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Brooktrout
White Paper
Beyond Dial Tone:
Opportunities for Value in IP Telephony
Josh
Adelson
Brooktrout Technology
October, 1998
| Abstract Though far from mainstream, IP telephony has progressed from its pioneering stages with a flood of technologies, products and service offerings. The market is now showing signs of maturing. This paper provides potential implementers with a model for potential IP telephony applications, and how the various technical approaches fit. Through an understanding of the applications and available technologies, users will be able to select and deploy solutions that provide tangible benefits. |
Introduction
Internet Protocol (IP) telephony was born commercially in early 1995 when VocalTec first
introduced its Internet Phone software. This allowed two Internet-connected individuals
anywhere in the world to have a live conversation using the Internet and their multimedia
PCs. Internet Phone bore numerous limitations vis-à-vis traditional telephonynot
the least of which was that the call participants had to prearrange their Internet call
via e-mail or a standard telephone call. However, in spite of the constraints, the concept
of Internet telephony captured the imaginations of those who saw the opportunity for
low-cost long-distance calling as well as voice-based community-building on the Web.
Since that time, IP telephony has become the focus of many technology vendors and service
providers. While the market models have matured considerably, the benefits sought by users
still tend to fall into two major categories:
In spite of the flood of product and service announcements, actual implementation of IP
telephony is only just beginning to take place at the enterprise level. This lag is
natural for any new technology. After all, MIS and telecom managers first priority
is to provide reliable services and applications to their user communities. Now, the IP
telephony marketplace has matured to the point where enterprise and carrier technology
implementers can begin to assess the opportunities and potential benefits of IP telephony
for their organizations.
Technology Basics
While this paper is not focused on the technical issues, it is useful to have a
familiarity with the technical underpinnings of IP telephony as a foundation for
understanding its applications and potential.
IP telephony can be defined simply as the transmission of voice and fax over IP data
networks. Originating and terminating devices may be traditional telephones and fax
machines, multimedia PCs, or a new class of "Internet aware" fax machines and
telephones. Within this paper, most of the discussion will focus on applications that have
traditional phones and fax machines at one or both ends of the call. This emphasis is
based on the pervasiveness of the PSTN and the corresponding requirement for most IP
telephony solutions to interact with it.
IP Telephony Gateways and Servers
Standard PSTN phones and fax machines connect to IP networks via IP telephony gateways and
servers (see figure below). These devices perform some or all of the following functions:

IP telephony gateways and servers link traditional phone devices to IP
networks
Both gateways and servers occupy the same positions within the network topology. The term
gateway refers to devices that largely provide a connectivity function, while servers
perform some value-added application in addition to providing connectivity.
Technical Challenges
Sending voice and fax over IP presents a number of technical challenges. Most of these
relate in some way to preserving the quality of experience to which PSTN users have become
accustomed over many years.
Delay is the biggest single concern for network managers trying to maintain good voice
quality. Delay is introduced at a number of points in the network, including
compression/decompression, buffering, and routing. Toll-quality voice service requires
delay not greater than about 250 milliseconds. Delays that exceed this threshold become
noticeable and annoying to callers. The public Internet is subject to highly variable
delay conditions, and as a result is not considered viable for quality voice
transmissions. Delay and other challenges, such as congestion and packet loss, must be
addressed through proper network design and capacity planning.
Technical Advantages of IP Telephony
Technical benefits are achieved on several levels by transmitting calls over IP networks.
From a utilization standpoint, packet-based IP networks are more efficient than
circuit-switched networks which allocate a full end-to-end circuit for the duration of a
call. IP telephony also saves network resources through compression, reducing 64 kbit/s
PCM streams down to typically 5-8 kbit/s for voice or 14.4 kbit/s for fax. Further savings
are made through silence suppression. It is estimated that up to 50% of a voice
conversation is silence; circuit-switched networks cannot reallocate silent intervals to
other calls, but packet switched networks can. These bandwidth savings are somewhat offset
by IP protocol overheads, but they are still significant.
At an application level, IP telephony brings the benefit that it treats voice as a data
type, alongside text, graphics, video, etc. This is the basis for value-added
applications, which will be discussed later.
