Internet Engineering Task Force                      Ken Carlberg
INTERNET DRAFT                                       G11
December 1, 2003                                     Ian Brown
                                                     Cory Beard

              Framework for Supporting ETS in IP Telephony

Status of this Memo

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

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   For potential updates to the above required-text see:


   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

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   service in the PSTN, contribute to a constrained solution space.

1.  Introduction

   The Internet has become the primary target for worldwide communica-
   tions.  This is in terms of recreation, business, and various ima-
   ginative 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.

   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 controlled load (bandwidth
   bounds) and guaranteed service (bandwidth & delay bounds) respec-
   tively, 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 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 esta-
   blished by IANA, or set aside as experimental.  It can be expected
   that sets of microflows, a granular identification of a set of pack-
   ets, will correspond to a given code point, thereby achieving an

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   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

   Beyond the introduction of new services, the continued pace of addi-
   tional 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 pro-
   duced in the area of Multi Protocol Label Switching (MPLS), exempli-
   fies 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 of differentiated services [9], and the on-
   going work in the inclusion of fast bandwidth recovery of routing
   failures for MPLS [10].

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.

   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 or the end-to-end applications and
   how it should be supported at any layer.  Given the emergence of IP
   telephony, a natural inclusion of its service is an ability to sup-
   port 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.

1.1.1.  Government Emergency Telecommunications Service (GETS)

   GETS is an emergency telecommunications service available in the U.S.
   and overseen by the National Communications System (NCS) -- an office
   established by the White House under an executive order [30] and now

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   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 but a few).  In addition, a
   select set of individuals from private industry (telecommunications
   companies, utilities, etc.) that are involved in criticial infras-
   tructure recovery operations are also provided access to GETS.

   The purpose of GETS is to achieve a high probability that phone ser-
   vice 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., conges-
   tion) on the U.S. Public Switched Telephone Network by the general

   GETS is based in part on the ANSI T1.631 standard, specifying a High
   Probability of Completion (HPC) for SS7 signaling [13].

1.1.2.  International Emergency Preparedness Scheme (IEPS)

   [18] is a recent ITU standard that describes emergency related com-
   munications over 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 queu-
   ing, 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, 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 case we assume
   some form of application layer signaling is used to explicitly estab-
   lish and maintain voice data traffic.  This explicit signaling capa-
   bility 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)
   [14] begins sending VoIP packets to a unicast (or multicast)

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   destination.  RAT does not use explicit signaling like SIP to estab-
   lish 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.  [31]
   makes an argument that there is a maximum packet loss and delay for
   VoIP traffic, and 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 considered
   acceptable.  [32] discusses the same topic and makes an arguement
   that packet size plays a significant role in what users tolerate as
   "intelligible" VoIP.  The larger the packet, correlating to longer
   sampling rate, the lower the acceptable rate of loss.  Note that
   [31,32] provide only two of several perspectives in examining VoIP.
   A more indepth discussion on this topic is outside the scope of this

   Regardless of a single and definitive characteristic for stressed
   conditions, it would seem that interactive voice has a lower thres-
   hold of some combinations of loss/delay/jitter than elastic applica-
   tions such as email or web browsers.  This places a higher burden on
   the problem space 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 so that they have
   an understanding of 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 expecta-
   tions.  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

   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 capa-
   bilities.  Section 3 presents additional value added objectives,
   which are viewed as useful, but not critical.  Section 4 presents

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   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 ([38]
   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 dedi-
   cated links.

2.  Objective

   The objective of this document is to present a framework that
   describes how various protocols and capabilities (or mechanisms) can
   be used to 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 more indepth 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 wide-spread 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 possi-
   bly partial support that need 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

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   related communications.  The effort in interoperating with these sys-
   tems, presumably through gateways or similar type 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 [39, 40] in
   discussing the protocols and mechanisms below in Secytion 4.  In
   doing this, we do not make a one-to-one mapping of protocol discus-
   sion with requirement.  Rather, we make sure the discussion of Sec-
   tion 4 does not violet any of the requirements in [39,40].

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
   need for all to exist given different scenarios of operation, and
   that ETS support is not expected to be a ubiquitously available ser-
   vice.  We divide this section into 4 areas:

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

4.1.  Signaling & State Information

   Signaling is used to convey various information to either intermedi-
   ate 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 dis-
   cuss potential augmentations to different types of signaling and
   state information to help support the distinction of emergency
   related communications in general.

