Internet Engineering Task Force Ken Carlberg
INTERNET DRAFT Ian Brown
January 24, 2002 UCL
Framework for Supporting ETS in IP Telephony
<draft-ietf-ieprep-framework-03.txt>
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Abstract
This document presents a framework for supporting authorized
emergency related communication within the context of IP telephony.
We present a series of objectives that reflect a general view of how
authorized emergency service, in line with the 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 PSTN, contribute to a constrained solution space.
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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, infers
that the network will attempt to forward traffic to the destination
as best as it can with no guarantees being made, nor any resources
reserved, to support specific measures of Quality of Service (QoS).
An underlying goal is to be "fair" to all the traffic in terms of the
resources used to forward it to the destination.
In an attempt to go beyond best effort service, [2] presented an
overview of Integrated Services (int-serv) and its inclusion into the
Internet architecture. This was followed by [3], which specified the
RSVP signaling protocol used to convey QoS requirements. With the
addition of [4] and [5], specifying control load (bandwidth bounds)
and guaranteed service (bandwidth & delay bounds) respectively, a
design existed to achieve specific measures of QoS for an end-to-end
flow of traffic traversing an IP network. In this case, our refer-
ence 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
aggregated treatment of data.
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One constant in the introduction of new service models has been the
designation of Best Effort as the default service model. If traffic
is not, or cannot be, associated as diff-serv or int-serv, then it is
treated as Best Effort and uses what resources are made available to
it.
Beyond the introduction of new services, the continued pace of 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 it as part of a telco carrier's
backbone network, or into the Internet as a whole, implies the abil-
ity to support existing emergency related services. Typically, one
associates emergency calls with "911" telephone service in the U.S.,
or "999" in the U.K. -- both of which are attributed to national
boundaries and accessible by the general public. Outside of this
exists emergency telephone services that involved authorized usage,
as described in the following subsection.
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]. Unlike
"911", it is only accessible by authorized individuals. The majority
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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 (telecom-
munications companies, utilities, etc.) that are involved in criti-
cial infrastructure recovery operations are also provided access to
GETS.
The purpose of GETS is to increase the probability that phone service
will be available to selected authorized personnel in times of emer-
gencies, 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 [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) destina-
tion. RAT does not use explicit signaling like SIP to establish an
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end-to-end call between two users. It simply sends data packets to
the target destination. On the other hand, "SIP phones" are host
devices that use a signaling protocol to establish a call signal
before sending data towards the destination.
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 arguement 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.
Regardless of a definitive drop rate, it would seem that interactive
voice has a lower threshold of loss than other elastic applications.
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 carriers so that they have an under-
standing 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 expectations. In
particular, we wish to avoid any temptation of trying to replicate
the exact capabilities of existing emergency voice service currently
available in the PSTN to that of IP and the Internet. If nothing
else, intrinsic differences between the two communications architec-
tures precludes this from happening. Note, this does not prevent us
from borrowing design concepts or objectives from existing systems.
Section 2 presents several primary objectives that articulate what is
considered important in supporting ETS related IP telephony traffic.
These objectives represent a generic set of goals and desired capa-
bilities. 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 ([38]
provides a more extensive discussion on the topology scenarios
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related to IP telephony). However, these scenarios do show cases
where some of the protocols of 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 carriers with regard to dedicated links.
2. Objective
The support of ETS within IP telephony can be realized in the form of
several primary objectives. From this set, we present protocols and
capabilities (presented below in section 3) to be considered by
clients and providers of ETS type services. This document uses the
IEPREP requirements of [39, 40] as a guide in specifying the objec-
tives listed in this section.
There are two underlying goals in the selection of these objectives.
One goal is to produce a design that maximizes the use of existing IP
protocols and minimizes the set of additional specifications needed
to support IP-telephony based ETS. Thus, with the inclusion of these
minimal augmentations, the bulk of the work in achieving ETS over an
IP network that is connected or unconnected to the Internet involves
operational issues. Examples of this would be the establishment of
Service Level Agreements (SLA) with ISPs, and/or the provisioning of
traffic engineered paths for ETS-related telephony traffic.
