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rfc:rfc4190

Network Working Group K. Carlberg Request for Comments: 4190 G11 Category: Informational I. Brown

                                                                   UCL
                                                              C. Beard
                                                                  UMKC
                                                         November 2005
                     Framework for Supporting
     Emergency Telecommunications Service (ETS) in IP Telephony

Status of This Memo

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

Copyright Notice

 Copyright (C) The Internet Society (2005).

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 Emergency
 Telecommunications Service (ETS), should be realized within today's
 IP architecture and service models.  From these objectives, we
 present a corresponding set of protocols and capabilities, 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 Public Switched Telephone Network (PSTN), contribute
 to a constrained solution space.

Carlberg, et al. Informational [Page 1] RFC 4190 IP Telephony Framework November 2005

Table of Contents

 1. Introduction ....................................................2
    1.1. Emergency Related Data .....................................4
         1.1.1. Government Emergency Telecommunications
                Service (GETS) ......................................4
         1.1.2. International Emergency Preparedness Scheme (IEPS) ..5
    1.2. Scope of This Document .....................................5
 2. Objective .......................................................7
 3. Considerations ..................................................7
 4. Protocols and Capabilities ......................................7
    4.1. Signaling and State Information ............................8
         4.1.1. SIP .................................................8
         4.1.2. Diff-Serv ...........................................8
         4.1.3. Variations Related to Diff-Serv and Queuing .........9
         4.1.4. RTP ................................................10
         4.1.5. GCP/H.248 ..........................................11
    4.2. Policy ....................................................12
    4.3. Traffic Engineering .......................................12
    4.4. Security ..................................................13
         4.4.1. Denial of Service ..................................13
         4.4.2. User Authorization .................................14
         4.4.3. Confidentiality and Integrity ......................15
    4.5. Alternate Path Routing ....................................16
    4.6. End-to-End Fault Tolerance ................................17
 5. Key Scenarios ..................................................18
    5.1. Single IP Administrative Domain ...........................18
    5.2. Multiple IP Administrative Domains ........................19
 6. Security Considerations ........................................20
 7. Informative References .........................................20
 Appendix A: Government Telephone Preference Scheme (GTPS) .........24
    A.1.  GTPS and the Framework Document ..........................24
 Appendix B: Related Standards Work ................................24
    B.1.  Study Group 16 (ITU) .....................................25
 Acknowledgements ..................................................26

1. Introduction

 The Internet has become the primary target for worldwide
 communications 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, implies
 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.

Carlberg, et al. Informational [Page 2] RFC 4190 IP Telephony Framework November 2005

 In an attempt to go beyond best effort service, [1] presented an
 overview of Integrated Services (int-serv) and its inclusion into the
 Internet architecture.  This was followed by [2], which specified the
 RSVP signaling protocol used to convey QoS requirements.  With the
 addition of [3] and [4], specifying controlled 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 applies to specific application sessions.
 From a deployment perspective (as of the date of this document),
 int-serv has been predominantly constrained to intra-domain paths, at
 best resembling isolated "island" reservations for specific types of
 traffic (e.g., audio and video) by stub domains.  [5] and [6] 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 [7], which presented an architecture for
 Differentiated Services (diff-serv).  This effort focused on a more
 aggregated perspective and classification of packets than that of
 [1].  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
 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 on 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
 related traffic engineering.  Specifically, we refer to MPLS support

Carlberg, et al. Informational [Page 3] RFC 4190 IP Telephony Framework November 2005

 of differentiated services [8], and the ongoing work in the inclusion
 of fast bandwidth recovery of routing failures for MPLS [9].

1.1. 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)
 that is being referred to.
 [10] has defined a priority field for SMTP, but it is only for
 mapping with X.400 and is not meant for general usage.  SIP [11] has
 an embedded field denoting "priority", but it is only targeted toward
 the end-user and is not meant to provide an indication to the
 underlying network or end-to-end applications.
 Given the emergence of IP telephony, a natural inclusion of its
 service is an 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 there exist emergency telephone services
 that involve authorized usage, as described in the following
 subsection.

1.1.1. Government Emergency Telecommunications Service (GETS)

 GETS is an emergency telecommunications service available in the U.S.
 and is overseen by the National Communications System (NCS) -- an
 office established by the White House under an executive order [27]
 and now a part of the Department of Homeland Security.  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 a few).  In
 addition, a select set of individuals from private industry
 (telecommunications companies, utilities, etc.) that are involved in
 critical infrastructure recovery operations are also provided access
 to GETS.
 The purpose of GETS is to achieve a high probability that phone
 service will be available to selected authorized 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.

Carlberg, et al. Informational [Page 4] RFC 4190 IP Telephony Framework November 2005

 GETS is based in part on the ANSI T1.631 standard, specifying a High
 Probability of Completion (HPC) for SS7 signaling [12][24].

1.1.2. International Emergency Preparedness Scheme (IEPS)

 [25] is a recent ITU standard that describes emergency-related
 communications over the international telephone service.  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).