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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".  These values are meant to convey
   importance to the end-user and have no additional sematics associated
   with them.

   [15] 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 dis-
   tinction that can influence the behavior of gateways and SIP proxies.

4.1.2.  Diff-Serv

   In accordance with [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 that have been specified: Default (correlating
   to best effort forwarding), Assured Forwarding, and Expedited For-
   warding.  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 in
   case the first set has been completely assigned.

   One approach discussed on the IEPREP mailing list is the specifica-
   tion 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"), MLPP, 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 pre-
   cedence approach of Assured Forwarding).  The one caveat to this
   statement are the set of DSCP bits set aside for experimental pur-
   poses. But as the name implies, experimental is for internal examina-
   tion and use and not for standardization.


   It is important to note that as of the time that this document was
   written, the IETF has been taking a conservative approach in

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   specifying new PHBs.  This is because the number of code points that
   can be defined is relatively small and 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 For-
   warding (AF) PHB [20] for distinguishing emergency traffic from other
   types of flows.  At a minimum, AF could be used for the different SIP
   call signaling messages.  If EF 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 back-
   logged.  Its purpose, therefore, it to manage the rates and delays
   experienced by each class.  But emergency traffic may not necessarily
   require QoS any better or different than non-emergency traffic.  It
   may just need higher probability of being forwarded to the next hop,
   which could be accomplished simply through drop precedences within a

   To implement preferential dropping between classes of traffic, one of
   which being emergency traffic, one would probably need to use a more
   advanced form of Active Queue Management (AQM).  Current implementa-
   tions use an overall queue fill measurement to make decisions; this
   might cause emergency classified packets to be dropped.  One new from
   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 where reasonably homogenous in the traffic mix.  But one might
   still have a need to be able to differentiate three drop precedences
   just within non-emergency traffic.  If so, more drop precedences
   could be implemented.  Also, if one wanted discrimination within
   emergency traffic, as with MLPPs five levels of precedence, more drop
   precedences might also be considered.  The five levels would also
   correlate to a recent effort in the Study Group 11 of the ITU to
   define 5 levels for Emergency Telecommunications Service.

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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 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 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 emer-
   gencies 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 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
   amount 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

   Discussions have been held in the Audio/Visual Transport (AVT) work-
   ing group of augmenting RTP so that it can carry a marking that dis-
   tinguishes emergency-related traffic from that which is not.  Specif-
   ically, these discussions centered on defining a new extention that
   contains a "classifier" field indicating the condition associated
   with the packet (e.g., authorized-emergency, emergency, normal) [29].
   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 pay-
   loads that it encapsulates.  However, the AVT group has expressed a

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   rough consensus that placing additional classifier state in the RTP
   header to denote the importance of one flow over another is not an
   approach that they wish to advance.  Objections ranging from relying
   on SIP to convey importance of a flow, as well as 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) [23] defines the interaction
   between a media gateway and a media gateway controller.  [23] is
   viewed as an updated version of 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
   GCP priority -- that 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- sig-
   naling, 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 pro-
   ducing a baseline capability to be supported; in our case, a default
   policy for non-preemptive call establishment of ETS signaling and

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   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 ETS related communication, then we
   stipulate that some type of policy mechanism is 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 [36] by clients and
   prospective providers of ETS service, which gives an overview and a
   set of principles of Internet traffic engineering.

   MPLS is generally the first protocol that comes to mind when the sub-
   ject 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 [33].

   However, having cited MPLS, we need to stress that it is an intra-
   domain 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, [41] presents an example by 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
   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 alloca-
   tion of (e.g., 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 have a limit with respect to the amount of
   traffic distinguished as being emergency related, and initiated by an
   authorized user.