A second underlying goal in selecting the following objectives is to
take into account experiences from an existing emergency-type commun-
ication system (as described in section 1.1) as well as the existing
restrictions and constraints placed by some countries. In the former
case, we do not attempt to mimic the system, but rather extract
information as a reference model. With respect to constraints based
on laws or agency regulations, this would normally be considered out-
side of the scope of any IETF document. However, these constraints
act as a means of determining the lowest common denominator in speci-
fying technical functional requirements. If such constraints do not
exist, then additional capabilities can be added to the baseline set.
This last item will be expanded upon in the description of Objective
#3 below.
The primary Objectives in support of authorized emergency calls:
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1) High Probability of Call Completion
2) No loss of information when interacting with
PSTN signaling
3) Distinction of ETS data traffic
4) Non-preemptive action
5) Non-ubiquitous support
6) Authenticated service
The first objective is the crux of our work because it defines our
expectations for both data and call signaling for IP telephony. As
stated, our objective is achieving a high probability that emergency
related calls (both data and signaling) will be forwarded through an
IP network. Specifically, we envision the relevance of this objec-
tive during times of congestion, the context of which we describe
further below in this section. The critical word in this objective
is "probability", as opposed to assurance or guarantee -- the latter
two placing a higher burden on the network. It stands to reason,
though, that the word "probability" is a less tangible description
that cannot be easily quantified. It is relative in relation to
other traffic transiting the same network. Objectives 4 and 5 listed
above help us to qualify the term probability in the context of other
objectives.
The second objective involves the interaction of IP telephony signal-
ing with existing PSTN support for emergency related voice communica-
tions. As mentioned above in Section 1.2, standard T1.631 [26] speci-
fies emergency code points for SS7. Specifically, the National Secu-
rity and Emergency Preparedness (NS/EP) Calling Party Category code
point is defined for ISUP IAM messages used by SS7 [26]. =A0Hence,
when IP providers choose to interconnect with the PSTN, it is our
objective that this interaction between the PSTN and IP telephony
with respect to ETS (and national indicators) is a semantically
straightforward, reversible mapping of comparable code points.
The third objective focuses on the ability to distinguish ETS data
packets from other types of VoIP packets. With such an ability,
transit providers can more easily ensure that pre-existing service
level agreements relating to ETS are adhered to. Note that we do not
assume that the actions taken to distinguish ETS type packets are
easy. Nor, in this section, do we state the form of this distinc-
tion. We simply present the objective of identifying flows that
relate to ETS versus others that traverse a transit network.
At an abstract level, the fourth objective pertains to the actions
taken when an IP telephony call, via a signaling protocol such as
SIP, cannot be forwarded because the network is experiencing a form
of congestion. We state this in general terms because of two
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reasons: a) there may exist applications other than SIP, like H.248,
used for call establishment, and b) congestion may come in several
forms. For example, congestion may exist at the IP packet layer with
respect to queues being filled to their configured limit. Congestion
may also arise from resource allocation (i.e., QoS) attributed per
call or aggregated sets of calls. In this latter case, while there
may exist resources to forward the packets, a stateful signaling
server may have reached its configured limit as to how many telephony
calls it will support while retaining toll-quality service per call.
Typically, one terms this form of congestion as call blocking. Note
that we do not address the case when congestion occurs at the bit
level below that of IP, due to the position that it is outside the
scope of IP and the IETF.
So, given the existence of congestion in its various forms, our
objective is to support ETS-related IP telephony call signaling and
data traffic via non-preemptive actions taken by the network. More
specifically, we associate this objective in the context of IP
telephony acting as part of the Public Telephone Network (PTN).
This, as opposed to the use of IP telephony within a private or stub
network. In section 5 below, we expand on this through the descrip-
tion of two distinct scenarios of IP telephony and its operation with
IEPS and the PSTN.
It is important to mention that the fourth objective is a default
position influenced by existing laws & regulations of some countries.
