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Internet Engineering Task Force                    Ken Carlberg
INTERNET DRAFT                                     UCL
February 15, 2002                                  Ian Brown
                                                   UCL



             Framework for Supporting IEPS in IP Telephony
                 <draft-carlberg-ieprep-framework-00.txt>



Status of this Memo

   This document is an Internet-Draft and is in full conformance with
   all provisions of Section 10 of RFC2026 [1].

   Internet-Drafts are working documents of the Internet Engineering
   Task Force (IETF), its areas, and its working groups. Note that other
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   For potential updates to the above required-text see:
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Abstract

   This document presents a framework for supporting authorized
   emergency related communication within the context of IP telephony.
   We present a series of objectives that reflect a general view of how
   authorized emergency service, in line with the International
   Emergency Preparedness Scheme (IEPS), should be realized within
   today's IP architecture and service models.  From these objectives,
   we present a corresponding set of functional requirements, which
   provide a more specific set of recommendations regarding existing
   IETF protocols.  Finally, we present two scenarios that act as
   guiding models for the objectives and functions listed in this
   document.  These, models, coupled with an example of an existing
   service in the PSTN, contribute to a constrained solution space.



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1. Introduction

   The Internet has become the primary target for worldwide
   communications.  This is in terms of recreation, business, and
   various imaginative reasons for information distribution. A constant
   fixture in the evolution of the Internet has been the support of Best
   Effort as the default service model.  Best Effort, in general terms,
   infers that the network will attempt to forward traffic to the
   destination as best as it can with no guarantees being made, nor any
   resources reserved, to support specific measures of Quality of
   Service (QoS). An underlying goal is to be 'fair' to all the traffic
   in terms of the resources used to forward it to the destination.

   In an attempt to go beyond best effort service, [2] presented an
   overview of Integrated Services (int-serv) and its inclusion into the
   Internet architecture.  This was followed by [3], which specified the
   RSVP signaling protocol used to convey QoS requirements.  With the
   addition of [4] and [5], specifying control load (bandwidth bounds)
   and guaranteed service (bandwidth & delay bounds) respectively, a
   design existed to achieve specific measures of QoS for an end-to-end
   flow of traffic traversing an IP network.  In this case, our
   reference to a flow is one that is granular in definition and
   applying to specific application sessions.

   From a deployment perspective (as of the date of this document),
   int-serv has been predominantly constrained to stub intra-domain
   paths, at best resembling isolated "island" reservations for specific
   types of traffic (e.g., audio and video) by stub domains.  [6] and
   [7] will probably contribute to additional deployment of int-serv to
   Internet Service Providers (ISP) and possibly some inter-domain
   paths, but it seems unlikely that the original vision of end-to-end
   int-serv between hosts in source and destination stub domains will
   become a reality in the near future (the mid- to far-term is a
   subject for others to contemplate).

   In 1998, the IETF produced [8], which presented an architecture for
   Differentiated Services (diff-serv).  This effort focused on a more
   aggregated perspective and classification of packets than that of
   [2].  This is accomplished with the recent specification of the
   diff-serv field in the IP header (in the case of IPv4, it replaced
   the old ToS field).  This new field is used for code points
   established by IANA, or set aside as experimental.  It can be
   expected that sets of microflows, a granular identification of a set
   of packets, will correspond to a given code point, thereby achieving
   an aggregated treatment of data.

   One constant in the introduction of new service models has been the
   designation of Best Effort as the default service model.  If traffic



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   is not, or cannot be, associated as diff-serv or int-serv, then it is
   treated as Best Effort and uses what resources are made available to
   it.

   Beyond the introduction of new services, the continued pace of
   additional traffic load experienced by ISPs over the years has
   continued to place a high importance for intra-domain traffic
   engineering.  The explosion of IETF contributions, in the form of
   drafts and RFCs produced in the area of Multi Protocol Label
   Switching (MPLS), exemplifies the interest in versatile and
   manageable mechanisms for intra-domain traffic engineering.  One
   interesting observation is the work involved in supporting QoS
   sensitive traffic like Voice over IP (VoIP). Specifically, we refer
   to the work in progress discussion of a framework to support VoIP
   using MPLS [9], and the inclusion of fault tolerance [10].  This
   latter item can be viewed as being similar to "crank-back", a term
   used to describe the means by which the Public Switched Telephone
   Network (PSTN) routes around congested switches.


1.2  Emergency Related Data

   The evolution of the IP service model architecture has traditionally
   centered on the type of application protocols used over a network.
   By this we mean that the distinction, and possible bounds on QoS,
   usually centers on the type of application (e.g., audio video tools).

   While protocols like SMTP [11] and SIP [12] have embedded fields
   denoting "priority", there has not been a previous IETF standards
   based effort to state or define what this distinction means with
   respect to the underlying network and how it should be supported.
   Given the emergence of IP telephony, a natural inclusion of it as
   part of a telco carriers backbone network, or into the Internet as a
   whole, implies the ability to support existing emergency related
   services.  Typically, one associates emergency calls with "911"
   telephone service in the U.S., or "999" in the U.K. -- both of which
   are attributed to national boundaries and accessible by the general
   public.  Outside of this exists emergency telephone services that
   involved authorized usage, as described in the following subsection.

   GETS is an emergency telecommunications service available in the U.S.
   and established by the National Communications System (NCS) -- an
   office established by the White House under an executive order.
   Unlike "911", it is only accessible by authorized individuals.  The
   majority of these individuals are from various government agencies
   like the Department of Transportation, NASA, the Department of
   Defense, and the Federal Emergency Management Agency (to name but a
   few).  In addition, individuals from private industry



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   (telecommunications companies, utilities, etc.) that are involved in
   criticial infrastructure recovery operations are also provided access
   to GETS.