1.2. Scope of This Document

 The scope of this document centers on the near and mid-term support
 of ETS within the context of IP telephony versus Voice over IP.  We
 make a distinction between these two by treating IP telephony as a
 subset of VoIP, where in the former case, we assume that 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 Robust Audio Tool (RAT)
 [13] 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
 before sending data towards the destination.
 One other aspect we should probably assume exists with IP Telephony
 is an association of a target level of QoS per session or flow.  [28]
 makes an argument that there is a maximum packet loss and delay for
 VoIP traffic, and that both are interdependent.  For delays of
 ~200ms, a corresponding drop rate of 5% is deemed acceptable.  When
 delay is lower, a 15-20% drop rate can be experienced and still be
 considered acceptable.  [29] discusses the same topic and makes an
 argument that packet size plays a significant role in what users
 tolerate as "intelligible" VoIP.  The larger the packet, correlating
 to a longer sampling rate, the lower the acceptable rate of loss.
 Note that [28, 29] provide only two of several perspectives in
 examining VoIP.  A more in-depth discussion on this topic is outside

Carlberg, et al. Informational [Page 5] RFC 4190 IP Telephony Framework November 2005

 the scope of this document, though it should be noted that the choice
 of codec can significantly alter the above results.
 Regardless of a single and definitive characteristic for stressed
 conditions, it would seem that interactive voice has a lower
 threshold of some combinations of loss/delay/jitter than elastic
 applications such as email or web browsers.  This places a higher
 burden on the problem of supporting VoIP over the Internet.  This
 problem is further compounded when toll-quality service is expected
 because it assumes a default service model that is better than best
 effort.  This, in turn, can increase the probability that a form of
 call-blocking can occur with VoIP or IP telephony traffic.
 Beyond this, part of our motivation in writing this document is to
 provide a framework for ISPs and telephony carriers to understand the
 objectives used to support ETS-related IP telephony traffic.  In
 addition, we also wish to provide a reference point for potential
 customers 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 that are 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 ETS-related IP telephony traffic.
 These objectives represent a generic set of goals and desired
 capabilities.  Section 3 presents additional value-added objectives,
 which are viewed as useful, but not critical.  Section 4 presents
 protocols and capabilities that relate or can play a role in support
 of the objectives articulated in Section 2.  Finally, Section 5
 presents two scenarios that currently exist or are being deployed in
 the near term over IP networks.  These are not all-inclusive
 scenarios, nor are they the only ones that can be articulated ([34]
 provides a more extensive discussion on the topology scenarios
 related to IP telephony).  However, these scenarios do show cases
 where some of the protocols discussed in Section 4 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 telephony carriers with regard to
 dedicated links.

Carlberg, et al. Informational [Page 6] RFC 4190 IP Telephony Framework November 2005

2. Objective

 The objective of this document is to present a framework that
 describes how various protocols and capabilities (or mechanisms) can
 facilitate and support the traffic from ETS users.  In several cases,
 we provide a bit of background in each area so that the reader is
 given some context and a more in-depth understanding.  We also
 provide some discussion on aspects about a given protocol or
 capability that could be explored and potentially advanced to support
 ETS.  This exploration is not to be confused with specific solutions
 since we do not articulate exactly what must be done (e.g., a new
 header field, or a new code point).

3. Considerations

 When producing a solution, or examining existing protocols and
 mechanisms, there are some things that should be considered.  One is
 that inter-domain ETS communications should not rely on ubiquitous or
 even widespread support along the path between the end points.
 Potentially, at the network layer there may exist islands of support
 realized in the form of overlay networks.  There may also be cases
 where solutions may be constrained on an end-to-end basis (i.e., at
 the transport or application layer).  It is this diversity and
 possibly partial support that needs to be taken into account by those
 designing and deploying ETS-related solutions.
 Another aspect to consider is that there are existing architectures
 and protocols from other standards bodies that support emergency-
 related communications.  The effort in interoperating with these
 systems, presumably through gateways or similar types of nodes with
 IETF protocols, would foster a need to distinguish ETS flows from
 other flows.  One reason would be the scenario of triggering ETS
 service from an IP network.
 Finally, we take into consideration the requirements of [35, 36] in
 discussing the protocols and mechanisms below in Section 4.  In doing
 this, we do not make a one-to-one mapping of protocol discussion a
 requirement.  Rather, we make sure the discussion of Section 4 does
 not violate any of the requirements in [35, 36].

4. Protocols and Capabilities

 In this section, we take the objectives presented above and present a
 set of protocols and capabilities that can be used to achieve them.
 Given that the objectives are predominantly atomic in nature, the
 measures used to address them are to be viewed separately with no
 specific dependency upon each other as a whole.  Various protocols
 and capabilities may be complimentary to each other, but there is no

Carlberg, et al. Informational [Page 7] RFC 4190 IP Telephony Framework November 2005

 need for all to exist, given different scenarios of operation; and
 ETS support is not expected to be an ubiquitously available service.
 We divide this section into 5 areas:
    1) Signaling
    2) Policy
    3) Traffic Engineering
    4) Security
    5) Routing

4.1. Signaling and State Information

 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 be in-band and part of the state information in a datagram
 containing the voice data.  This latter example could be realized in
 the form of diff-serv code points in the IP packet.
 In the following subsections, we discuss the current state of some
 protocols and their use in providing support for ETS.  We also
 discuss potential augmentations to different types of signaling and
 state information to help support the distinction of emergency-
 related communications in general.