Those countries, regions, or private networks not bound by these res-
trictions can remove this objective and make provisions to enforce
preemptive action. In this case, it would probably be advantageous
to deploy a signaling system similar to that proposed in [15],
wherein multiple levels of priority are defined and preemption via
admission control from SIP servers is enforced.
The fifth objective stipulates that we do not advocate the need or
expectation for ubiquitous support of ETS across all administrative
domains of the Internet. While it would be desirable to have ubiqui-
tous support, we feel the reliance of such a requirement would doom
even the contemplation of supporting ETS by the IETF and the expected
entities (e.g., ISPs and vendors) involved in its deployment.
We use the existing GETS service in the U.S. as an existing example
in which emergency related communications does not need to be ubiqui-
tous. As mentioned previously, the measure and amount of support
provided by the U.S. PSTN for GETS does not exist for all U.S. IXCs
nor LECs. Given the fact that GETS still works within this context,
it is our objective to follow this deployment model such that we can
accomplish the first objective listed above -- a higher probability
of call completion than that of normal IP telephony call traffic.
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Our final objective is that only authorized users may use the ser-
vices outlined in this framework. GETS users are authenticated using
a PIN provided to the telecommunications carrier, which signals
authentication to subsequent networks via the HPC class mark. In an
IP network, the authentication center will need to securely signal
back to the IP ingress point that a given user is authorized to send
ETS related flows. Similarly, transit networks that chose to support
ETS SLAs must securely interchange authorized ETS traffic. In both
cases, IPSec authentication transforms may be used to protect this
traffic. This is entirely separate from end-to-end IPSec protection
of user traffic, which will be configured by users. IP-PSTN gateways
must also be able to securely signal ETS authorization for a given
flow. As these gateways are likely to act as SIP servers, we further
consider the use of SIP's security functions to aid this objective.
3. Value Added Objective
This objective is viewed as being helpful in achieving a high proba-
bility of call completion. Its realization within an IP network
would be in the form of new protocols or enhancements to existing
ones. Thus, objectives listed in this section are treated as value
added -- an expectation that their existence would be beneficial, and
yet not viewed as critical to support ETS related IP telephony
traffic.
3.1. Alternate Path Routing
This objective involves the ability to discover and use a different
path to route IP telephony traffic around congestion points and thus
avoid them. Ideally, the discovery process would be accomplished in
an expedient manner (possibly even a priori to the need of its
existence). At this level, we make no 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 call completion of IEPS traffic by mak-
ing 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 messages).
At the time that this document was written, we can identify two
work-in-progress areas in the IETF that can be helpful in providing
alternate paths for call signaling. The first is [10], which is
focused on network layer routing and describes a framework for
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enhancements to the LDP specification of MPLS to help achieve fault
tolerance. This in itself does not provide alternate path routing,
but rather helps minimize loss in intradomain connectivity when MPLS
is used within a domain.
The second effort comes from the IP Telephony working group and
involves Telephony Routing over IP (TRIP). To date, a framework
document [19] has been published as an RFC which describes the
discovery and exchange of IP telephony gateway routing tables between
providers. The TRIP protocol [22] specifies application level
telephony routing regardless of the signaling protocol being used
(e.g., SIP or H.323). TRIP is modeled after BGP-4 and advertises
reachability and attributes of destinations. In its current form,
several attributes have already been defined, such as LocalPreference
and MultiExitDisc. Additional attributes can be registered with
IANA.
3.2. 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
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
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original data packet is followed by one or more redundant packets.
At first glance, this would appear to be even less friendly to the
network than that of adding FEC packets. However, the encodings of
the redundant packets can be of a different type (or even transcoded
into a lower quality) that produce redundant data packets that are
significantly smaller than the original packet.
Two RFCs [24, 25] have been produced that define RTP payloads for FEC
and redundant audio data. An implementation example of a redundant
audio application can be found in [14]. We note that both FEC and
redundant transmissions can be viewed as rather specific and to a
degree tangential solutions regarding packet loss and emergency com-
munications. Hence, these topics are placed under the category of
value added objectives.