   The purpose of GETS is to increase the probability that phone service
   will be available to selected government agency personnel in times of
   emergencies, such as hurricanes, earthquakes, and other disasters
   that may produce a burden in the form of call blocking (i.e.,
   congestion) on the U.S. Public Switched Telephone Network by the
   general public.

   The key aspect is that GETS only supports a probabilistic approach to
   call completion, as opposed to call preemption.  This distinction is
   important because emergency systems like GETS are not allowed to
   terminate existing calls in order to allow a GETS call to be
   established.  Thus, the mechanisms and specifications that comprise
   GETS only focus on increasing the chances that a particular telephone
   call will be established.

   The basis for GETS with respect to Signaling System 7 (SS7) support
   is found in the T1.631 protocol on High Probability of Completion
   (HPC) network capability [13].  This document describes the
   specification of a National Security and Emergency Preparedness
   (NS/EP) Calling Party Category (CPC) code point used for SS7 ISDN
   User Part (ISUP) Initial Address Message (IAM).  In the presense of
   this code point, Local Exchange Carriers (LEC) will attempt (if
   necessary and if the option is supported) to route the call through
   alternate inter-exchange carriers (IXC) if it cannot complete the
   call through the default IXC.

   The procedure for a user (i.e., a person) establishing a GETS call is
   as follows:

       1) Dial a non-geographical area code number: 710-XXX-XXXX
       2) Dial a PIN used to authenticate the call
       3) Dial the actual destination number to be reached

   In conjunction with the above, the source LEC (where the call
   originated) attempts to establish the call through an IXC.  This is
   done even if the destination number is within the LEC itself.  If the
   IXC cannot forward the call to the destination LEC, then the source
   LEC attempts to route the call through an alternate IXC.  If
   alternate IXCs cannot help establish the call, then a busy signal is
   finally returned to the user.  Otherwise, the call is completed and
   retains the same quality of service as all other telephone calls.

   The HPC component of GETS is not ubiquitously supported by the U.S.
   PSTN.  The only expectation is that the 710 area code is recognized



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   by all carriers.  Additional support is conditional and dependent
   upon the equivalent service level agreements established between the
   U.S. Government and various telco carriers.  Thus, the default end-
   to-end service for establishing a GETS call can be viewed as best
   effort and associated with the same priority as calls from the
   general public.  The exception to this rule is when the call is
   forwarded through carriers that have been contracted through a
   service level agreement to support HPC, which results in a higher
   probability that the GETS call will be established.

   It should be noted from the above description that GETS is separate
   and unrelated to other emergency services like "911".


1.2.1  International Emergency Preparedness Scheme (IEPS)

   [18] is a recent ITU standard that describes emergency related
   communications over international telephone service (Note, this
   document has also been published as a draft-RFC in [28].  While
   systems like GETS are national in scope, IEPS acts as an extension to
   local or national authorized emergency call establishment and
   provides a building block for a global service.

   As in the case of GETS, IEPS promotes mechanisms like extended
   queuing, alternate routing, and exemption from restrictive management
   controls in order to increase the probability that international
   emergency calls will be established.  The specifics of how this is to
   be accomplished are to be defined in future ITU document(s).


1.3  Scope of this Document

   The scope of this document centers on the support of IEPS within the
   context of IP telephony, though not necessarily Voice over IP.  We
   make a distinction between these two by treating IP telephony as a
   subset of VoIP, where in the former we assume some form of
   application layer signaling is used to explicitly establish and
   maintain voice data traffic.  This explicit signaling capability
   provides the hooks from which VoIP traffic can be bridged to the
   PSTN.

   An example of this distinction is when the Redundant Audio Tool (RAT)
   [14] begins sending VoIP packets to a unicast (or multicast)
   destination.  RAT does not use explicit signaling like SIP to
   establish an end-to-end call between two users.  It simply sends data
   packets to the target destination.  On the other hand, "SIP phones"
   are host devices that use a signaling protocol to establish a call
   signal before sending data towards the destination.



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   Beyond this, part of our motivation in writing this document is to
   provide a reference point for ISPs and carriers so that they have an
   understanding of objectives and accompanying functional requirements
   used to support IEPS related IP telephony traffic.  In addition, we
   also wish to provide a reference point for potential customers (users
   of IEPS) in order to constrain their expectations.  In particular, we
   wish to avoid any temptation of trying to replicate the exact
   capabilities of existing emergency voice service currently available
   in the PSTN to that of IP and the Internet.  If nothing else,
   intrinsic differences between the two communications architectures
   precludes this from happening.  Note, this does not prevent us from
   borrowing design concepts or objectives from existing systems.

   Section 2 presents several primary objectives that articulate what is
   considered important in supporting IEPS related IP telephony traffic.
   These objectives represent a generic set of goals and capabilities
   attributed to supporting IEPS based IP telephony.  Section 3 presents
   additional value added objectives.  These are capabilities that are
   viewed as useful, but not critical in support of IEPS.  Section 4
   presents a series of functional requirements that stem from the
   objectives articulated in section 2.  Finally, Section 5 presents two
   scenarios in IEPS that exist or are being deployed over IP networks.
   These are not all-inclusive scenarios, nor are they the only ones
   that can be articulated.  However, they do show cases where some of
   the functional requirements apply, and where some do not.

   Finally, we need to state that this document focuses its attention on
   the IP layer and above.  Specific operational procedures pertaining
   to Network Operation Centers (NOC) or Network Information Centers
   (NIC) are outside the scope of this document.  This includes the
   "bits" below IP, other specific technologies, and service level
   agreements between ISPs and carriers with regard to dedicated links.


2. Objective

   The support of IEPS within IP telephony can be realized in the form
   of several primary objectives.  These objectives define the generic
   functions or capabilities associated with IEPS, and the scope of the
   support needed to achieve these capabilities.  From this generic set
   of objectives, we present specific functional requirements of
   existing IP protocols (presented below in section 3).

