Transport Working Group                                         F. Baker
Internet-Draft                                                   J. Polk
Updates: 4542,4594                                         Cisco Systems
(if approved)                                                   M. Dolly
Intended status: Standards Track                               AT&T Labs
Expires: August 28, 2008                               February 25, 2008

                  DSCPs for Capacity-Admitted Traffic

Status of this Memo

   By submitting this Internet-Draft, each author represents that any
   applicable patent or other IPR claims of which he or she is aware
   have been or will be disclosed, and any of which he or she becomes
   aware will be disclosed, in accordance with Section 6 of BCP 79.

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   This Internet-Draft will expire on August 28, 2008.

Copyright Notice

   Copyright (C) The IETF Trust (2008).


   This document requests one Differentiated Services Code Point (DSCP)
   from the Internet Assigned Numbers Authority (IANA) for real-time
   traffic classes similar to voice conforming to the Expedited
   Forwarding Per Hop Behavior, and admitted using a call admission
   procedure involving authentication, authorization, and capacity

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   It also recommends that certain classes of video traffic described in
   RFC 4594 and which have similar requirements be changed to require
   admission using a Call Admission Control (CAC) procedure involving
   authentication, authorization, and capacity admission.

Requirements Language

   The key words "MUST", "MUST NOT", "REQUIRED", "SHALL", "SHALL NOT",
   document are to be interpreted as described in RFC 2119 [RFC2119].

Table of Contents

   1.  Introduction . . . . . . . . . . . . . . . . . . . . . . . . .  3
     1.1.  Definitions  . . . . . . . . . . . . . . . . . . . . . . .  4
     1.2.  Problem  . . . . . . . . . . . . . . . . . . . . . . . . .  6
     1.3.  Proposed Solution  . . . . . . . . . . . . . . . . . . . .  7
   2.  Implementation of the Admitted Service Classes . . . . . . . .  7
     2.1.  Potential implementations of EF in this model  . . . . . .  7
     2.2.  Capacity admission control . . . . . . . . . . . . . . . .  9
       2.2.1.  Capacity admission control by assumption . . . . . . . 10
       2.2.2.  Capacity admission control by call counting  . . . . . 10
       2.2.3.  End-point capacity admission performed by probing
               the network  . . . . . . . . . . . . . . . . . . . . . 11
       2.2.4.  Centralized capacity admission control . . . . . . . . 11
       2.2.5.  Distributed capacity admission control . . . . . . . . 12
     2.3.  Prioritized capacity admission control . . . . . . . . . . 12
   3.  Recommendations on implementation of an Admitted Telephony
       Service Class  . . . . . . . . . . . . . . . . . . . . . . . . 13
   4.  IANA Considerations  . . . . . . . . . . . . . . . . . . . . . 14
   5.  Security Considerations  . . . . . . . . . . . . . . . . . . . 14
   6.  Acknowledgements . . . . . . . . . . . . . . . . . . . . . . . 15
   7.  References . . . . . . . . . . . . . . . . . . . . . . . . . . 15
     7.1.  Normative References . . . . . . . . . . . . . . . . . . . 15
     7.2.  Informative References . . . . . . . . . . . . . . . . . . 15
   Authors' Addresses . . . . . . . . . . . . . . . . . . . . . . . . 17
   Intellectual Property and Copyright Statements . . . . . . . . . . 19

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1.  Introduction

   This document requests one Differentiated Services Code Point (DSCP)
   from the Internet Assigned Numbers Authority (IANA) for a class of
   real-time traffic.  This class conforms to the Expedited Forwarding
   [RFC3246] [RFC3247] Per Hop Behavior.  It is also admitted using a
   CAC procedure involving authentication, authorization, and capacity
   admission.  This differs from a real-time traffic class conforming to
   the Expedited Forwarding Per Hop Behavior but not subject to capacity
   admission or subject to very coarse capacity admission.

   It also recommends that certain classes of video described in
   [RFC4594] be treated as requiring capacity admission as well.