4.1.1. SIP

 With respect to application-level signaling for IP telephony, we
 focus our attention on 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 values currently defined are: "emergency", "urgent", "normal",
 "non-urgent", "other-priority".  These values are meant to convey
 importance to the end-user and have no additional semantics
 associated with them.
 [14] is an RFC that defines the requirements for a new header field
 for SIP in reference to resource priority.  The requirements are
 meant to lead to a means of providing an additional measure of
 distinction that can influence the behavior of gateways and SIP
 proxies.

4.1.2. Diff-Serv

 In accordance with [15], 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 that have been specified: Default (correlating

Carlberg, et al. Informational [Page 8] RFC 4190 IP Telephony Framework November 2005

 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 aside
 for local or experimental use, but may later be reassigned to IANA if
 the first set has been completely assigned.
 One approach discussed on the IEPREP mailing list is the
 specification of a new Per-Hop Behaviour (PHB) for emergency-related
 flows.  The rationale behind this idea is that it would provide a
 baseline by which specific code points may be defined for various
 emergency-related traffic: authorized emergency sessions (e.g., ETS),
 general public emergency calls (e.g., "911"), Multi-Level Precedence
 and Preemption (MLPP) [19], etc.  However, in order to define a new
 set of code points, a forwarding characteristic must also be defined.
 In other words, one cannot simply identify a set of bits without
 defining their intended meaning (e.g., the drop precedence approach
 of Assured Forwarding).  The one caveat to this statement are the set
 of DSCP bits set aside for experimental purposes.  But as the name
 implies, experimental is for internal examination and use and not for
 standardization.
    Note:
       It is important to note that at the time this document was
       written, the IETF had been taking a conservative approach in
       specifying new PHBs.  This is because the number of code points
       that can be defined is relatively small and is 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 near term, we would initially suggest using the Assured
       Forwarding (AF) PHB [18] for distinguishing emergency traffic
       from other types of flows.  At a minimum, AF could be used for
       the different SIP call signaling messages.  If the Expedited
       Forwarding (EF) PHB [40] was also supported by the domain, then
       it would be used for IP telephony data packets.  Otherwise,
       another AF class would be used for those data flows.

4.1.3. Variations Related to Diff-Serv and Queuing

 Scheduling mechanisms like Weighted Fair Queueing and Class Based
 Queueing are used to designate a percentage of the output link
 bandwidth that would be used for each class if all queues were
 backlogged.  Its purpose, therefore, is to manage the rates and
 delays experienced by each class.  But emergency traffic may not
 necessarily require QoS perform any better or differently than non-

Carlberg, et al. Informational [Page 9] RFC 4190 IP Telephony Framework November 2005

 emergency traffic.  It may just need higher probability of being
 forwarded to the next hop, which could be accomplished simply by
 dropping precedences within a class.
 To implement preferential dropping between classes of traffic, one of
 which is emergency traffic, one would probably need to use a more
 advanced form of Active Queue Management (AQM).  Current
 implementations use an overall queue fill measurement to make
 decisions; this might cause emergency classified packets to be
 dropped.  One new form of AQM could be a Multiple Average-Multiple
 Threshold approach, instead of the Single Average-Multiple Threshold
 approach used today.  This allows creation of drop probabilities
 based on counting the number of packets in the queue for each drop
 precedence individually.
 So, it could be possible to use the current set of AF PHBs if each
 class were reasonably homogenous in the traffic mix.  But one might
 still have a need to differentiate three drop precedences within
 non-emergency traffic.  If so, more drop precedences could be
 implemented.  Also, if one wanted discrimination within emergency
 traffic, as with MLPP's five levels of precedence, more drop
 precedences might also be considered.  The five levels would also
 correlate to a recent effort in Study Group 11 of the ITU to define 5
 levels for Emergency Telecommunications Service.

4.1.4. 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 is
 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 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, present 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

Carlberg, et al. Informational [Page 10] RFC 4190 IP Telephony Framework November 2005

 emergency related data packets, we are still missing an in-band
 marking in a data packet that stays constant on an end-to-end basis.
 There are three choices in defining a persistent marking of data
 packets and thus avoiding the transitory marking of diff-serv code
 points.  One can propose a new PHB dedicated for emergency type
 traffic as discussed in 4.1.2.  One can propose a specification of a
 new shim layer protocol at some location above IP.  Or, one can add a
 new specification to an existing application layer protocol.  The
 first two cases are probably the "cleanest" architecturally, but they
 are long term efforts that may not come to pass because of a limited
 number of diff-serv code points and the contention that yet another
 shim layer will make the IP stack too large.  The third case, placing
 a marking in an application layer packet, also has drawbacks; the key
 weakness being the specification of a marking on a per-application
 basis.
 Discussions have been held in the Audio/Visual Transport (AVT)
 working group on augmenting RTP so that it can carry a marking that
 distinguishes emergency-related traffic from that which is not.
 Specifically, these discussions centered on defining a new extension
 that contains a "classifier" field indicating the condition
 associated with the packet (e.g., authorized-emergency, emergency,
 normal) [26].  The rationale behind this idea was that focusing on
 RTP would allow one to rely on a point of aggregation that would
 apply to all payloads that it encapsulates.  However, the AVT group
 has expressed a rough consensus that placing an additional classifier
 state in the RTP header to denote the importance of one flow over
 another is not an approach they wish to advance.  Objections ranging
 from relying on SIP to convey the importance of a flow, to the
 possibility of adversely affecting header compression, were
 expressed.  There was also the general feeling that the extension
 header for RTP that acts as a signal should not be used.