   There are two underlying goals in the selection of these objectives.
   One goal is to produce a design that maximizes the use of existing IP
   protocols and minimizes the set of additional specifications needed
   to support IP-telephony based IEPS.  Thus, with the inclusion of
   these minimal augmentations, the bulk of the work in achieving IEPS



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   over an IP network that is connected or unconnected to the Internet
   involves operational issues.  Examples of this would be the
   establishment of Service Level Agreements (SLA) with ISPs, and/or the
   provisioning of traffic engineered paths for IEPS-related telephony
   traffic.

   A second underlying goal in selecting the following objectives is to
   take into account experiences from an existing emergency-type
   communication system (as described in section 1.2.1) as well as the
   existing restrictions and constraints placed by some countries.  In
   the former case, we do not attempt to mimic the system, but rather
   extract information as a reference model.  With respect to
   constraints based on laws or agency regulations, this would normally
   be considered outside of the scope of any IETF document.  However,
   these constraints act as a means of determining the lowest common
   denominator in specifying technical functional requirements.  If such
   constraints do not exist, then additional functions can be added to
   the baseline set of functions.  This last item will be expanded upon
   in the description of Objective #3 below.

   The following list of objectives are termed primary because they
   pertain to that which defines the underlying goals of IEPS in
   relation to IP telephony.  However, the primary objectives are not
   meant to dictate major overhauls of existing IP protocols, nor do
   they require new protocols to be developed.

   Primary Objectives in support of authorized emergency calls:

       1) High Probability of Call Completion
       2) Interaction with PSTN
       3) Distinction of IEPS data traffic
       4) Non-preemptive action
       5) Non-ubiquitous support
       6) Authenticated service

   The first objective is the crux of our work because it defines our
   expectations for both data and call signaling for IP telephony.  As
   stated, our objective is achieving a high probability that emergency
   related calls (both data and signaling) will be forwarded through an
   IP network.  Specifically, we envision the relevance of this
   objective during times of congestion, the context of which we
   describe further below in this section.  The critical word in this
   objective is "probability", as opposed to assurance or guarantee --
   the latter two placing a higher burden on the network.  It stands to
   reason, though, that the word "probability" is a less tangible
   description that cannot be easily quantified.  It is relative in
   relation to other traffic transiting the same network.  Objectives 3
   through 5 below help us to qualify the term probability in the



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   context of other objectives.

   The second objective involves the interaction of IP telephony
   signaling with existing PSTN support for emergency related voice
   communications.  As mentioned above in Section 1.2, standard T1.631
   [26] specifies emergency code points for SS7.  Specifically, the
   National Security and Emergency Preparedness (NS/EP) Calling Party
   Category code point is defined for ISUP IAM messages used by SS7
   [26].  Hence, our objective in the interaction between the PSTN and
   IP telephony with respect to IEPS (and national indicators) is a
   direct mapping between related code points.

   The third objective focuses on the ability to distinguish IEPS data
   packets from other types of VoIP packets.  With such an ability,
   transit providers can more easily ensure that service level
   agreements relating to IEPS are adhered.  Note that we do not assume
   that the actions taken to distinguish IEPS type packets is easy.
   Nor, in this section, do we state the form of this distinction.  We
   simply present the objective of identifying flows that relate to IEPS
   versus others that traverse a transit network.

   At an abstract level, the forth objective pertains to the actions
   taken when an IP telephony call, via a signaling protocol such as
   SIP, cannot be forwarded because the network is experiencing a form
   of congestion.  We state this in general terms because of two
   reasons: a) there may exist applications other than SIP, like H.248,
   used for call establishment, and b) congestion may come in several
   forms.  For example, congestion may exist at the IP packet layer with
   respect to queues being filled to their configured limit. Congestion
   may also arise from resource allocation attributed per call or
   aggregated sets of calls.  In this latter case, while there may exist
   resources to forward the packets, a signaling server may have reached
   its limit as to how many telephony calls it will support while
   retaining toll-quality service per call.  Typically, one terms this
   form of congestion as call blocking.  Note that we do not address the
   case when congestion occurs at the bit level below that of IP, due to
   the position that it is outside the scope of IP and the IETF.

   So, given the existence of congestion in its various forms, our
   objective is to support IEPS-related IP telephony call signaling and
   data traffic via non-preemptive actions taken by the network.  More
   specifically, we associate this objective in the context of IP
   telephony acting as part of the Public Telephone Network (PTN).
   This, as opposed to the use of IP telephony within a private or stub
   network.  In section 5 below, we expand on this through the
   description of two distinct scenarios of IP telephony and its
   operation with IEPS and the PSTN.




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   It is important to mention that this is a default objective
   influenced by existing laws & regulations.  Those countries not bound
   by these restrictions can remove this objective and make provisions
   to enforce preemptive action.  In this case, it would probably be
   advantageous to deploy a signaling system similar to that proposed in
   [15], wherein multiple levels of priority are defined and preemption
   via admission control from SIP servers is enforced.

   The fifth objective stipulates that we do not advocate the need or
   expectation for ubiquitous support of IEPS across all administrative
   domains of the Internet.  While it would be desirable to have
   ubiquitous support, we feel the reliance of such a requirement would
   doom even the contemplation of supporting IEPS by the IETF and the
   expected entities (e.g., ISPs and vendors) involved in its
   deployment.

   We use the existing GETS service in the U.S. as an existing example
   in which emergency related communications does not need to be
   ubiquitous.  As mentioned previously, the measure and amount of
   support provided by the U.S. PSTN for GETS is not ubiquitous across
   all U.S. Inter-exchange Carriers (IXC) nor Local Exchange Carriers
   (LEC).  Given the fact that GETS still works within this context, it
   is our objective to follow this deployment model such that we can
   accomplish the first objective listed above -- a higher probability
   of call completion than that of normal IP telephony call traffic.