   These applications have one or more potential congestion points
   between the video distribution/conferencing bridge or gaming server
   and the user(s), and reserving capacity for them is important to
   application performance.  All of these applications have low
   tolerance to jitter (aka delay variation) and loss, as summarized in
   Section 2, and most (except for multimedia conferencing) have
   inelastic flow behavior from Figure 1 of [RFC4594].  Inelastic flow
   behavior and low jitter/loss tolerance are the service
   characteristics that define the need for admission control behavior.

   One of the reasons behind this is the need for classes of traffic
   that are handled under special policies, such as the non-preemptive
   Emergency Telecommunication Service, the US Department of Defense's
   Assured Service (which is similar to Multi-Level Precedence and
   Preemption [ITU.MLPP.1990] procedure), or e-911, in addition to
   normal routine calls that use call admission.  It is possible to use
   control plane protocols to generally restrict session admission such
   that admitted traffic should receive the desired service, and the
   policy (e.g.  Routine, National Security or Emergency Preparedness
   [NS/EP] communications, e-911, etc) need not be signaled in a DSCP.
   However, service providers need to distinguish between special-policy
   traffic and other classes, particularly the existing VoIP services
   that perform no capacity admission or only very coarse capacity
   admission and can exceed their allocated resources.

   The requested DSCP applies to the Telephony Service Class described
   in [RFC4594].  The video classes addressed include the

   o  interactive real-time traffic (CS4, used for Video conferencing
      and Interactive gaming),

   o  broadcast TV (CS3) for use in a video on demand context, and

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   o  AF4 Multimedia conferencing (video conferencing).

   Since the admitted video classes have not had the history of mixing
   admitted and non-admitted traffic in the same Per-Hop Behavior (PHB)
   as has occurred for EF, an additional DSCP code point is not
   recommended.  Instead, the recommended "best practice" is to perform
   admission control for the above video classes.

   Other video classes are not believed to be required by the targeted
   services and to not have the current problem of confusion with
   unadmitted traffic.  Within an ISP and on inter-ISP links (i.e.
   within networks whose internal paths are uniform at hundreds of
   megabits or faster), one would expect all of this traffic to be
   carried in the Real Time Traffic Class described in [RFC5127].

1.1.  Definitions

   The following terms and acronyms are used in this document.

   PHB:  A Per-Hop-Behavior (PHB) is the externally observable
      forwarding behavior applied at a Differentiated Services compliant
      node to a DS behavior aggregate [RFC2475].  It may be thought of
      as a program configured on the interface of an Internet host or
      router, specified drop probabilities, queuing priorities or rates,
      and other handling characteristics for the traffic class.

   DSCP:  The Differentiated Services Code Point (DSCP), as defined in
      [RFC2474], is a value which is encoded in the DS field, and which
      each DS Node MUST use to select the PHB which is to be experienced
      by each packet it forwards [RFC3260].  It is a 6-bit number
      embedded into the 8-bit TOS field of an IPv4 datagram or the
      Traffic Class field of an IPv6 datagram.

   CAC:  Call Admission Control includes concepts of authorization and
      capacity admission.  "Authorization" includes and procedure
      identifies a user, verifies the authenticity of the
      identification, and determines whether the user is authorized to
      use the service.  "Capacity Admission" refers to any procedure
      that determines whether capacity exists supporting a session's
      requirements under some policy.

      In the Internet, these are separate functions, while in the PSTN
      they and call routing are carried out together.

   UNI:  A User/Network Interface (UNI) is the interface (often a
      physical link or its virtual equivalent) that connects two
      entities that do not trust each other, and in which one (the user)
      purchases connectivity services from the other (the network).

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      Figure 1 shows two user networks connected by what appears to each
      of them to be a single network ("The Internet", access to which is
      provided by their service provider) that provides connectivity
      services to other users.