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 viewed as a ubiquitously available service.
We divide this section into 4 areas:
1) Signaling
2) Policy
3) Traffic Engineering
4) Security
4.1. Signaling
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 potential augmentations to
different types of signaling and state information to help support
the distinction of emergency related communications in general, and
IEPS specifically.
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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 a (soon to be) RFC that defines the requirements for a new
header field for SIP in reference to resource priority. This new
header field is meant to provide an additional measure of distinction
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 candidate approach to consider involves the specification of a
new type of Per-Hop Behavior (PHB). This would provide a specific
means of distinguishing emergency related traffic (signaling and user
data) from other traffic. The existence of this PHB then provides a
baseline by which specific code points may be defined related to
various emergency related traffic: authorized emergency sessions
(e.g., ETS), general public emergency calls (e.g., "911"), MLPP.
Aggregates would still exist with respect to the bundling of applica-
tions per code point. Further, one would associate a forwarding
paradigm aimed at a low loss rate reflective of the code point
selected. The new PHB could be in the form of a one or more code
points that duplicate EF-type traffic characteristics. Policies
would determine IF a measure of importance exists per EF-type code-
point.
A potential issue that could be addressed by a new PHB involves merge
points of flows within a diff-serv domain. With EF, one can expect
admission control being performed at the edges of the domain.
Presumably, careful traffic engineering would be applied to avoid
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congestion of EF queues at internal/core merge points stemming from
flows originating from different ingress nodes of the diff-serv
domain. However, traffic engineering may not be able to compensate
for congestion of EF-type traffic at the domain's core routers.
Hence, a new PHB that has more than one code point to identify EF-
type traffic may address congestion by associating a drop precedence
for certain types of EF-type datagrams. Note that local policy and
SLAs would define which EF-type of traffic, if any, would be associ-
ated with a specific drop precedence.
Another approach to consider would be to define a new or fifth class
for the existing AF PHB. Unlike the other currently defined classes,
this new one would be based on five levels of drop precedence. This
increase in the number of levels would conveniently correlate to the
levels of MLPP, which has five types of priorities. 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
(ETS). Beyond these other standardization efforts, the 5 levels
would provide a higher level of variance that could be used to super-
cede the existing 3 levels used in the other classes. Hence, if
other non-emergency aggregate traffic were assigned to the new class,
the highest drop precedence they are assigned to is (3) --
corresponding to the other four currently defined classes. Emergency
traffic would be set to (4) or (5), depending on the SLA that has
been defined.
It is important to note that as of the time that this document was
written, the IETF is taking a conservative approach in specifying new
PHBs. This is because the number of code points that can be defined
is relatively small, and 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 recommend
using the Assured Forwarding (AF) PHB [20] for distinguishing emer-
gency 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.
It is critical to note that one cannot specify an exact code point
used for emergency related data flows because the relevance of a code
point is local to the given diff-serv domain (i.e., they are not glo-
bally unique per micro-flow or aggregate of flows). In addition, we
can expect that the existence of a codepoint for emergency related
flows is based on the service level agreements established with a
given diff-serv domain.
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4.1.3. RTP
The Real-Time Transport Protocol (RTP) provides end-to-end delivery
services for data with real-time characteristics. The type of data
is generally in the form of audio or video type applications, and are
frequently interactive in nature. RTP is typically run over UDP and
has been designed with a fixed header that identifies a specific type
of payload 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 avoid 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
basis.
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.
4.1.4. MEGACO/H.248
The Media Gateway Control protocol (MEGACO) [23] defines the interac-
tion between a media gateway and a media gateway controller. [23] is
viewed as common text with ITU-T Recommendation H.248 and is a result
of applying the changes of RFC 2886 (Megaco Errata) to the text of
RFC 2885 (Megaco Protocol version 0.8).
In [23], the protocol specifies a Priority and Emergency field for a
context attribute and descriptor. The Emergency is an optional
boolean (True or False) condition. The Priority value, which ranges
from 0 through 15, specifies the precedence handling for a context.