   Our final objective is that only authorized users may use the
   services outlined in this framework.  GETS users are authenticated
   using a PIN provided to the telecommunications carrier, which signals
   approval back to the user's local exchange over SS7.  In an IP
   network, the authentication center will need to securely signal back
   to the IP ingress point that a given user is authorized to send IEPS
   related flows.  Similarly, transit networks with IEPS SLAs must
   securely interchange authorized IEPS traffic.  In both cases, IPSec
   authentication transforms may be used to protect this traffic.  This
   is entirely separate from end-to-end IPSec protection of user
   traffic, which will be configured by users.  IP-PSTN gateways must
   also be able to securely signal IEPS authorization for a given flow.
   As these gateways are likely to act as SIP servers, we further
   consider the use of SIP's security functions to aid this objective.


3. Value Added Objective

   This objective is viewed as being helpful in achieving a high
   probability of call completion.  Its realization within an IP network
   would be in the form of new protocols or enhancements to existing
   ones.  Thus, objective listed in this section are treated as value



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   added -- an expectation that their existence would be beneficial, and
   yet not viewed as critical to support IEPS related IP telephony
   traffic.


3.1 Alternate Path Routing

   This objective involves the ability to discover and use a different
   path to route IP telephony traffic around congestion points and thus
   avoid them.  Ideally, the discovery process would be accomplished in
   an expedient manner (possibly even a priori to the need of its
   existence).  At this level, we make no requirements as to how the
   alternate path is accomplished, or even at which layer it is achieved
   -- e.g., the network versus the application layer.  But this kind of
   capability, at least in a minimal form, would help contribute to
   increasing the probability of call completion of IEPS traffic by
   making use of noncongested alternate paths.  We use the term "minimal
   form" to concede the fact that care must be taken in how the system
   provides alternate paths so it does not significantly contribute to
   the congestion that is to be avoided (e.g., via excess
   control/discovery messages).

   At the time that this document was written, we can identify two
   work-in-progress areas in the IETF that can be helpful in providing
   alternate paths for call signaling.  The first is [21], which is
   focused on network layer routing and describes enhancements to the
   LDP specification of MPLS to help achieve fault tolerance.  This in
   itself does not provide alternate path routing, but rather helps
   minimize loss in intradomain connectivity when MPLS is used within a
   domain.

   The second effort comes from the IP Telephony working group and
   involves Telephony Routing over IP (TRIP).  To date, a framework
   document [19] has been published as an RFC which describes the
   discovery and exchange of IP telephony gateway routing tables between
   providers.  The TRIP protocol [22], a supplemental work in progress,
   specifies application level telephony routing regardless of the
   signaling protocol being used (e.g., SIP or H.323).  TRIP is modeled
   after BGP-4 and advertises reachability and attributes of
   destinations.  In its current form, several attributes have already
   been defined, such as LocalPreference and MultiExitDisc.  Upon
   standardization of TRIP, additional attributes can be registered with
   IANA.  Initially, we would recommend two additional attributes that
   would relate to emergency related flows.  These being:







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      EmergencyMultiExitDisc

        The EmergencyMultiExitDisc attribute is similar to the
        MultiExitDisc in that it is an inter-domain attribute used
        to express a preference of one or more links over others
        between domains.  Unlike the MultiExitDisc, this attribute
        specifically identifies links that are preferred for emergency
        related calls that span domains.

      EmergencyLocalPreference

        The EmergencyLocalPreference attribute is similar to the
        LocalPreference in that it is an intra-domain attribute used
        to inform other LSs of the local LSs preference for a given
        route.  The difference between the two types attributes is
        that the preferred route specifically relates to emergency-type
        calls (e.g., 911).  This attribute has no significance between
        domains.  Local policy determines if there is an association
        between the EmergencyLocalPreference and the
        EmergencyMultiExitDisc attribute.



3.2 End-to-End Fault Tolerance

   This topic involves the work that has been done in trying to
   compensate for lossy networks providing best effort service.  In
   particular, we focus on the use of a) Forward Error Correction (FEC),
   and b) redundant transmissions that can be used to compensate for
   lost data packets.  (Note that our aim is fault tolerance, as opposed
   to an expectation of always achieving it).

   In the former case, additional FEC data packets are constructed from
   a set of original data packets and inserted into the end-to-end
   stream.  Depending on the algorithm used, these FEC packets can
   reconstruct one or more of the original set that were lost by the
   network.  An example may be in the form of a 10:3 ratio, in which 10
   original packets are used to generate three additional FEC packets.
   Thus, if the network loses 30% or less number of packets, then the
   FEC scheme will be able to compensate for that loss.  The drawback to
   this approach is that to compensate for the loss, a steady state
   increase in offered load has been injected into the network.  This
   makes an arguement that the act of protection against loss has
   contributed to additional pressures leading to congestion, which in
   turn helps trigger packet loss.  In addition, in using a ratio of
   10:3, the source (or some proxy) must 'hold' all 10 packets in order
   to construct the three FEC packets.  This contributes to the end-to-
   end delay of the packets as well as minor bursts of load in addition



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   to changes in jitter.

   The other form of fault tolerance we discuss involves the use of
   redundant transmissions. By this we mean the case in which an
   original data packet is followed by one or more redundant packets.
   At first glance, this would appear to be even less friendly to the
   network than that of adding FEC packets.  However, the encodings of
   the redundant packets can be of a different type (or even transcoded
   into a lower quality) that produce redundant data packets that are
   significantly smaller than the original packet.