      UNIs tend to be the bottlenecks in the Internet, where users
      purchase relatively low amounts of bandwidth for cost or service
      reasons, and as a result are most subject to congestion issues and
      therefore issues requiring traffic conditioning and service

   NNI:  A Network/Network Interface (NNI) is the interface (often a
      physical link or its virtual equivalent) that connects two
      entities that trust each other within limits, and in which the two
      are seen as trading services for value.  Figure 1 shows three
      service networks that together provide the connectivity services
      that we call "the Internet".  They are different administrations
      and are very probably in competition, but exchange contracts for
      connectivity and capacity that enable them to offer specific
      services to their customers.

      NNIs may not be bottlenecks in the Internet if service providers
      contractually agree to provision excess capacity at them, as they
      commonly do.  However, NNI performance may differ by ISP, and the
      performance guarantee interval may range from a month to a much
      shorter period.  Furthermore, a peering point NNI may not have
      contractual performance guarantees or may become overloaded under
      certain conditions.  They are also policy-controlled interfaces,
      especially in BGP.  As a result, they may require traffic
      prioritization policy.

   Queue:  There are multiple ways to build a multi-queue scheduler.
      Weighted Round Robin (WRR) literally builds multiple lists and
      visits them in a specified order, while a calendar queue (often
      used to implement Weighted Fair Queuing, or WFQ) builds a list for
      each time interval and enqueues at most a stated amount of data in
      each such list for transmission during that time interval.  While
      these differ dramatically in implementation, the external
      difference in behavior is generally negligible when they are
      properly configured.  Consistent with the definitions used in the
      Differentiated Services Architecture [RFC2475], these are treated
      as equivalent in this document, and the lists of WRR and the
      classes of a calendar queue will be referred to uniformly as

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                                    ,-''           `--.
                                 ,-'                   `-.
           ,-------.           ,',-------.                `.
         ,'         `.       ,','         `.                `.
        /  User       \ UNI / /   Service   \                 \
       (    Network    +-----+    Network    )                 `.
        \             /  ;    \             /                    :
         `.         ,'   ;     `.         .+                     :
           '-------'    /        '-------'  \ NNI                 \
                       ;                     \                     :
                       ;     "The Internet"   \  ,-------.         :
                      ;                        +'         `.        :
        UNI: User/Network Interface           /   Service   \       |
                     |                       (    Network    )      |
        NNI: Network/Network Interface        \             /       |
                      :                        +.         ,'        ;
                       :                      /  '-------'         ;
                       :                     /                     ;
           ,-------.    \        ,-------.  / NNI                 /
         ,'         `.   :     ,'         `+                     ;
        /  User       \ UNI   /   Service   \                    ;
       (    Network    +-----+    Network    )                 ,'
        \             /     \ \             /                 /
         `.         ,'       `.`.         ,'                ,'
           '-------'           `.'-------'                ,'
                                 `-.                   ,-'
                                    `--.           _.-'

                     Figure 1: UNI and NNI interfaces

1.2.  Problem

   In short, the Telephony Service Class described in [RFC4594] permits
   the use of capacity admission in implementing the service, but
   present implementations either provide no capacity admission services
   or do so in a manner that depends on specific traffic engineering.
   In the context of the Internet backbone, the two are essentially
   equivalent; the edge network is depending on specific engineering by
   the service provider that may not be present.

   However, services are being requested of the network that would
   specifically make use of capacity admission, and would distinguish
   among users or the uses of available Voice-on-IP or Video-on-IP
   capacity in various ways.  Various agencies would like to provide
   services as described in section 2.6 of [RFC4504] or in [RFC4190].
   This requires the use of capacity admission to differentiate among

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   users (which might be 911 call centers, other offices with
   preferential service contracts, or individual users gaining access
   with special credentials) to provide services to them that are not
   afforded to non-capacity admitted customer-to-customer IP telephony

1.3.  Proposed Solution

   The IETF is asked to differentiate, in the Telephony Service, between
   sessions that are originated without capacity admission or using
   traffic engineering and sessions that are originated using more
   robust capacity admission procedures.  Sessions of the first type use
   a traffic class in which they compete without network-originated
   control as described in Section 2.2.1 or Section 2.2.2, and in the
   worst case lose traffic due to policing.  Sessions of the second type
   cooperate with network control, and may be given different levels of
   preference depending on the policies that the network applies.  In
   order to provide this differentiation, the IETF requests that the
   IANA assign a separate DSCP value to admitted sessions using the
   Telephony service (see Section 4 ).

