You are on page 1of 23

CHAPTER 1

INTRODUCTION

Since the 1960's when digital voice communication first emerged, the Public Switched
Transfer Network (PSTN) has been supported worldwide as the primary means of voice
communication. The PSTN is a connection-oriented, circuit-switched network in which a
dedicated channel is established for the duration of a transmission. Originally transmitting
only analog signals, the PSTN ultimately switched to using digital communication, which
offered solutions to the attenuation, noise and interference problems inherent in the analog
system. The modern PSTN uses Pulse Code Modulation (PCM) to convert all analog signals
into digital transmissions at the calling end office that initiated communication and reverses
the processes at the receiving end office. Although highly rated for reliability and Quality of
Service (QoS), the PSTN has two significant disadvantages:
 Expensive bandwidth, which results in high telephone bills for individuals and
businesses alike;
 Inefficient use of networking channels, which results from dedicating an entire channel for
each conversation
Internet Protocol (IP)-based networks are the latest innovation to offer solutions to such
problems and are increasingly being used as alternatives to the traditional circuit switched
telephone service. IP Telephony provides alternative means of originating, transmitting, and
terminating voice and data transmissions that would otherwise be carried by the public
switched telephone network (PSTN).IP telephony has a relatively short history, which began
with Vocaltec Inc.'s introduction of its Internet Phone software in February 1995. For the first
time, this software enabled analog signals converted into digital IP packets to be transmitted
over the Internet. By using the Internet, which is connectionless-oriented and packet-switched
based, businesses could eliminate much of their communications expenses. In addition, they
could use the bandwidth of their communications channels more efficiently through
multiplexing of audio, video and data. The new software quickly captured industry interest,
created market demand for the communications capability and sparked competition among IT
vendors.
Like virtually all-new technology, the first VoIP software had problems. First, there was

1
a "lack of connectivity between an IP telephony network and the PSTN"; IP telephony could
successfully occur only when both parties were using the same software. However, this
problem was remedied by the development of gateways, which contained voice processing
cards and provided the necessary interface between the IP telephony software and the PSTN.
The second hurdle was the QoS standard that most businesses expected for communication.
Unlike PSTN, the packet-switched nature of the IP protocol hindered reliability and posed a
significant concern for the level of quality.

Although newly developed network architecture components, such as gateways and routers,
have improved overall quality of communications for intranets, the lack of standards
governing VoIP and the current status of the Internet architecture continue to present
problems. Nonetheless, VoIP is still considered a powerful communications tool, and new
advances in the technology are being made quickly. Thus, the perpetual search for
competitive advantage continues to serve as the primary motivation behind increased
investment and research into this new technology. Internet Telephony Service Providers
(ITSPs) can provide voice and fax services that are close to becoming functionally equivalent
to those provided by public telecommunication operators (PTOs). However, national
authorities license few ITSPs and they generally do not have any universal service
obligations. Many countries ban IP telephony completely, yet IP calls can be made to almost
any telephone in the world. Many PTOs are establishing their own IP telephony services,
and/or using IP-based networks as alternative transmission platforms.

1.1. Internet Service Providers v/s Public Telecommunications Operators

In order to approach the regulatory discussion, is important to indicate the general


comparative advantages of Internet Service Providers over Public Telecommunications
Operators (PTOs). While some of these advantages are technical or economical, a great part
is based on regulatory discrepancies that will be analyzed in the next section of this paper.
 Licensing Advantages:-
While the PTO has to compete for a license issued by the national regulator or ministry, to be
an ISP rarely requires a license. Further, frequently the PTO has to pay in order to get that
license, which also often comes with regulatory obligations like price control and universal
services contributions. This is a major regulatory concern that will be analyzed below.

2
However, it is not a common problem world-wide, rather it is a major concern for the
countries that have not committed themselves to a total liberalization.
 Networking Requirements:-
The PTO has to establish a network of fiber, copper, microwave or satellite links, while the
ISP can lease a circuit to the closest PoP. Moreover, the PSTN technology obligates the PTO
to have excess of capacity to meet peak loads, in comparison to the ISP who can add extra
bandwidth as demand grows.
 Pricing Concerns:-
The PTO has to connect each subscriber at the same price, regardless of location and
inconformity with the regulator price structure. In contrast, the ISP can introduce numerous
price structures depending on the customer and whether the customer is willing to lease his
or her own line. Further PTO has to bill customers according to their level of usage while ISP
can charge customers at a flat rate depending on the bandwidth capacity of their connection.
 Traffic Concerns:-
The PTO has to establish a hierarchy of central office exchanges and an international
gateway for international traffic. In the other hand, the ISP can lease routers and servers,
adding modem as needed.
 Accounting Rate Concerns:-
The PTO has to pay accounting rates and negotiate settlement arrangements with carriers in
each country where the traffic is directed .Instead, the ISP can negotiate peering and transit
arrangements with local point of presence. However, in the case of computer to phone or
phone-to-phone IP telephony the ISP will have to negotiate arrangements for termination
with other ISP. In any case, the ISPs do not pay accounting rates. This is a second major
issue, especially for developing countries. The FCC under ITU and bilaterally has been
pushing hard in order to lower the accounting rates to cost levels. However, the process has
been slow due to the big opposition from the regulators and operators in these countries.