Evaluating Opportunities for IP Telephony
Implementation
A plethora of products and services are now offered under the heading of IP telephony. One
way to bring order to the confusion is to map each solution along two dimensions:
The figure below illustrates this categorization graphically, with one
of the dimensions on each axis, and representative solutions in each of the four
quadrants.
| Applications Dial Tone |
|
||||
Carrier |
IP Telephony Solutions Classified by Venue and Value-Add
By classifying solutions within this model, we gain a better understanding of their value
to the user, and of the requirements of the system being deployed. This will become
evident as we take a look at each quadrant and its solutions.
Carrier-Based Dial-Tone Solutions
This set of solutions, shown in the lower left-hand corner of the matrix, essentially
strives to replicate existing PSTN voice and/or fax services at lower cost by using IP as
a transport medium. These services may take the form of consumer dialing plans and
calling-cards. They may also be offered to enterprises as an IP voice backbone, similar to
circuit-switched virtual private networks that have existed for some years.
From a user perspective, the main requirements of carrier-based dial-tone IP solutions is
that they be virtually indistinguishable from the PSTN services they are replacing in
terms of voice quality. Consumers may be willing to tolerate minor inconveniences such as
two-stage dialing if they perceive that they are obtaining significant price reductions;
enterprise users will expect transparent dialing.
From a system perspective, carrier-based IP dial tone solutions need to be large scale so
that they can have termination points close to the majority of call destinations and also
so they can attract enough traffic to gain economies of scale sufficient to achieve low
cost. IP telephony gateways used in these solutions must offer high densityhundreds
or even thousands of ports per systemat the lowest possible per-port cost. These
gateways need not support fancy user applications, but they do require real time network
monitoring tools to ensure service availability, and sophisticated billing systems to
support the business.
Requirements for fax-only services differ from those for voice in a couple of important
ways. Fax-only gateways can be less dense because fax calling volumes are less than voice
volumes. Also, networks that support only fax can tolerate significantly higher
delaysup to three seconds or morethanks to spoofing technologies employed on
intelligent fax boards such as the Brooktrout TR114, and utilized in carrier class
gateways such as those from Clarent Corporation and Inter-Tel.
Enterprise Dial-Tone Solutions
Enterprise solutions seek to save toll charges by routing long-distance calls over
dedicated "data" lines between company offices. Even external calls may be
routed over the company network to the network location nearest the call-destination,
saving toll charges on all but the "last mile." In terms of quality and user
experience, the requirements of these solutions are similar to those of carrier-based
systems. Additionally, these systems must support existing PBX-based functions, such as
call-forwarding and conference-calling. Enterprise solutions tend to be much smaller in
scale, requiring no more than hundreds of ports at all but the largest sites, and as few
as one or two ports at small branch offices.

Enterprise Dial-Tone Solution: Saving Toll Charges By Transporting
Voice and Fax Over the Corporate Data Network
Carrier-Based Value-Added Solutions
A number of applications are now being introduced by service providers in which the main
benefit is not connectivity per se, but some value-added function related to the
connection. A good example is Internet call-waiting, developed by companies such as
Nortel. Internet call waiting allows home-based Web surfers to act upon incoming calls
without interrupting their Internet connection. A window pops up in the browser, notifying
of the incoming call and providing several options for dealing with it. Another example of
carrier-based value-added solutions is messaging. A number of services, developed by
companies such as Voice and Data Systems, exist that allow users to access voice-mail,
e-mail and faxes through a single desktop or telephony interface.
Systems for carrier-based value-added solutions are generally smaller in scale than those
for dial-tone solutions. While they may need to handle a large number of subscribers, only
a subset of subscribers will be using the value-added function at any given time. Because
of the smaller scale and higher intrinsic value-add, these solutions are not as
cost-sensitive on a per-port basis. They do need to be very flexible, however, to support
rapid development and modification of applications. Carrier-based value-added systems may
need to support more computing functions, such as database lookups or message storage and
retrieval. Finally, these systems must incorporate billing and accounting functions that
feed the appropriate service usage data into the carriers mainline billing systems.

Carrier-Based Value-Added Solution: Internet Call Waiting
Enterprise Value-Added Solutions
IP telephony provides a number of opportunities for value-added communications within the
enterprise, or between an enterprise and its customers. One example is call-enabled web
pages, variously called "voice-button" or "push-to-talk," as developed
by Nortel and eFusion. This application allows visitors to a company Web site to select a
button on the Web page which automatically establishes a call into a pre-programmed
corporate location, such as the ordering department or customer service. Depending on the
implementation, remote callers may speak through their multimedia PC or through their
traditional desktop phone.