2.  Implementation of the Admitted Service Classes

2.1.  Potential implementations of EF in this model

   There are at least two possible ways to implement the Expedited
   Forwarding PHB in this model.  They are to implement separate classes
   as a set of

   o  Multiple data plane traffic classes, each consisting of a policer
      and a queue, and the queues enjoying different priorities, or

   o  Multiple data plane traffic classes, each consisting of a policer
      but feeding into a common queue or multiple queues at the same

   We will explain the difference, and describe in what way they differ
   in operation.  The reason this is necessary is that there is current
   confusion in the industry, including a widely reported test for NS/EP
   services that implemented the policing model and described it as an
   implementation of the multi-priority model, and discussion in other
   environments of the intermixing of voice and video traffic at
   relatively low bandwidths in the policing model.

   The multi-priority model is shown in Figure 2.  In this model,
   traffic from each service class is placed into a separate priority
   queue.  If data is present in both queues, traffic from one of them

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   will always be selected for transmission.  This has the effect of
   transferring jitter from the higher priority queue to the lower
   priority queue, and reordering traffic in a way that gives the higher
   priority traffic a smaller average queuing delay.  Each queue must
   have its own policer, however, to protect the network from errors and
   attacks; if a traffic class thinks it is carrying a certain data rate
   but an abuse sends significantly more, the effect of simple
   prioritization would not preserve the lower priorities of traffic,
   which could cause routing to fail or otherwise impact an SLA.

                     policers    priorities  |`.
           EF ---------> <=> ----------||----+  `.
                                         high|    `.
           EF2---------> <=> ----------||----+     .'-----------
                           .             medium  .'
              rate queues  |`.         +-----+ .' Priority
           AF1------>||----+  `.      /  low |'   Scheduler
                           |    `.   /
           AF2------>||----+     .'-+
                           |   .'
           CS0------>||----+ .' Rate Scheduler
                           |'   (WFQ, WRR, etc)

             Figure 2: Implementation as a data plane priority

   The multi-policer model is shown in Figure 3.  In this model, traffic
   from each service class is policed according to its SLA requirements,
   and then placed into a common priority queue.  Unlike the multi-
   priority model, the jitter experienced by the traffic classes in this
   case is the same, as there is only one queue, but the sum of the
   traffic in this higher priority queue experiences less average jitter
   than the elastic traffic in the lower priority.

                       policers    priorities  .
             EF ---------> <=> -------\        |`.
                                       --||----+  `.
             EF2---------> <=> -------/    high|    `.
                             .                 |     .'--------
                rate queues  |`.         +-----+   .'
             AF1------>||----+  `.      /  low | .' Priority
                             |    `.   /       |'   Scheduler
             AF2------>||----+     .'-+
                             |   .'
             CS0------>||----+ .' Rate Scheduler
                             |'   (WFQ, WRR, etc)

             Figure 3: Implementation as a data plane policer

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   The difference between the two operationally is, as stated, the
   issues of loss due to policing and distribution of jitter.

   If the two traffic classes are, for example, voice and video,
   datagrams containing video data are relatively large (generally the
   size of the path MTU) while datagrams containing voice are relatively
   small, on the order of only 40 to 200 bytes, depending on the codec.
   On lower speed links (less than 10 MBPS), the jitter introduced by
   video to voice can be disruptive, while at higher speeds the jitter
   is nominal compared to the jitter requirements of voice.  At access
   network speeds, therefore, [RFC4594] recommends separation of video
   and voice into separate queues, while at optical speeds [RFC5127]
   recommends that they use a common queue.