   Two RFCs [24, 25] have been produced that define RTP payloads for FEC
   and redundant audio data.  An implementation example of a redundant
   audio application can be found in [14].  We note that both FEC and
   redundant transmissions can be viewed as rather specific and to a
   degree tangential solutions regarding packet loss and emergency
   communications.  Hence, these topics are placed under the category of
   value added objectives.


4. Functional Requirements

   In this section, we take the objectives presented above and specify a
   corresponding set of functional requirements to achieve them.  Given
   that the objectives are predominantly atomic in nature, the
   corresponding functional requirements are to be viewed separately
   with no specific dependency upon each other as a whole.  They may be
   complimentary with each other, but there is no need for all to exist
   given different scenarios of operation, and that IEPS support is not
   viewed as a ubiquitously available service.  We divide the functional
   requirements into 4 areas:

        1) Signaling
        2) Policy
        3) Traffic Engineering
        4) Security


4.1 Signaling

   Signaling is used to convey various information to either
   intermediate nodes or end nodes.  It can be out-of-band of a data
   flow, and thus in a separate flow of its own, such as SIP messages.
   It can also be in-band and part of the datagram containing the voice
   data.  This latter example could also be in the form of diff-serv
   code points in the IP packet, and/or in an extension to the RTP
   header denoting an additional classification of the payload -- the
   latter predominantly being used on an end-to-end basis.



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   In the following subsections, we discuss augmentations to three
   specific types of signaling to help support the distinction of
   emergency related communications in general, and IEPS specifically.
   We also discuss Media Gateway Control (MEGACO).


4.1.1 SIP

   With respect to application level signaling for IP telephony, we
   focus our attention to the Session Initiation Protocol (SIP).
   Currently, SIP has an existing "priority" field in the Request-
   Header-Field that distinguishes different types of sessions.  The
   five currently defined values are: "emergency", "urgent", "normal",
   "non-urgent", "other-priority".

   It is understood that the IETF prefers that no changes or additions
   be made to these existing values.  Hence, we shall follow the
   approach taken in [15] and propose the specification of a new field
   in the "Request-Header-Field" titled "Emergency-State".  This new
   field provides an additional level in distinguishing types of
   emergencies.  Currently, we would propose defining two values for
   this field:

           1) "authorized-emergency"
           2) "general-emergency"

   The former would correlate to calls that have been initiated by an
   authorized individual.  Specifically, this single SIP value would
   correlate to other authorized PSTN based code points like NS/EP and
   IEPS.  The second defined value would correlate to the more commonly
   known type of local emergency calls initiated by the general public
   (e.g., "911" in the U.S., "999", in the UK, and "112" in Germany).
   The objective is to define a single generic value that correlates to
   several similar but different types of emergency calls.

   It is important to note that this is the one functional requirement
   that is considered mandatory with respect to supporting IEPS within
   IP telephony.  We take this position because regardless of the extent
   by which the underlying network supports IEPS-based traffic, there is
   a need to distinguish IEPS sessions (i.e., authorized-emergency
   calls) from others.

   The existence of this new value in the SIP priority field allows an
   IP telephony domain to map an IEPS call to the existing NS/EP code
   point from an SS7 telephony domain.  This will help facilitate a
   seamless interaction between the PSTN and the an IP network acting as
   either an internal backbone or as a peering ISP.




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   Author's Note: The work put forth by James Polk in [15] is quite
   similar to our own in that both articulate a need to specify a more
   granular and specific means of identifying different types of
   emergencies.  Beyond the different values specified for MLPP, the
   main difference between the two efforts involves the use of
   preemption for [15], as opposed to our need to simply increase the
   probability of call completion.


4.1.2  Diff-Serv

   In accordance to [16], the differentiated services code point (DSCP)
   field is divided into three sets of values.  The first set is
   assigned by IANA.  Within this set, there are currently, three types
   of Per Hop Behaviors have been specified: Default (correlating to
   best effort forwarding), Assured Forwarding, and Expedited
   Forwarding.  The second set of DSCP values are set aside for local or
   experimental use.  The third set of DSCP values are also set local or
   experimental use, but may later be reassigned to IANNA in case the
   first set has been completely assigned.

   One recomendation involves the specification of a new type of Per-Hop
   Behavior (PHB) we term Emergency Related Forwarding (ERF).  This
   would provide a specific means of distinguishing emergency related
   traffic (signaling and user data) from other traffic.  The existence
   of this PHB then provides a baseline by which specific code points
   may be defined related to various emergency related traffic:
   authorized emergency sessions (e.g., IEPS), general public emergency
   calls (e.g., "911"), MLPP.  Aggregates would still exist with respect
   to the bundling of applications per code point.  Further, one would
   associate a forwarding paradigm aimed at a low loss rate reflective
   of the code point selected.  Hence, SIP or H.323 messages marked with
   "authorized-emergency" or "emergency" may be assigned a code point
   reflecting a lower loss rate than other type of traffic (even the
   emergency-related data flow itself).  The jitter associated with
   application layer signaling of IP telephony would be inversely
   important with respect to loss rate, and thus would be reflective of
   the code points defined for the proposed PHB.

   Another recomendation would be to define a new or fifth class for the
   existing AF PHB.  Unlike the other currently defined classes, this
   new one would be based on five levels of drop precedence.  This
   increase in the number of levels would conveniently correlate to the
   worst case scenario posed by MLPP, which has five types of
   priorities.  In addition, it would provide a higher level of variance
   that could be used to supercede the existing 3 levels used in the
   other classes.  Hence, if other non-emergency aggregate traffic where
   assigned to the class, the highest drop precedence they are assigned



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   to is (3) -- corresponding to the other four currently defined
   classes.  Emergency traffic would be set to (4) or (5), depending on
   the SLA tht has been defined.