4.1.5. GCP/H.248

 The Gateway Control Protocol (GCP) [21] defines the interaction
 between a media gateway and a media gateway controller.  [21] is
 viewed as an updated version of common text with ITU-T Recommendation
 H.248 [41] and is a result of applying the changes of RFC 2886
 (Megaco Errata) [43] to the text of RFC 2885 (Megaco Protocol version
 0.8) [42].
 In [21], 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.

Carlberg, et al. Informational [Page 11] RFC 4190 IP Telephony Framework November 2005

 The protocol does not specify individual values for priority.  We
 also do not recommend the definition of a well known value for the
 GCP priority as this is out of scope of this document.  Any values
 set should be a function of any SLAs that have been established
 regarding the handling of emergency traffic.

4.2. Policy

 One of the objectives listed in Section 3 above is to treat ETS
 signaling, and related data traffic, as non-preemptive in nature.
 Further, 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 capability to be supported; in our case, a
 default policy for non-preemptive call establishment of ETS 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 [16].  There is no requirement as to how policy is
 accomplished.  Instead, if a domain follows actions outside of the
 default non-preemptive action of ETS-related communication, then we
 stipulate that some type of policy mechanism be in place to satisfy
 the local policies of an SLA established for ETS-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 ETS-based traffic, it can be
 expected that some form of traffic engineering is applied to the
 operation of the network.  We make no recommendations as to which
 type of traffic engineering mechanism is used, but that such a system
 exists in some form and can distinguish and support ETS signaling
 and/or data traffic.  We recommend a review of [32] by clients and
 prospective providers of ETS service that gives an overview and a set
 of principles of Internet traffic engineering.

Carlberg, et al. Informational [Page 12] RFC 4190 IP Telephony Framework November 2005

 MPLS is generally the first protocol that comes to mind when the
 subject of traffic engineering is brought up.  This notion is
 heightened concerning the subject of IP telephony because of MPLS's
 ability to permit a quasi-circuit switching capability to be
 superimposed on the current Internet routing model [30].
 However, having cited MPLS, we need to stress that it is an
 intradomain protocol, and so may or may not exist within a given ISP.
 Other forms of traffic engineering, such as weighted OSPF, may be the
 mechanism of choice by an ISP.
 As a counter example of using a specific protocol to achieve traffic
 engineering, [37] presents an example of one ISP relying on a high
 amount of overprovisioning within its core to satisfy potentially
 dramatic spikes or bursts of traffic load.  In this approach, any
 configuring of queues for specific customers (neighbors) to support
 the target QoS is done on the egress edge of the transit network.
 Note: As a point of reference, existing SLAs established by the NCS
 for GETS service tend to focus on a loosely defined maximum
 allocation of, for example, 1% to 10% of calls allowed to be
 established through a given LEC using HPC.  It is expected, and
 encouraged, that ETS related SLAs of ISPs will be limited with
 respect to the amount of traffic distinguished as being emergency
 related and initiated by an authorized user.

4.4. Security

 This section provides a brief overview of the security issues raised
 by ETS support.

4.4.1. Denial of Service

 Any network mechanism that enables a higher level of priority for a
 specific set of flows could be abused to enhance the effectiveness of
 denial of service (DoS) attacks.  Priority would magnify the effects
 of attack traffic on bandwidth availability in lower-capacity links,
 and increase the likelihood of it reaching its target(s).  An attack
 could also tie up resources such as circuits in a PSTN gateway.
 Any provider deploying a priority mechanism (such as the QoS systems
 described in Section 4.1) must therefore carefully apply the
 associated access controls and security mechanisms.  For example, the
 priority level for traffic originating from an unauthorized part of a
 network or ingress point should be reset to normal.  Users must also
 be authenticated before being allowed to use a priority service (see
 Section 4.4.2).  However, this authentication process should be
 lightweight to minimise opportunities for denial of service attacks

Carlberg, et al. Informational [Page 13] RFC 4190 IP Telephony Framework November 2005

 on the authentication service itself, and ideally should include its
 own anti-DoS mechanisms.  Other security mechanisms may impose an
 overhead that should be carefully considered to avoid creating other
 opportunities for DoS attacks.
 As mentioned in Section 4.3, SLAs for ETS facilities often contain
 maximum limits on the level of ETS traffic that should be prioritised
 in a particular network (say 1% of the maximum network capacity).
 This should also be the case in IP networks to again reduce the level
 of resources that a denial of service attack can consume.
 As of this writing, a typical inter-provider IP link uses 1 Gbps
 Ethernet, OC-48 SONET/SDH, or some similar or faster technology.
 Also, as of this writing, it is not practical to deploy per-IP packet
 cryptographic authentication on such inter-provider links, although
 such authentication might well be needed to provide assurance of IP-
 layer label integrity in the inter-provider scenario.
 While Moore's Law will speed up cryptographic authentication, it is
 unclear whether that is helpful because the speed of the typical
 inter-domain link is also increasing rapidly.