   If, on the other hand, the two traffic classes are carrying the same
   type of application with the same jitter requirements, then giving
   one preference in this sense does not benefit the higher priority
   traffic and may harm the lower priority traffic.  In such a case,
   using separate policers and a common queue is a superior approach.

2.2.  Capacity admission control

   There are five major ways that capacity admission is done or has been
   proposed to be done in real-time applications:

   o  Capacity admission control by assumption,

   o  Capacity admission control by call counting,

   o  End-point capacity admission performed by probing the network,

   o  Centralized capacity admission control, and

   o  Distributed capacity admission control.

   There is also a mechanism that has been proposed for enhancing the
   probability of call completion in a preferential manner.  This is not
   capacity admission per se, since it never actually refuses a call
   (although a session may be dropped by its user when the user finds
   continuation untenable).  The central notion is that when the
   capacity available for a set of variable rate sessions has been
   overbooked, traffic may be randomly dropped from lower precedence
   sessions to allow a higher precedence session in.  This has a number
   of ramifications that make it inappropriate in the Internet.  A key
   issue is that it affects not a single session but a class of sessions
   - all sessions of lower precedence than the protected session(s).  A
   video example will suffice for the present.  Multimedia data streams
   and sensor traffic often build on information in previous frames, and

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   their content spans multiple datagrams in the same frame.  The loss
   of a datagram forces the codec into a recovery mode that reduces
   image quality for at least one frame, and may cause the image to
   freeze for multiple seconds.  This is readily observed on television,
   where screen artifacts are very visible.  Scattered random datagram
   loss results in all sessions in the class being impacted to some
   degree.  Hence, it is far more suitable to drop an entire session
   (and therefore impact only one session) than to impact all sessions
   in a class in a manner that consumes the available bandwidth but
   delivers sub-SLA service to an entire class of sessions.  It also
   exposes the precedence level of each session in the clear.

2.2.1.  Capacity admission control by assumption

   The first approach is to ignore the matter entirely.  If one assumes
   that the capacity available to the application is uniformly far in
   excess of its requirements, it is perhaps overhead that can be
   ignored.  This assumption is currently made in Internet VoIP
   offerings such as Skype and Vonage; the end user is responsible to
   place his service on a LAN connected to the Internet backbone by a
   high speed broadband connection and use capable ISPs to deliver the
   service.  The only "authorization" verified is that the user pays his
   bills; no capacity admission is considered because there is a clear
   separation from the application service provider admitting the calls
   and the access network provider admitting the traffic.  The two have
   no way of knowing about each other, except in the abstract sense.

2.2.2.  Capacity admission control by call counting

   The H.323 gatekeeper, originally specified in 1996, operates on the
   model that the considerations of Section 2.2.1 generally apply, and
   that it is therefore sufficient to count calls in order to ensure
   that any bottlenecks in the network are never overloaded.  Which
   phone is calling which phone is configured information into the
   Gatekeeper, ensuring it doesn't admit too many calls across a low
   speed link.  The area of influence of a Gatekeeper is called a Zone,
   and limits how far away a Gatekeeper can influence calls.  This is
   because call counting doesn't scale when more than one server is
   admitting flows across the same limited speed links.  This approach
   is consistent with the original design of H.323, which in 1996 was a
   mechanism for connecting H.320 media gateways across a LAN.  VoIP has
   come a long way since then, however, and the engineering trade-offs
   this approach requires in complex networks are unsatisfactory.

   SIP provides the option to go down another path, to admit its servers
   at layer 7, have no awareness of lower layer connectivity, resulting
   in a divorce from infrastructure knowledge - save for [RFC3312],
   which binds the two, but only at the endpoints.

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   In short, if there is a bottleneck anywhere in the network that might
   be used to connect two gatekeepers, SIP proxies that do not implement
   or do not configure the use of [RFC3312], or other call management
   systems, the amount of traffic between the two must be contained
   below that bottleneck even if the normal path is of much higher
   bandwidth.  In addition, the multiplexing of traffic streams between
   different pairs of gatekeepers over a common LAN infrastructure is
   not handled by the application, and so must be managed in the
   engineering of the network.