The protocol does not specify individual values for priority. We
also do not recommend the definition of a well known value for the
MEGAGO priority. Any values set should be a function of any SLAs
that have been established regarding the handling of emergency
traffic. In addition, given that priority values denote precedence
(according to the Megaco protocol), then by default the ETS telephony
data flows should probably receive the same priority as other non-
emergency calls. This approach follows the objective of not relying
on preemption as the default treatment of emergency-related.
4.2. Policy
One of the objectives listed in section 3 above is to treat 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
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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
data.
Policy can be in the form of static information embedded in various
components (e.g., SIP servers or bandwidth brokers), or it can be
realized and supported via COPS with respect to allocation of a
domain's resources [17]. There is no requirement as to how policy is
accomplished. Instead, if a domain follows actions outside of the
default non-preemptive action of 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.
Note: As a point of reference, existing SLAs established by the NCS
for GETS service tend to focus on a maximum allocation of (e.g., 1%)
of calls allowed to be established through a given LEC using HPC.
Once this limit is reached, all other GETS calls experience the same
probability of call completion as the general public. It is
expected, and encouraged, that ETS related SLAs will have a limit
with respect to the amount of traffic distinguished as being emer-
gency 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 carrier's backbone IP network, very little IP-specific security
support is required. The user authenticates themself as usual to the
network using a PIN. 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. If the endpoint is
an IP device on the carrier's network, the link may be set up
securely from the ingress gateway to the end device.
As flows traverse more than one IP network, domains whose peering
agreements include 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-
tions.
When a call originates from an IP device, the ingress network may
authorize IEPS traffic over that link as part of its user authentica-
tion procedures without necessarily communicating with a central ETS
authentication center as happens with POTS-originated calls. These
authentication procedures may occur at the link or network layers,
but are entirely at the discretion of the ingress network. That net-
work 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.
5. Key Scenarios
There are various scenarios in which IP telephony can be realized,
each of which can infer a unique set of functional requirements that
may include just a subset of those listed above. We acknowledge that
a scenario may exist whose functional requirements are not listed
above. Our intention is not to consider every possible scenario by
which support for emergency related IP telephony can be realized.
Rather, we narrow our scope using a single guideline; we assume there
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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
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 represents a single isolated IP
administrative domain that has no directly adjacent IP domains con-
nected to it. We show an example of this scenario below in Figure 1.
In this example, we show two types of carriers. One is the legacy
carrier, whose infrastructure retains the classic switching architec-
ture attributed to the PSTN. The other is the next generation car-
rier, which uses a data network (e.g., IP) as its core infrastruc-
ture, and Signaling Gateways at its edges. These gateways "speak"
SS7 externally with peering carriers, and another protocol (e.g.,
SIP) internally, which rides on top of the IP infrastructure.
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 Grade of Ser-
vice as the voice traffic (data and signaling) exits the IP network
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because of the existing SS7 provisioned service between carriers.
Thus, the need for support of mechanisms like diff-serv, and an
expansion of the defined set of Per-Hop Behaviors is reduced (if not
eliminated) under this scenario.
Another function that has little or no importance within the closed
IP environment of Figure 1 is that of IP security. The fact that
each administrative domain peers with each other as part of the PSTN,
means that existing security, in the form of Personal Identification
Number (PIN) authentication (under the context of telephony 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 sig-
naling 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
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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
anticipate the need for an additional document to specify the mapping
between new SIP labels and existing PSTN code points like NS/EP and
MLPP.
6. Security Considerations
Information on this topic is presented in sections 2 and 4.
7. 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",
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Standards Track, RFC 3270, May 2002.
10 Sharma, V., Hellstrand, F., Framework for MPLS-Based Recovery,
Internet-Draft, Work in Progress, Oct, 2002
11 Postel, J., "Simple Mail Transfer Protocol", Standard, RFC 821,
August 1982.