Another enterprise value-added solution is teleconferencing, by which geographically
separate employees can hold an online meeting, using voice, data and possibly video to
replicate as well as possible the face-to-face meeting experience. Teleconferencing is not
new, but ubiquitous IP networks and OS-independent browsers remove barriers to widespread
adoption.
Yet another class of enterprise applications use the flexibility of LAN servers to act as
programmable call-switching and routing nodes. For example, in an application developed by
InVADE, IP telephony and wireless LAN technologies are combined to support facility-wide
wireless phone service within environments such as entertainment centers and warehouses
where staff are continually moving about.
System requirements for enterprise value-added solutions are similar to those for carrier
value-added solutions, though they may be smaller in scale. Applications that interface
with the PSTN will require gateways that support the chosen application. Because the value
is in the application, flexibility and rapid development are at a premium. Depending on
the application, enterprises may be willing to forego so-called "carrier class"
attributes such as hot-swap and power redundancy, but enterprise systems still need to
operate reliably and consume minimum possible MIS resources.
Little Things Mean A Lot
It is always tempting to paint grand future visions, but small new applications are often
the ones that take off, especially if they efficiently address a particular need. For
example, Internet call waiting not only solves the problem of busy phone lines, it also
provides an ongoing revenue opportunity for service providers. Likewise, push-to-talk
technologies may in future become the basis for intranet-enabled company call directories.
Employees could be searched by name or department, and then phoned with the push of a
button.
Its still too early in the market to say which applications will catch on and which
ones will never go beyond the trade show floor. However, its important to keep in
mind that some of the more mundane-sounding applications can end up being the most
popular, because they fill a specific need and they do not require a major change to work
behavior or business practices.
Considerations for Implementation
Organizations that choose to implement IP telephony applications face a number of choices
in terms of technologies, standards and business practices. In this section we will
examine two of the major choices: open vs. embedded systems, and standards-based vs.
proprietary systems.
Comparing Open and Embedded Systems
Many of the early entrants to the IP telephony gateway and server market have based their
products on open systems. These are PC or workstation chassis with computer telephony
boards that perform key functions such as voice compression and line interfaces. Some
products, such as the Brooktrout TR2001, combine multiple functions on a single board. By
using open systems, these vendors have been able to take advantage of hardware available
on the open market while focusing their own development efforts on application software,
hardware integration, and network implementation.
In parallel, many voice and data networking vendors have focused on adding IP telephony
gateway capability to their existing network devicesPBXs, routers and frame relay
access devices (FRADs). These vendors are looking to leverage their existing product
architectures and installed bases. Both approaches have their merits, depending on the
existing network environment and the intended applications.
Embedded systems may offer significant cost advantages, assuming that the IP telephony
function can be added as a board or firmware upgrade to existing equipment. If this is not
the case, then a new box must be purchased regardless. Embedded systems may also have a
cost advantage and very low and very high port counts. The overhead cost of the PC
motherboard and chassis must be spread over at least eight ports, preferably 24, to
provide a competitive per-port system cost. At the very high end (DS3 and above)
open-standard form factors such as ISA and PCI cards do not support high port counts as
efficiently as do larger, proprietary forms. The emerging compact PCI form factor
increases the port scale at which open systems are economically competitive.
Open systems have the chief advantage of supporting value-added applications. These
systems are based on popular operating systems such as Windows NT and UNIX. Not only can
applications be more easily developed on these operating systems, but enterprises and
service providers are not reliant upon the equipment vendor for the applications;
applications can be developed internally or procured in the open market. Likewise, open
systems also take advantage of the competitive market for computer telephony hardware.
Open system devices are most cost competitive in the mid-range scalefrom one to
eight T1/E1 spans, or 24 to 240 ports per chassis.
Many network managers consider a PC to be insufficiently reliable for use in network
applications, particularly voice telephony, where users expect uninterrupted service.
