to 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 - this means Alice's
   UA (her phone) in the text here.

   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 requestor 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



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   parameters of the session in an INVITE message, and that the INVITE
   recipient (the UAS) must follow the parameters established in that
   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:

   The reservation from Carol to Dave be through routers:

      Carol -> R6 -> R2 -> R3 -> R7 -> R11 -> Dave

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




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      Dave -> R11 -> R7 -> R3 -> R2 -> R6 -> 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
   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 Int7 to be what will first know about a
   congestion condition.

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


                 Figure 6: Reduced Reservation Topology

   From Figure 6, 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] "SIP Preconditions for QoS".

   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 are 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------------------>|  *
       *  |                                            |  * When
       *  |<-----------(4) 200 OK (PRACK)--------------|  * RSVP
       *  |                                            |  * is
       *  |                                            |  * signaled
    +->***|********************************************|***
    |     |-------------(5) UPDATE SDP3--------------->|
   Note 2 |                                            |
          |<--------(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 MLPP 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 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
      Requires: 100rel
      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 (before the empty line in the middle) are SIP
   headers and header values, the lower half of the lines above are
   Session Description Protocol (SDP) lines.  SIP headers (after the
   first line) are not to be 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 proxy
   this message traverses.  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 not 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 that is not worth the space here to
   replicate what has been done in SIP RFCs already.

   SIP headers that are used for Preconditions are the:

      Requires header - which mandates a reliable provisional response
      message to the conditions requesting in this INVITE (knowing they
      are special).

   This will result in the 183 "Session Progress" message from Bob's UA



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   as a reliable confirmation that preconditions are required for this
   call.

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

   And that's it for SIP.  Preconditions is 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 reasons by endsystems.
   Dialog (Call) generating SDP message bodies all must have an "m" line
   (or media description line).  Following the "m" line is 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 8 = 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 RSVP is required from Alice's UA to Bob's UA; meaning
      an RSVP failure in either direction will fail the call attempt.

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

   The final a-line (a=des):

      a=des:qos mandatory e2e sendrecv

   identifies the "desired" level of qos

      "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.



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   As discussed, RSVP does not reserve bandwidth in both directions, and
   that 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
   that 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 172.16.1.36
      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


                                Figure 9

   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
   set-up (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).




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   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 set-up.

   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 set up 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.  But 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
   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 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>
      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


                               Figure 10

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






















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      [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>
      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


                               Figure 11

   At this point, the reservation is operational and both UA's know it,
   and Bob's UA now rings ([M7] is the SIP indication to Alice this is
   taking place) telling Bob the user that Alice is calling her.
   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 here.

   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 is can allocate for RSVP traffic and has reached
   its maximum with 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



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   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 characteristics important here:

   [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 area 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

   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 set-up 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 reservation above in this direction (Dave to will not traverse
   any of the same routers as the Alice to Bob reservations.  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:





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      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 Int-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 or the RESV message (Bob).
   This clears the reservation in all routers downstream of R2 (meaning
   R3 and R4).  Once Bob receives the ResvErr message indicating
   preemption has occur 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
   set-up path that wanted to remain "dialog stateful" per [RFC3261]],
   and 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,
   "RSVP_Preemption".  Here is that 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: cause=2 ;text=RSVP preemption
      Call-ID: a84b4c76e66710@swp34.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.




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   The RESV message from Dave successfully traverses R2 and Carol's UA
   receives it.  Just as with the Alice to Bob call set-up, 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 set-up 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
   will preempt any lower priority calls where these is 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 R2 (in
   this example) generating a syslog message to the domain administrator
   to properly manage and track such events within this domain.  This
   will ensure the domain administrators have recorded knowledge of
   where such events occur, and what the conditions were that caused
   them.






























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