4.4.2. User Authorization

 To prevent theft of service and reduce the opportunities for denial
 of service attacks, it is essential that service providers properly
 verify the authorization of a specific traffic flow before providing
 it with ETS facilities.
 Where an ETS call is carried from PSTN to PSTN via one telephony
 carrier's backbone IP network, very little IP-specific user
 authorization support is required.  The user authenticates itself to
 the PSTN as usual -- for example, using a PIN in the US GETS.  The
 gateway from the PSTN connection into the backbone IP network must be
 able to signal that the flow has an ETS label.  Conversely, the
 gateway back into the PSTN must similarly signal the call's label.  A
 secure link between the gateways may be set up using IPSec or SIP
 security functionality to protect the integrity of the signaling
 information against attackers who have gained access to the backbone
 network, and to prevent such attackers from placing ETS calls using
 the egress PSTN gateway.  If the destination of a call is an IP
 device, the signaling should be protected directly between the IP
 ingress gateway and the end device.
 When ETS priority is being provided to a flow within one domain, that
 network must use the security features of the priority mechanism
 being deployed to ensure that the flow has originated from an
 authorized user or process.

Carlberg, et al. Informational [Page 14] RFC 4190 IP Telephony Framework November 2005

 The access network may authorize ETS traffic over a link as part of
 its user authentication procedures.  These procedures may occur at
 the link, network, or higher layers, but are at the discretion of a
 single domain network.  That network must decide how often it should
 update its list of authorized ETS users based on the bounds it is
 prepared to accept on traffic from recently-revoked users.
 If ETS support moves from intra-domain PSTN and IP networks to
 inter-domain end-to-end IP, verifying the authorization of a given
 flow becomes more complex.  The user's access network must verify a
 user's ETS authorization if network-layer priority is to be provided
 at that point.
 Administrative domains that agree to exchange ETS traffic must have
 the means to securely signal to each other a given flow's ETS status.
 They may use physical link security combined with traffic
 conditioning measures to limit the amount of ETS traffic that may
 pass between the two domains.  This agreement must require the
 originating network to take responsibility for ensuring that only
 authorized traffic is marked with ETS priority, but the recipient
 network cannot rely on this happening with 100% reliability.  Both
 domains should perform conditioning to prevent the propagation of
 theft and denial of service attacks.  Note that administrative
 domains that agree to exchange ETS traffic must deploy facilities
 that perform these conditioning and security services at every point
 at which they interconnect with one another.
 Processes using application-layer protocols, such as SIP, should use
 the security functionality in those protocols to verify the
 authorization of a session before allowing it to use ETS mechanisms.

4.4.3. Confidentiality and Integrity

 When ETS communications are being used to respond to a deliberate
 attack, it is important that they cannot be altered or intercepted to
 worsen the situation -- for example, by changing the orders to first
 responders such as firefighters, or by using knowledge of the
 emergency response to cause further damage.
 The integrity and confidentiality of such communications should
 therefore be protected as far as possible using end-to-end security
 protocols such as IPSec or the security functionality in SIP and SRTP
 [39].  Where communications involve other types of networks such as
 the PSTN, the IP side should be protected and any security
 functionality available in the other network should be used.

Carlberg, et al. Informational [Page 15] RFC 4190 IP Telephony Framework November 2005

4.5. Alternate Path Routing

 This subject 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 assumptions 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 ETS call completion by making use of
 noncongested alternate paths.  We use the term "minimal form" to
 emphasize the fact that care must be taken in how the system provides
 alternate paths so that it does not significantly contribute to the
 congestion that is to be avoided (e.g., via excess control/discovery
 messages).
 Routing protocols at the IP network layer, such as BGP and OSPF,
 contain mechanisms for determining link failure between routing
 peers.  The discovery of this failure automatically causes
 information to be propagated to other routers.  The form of this
 information, the extent of its propagation, and the convergence time
 in determining new routes is dependent on the routing protocol in
 use.  In the example of OSPF's Equal Cost Multiple Path (ECMP), the
 impact of link failure is minimized because of pre-existing alternate
 paths to a destination.
 At the time this document was written, we can identify two additional
 areas in the IETF that can be helpful in providing alternate paths
 for the specific case of call signaling.  The first is [9], which is
 focused on network layer routing and describes a framework for
 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 [17] has been published as an RFC that describes the
 discovery and exchange of IP telephony gateway routing tables between
 providers.  The TRIP protocol [20] 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.  Additional attributes can be registered with
 IANA.