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4.4.  Security

   If ETS support moves from intra-domain PSTN and IP networks to
   inter-domain end-to-end IP, authenticated service becomes more com-
   plex to provide.  Where an ETS call is carried from PSTN to PSTN via
   one telephony carrier's backbone IP network, very little IP-specific
   security support is required.  The user authenticates themself as
   usual to the network.  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 sig-
   nal the call's label. A secure link between the gateways may be set
   up using IPSec or SIP security functionality. If the endpoint is an
   IP device, 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 ETS support must have the means to securely signal
   a given flow's ETS status. They may choose to use physical link secu-
   rity and/or IPSec authentication, combined with traffic conditioning
   measures to limit the amount of ETS 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 ETS 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 func-

   When a call originates from an IP device, the ingress network may
   authorize ETS traffic over that link as part of its user authentica-
   tion procedures. 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 ETS users based on the bounds it is prepared
   to accept on traffic from recently-revoked users.

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

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   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 it does not significantly contribute to the
   congestion that is to be avoided (e.g., via excess control/discovery

   At the time that this document was written, we can identify two areas
   in the IETF that can be helpful in providing alternate paths for call
   signaling.  The first is [10], which is focused on network layer
   routing and describes a framework for enhancements to the LDP specif-
   ication 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] 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

   Inter-domain routing is not an area that should be considered in
   terms of 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
   [42] for a commentary on Inter-Domain routing.

4.6.  End-to-End Fault Tolerance

   This topic involves the work that has been done in trying to compen-
   sate for lossy networks providing best effort service.  In particu-
   lar, 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

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   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 con-
   tributed 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 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 origi-
   nal data packet is followed by one or more redundant packets.  At
   first glance, this would appear to be even less friendly to the net-
   work 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 signi-
   ficantly 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 com-
   munications.  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 & 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, [36] provides a more extensive set of

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   scenarios in which IP telephony can be deployed.  Our selected set
   below is only meant to provide an couple of examples of how the pro-
   tocols and capabilities presented in Section 3 can play a role.

   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 iso-
   lated 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 car-
   riers.  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 infras-

     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 not a problem of retaining toll quality Quality of
   Service as the voice traffic (data and signaling) exits the IP net-
   work because of the existing SS7 provisioned service between
   telephony carriers.  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

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   Number (PIN) authentication (under the context of telephony infras-
   tructure 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 exist-
   ing 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.

     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 emer-
   gency 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 complimentary work in pro-
   gress in providing ISUP transparency across SS7-SIP interworking
   [37].  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 [37], so we

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   anticipate the need for an additional document to specify the mapping
   between new SIP labels and existing PSTN code points like NS/EP and

6.  Security Considerations

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

7.  References

   This is an informational document.  The following are Informative
   References, and there are no Normative 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  Baker, F., et. al., "Aggregation of RSVP for IPv4 and IPv6
      Reservations", Proposed Standard, RFC 3175, September 2001.

   7  Berger, L, et. al., "RSVP Refresh Overhead Reduction Extensions",
      Proposed Standard, RFC 2961, April, 2001.

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

   9  Faucheur, F., et. al., "MPLS Support of Differentiated Services",
      Standards Track, RFC 3270, May 2002.

   10 Sharma, V., Hellstrand, F., "Framework for MPLS-Based Recovery",
      Informational, RFC 3469, February 2003

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

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      August 1982.

   12 Rosenberg, J., et. al., "SIP: Session Initiation Protocol",
      Proposed Standard, RFC 3261, June 2002.

   13 ANSI, "Signaling System No. 7(SS7), High Probability of
      Completion (HPC) Network Capability", ANSI T1.631-1993, (R1999).

   14 Robust Audio Tool (RAT):

   15 Schulzrinne, H, "Requirements for Resource Priority Mechanisms for
      the Session Initiation Protocol", Informational, RFC 3487,
      February 2003

   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 ITU, "Multi-Level Precedence and Preemption Service, ITU,
      Recomendation, I.255.3, July, 1990.

   22 Rosenburg, J, et. al, "Telephony Routing over IP (TRIP)",
      Standards Track, RFC 3219, January 2002.

   23 Cuervo, F., et. al, "Gateway Control Protocol Version 1",
      Standards Track, RFC 3525, June 2003

   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,

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   27 Brown, I., "Securing IEPS over IP", White Paper,

   28 "Description of an International Emergency Preference
      Scheme (IEPS)", ITU-T Recommendation  E.106 March, 2002

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

   30 National Communications System:

   31 Bansal, R., Ravikanth, R., "Performance Measures for Voice on IP",,
      IETF Presentation: IPPM-Voiceip, Aug, 1997

   32 Hardman, V., et al, "Reliable Audio for Use over the Internet",
      Proceedings, INET'95, Aug, 1995.

   33 Awduche, D, et al, "Requirements for Traffic Engineering Over
      MPLS", Informational, RFC 2702,  September, 1999.

   34 Polk, J., "An Architecture for Multi-Level Precedence and
      Preemption over IP", Internet Draft, Work In Progress,
      November, 2001.