   It is important to note that as of the time that this document was
   written, the IETF is taking a conservative approach in specifying new
   PHBs.  This is because the number of code points that can be defined
   is relatively small, and thus understandably considered a scarce
   resource.  Therefore, the possibility of a new PHB being defined for
   emergency-related traffic is at best a long term project that may or
   may not be accepted by the IETF.  In the meantime, we would initially
   recommend using the Assured Forwarding (AF) PHB [20] for
   distinguishing emergency traffic from other types of flows.
   Specifically, we would suggest the use the low drop precedence of one
   of the four defined classes of AF codepoints.  It is critical to note
   that one cannot specify an exact code point used for emergency
   related data flows because the relevance of a code point is local to
   the given diff-serv domain (i.e., they are not globally unique per
   micro-flow or aggregate of flows).  In addition, we can expect that
   the existence of a codepoint for emergency related flows is based on
   the service level agreements established with a given diff-serv
   domain.


4.1.3  RTP

   The Real-Time Transport Protocol (RTP) provides end-to-end delivery
   services for data with real-time characteristics.  The type of data
   is generally in the form of audio or video type applications, and are
   frequently interactive in nature.  RTP is typically run over UDP and
   has been designed with a fixed header that identifies a specific type
   of payload -- typically representing a specific form of application
   media.  The designers of RTP also assumed an underlying network
   providing best effort service.  As such, RTP does not provide any
   mechanism to ensure timely delivery or provide other QoS guarantees.
   However, the emergence of applications like IP telephony, as well as
   new service models, presents new environments where RTP traffic may
   be forwarded over networks that support better than best effort
   service.  Hence, the original scope and target environment for RTP
   has expanded to include networks providing services other than best
   effort.

   In 4.1.2, we discussed one means of marking a data packet for
   emergencies under the context of the diff-serv architecture.
   However, we also pointed out that diff-serv markings for specific
   PHBs are not globally unique, and may be arbitrarily removed or even
   changed by intermediary nodes or domains.  Hence, with respect to
   emergency related data packets, we are still missing an in-band



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   marking in a data packet that stays constant on an end-to-end basis.

   We have three choices in defining a persistent marking of data
   packets and thus avoid the transitory marking of diff-serv code
   points.  We can propose a new PHB dedicated for emergency type
   traffic as discussed in 4.1.2.  We can propose a specification of a
   new shim layer protocol at some location above IP.  Or, we can add a
   new specification to an existing upper layer protocol.  The first two
   cases are probably the "cleanest" architecturally, but they are long
   term efforts that will take time to support in terms of design and
   implementation.  It also may be argued that yet another shim layer
   will make the IP stack too large.  The third case, placing a marking
   in an application layer packet, has the potential to be more
   appealing depending on where the augmentation is targeted.

   An approach in realizing this third case involves an augmentation to
   RTP so that it can carry a marking that distinguishes emergency-
   related traffic from that which is not.  Specifically, one would
   define a new extention that contains a "classifier" field indicating
   the condition associated with the packet (e.g., authorized-emergency,
   emergency, normal) [29].

   An issue in defining a new extension to RTP is that its presence may
   adversely affect header compression for those implementations that
   are not expecting added optional octets in RTP packets.  In addition,
   the issue of security and authentication of such a marking remains an
   important issue and is subject to the constraints discussed below in
   section 4.4, and in more detail in [27].


4.1.4  MEGACO/H.248

   The Media Gateway Control protocol (MEGACO) [23] defines the
   interaction between a media gateway and a media gateway controller.
   [23] is viewed as common text with ITU-T Recommendation H.248 and is
   a result of applying the changes of RFC 2886 (Megaco Errata) to the
   text of RFC 2885 (Megaco Protocol version 0.8).

   In [23], the protocol specifies a Priority and Emergency field for a
   context attribute and descriptor.  The Emergency is an optional
   boolean (True or False) condition.  The Priority value, which ranges
   from 0 through 15, specifies the precedence handling for a context.

   The protocol does not specify individual values for priority.  We
   also do not recommend the definition of a well known value for the
   MEGAGO priority.  Any values set should be a function of any SLAs
   that have been established regarding the handling of emergency
   traffic.  In addition, given that priority values denote precedence



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   (according to the Megaco protocol), then by default the IEPS flows
   should probably receive the same priority as other non-emergency
   calls.  This approach follows the objective of not relying on
   preemption as the default treatment of emergency-related.


4.2  Policy

   One of the objectives listed in section 3 above is to treat IEPS-
   signaling, and related data traffic, as non-preemptive in nature.
   Further, that this treatment is to be the default mode of operation
   or service.  This is in recognition that existing regulations or laws
   of certain countries governing the establishment of SLAs may not
   allow preemptive actions (e.g., dropping existing telephony flows).
   On the other hand, the laws and regulations of other countries
   influencing the specification of SLA(s) may allow preemption, or even
   require its existence.  Given this disparity, we rely on local policy
   to determine the degree by which emergency related traffic affects
   existing traffic load of a given network or ISP.  Important note: we
   reiterate our earlier comment that laws and regulations are generally
   outside the scope of the IETF and its specification of designs and
   protocols.  However, these constraints can be used as a guide in
   producing a baseline function to be supported; in our case, a default
   policy for non-preemptive call establishment of IEPS-signaling and
   data.

   Policy can be in the form of static information embedded in various
   components (e.g., SIP servers or bandwidth brokers), or it can be
   realized and supported via COPS with respect to allocation of a
   domain's resources [17].  There is no requirement as to how policy is
   accomplished.  Instead, if a domain follows actions outside of the
   default non-preemptive action of IEPS-related communication, then we
   stipulate a functional requirement that some type of policy mechanism
   is in place to satisfy the local policies of an SLA established for
   IEPS type traffic.