Carlberg, et al. Informational [Page 16] RFC 4190 IP Telephony Framework November 2005

 Inter-domain routing is not an area that should be considered in
 terms of additional alternate path routing support for ETS.  The
 Border Gateway Protocol is currently strained in meeting its existing
 requirements, and thus adding additional features that would generate
 an increase in advertised routes will not be well received by the
 IETF.  Refer to [38] for a commentary on Inter-Domain routing.

4.6. End-to-End Fault Tolerance

 This topic involves 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% of packets or less, 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 argument that the act of protection against loss has
 contributed to additional pressures leading to congestion, which in
 turn helps trigger packet loss.  In addition, by 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 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 [22, 23] 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 [13].  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

Carlberg, et al. Informational [Page 17] RFC 4190 IP Telephony Framework November 2005

 communications.  Hence, these topics are placed under the category of
 value-added objectives.

5. Key Scenarios

 There are various scenarios in which IP telephony can be realized,
 each of which can imply 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 and 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.
 Note: as stated in Section 1.2, [32] provides a more extensive set of
 scenarios in which IP telephony can be deployed.  Our selected set
 below is only meant to provide a couple of examples of how the
 protocols and capabilities presented in Section 3 can play a role.

5.1. 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 scenario 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 telephony
 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.

Carlberg, et al. Informational [Page 18] RFC 4190 IP Telephony Framework November 2005

  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 f
 each IP "island" falls within a given administrative authority.
 Hence, there is not a problem in retaining PSTN type QoS for voice
 traffic (data and signaling) exiting the IP network.  Thus, the need
 for support of mechanisms like diff-serv in the presence of
 overprovisioning, and an expansion of the defined set of Per-Hop
 Behaviors, is reduced 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.

5.2. 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.

Carlberg, et al. Informational [Page 19] RFC 4190 IP Telephony Framework November 2005

  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
 ETS 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 ETS-type traffic is indeed valid for
 the given source.
 We conclude this section by mentioning a complementary work in
 progress in providing ISUP transparency across SS7-SIP interworking
 [33].  The objective of this effort is to access services in the SIP
 network and yet maintain transparency of end-to-end PSTN services.
 Not all services are mapped (as per the design goals of [33]), so we
 anticipate the need for an additional document to specify the mapping
 between new SIP labels and existing PSTN code points like NS/EP and
 MLPP.

6. Security Considerations

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

7. Informative References

 [1]  Braden, R., Clark, D., and S. Shenker, "Integrated Services in
      the Internet Architecture: an Overview", RFC 1633, June 1994.
 [2]  Braden, R., Zhang, L., Berson, S., Herzog, S., and S. Jamin,
      "Resource ReSerVation Protocol (RSVP) -- Version 1 Functional
      Specification", RFC 2205, September 1997.
 [3]  Shenker, S., Partridge, C., and R. Guerin, "Specification of
      Guaranteed Quality of Service", RFC 2212, September 1997.

Carlberg, et al. Informational [Page 20] RFC 4190 IP Telephony Framework November 2005

 [4]  Wroclawski, J., "Specification of the Controlled-Load Network
      Element Service", RFC 2211, September 1997.
 [5]  Baker, F., Iturralde, C., Le Faucheur, F., and B. Davie,
      "Aggregation of RSVP for IPv4 and IPv6 Reservations", RFC 3175,
      September 2001.
 [6]  Berger, L., Gan, D., Swallow, G., Pan, P., Tommasi, F., and S.
      Molendini, "RSVP Refresh Overhead Reduction Extensions", RFC
      2961, April 2001.
 [7]  Blake, S., Black, D., Carlson, M., Davies, E., Wang, Z., and W.
      Weiss, "An Architecture for Differentiated Service", RFC 2475,
      December 1998.
 [8]  Le Faucheur, F., Wu, L., Davie, B., Davari, S., Vaananen, P.,
      Krishnan, R., Cheval, P., and J. Heinanen, "Multi-Protocol Label
      Switching (MPLS) Support of Differentiated Services", RFC 3270,
      May 2002.
 [9]  Sharma, V. and F. Hellstrand, "Framework for Multi-Protocol
      Label Switching (MPLS)-based Recovery", RFC 3469, February 2003.
 [10] Kille, S., "MIXER (Mime Internet X.400 Enhanced Relay): Mapping
      between X.400 and RFC 822/MIME", RFC 2156, January 1998.
 [11] Rosenberg, J., Schulzrinne, H., Camarillo, G., Johnston, A.,
      Peterson, J., Sparks, R., Handley, M., and E. Schooler, "SIP:
      Session Initiation Protocol", RFC 3261, June 2002.
 [12] ANSI, "Signaling System No. 7(SS7), High Probability of
      Completion (HPC) Network Capability", ANSI T1.631-1993, (R1999).
 [13] Robust Audio Tool (RAT):  http://www-
      mice.cs.ucl.ac.uk/multimedia/software/rat
 [14] Schulzrinne, H., "Requirements for Resource Priority Mechanisms
      for the Session Initiation Protocol (SIP)", RFC 3487, February
      2003.
 [15] 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.
 [16] Durham, D., Boyle, J., Cohen, R., Herzog, S., Rajan, R., and A.
      Sastry, "The COPS (Common Open Policy Service) Protocol", RFC
      2748, January 2000.