However, when properly configured for telecom environments, open systems can be extremely
reliable. Many vendors offer industrial-class rack-mounted PCs, with features such as
built-in redundancy. Compact PCI offers telecom rack-style form factors with higher-grade
physical connectors and support for features such as hot-swappability, power redundancy,
and extensive cooling.
| Capability | Open Systems | Embedded Systems |
Add-on to existing infrastructure |
|
X |
Dial-tone platform |
X | X |
Applications Platform |
X | |
Flexibility |
X | |
Economic Scale (ports) |
CPCIX X |
X X |
Open vs. Embedded Systems: Pros and Cons
Standards-Based vs. Proprietary Systems
Enterprises and service providers considering IP telephony implementation are also
confronted with competing standards as well as proprietary systems. As with the open
systems vs. embedded systems debate, there is no single right answer; the choice depends
on the application.
The major benefit of standards is interoperability. The best-known standard for IP
telephony is ITU-H.323. H.323 is a far-reaching umbrella standard which includes
coding/compression algorithms for voice, data and video, as well as call-establishment and
switching functions, and recently incorporation of T.38 real time fax. Its important
to recognize that H.323 was originally developed for multimedia teleconferencing over IP;
it is part of a family of H.32x standards which includes conferencing over PSTN (H.324)
and ISDN (H.320). As such, H.323 defines more than is required for many voice over IP
applications. The chief advantage of this is that H.323 provides room for growth from
early single-media applications to multimedia applications in the future. Microsoft and
Netscape are currently supporting H.323 in their multimedia browser extensions.
The disadvantage of H.323 is that its call-establishment methodsbecause they
incorporate such rich application supportmay be inefficient for certain high-scale
voice/fax applications. The H.323 standards groups have work to address this with new
fast setup versions of the call establishment protocols in a the latest H.323
Version 2. It remains to be seen whether these improvements will support carrier-scale
environments in which tens of thousands or even millions of calls must be set up and
routed simultaneously.
There are also alternate proposals for standards aimed at providing more efficient,
scalable call establishment methods, in addition to H.323 V2. The Internet Engineering
Task Force (IETF) has proposed the Session Initiation Protocol, or SIP, and Bellcore has
proposed its Simple Gateway Control Protocol, or SGCP. It is too early to tell whether
these will take hold in the marketplace, but both of them address a requirement that needs
to be filled.
At the media coding level, the compression methods called within H.323 are gaining
widespread adoption across the board. These include G.711, G.723.1 and G.729A for voice,
and H.263 for video. Even those products that shun H.323 for call establishment generally
support at least one of the G.-series vocoders. These compression algorithms are already
supported by multiple vendors, with implementations on digital signal processors as well
as microprocessors. For example, G.723.1 speech compression/decompression can run on a DSP
in a high-density multi-port server, but it also runs on a single-session client software
on a Pentium-based desktop PC.
For provisioning of IP-based dial tone, it is not necessary to use the upper layers of the
H.323 standard, and alternative approaches, including proprietary ones, may be
advantageous as discussed above. In the long run some standard will be desirable to allow
networks to inter-work with one another, and to ensure an open market for equipment. For
provisioning of value-added applications, H.323 is important, because it takes advantage
of existing desktop H.323 compliant client software to support the application. H.323 is
with us to stay; there is already a lot of compliant software and a lot of industry energy
to evolve the standard to support more applications. But alternative standards will no
doubt also gain critical mass for carrier-scale dial-tone applications.
| Standard | Organization | Status | Function |
H.323 |
ITU |
Approved |
Network call control Multimedia conferencing |
G.711 |
ITU |
Approved |
64 kbit/s PCM voice coding |
G.723.1 |
ITU |
Approved |
5.3 and 6.4 kbit/s voice coding |
G.729A |
ITU and Frame Relay Forum |
Approved |
8 kbit/s voice coding |
SIP |
IETF |
Proposed |
Network call control |
SGCP |
Bellcore |
Proposed |
Network call control |
T.37 |
ITU |
Approved |
Store-and-Forward Fax |
T.38 |
ITU |
Approved |
Real Time Fax |
Key Standards for IP Telephony
Conclusions
The IP telephony marketplace has matured considerably in the past year. It is no longer
simply a technological phenomenon; real products are available that target specific
solutions for value-add. Enterprises and service providers, armed with an understanding of
what they need to achieve, can make intelligent choices among the products and
technologies that are available.
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