2.2.3.  End-point capacity admission performed by probing the network

   The IETF started looking into the use of Pre-Congestion Notification
   mechanism to full fill the need of admission control for real-time
   traffic.  The main contribution of this work for admission control is
   to allow the network to provide the network's pre-congestion
   information using encoding of a field in the IP header.  This network
   pre-congestion information is then used for making admission control
   decisions.  With the decision influenced by this network pre-
   congestion notification information and any applicable policy
   information with possible user credentials and situational
   information.  The pre-congestion notification mechanism does not
   limit the placement of the admission control decision point or the
   signaling protocol used.

   The overview of one of the current proposals is provided by
   [I-D.chan-pcn-problem-statement].  With the pre-congestion
   notification encoding described in [I-D.briscoe-tsvwg-cl-phb].  An
   initial deployment model provided by
   [I-D.briscoe-tsvwg-cl-architecture].  Another proposal is embodied in
   [I-D.charny-pcn-single-marking].  Similar approaches have been
   proposed in [I-D.morita-tsvwg-pps] and its related drafts, by Ivars
   and Karlsson in their PBAC work, and many others.

2.2.4.  Centralized capacity admission control

   The concept of a Bandwidth Broker was first discussed in the research
   world surrounding ESNET and Internet II in the late 1990's, and has
   been discussed in the literature pertaining to the Differentiated
   Services Architecture [RFC2475].  It is, in short, a central system
   that performs a variety of services on behalf of clients of a network
   including applying AAA services (as in [RFC2904] ) and authorizing
   them to use specified capacity at specified times.  Its strength is
   that it is relatively simple, at least in concept, and can keep track
   of simple book-keeping functions apart from network elements such as
   routers.  Its weakness is that it has no idea what the specific
   routing of any stated data flow is, or its capacity apart from
   services such as MPLS Traffic Engineering or engineering assumptions

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   specified by the designers of a network.  Obtaining that information
   from the network via SNMP GET or other network management action can
   impose a severe network overhead, and is obviously not real-time.

   For scaling reasons, operational Bandwidth Brokers generally take on
   a semi-distributed or fully distributed nature.  They are implemented
   on a per-point-of-presence basis, and in satellite networks might be
   implemented in each terminal.  At this point, they become difficult
   to operationally distinguish from distributed capacity admission
   services such as described in Section 2.2.5.

2.2.5.  Distributed capacity admission control

   The IETF developed the Integrated Services Model [RFC1633] and the
   RSVP capacity admission protocol [RFC2205] in the early 1990's, and
   then integrated it with the Differentiated Services Architecture in
   [RFC2998].  Since then, the IETF has been working on a next
   generation signaling protocol called NSIS [RFC4080] that can be used
   for capacity admission protocol, and which is limited in scope to
   considering unicast sessions.  [RFC4542] looked at the issue of
   providing preferential services in the Internet, and determined that
   RSVP with its defined extensions could provide those services to
   unicast and multicast sessions.

   As with the Bandwidth Broker model, there are concerns regarding
   scaling, mentioned in [RFC2208].  Present implementations that have
   been measured have been found to not display the scaling concerns,
   however, and in any event the use of RSVP Aggregation enables the
   backbone to handle such sessions in a manner similar to an ATM
   Virtual Path, bundling sessions together for capacity management

2.3.  Prioritized capacity admission control

   Emergency Telecommunication Service, the US Department of Defense's
   Assured Service, and e-911 each call for some form of prioritization
   of some calls over others.  Prioritization of the use of bandwidth is
   fundamentally a matter of choices - at a point where one has multiple
   choices, applying a policy that selects among them.  In the PSTN,
   GETS operates in favor of an authorized caller either by routing a
   call that would otherwise be refused by a path unavailable to the
   general public or by queuing the call until some existing call
   completes and bandwidth becomes available. e-911 is similar, but the
   policy is based on the called party, the emergency call center.  MLPP
   operates by preempting an existing call to make way for the new one.