   35 "Service Class Designations for H.323 Calls", ITU
      Recommendation H.460.4, November, 2002

   36 Awduche, D., et. al., "Overview and Principles of Internet Traffic
      Engineering", Informational, RFC 3272, May 2002.

   37 Vemuri, A., Peterson, J., "SIP for Telephones (SIP-T): Context and
      Architectures", Best Current Practice, RFC 3372, September 2002

   38 Polk, J., "IEPREP Telephony Topology Terminology", Informational,
      RFC 3523, April 2003

   39 Carlberg, K., Atkinson, R., "General Requirements for Emergency
      Telecommunications Service", Work in Progress, Internet-Draft,
      January, 2003

   40 Carlberg, K., Atkinson, R., "IP Telephony Requirements for
      Emergency Telecommunications Service", Work In Progress, Internet-
      Draft, January, 2003

   41 Meyers, D., "Some Thoughts on CoS and Backbone Networks"

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      IETF Presentation: IEPREP, Dec, 2002

   42 Huston, G., "Commentary on Inter-Domain Routing In the Internet",
      Informational, RFC 3221, December 2001.

8.  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 emer-
   gency related communication within the context of IP telephony.  We
   also use GETS as a helpful model to draw experience from.  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 of
   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-gaping for a given Category.  The authority to activate GTPS has
   been extended to either a central or delegated authority.

8.1.  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.

9.  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
   SS7 ISUP Initial Address Message.  This single label or value is

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   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 snap shot 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.

9.1.  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 [35] 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 tran-
   sported via the Resource Priority field for SIP [15].

   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 Priori-
   tyClassGenericData.  The PriorityClassInfo of the PriorityClassGener-
   icData 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)}


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     CallPriorityInfo::= SEQUENCE
       priorityValue  CHOICE
          emergencyAuthorized     NULL,
          emergencyPublic         NULL,
          high                    NULL,
          normal                  NULL,

       priorityExtension   INTEGER (0..255)  OPTIONAL,
       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 in 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.

10.  Acknowledgments

   The authors would like to acknowledge the helpful comments, opinions,
   and clarifications of Stu Goldman, James Polk, Dennis Berg, 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.

11.  Author's Addresses

   Ken Carlberg                            Ian Brown
   G11                                     University College London
   Department of Computer Science          Department of Computer Science
   Gower Street                            Gower Street
   London, WC1E 6BT                        London, WC1E 6BT
   United Kingdom                          United Kingdom

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   Cory Beard
   University of Missouri-Kansas City
   Division of Computer Science
   Electrical Engineering
   5100 Rockhill Road
   Kansas City, MO  64110-2499

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                           Table of Contents

1. Introduction ...................................................    2
1.1  Emergency Related Data .......................................    3
1.1.1  Government Emergency Telecommunications Service (GETS) .....    3
1.1.2  International Emergency Preparedness Scheme (IEPS) .........    4
1.2  Scope of this Document .......................................    4
2.  Objective .....................................................    6
3.  Considerations ................................................    6
4.  Protocols and Capabilities ....................................    7
4.1  Signaling & State Information ................................    7
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 .......................................................   11
4.3  Traffic Engineering ..........................................   12
4.4  Security .....................................................   13
4.5  Alternate Path Routing .......................................   13
4.6  End-to-End Fault Tolerance ...................................   14
5.  Key Scenarios .................................................   15
6.  Security Considerations .......................................   18
7.  References ....................................................   18
8.  Appendix A: Government Telephone Preference Scheme (GTPS) .....   21
8.1  GTPS and the Framework Document ..............................   21
9.  Appendix B: Related Standards Work ............................   21
9.1  Study Group 16 (ITU) .........................................   22
10.  Acknowledgments ..............................................   23
11.  Author's Addresses ...........................................   23

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