4.3  Traffic Engineering

   In those cases where a network operates under the constraints of
   SLAs, one or more of which pertains to IEPS based traffic, it can be
   expected that some form of traffic engineering is applied to the
   operation of the network.  We make no requirements as to which type
   of traffic engineering mechanism is used, but that such a system
   exists and can distinguish and support IEPS signaling and data
   traffic.

   A potentially complimentary work in progress can be found in [9],



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   which articulates a framework for Voice over MPLS.  We cite the draft
   only as a point of reference, with the idea that it may be augmented
   to reflect labeled path(s) dedicated to different values in the SIP
   priority field -- such as those pertaining to emergencies.  But of
   more significance, [9] presents a specific framework for traffic
   engineering support of toll quality (i.e., a particular grade of
   service) IP telephony.

   Note: As a point of reference, existing SLAs established by the NCS
   for GETS service tend to focus on a maximum allocation of 1% of calls
   allowed to be established through a given LEC using HPC.  Once this
   limit is reached, all other GETS calls experience the same probably
   of call completion as the general public.  It is expected, and
   encouraged, that IEPS related SLAs will have a limit with respect to
   the amount of traffic distinguished as being emergency related, and
   initiated by an authorized user.


4.4  Security

   As IEPS support moves from intra-domain PSTN and IP networks to
   diffuse inter-domain pure IP, authenticated service becomes more
   complex to provide.  Where an IEPS call is carried from PSTN to PSTN
   via one carrier's backbone IP network, very little IP-specific
   security support is required.  The user authenticates herself as
   usual to the network using a PIN.  The gateway from her PSTN
   connection into the backbone IP network must be able to signal that
   the flow has IEPS priority.  Conversely, the gateway back into the
   PSTN must similarly signal the call's higher priority. A secure link
   between the gateways may be set up using IPSec or SIP security
   functionality. If the endpoint is an IP device on the carrier's
   network, the link may be set up securely from the ingress gateway to
   the end device.

   As flows traverse more than one IP network, domains whose peering
   agreements include IEPS support must have means to securely signal a
   given flow's IEPS status. They may choose to use physical link
   security and/or IPSec authentication, combined with traffic
   conditioning measures to limit the amount of IEPS traffic that may
   pass between the two domains. The inter-domain agreement may require
   the originating network to take responsibility for ensuring only
   authorized traffic is marked with IEPS priority; the downstream
   domain may still perform redundant conditioning to prevent the
   propagation of theft and denial of service attacks.  Security may be
   provided between ingress and egress gateways or IP endpoints using
   IPSec or SIP security functions.

   When a call originates from an IP device, the ingress network may



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   authorize IEPS traffic over that link as part of its user
   authentication procedures without necessarily communicating with a
   central IEPS authentication center as happens with POTS-originated
   calls. These authentication procedures may occur at the link or
   network layers, but are entirely at the discretion of the ingress
   network. That network must decide how often it should update its list
   of authorized IEPS users based on the bounds it is prepared to accept
   on traffic from recently-revoked users.


5. Key Scenarios

   There are various scenarios in which IP telephony can be realized,
   each of which can infer a unique set of functional requirements that
   may include just a subset of those listed above.  We acknowledge that
   a scenario may exist whose functional requirements are not listed
   above.  Our intention is not to consider every possible scenario by
   which support for emergency related IP telephony can be realized.
   Rather, we narrow our scope using a single guideline; we assume there
   is a signaling & data interaction between the PSTN and the IP network
   with respect to supporting emergency-related telephony traffic.  We
   stress that this does not preclude an IP-only end-to-end model, but
   rather the inclusion of the PSTN expands the problem space and
   includes the current dominant form of voice communication.

   There are two scenarios that we use as a model for determining our
   objectives and subsequent functional requirements.  These are:



   Single IP Administrative Domain
   -------------------------------

   This scenario is a direct reflection of the evolution of the PSTN.
   Specifically, we refer to the case in which data networks have
   emerged in various degrees as a backbone infrastructure connecting
   PSTN switches at its edges.  This represents a single isolated IP
   administrative domain that has no directly adjacent IP domains
   connected to it.  We show an example of this scenario below in Figure
   1.  In this example, we show two types of carriers.  One is the
   legacy carrier, whose infrastructure retains the classic switching
   architecture attributed to the PSTN.  The other is the next
   generation carrier, which uses a data network (e.g., IP) as its core
   infrastructure, and Signaling Gateways at its edges.  These gateways
   "speak" SS7 externally with peering carriers, and another protocol
   (e.g., SIP) internally, which rides on top of the IP infrastructure.





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     Legacy            Next Generation            Next Generation
     Carrier              Carrier                    Carrier
     *******          ***************             **************
     *     *          *             *     ISUP    *            *
    SW<--->SW <-----> SG <---IP---> SG <--IAM--> SG <---IP---> SG
     *     *   (SS7)  *     (SIP)   *    (SS7)    *    (SIP)   *
     *******          ***************             **************

                                        SW - Telco Switch
                                        SG - Signaling Gateway

                            Figure 1


   The significant aspect of this scenario is that all the resources of
   each IP "island" fall within a given administrative authority.
   Hence, there is no problem of retaining toll quality Grade of Service
   as the voice traffic (data and signaling) exits the IP network
   because of the existing SS7 provisioned service between carriers.
   Thus, the need for support of mechanisms like diff-serv, and an
   expansion of the defined set of Per-Hop Behaviors is reduced (if not
   eliminated) under this scenario.

   Another function that has little or no importance within the closed
   IP environment of Figure 1 is that of IP security.  The fact that
   each administrative domain peers with each other as part of the PSTN,
   means that existing security, in the form of Personal Identification
   Number (PIN) authentication (under the context of telephony
   infrastructure protection), is the default scope of security.  We do
   not claim that the reliance on a PIN based security system is highly
   secure or even desirable.  But, we use this system as a default
   mechanism in order to avoid placing additional requirements on
   existing authorized emergency telephony systems.