Carlberg, et al. Informational [Page 21] RFC 4190 IP Telephony Framework November 2005

 [17] Rosenberg, J. and H. Schulzrinne, "A Framework for Telephony
      Routing over IP", RFC 2871, June 2000.
 [18] Heinanen, J., Baker, F., Weiss, W., and J. Wroclawski, "Assured
      Forwarding PHB Group", RFC 2597, June 1999.
 [19] ITU, "Multi-Level Precedence and Preemption Service, ITU,
      Recommendation, I.255.3, July, 1990.
 [20] Rosenberg, J., Salama, H., and M. Squire, "Telephony Routing
      over IP (TRIP)", RFC 3219, January 2002.
 [21] Groves, C., Pantaleo, M., Anderson, T., and T. Taylor, "Gateway
      Control Protocol Version 1", RFC 3525, June 2003.
 [22] Perkins, C., Kouvelas, I., Hodson, O., Hardman, V., Handley, M.,
      Bolot, J., Vega-Garcia, A., and S. Fosse-Parisis, "RTP Payload
      for Redundant Audio Data", RFC 2198, September 1997.
 [23] Rosenberg, J. and H. Schulzrinne, "An RTP Payload Format for
      Generic Forward Error Correction", RFC 2733, December 1999.
 [24] ANSI, "Signaling System No. 7, ISDN User Part", ANSI T1.113-
      2000, 2000.
 [25] "Description of an International Emergency Preference Scheme
      (IEPS)", ITU-T Recommendation  E.106 March, 2002
 [26] Carlberg, K., "The Classifier Extension Header for RTP", Work In
      Progress, October 2001.
 [27] National Communications System: http://www.ncs.gov
 [28] Bansal, R., Ravikanth, R., "Performance Measures for Voice on
      IP", http://www.ietf.org/proceedings/97aug/slides/tsv/ippm-
      voiceip/, IETF Presentation: IPPM-Voiceip, Aug, 1997
 [29] Hardman, V., et al, "Reliable Audio for Use over the Internet",
      Proceedings, INET'95, Aug, 1995.
 [30] Awduche, D., Malcolm, J., Agogbua, J., O'Dell, M., and J.
      McManus, "Requirements for Traffic Engineering Over MPLS", RFC
      2702, September 1999.
 [31] "Service Class Designations for H.323 Calls", ITU Recommendation
      H.460.4, November, 2002.

Carlberg, et al. Informational [Page 22] RFC 4190 IP Telephony Framework November 2005

 [32] Awduche, D., Chiu, A., Elwalid, A., Widjaja, I., and X. Xiao,
      "Overview and Principles of Internet Traffic Engineering", RFC
      3272, May 2002.
 [33] Vemuri, A. and J. Peterson, "Session Initiation Protocol for
      Telephones (SIP-T): Context and Architectures", BCP 63, RFC
      3372, September 2002.
 [34] Polk, J., "Internet Emergency Preparedness (IEPREP) Telephony
      Topology Terminology", RFC 3523, April 2003.
 [35] Carlberg, K. and R. Atkinson, "General Requirements for
      Emergency Telecommunication Service (ETS)", RFC 3689, February
      2004.
 [36] Carlberg, K. and R. Atkinson, "IP Telephony Requirements for
      Emergency Telecommunication Service (ETS)", RFC 3690, February
      2004.
 [37] Meyers, D., "Some Thoughts on CoS and Backbone Networks"
      http://www.ietf.org/proceedings/02nov/slides/ieprep-4.pdf IETF
      Presentation: IEPREP, Dec, 2002.
 [38] Huston, G., "Commentary on Inter-Domain Routing in the
      Internet", RFC 3221, December 2001.
 [39] Baugher, M., McGrew, D., Naslund, M., Carrara, E., and K.
      Norrman, "The Secure Real-time Transport Protocol (SRTP)", RFC
      3711, March 2004.
 [40] Davie, B., Charny, A., Bennet, J.C., 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.
 [41] ITU, "Gateway Control Protocol", Version 3, ITU, September,
      2005.
 [42] Cuervo, F., Greene, N., Huitema, C., Rayhan, A., Rosen, B., and
      J. Segers, "Megaco Protocol version 0.8", RFC 2885, August 2000.
 [43] Taylor, T., "Megaco Errata", RFC 2886, August 2000.

Carlberg, et al. Informational [Page 23] RFC 4190 IP Telephony Framework November 2005

Appendix A: Government Telephone Preference Scheme (GTPS)

 This framework document uses the T1.631 and ITU IEPS standard as a
 target model for defining a framework for supporting authorized
 emergency-related communication within the context of IP telephony.
 We also use GETS as a helpful model from which to draw experience.
 We take this position because of the various areas that must be
 considered; from the application layer to the (inter)network layer,
 in addition to policy, security (authorized access), and traffic
 engineering.
 The U.K. has a different type of authorized use of telephony
 services, referred to as the Government Telephone Preference Scheme
 (GTPS).  At present, GTPS only applies to a subset of the local loop
 lines within the UK.  The lines are divided into Categories 1, 2, and
 3.  The first two categories involve authorized personnel involved in
 emergencies such as natural disasters.  Category 3 identifies the
 general public.  Priority marks, via C7/NUP, are used to bypass
 call-gapping for a given Category.  The authority to activate GTPS
 has been extended to either a central or delegated authority.