12 Handley, M., et. al., "SIP: Session Initiation Protocol",
Proposed Standard, RFC 2543, March 1999.
13 ANSI, "Signaling System No. 7(SS7) _ High Probability of
Completion (HPC) Network Capability_, ANSI T1.631-1993, (R1999).
14 Robust Audio Tool (RAT):
http://www-mice.cs.ucl.ac.uk/multimedia/software/rat
15 Schulzrinne, H, "Requirements for Resource Priority Mechanisms for
the Session Initiation Protocol", Internet Draft, Work In Pro-
gress,
December, 2001.
16 Nichols, K., et. al.,"Definition of the Differentiated Services
Field (DS Field) in the Ipv4 and Ipv6 Headers", Proposed
Standard, RFC 2474, December 1998.
17 Durham, D., "The COPS (Common Open Policy Service) Protocol",
Proposed Standard, RFC 2748, Jan 2000.
18 ITU, "International Emergency Preparedness Scheme", ITU
Recommendation, E.106, March 2000.
19 Rosenburg, J., Schulzrinne, H., "A Framework for Telephony Routing
Over IP", Informational, RFC 2871, June 2000
20 Heinanen. et. al, "Assured Forwarding PHB Group", Proposed
Standard, RFC 2597, June 1999
21 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, "Megaco Protocol Version 1.0", Standards
Track, RFC 3015, November 2000
24 Perkins, C., et al., "RTP Payload for Redundant Audio Data",
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Standards Track, RFC 2198, September, 1997
25 Rosenburg, J., Schulzrinne, H., "An RTP Payload Format for
Generic Forward Error Correction", Standards Track, RFC 2733,
December, 1999.
26 ANSI, "Signaling System No. 7, ISDN User Part", ANSI T1.113-2000,
2000.
27 Brown, I., "Securing IEPS over IP", White Paper,
http://iepscheme.net/docs/secure_IEPS.doc
28 "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: http://www.ncs.gov
31 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
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", work in progress, Internet-Draft, June, 2002.
38 Polk, J., "IEPREP Topology Scenarios", Work in Progress, Internet-
Draft, December, 2002
39 Carlberg, K., Atkinson, R., "General Requirements for Emergency
Telecommunications Service", Work in Progress, Internet-Draft,
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January, 2003
40 Carlberg, K., Atkinson, R., "IP Telephony Requirements for
Emergency Telecommunications Service", Work In Progress, Internet-
Draft, January, 2003
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, 5 levels have been defined for the Priority Value part of
the Priority Class parameter: Low, 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:
ServiceClassInfo ::= SEQUENCE
{
priority CHOICE
{
emergencyAuthorized NULL,
emergencyPublic NULL,
high NULL,
normal NULL,
low NULL
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}
priorityExtension INTEGER (0..255) OPTIONAL;
requiredClass NULL OPTIONAL
tokens SEQUENCE OF ClearToken OPTIONAL
cryptoTokens SEQUENCE OF CryptoH323Token OPTIONAL
}
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
University College London 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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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. Value Added Objective ......................................... 9
3.1 Alternate Path Routing ....................................... 9
3.2 End-to-End Fault Tolerance ................................... 10
4. Protocols and Capabilities .................................... 11
4.1 Signaling & State Information ................................ 11
4.1.1 SIP ........................................................ 12
4.1.2 Diff-Serv .................................................. 12
4.1.3 RTP ........................................................ 14
4.1.4 MEGACO/H.248 ............................................... 15
4.2 Policy ....................................................... 15
4.3 Traffic Engineering .......................................... 16
4.4 Security ..................................................... 17
5. Key Scenarios ................................................. 17
6. Security Considerations ....................................... 20
7. References .................................................... 20
8. Appendix A: Government Telephone Preference Scheme (GTPS) ..... 23
8.1 GTPS and the Framework Document .............................. 23
9. Appendix B: Related Standards Work ............................ 23
9.1 Study Group 16 (ITU) ......................................... 24
10. Acknowledgments .............................................. 25
11. Author's Addresses ........................................... 25
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