   In the Internet, routing is not performed on a per-call basis, so,
   apart from interconnections to the PSTN, re-routing isn't an option.

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   On the other hand, in the Internet there are more classes of traffic
   than in the PSTN.  In the PSTN, all calls are uses of circuits, while
   in the Internet some bandwidth is always reserved for elastic
   applications - at least, it must be available for routing, and there
   is generally significant consideration of the web, instant messaging,
   and other applications.  In essence, any capacity admission policy
   that differentiates between calls has the option of temporarily
   borrowing bandwidth from the capacity reserved for elastic traffic by
   accepting new sessions under some prioritized policy while refusing
   sessions of lower priority because the threshold at that priority has
   been reached.

   For example, regardless of the type of capacity admission that is
   used (apart from "no admission process"), one might admit prioritized
   sessions using a higher bandwidth threshold than one admits lower
   priority sessions.

   If capacity admission as described in Section 2.2.2 is in use, the
   thresholds must be set low enough that bandwidth would be available
   anywhere in the network.  This greatly limits the utility of such a
   service due to the level of bandwidth waste that results.

   If capacity admission as described in Section 2.2.3 is in use, then
   multiple thresholds must be applied in marking the traffic, multiple
   traffic marks must be applied, or there must be multiple ways to
   interpret the result.  In any event, this is only applicable in
   domains in which the law of large numbers applies.

   If capacity admission as described in Section 2.2.4 is in use,
   thresholds can be applied related to a general policy or SLA, or
   related to the network ingress and egress in use.  It requires them
   to maintain state regarding network traffic routing separate from the
   network; to the extent that is variable, it requires direct
   monitoring in the OSS.

   If capacity admission as described in Section 2.2.5 is in use,
   thresholds can be applied to the critical points of the path that the
   traffic in question actually takes because one is asking the
   equipment that the path traverses.

3.  Recommendations on implementation of an Admitted Telephony Service

   It is the belief of the authors that either data plane PHB described
   in Section 2.1, if coupled with adequate AAA and capacity admission
   procedures as described in Section 2.2.5, are sufficient to provide
   the services required for an Admitted Telephony service class and an

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   Admitted Multimedia Conferencing Service Class.  If preemption is
   required, as described in section of [RFC4542], this provides
   the tools for carrying out the preemption.  If preemption is not in
   view, or in addition to preemptive services, the application of
   different thresholds depending on call precedence has the effect of
   improving the probability of call completion by admitting preferred
   calls at a time that other calls are being refused.  Routine and
   priority traffic can be admitted using the same DSCP value, as the
   choice of which calls are admitted is handled in the admission
   procedure executed in the control plane, not the policing of the data

   On the point of what protocols and procedures are required for
   authentication, authorization, and capacity admission, we note that
   clear standards do not at this time exist for bandwidth brokers, NSIS
   has not at this time been finalized and in any event is limited to
   unicast sessions, and that RSVP has been standardized and has the
   relevant services.  We therefore recommend the use of RSVP at the
   UNI.  Procedures at the NNI are business matters to be discussed
   between the relevant networks, and are recommended but not required.

4.  IANA Considerations

   This note requests that IANA assign a DSCP value to a second EF
   traffic class consistent with [RFC3246] and [RFC3247] in the
   "Differentiated Services Field Codepoints" registry.  It implements
   the Telephony Service Class described in [RFC4594] at lower speeds
   and is included in the Real Time Treatment Aggregate [RFC5127] at
   higher speeds.  The recommended value for the code point 101100,
   paralleling the EF code point, which is 101110.  The code point
   should be referred to as VOICE-ADMIT.

   This traffic class requires the use of capacity admission such as
   RSVP services together with AAA services at the User/Network
   Interface (UNI); the use of such services at the NNI is at the option
   of the interconnected networks.