A.1. GTPS and the Framework Document

 The design of the current GTPS, with its designation of preference
 based on physical static devices, precludes the need for several
 aspects presented in this document.  However, one component that can
 have a direct correlation is the labeling capability of the proposed
 Resource Priority extension to SIP.  A new label mechanism for SIP
 could allow a transparent interoperation between IP telephony and the
 U.K. PSTN that supports GTPS.

Appendix B: Related Standards Work

 The process of defining various labels to distinguish calls has been,
 and continues to be, pursued in other standards groups.  As mentioned
 in Section 1.1.1, the ANSI T1S1 group has previously defined a label
 in the SS7 ISUP Initial Address Message.  This single label or value
 is referred to as the National Security and Emergency Preparedness
 (NS/EP) indicator and is part of the T1.631 standard.  The following
 subsections presents a snapshot of parallel, on-going efforts in
 various standards groups.
 It is important to note that the recent activity in other groups have
 gravitated to defining 5 labels or levels of priority.  The impact of
 this approach is minimal in relation to this ETS framework document
 because it simply generates a need to define a set of corresponding
 labels for the resource priority header of SIP.

Carlberg, et al. Informational [Page 24] RFC 4190 IP Telephony Framework November 2005

B.1. Study Group 16 (ITU)

 Study Group 16 (SG16) of the ITU is responsible for studies relating
 to multimedia service definition and multimedia systems, including
 protocols and signal processing.
 A contribution [31] has been accepted by this group that adds a
 Priority Class parameter to the call establishment messages of H.323.
 This class is further divided into two parts; one for Priority Value
 and the other is a Priority Extension for indicating subclasses.  It
 is this former part that roughly corresponds to the labels
 transported via the Resource Priority field for SIP [14].
 The draft recommendation advocates defining PriorityClass information
 that would be carried in the GenericData parameter in the H323-UU-PDU
 or RAS messages.  The GenericData parameter contains
 PriorityClassGenericData.  The PriorityClassInfo of the
 PriorityClassGenericData contains the Priority and Priority Extension
 fields.
 At present, 4 levels have been defined for the Priority Value part of
 the Priority Class parameter: Normal, High, Emergency-Public,
 Emergency-Authorized.  An additional 8-bit priority extension has
 been defined to provide for subclasses of service at each priority.
 The suggested ASN.1 definition of the service class is the following:
    CALL-PRIORITY {itu-t(0) recommendation(0) h(8) 460 4 version1(0)}
    DEFINITIONS AUTOMATIC TAGS::=
    BEGIN
    IMPORTS
       ClearToken,
       CryptoToken
        FROM H235-SECURITY-MESSAGES;
    CallPriorityInfo::= SEQUENCE
    {
      priorityValue  CHOICE
       {
         emergencyAuthorized     NULL,
         emergencyPublic         NULL,
         high                    NULL,
         normal                  NULL,
         ...
       },
      priorityExtension   INTEGER (0..255)  OPTIONAL,

Carlberg, et al. Informational [Page 25] RFC 4190 IP Telephony Framework November 2005

      tokens              SEQUENCE OF ClearToken       OPTIONAL,
      cryptoTokens        SEQUENCE OF CryptoToken    OPTIONAL,
      rejectReason        CHOICE
      {
          priorityUnavailable         NULL,
          priorityUnauthorized        NULL,
          priorityValueUnknown        NULL,
          ...
      } OPTIONAL,        -- Only used in CallPriorityConfirm
      ...
    }
 The advantage of using the GenericData parameter is that an existing
 parameter is used, as opposed to defining a new parameter and causing
 subsequent changes in existing H.323/H.225 documents.

Acknowledgements

 The authors would like to acknowledge the helpful comments, opinions,
 and clarifications of Stu Goldman, James Polk, Dennis Berg, Ran
 Atkinson as well as those comments received from the IEPS and IEPREP
 mailing lists.  Additional thanks to Peter Walker of Oftel for
 private discussions on the operation of GTPS, and Gary Thom on
 clarifications of the SG16 draft contribution.

Carlberg, et al. Informational [Page 26] RFC 4190 IP Telephony Framework November 2005

Authors' Addresses

 Ken Carlberg
 University College London
 Department of Computer Science
 Gower Street
 London, WC1E 6BT
 United Kingdom
 EMail: k.carlberg@cs.ucl.ac.uk
 Ian Brown
 University College London
 Department of Computer Science
 Gower Street
 London, WC1E 6BT
 United Kingdom
 EMail: I.Brown@cs.ucl.ac.uk
 Cory Beard
 University of Missouri-Kansas City
 Division of Computer Science
 Electrical Engineering
 5100 Rockhill Road
 Kansas City, MO  64110-2499
 USA
 EMail: BeardC@umkc.edu

Carlberg, et al. Informational [Page 27] RFC 4190 IP Telephony Framework November 2005

Full Copyright Statement

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Carlberg, et al. Informational [Page 28]

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