   Multiple IP Administrative Domains
   ----------------------------------

   We view the scenario of multiple IP administrative domains as a
   superset of the previous scenario.  Specifically, we retain the
   notion that the IP telephony system peers with the existing PSTN.  In
   addition, segments (i.e., portions of the Internet) may exchange
   signaling with other IP administrative domains via non-PSTN signaling
   protocols like SIP.






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     Legacy           Next Generation            Next Generation
     Carrier              Carrier                    Carrier
     *******          ***************            **************
     *     *          *             *            *            *
    SW<--->SW <-----> SG <---IP---> SG <--IP--> SG <---IP---> SG
     *     *   (SS7)  *     (SIP)   *    (SIP)   *    (SIP)   *
     *******          ***************            **************


                                          SW - Telco Switch
                                          SG - Signaling Gateway

                           Figure 2


   Given multiple IP domains, and the presumption that SLAs relating to
   IEPS traffic may exist between them, the need for something like
   diff-serv grows with respect to being able to distinguish the
   emergency related traffic from other types of traffic.  In addition,
   IP security becomes more important between domains in order to ensure
   that the act of distinguishing IEPS-type traffic is indeed valid for
   the given source.


8. Security Considerations

   Information on this topic is presented in sections 2 and 4.


9. References

   1  Bradner, S., "The Internet Standards Process -- Revision 3", BCP
      9, RFC 2026, October 1996.

   2  Braden, R., et. al., "Integrated Services in the Internet
      Architecture: An Overview", Informational, RFC 1633, June 1994.

   3  Braden, R., et. al., "Resource Reservation Protocol (RSVP) _
      Version 1, Functional Specification", Proposed Standard, RFC
      2205, Sept. 1997.

   4  Shenker, S., et. al., "Specification of Guaranteed Quality of
      Service", Proposed Standard, RFC 2212, Sept 1997.

   5  Wroclawski, J., "Specification for Controlled-Load Network
      Service Element", Proposed Standard, RFC 2211, Sept 1997.

   6  Gai, S., et. al., "RSVP Proxy", Internet Draft, Work in



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      Progress, July 2000.

   7  Wang, L, et. al., "RSVP Refresh Overhead Reduction by State
      Compression", Internet Draft, Work In Progress, March 2000.

   8  Blake, S., et. al., "An Architecture for Differentiated
      Service", Proposed Standard, RFC 2475, Dec. 1998.

   9  Kankkunen, A., et. al., "VoIP over MPLS Framework", Internet
      Draft, Work In Progress, July 2000.

   10 Sharma, V., et. al., "Framework for MPLS-based Recovery",
      Internet Draft, Work In Progress, September 2000.

   11 Postel, J., "Simple Mail Transfer Protocol", Standard, RFC 821,
      August 1982.

   12 Handley, M., et. al., "SIP: Session Initiation Protocol",
      Proposed Standard, RFC 2543, March 1999.

   13 ANSI, "Signaling System No. 7(SS7) _ High Probability of
      Completion (HPC) Network Capability_, ANSI T1.631, 1993.

   14 Reliable Audio Tool (RAT):
      http://www-mice.cs.ucl.ac.uk/multimedia/software/rat

   15 Polk, J., "SIP Extension for MLPP", Internet Draft, Work In
      Progress, March, 2001.

   16 Nichols, K., et. al.,"Definition of the Differentiated Services
      Field (DS Field) in the Ipv4 and Ipv6 Headers", Proposed
      Standard, RFC 2474, December 1998.

   17 Durham, D., "The COPS (Common Open Policy Service) Protocol",
      Proposed Standard, RFC 2748, Jan 2000.

   18 ITU, "International Emergency Preparedness Scheme", ITU
      Recommendation, E.106, March 2000.

   19 Rosenburg, J., Schulzrinne, H., "A Framework for Telephony Routing
      Over IP", Informational, RFC 2871, June 2000

   20 Heinanen. et. al, "Assured Forwarding PHB Group", Proposed
      Standard, RFC 2597, June 1999

   21 Farrel, A, et. al, "Fault Tolerance for LDP and CR-LDP", Internet
      Draft, Work In Progress, February 2001.




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   22 Rosenburg, J, et. al, "Telephony Routing over IP (TRIP)", Internet
      Draft, Work In Progress, April 2001.

   23 Cuervo, F., et. al, "Megaco Protocol Version 1.0", Standards
      Track, RFC 3015, November 2000

   24 Perkins, C., et al., "RTP Payload for Redundant Audio Data",
      Standards Track, RFC 2198, September, 1997

   25 Rosenburg, J., Schulzrinne, H., "An RTP Payload Format for
      Generic Forward Error Correction", Standards Track, RFC 2733,
      December, 1999.

   26 ANSI, "Signaling System No. 7, ISDN User Part", ANSI T1.113-2000,
      2000.

   27 Brown, I., "Securing IEPS over IP", White Paper,
      http://iepscheme.net/docs/secure_IEPS.doc

   28 Folts, H., "Description of an International Emergency Preference
      Scheme (IEPS) ITU-T Recommendation  E.106 (Formerly CCITT
      Recomendation), Internet Draft, Work In Progress, February 2001

   29 Carlberg, K., "The Classifier Extension Header for RTP", Internet
      Draft, Work In Progress, October 2001.


10.  Acknowledgments

   The authors would like to acknowledge the helpful comments, opinions,
   and clarifications of Stu Goldman and James Polk, as well as those
   comments received from the IEPS mailing list.


11. Author's Addresses


   Ken Carlberg
   University College London
   Department of Computer Science
   Gower Street
   London, WC1E 6BT
   United Kingdom


   Ian Brown
   University College London
   Department of Computer Science



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   Gower Street
   London, WC1E 6BT
   United Kingdom


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