draft-ietf-ieprep-framework-11.txt   rfc4190.txt 
This Internet-Draft, draft-ietf-ieprep-framework-10.txt, was published as an Informational RFC, RFC 4190 Network Working Group K. Carlberg
(http://www.ietf.org/rfc/rfc4190.txt), on 2005-11-21. Request for Comments: 4190 G11
Category: Informational I. Brown
C. Beard
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).
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.
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.
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
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
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
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.
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
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
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
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.
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
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
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
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
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-
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
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
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.
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.
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
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.
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.
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
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
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
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
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.
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
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.
[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-
[14] Schulzrinne, H., "Requirements for Resource Priority Mechanisms
for the Session Initiation Protocol (SIP)", RFC 3487, February
[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.
[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.
[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
[36] Carlberg, K. and R. Atkinson, "IP Telephony Requirements for
Emergency Telecommunication Service (ETS)", RFC 3690, February
[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
[41] ITU, "Gateway Control Protocol", Version 3, ITU, September,
[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.
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
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.
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
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)}
CallPriorityInfo::= SEQUENCE
priorityValue CHOICE
emergencyAuthorized NULL,
emergencyPublic NULL,
high NULL,
normal NULL,
priorityExtension INTEGER (0..255) 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.
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.
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
EMail: BeardC@umkc.edu
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