5.  Security Considerations

   A major requirement of this service is effective use of a signaling
   protocol such as RSVP, with the capabilities to identify its user
   either as an individual or as a member of some corporate entity, and
   assert a policy such as "routine" or "priority".

   This capability, one has to believe, will be abused by script kiddies
   and others if the proof of identity is not adequately strong or if

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   policies are written or implemented improperly by the carriers.  This
   goes without saying, but this section is here for it to be said...

6.  Acknowledgements

   Kwok Ho Chan offered some textual comments and rewrote Section 2.2.3.
   Georgios Karagiannis offered additional comments on the same section.
   The impetus for including Video in the discussion, which initially
   only targeted voice, is from Dave McDysan, and text he suggested was
   included.  Dan Voce also commented.

7.  References

7.1.  Normative References

   [RFC2119]  Bradner, S., "Key words for use in RFCs to Indicate
              Requirement Levels", BCP 14, RFC 2119, March 1997.

   [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.

   [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.

7.2.  Informative References

              Briscoe, B., "An edge-to-edge Deployment Model for Pre-
              Congestion Notification: Admission  Control over a
              DiffServ Region", draft-briscoe-tsvwg-cl-architecture-04
              (work in progress), October 2006.

              Briscoe, B., "Pre-Congestion Notification marking",
              draft-briscoe-tsvwg-cl-phb-03 (work in progress),
              October 2006.

              Chan, K., "Pre-Congestion Notification Problem Statement",
              draft-chan-pcn-problem-statement-01 (work in progress),
              October 2006.

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              Charny, A., Zhang, X., Faucheur, F., and V. Liatsos, "Pre-
              Congestion Notification Using Single Marking for Admission
              and  Termination", draft-charny-pcn-single-marking-03
              (work in progress), November 2007.

              Morita, N. and G. Karlsson, "Framework of Priority
              Promotion Scheme", draft-morita-tsvwg-pps-01 (work in
              progress), October 2003.

              International Telecommunications Union, "Multilevel
              Precedence and Preemption Service", 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.

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

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

   [RFC2904]  Vollbrecht, J., Calhoun, P., Farrell, S., Gommans, L.,
              Gross, G., de Bruijn, B., de Laat, C., Holdrege, M., and
              D. Spence, "AAA Authorization Framework", RFC 2904,
              August 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.

   [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

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              Behavior)", RFC 3247, March 2002.

   [RFC3260]  Grossman, D., "New Terminology and Clarifications for
              Diffserv", RFC 3260, April 2002.

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

   [RFC4080]  Hancock, R., Karagiannis, G., Loughney, J., and S. Van den
              Bosch, "Next Steps in Signaling (NSIS): Framework",
              RFC 4080, June 2005.

   [RFC4190]  Carlberg, K., Brown, I., and C. Beard, "Framework for
              Supporting Emergency Telecommunications Service (ETS) in
              IP Telephony", RFC 4190, November 2005.

   [RFC4504]  Sinnreich, H., Lass, S., and C. Stredicke, "SIP Telephony
              Device Requirements and Configuration", RFC 4504,
              May 2006.

   [RFC4542]  Baker, F. and J. Polk, "Implementing an Emergency
              Telecommunications Service (ETS) for Real-Time Services in
              the Internet Protocol Suite", RFC 4542, May 2006.

   [RFC4594]  Babiarz, J., Chan, K., and F. Baker, "Configuration
              Guidelines for DiffServ Service Classes", RFC 4594,
              August 2006.

   [RFC5127]  Chan, K., Babiarz, J., and F. Baker, "Aggregation of
              DiffServ Service Classes", RFC 5127, February 2008.

Authors' Addresses

   Fred Baker
   Cisco Systems
   Santa Barbara, California  93117

   Phone: +1-408-526-4257

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   James Polk
   Cisco Systems
   Richardson, Texas  75082

   Phone: +1-817-271-3552

   Martin Dolly
   AT&T Labs
   Middletown Township, New Jersey  07748

   Phone: +1-732-420-4574

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Full Copyright Statement

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