Fig.1 – VoIP phone

3
CHAPTER 2
What is IP Telephony?

Internet Protocol (IP) Telephony is the transmission of voice, fax and related services over
packet-switched IP-based networks. In the longer term, as more and more voice traffic
becomes IP data traffic, there will be little to distinguish between IP telephony and circuit-
switched telephony. However, many telecommunications regulatory schemes depend upon
such a distinction, both physically and as a matter of policy and law.
In order to address the regulatory battle between traditional PSTN, Public Switched
Telephone Network, voice services and IP telephony, it is important to underline the
differences between these two technologies. The bases of future regulations will have to
attend to not just the final product to the user, but also to the technological structure of the
services provided. The following description provides a basic introduction for those not
familiar with voice technologies.
 Traditional Telephony
The widespread traditional telephone service is based on the PSTN, relying on circuit
switching to reach end users. Every time that the user makes a call a dedicated circuit
transports the sound waves in form of electric signals. In user’s terms a "line" is dedicated
exclusively for the communication between the two actual users, regardless show efficiently
they use it. In other words, even if the users keep silence for a period of time, the line
remains dedicated to these particular users, at the same time is unavailable to others. This
inefficiency is unavoidable under PSTN.
Long distance service based on PSTN begins at the user's phone, then the signal travels
through local telephone company lines to the long distance company’s point of presence
(PoP), where the signal is carried by the long distance company to the PoP based on the
called local area. The local telephone company carries the signal finally to the other user’s
phone.
 IP Telephony
IP Telephony rather than circuit switching is based on packet switching. The voice
transmission, instead of using a dedicated line, is broken in numerous packets. These packets
look for the most efficient route individually, and later are reorganized at the IP address
where the voice was originally sent. This technology allows numerous users to send

4
information over the same line, providing with a more efficient utilization of the
telecommunications infrastructure.

2.1. Internet Telephony v/s VoIP

The most important threshold issue relating to IP Telephony concerns definitions. Divergent
views are expressed over the definition of IP Telephony, whether as a technology or as
service concept. IP Telephony can be subdivided into two major groups: Internet Telephony
and Voice-over-IP (VoIP). The distinction between Internet Telephony and VoIP lies in the
nature of the underlying means of transmission or the underlying IP Network. IP Telephony is
generally employed as a generic term covering both. The following definitions are offered as
a means of interpreting the many different terms that are thrown about in this field:
 Internet Telephony:
IP Telephony is the one, in which the principal transmission network is the public Internet.
(Internet Telephony is also commonly referred to as “Voiceover-the-Net” (VON), “Internet
Phone,” and “Net Telephony” – with appropriate modifications to refer to fax as well, such as
“Internet Fax”).
 Voice-over-IP (VoIP):
IP Telephony, in which the principal transmission network or networks are private, managed
IP-based networks.
Even within these two broad groups, there is a potentially infinite number of ways to use IP
technology to provide different services in different ways. Therefore, services are further
classified according to the nature of the terminal devices used (e.g., computer or telephone).

2.2. Types of IP Telephony

While the emergence of IP Telephony is often associated with the rise of the Internet itself, it
is important to appreciate that IP Telephony often does not involve the public Internet at all –
but rather only its underlying technology, the Internet Protocol suite. Different types of IP
Telephony can be identified according to the type of terminal used, where gateways are
located, and the underlying transmission means.
IP telephony has four main categories:
 Personal Computer (PC) to Personal Computer,

5
 Personal Computer to Telephone or Fax,
 Personal Computer to a content provider’s call center or website
 Telephone-to-Telephone communications via the Internet.

2.2.1 PC-to-PC:
The calling and called parties both have computers that enable them to connect to the
Internet, usually via the network of an Internet service provider (ISP). The two
correspondents are able to establish voice communication. Both users have to be connected to
the Internet at that time and use IP telephony software. In this the caller must know the IP
address of the called party.

2.2.2 Phone-to-phone over IP:


The calling and called parties are both subscribers to the public telephony network (fixed or
mobile) and use their telephone set for voice communication in the normal way. There are
two methods for communicating by means of two ordinary telephone sets via an IP or
Internet network.
Use of gateways:
One or more telecommunication players have established gateways that enable the
transmission of voice over an IP network in a way that is transparent to telephone users.

Fig.2 – PC To PC Communication

6
It works in “managed IP network” i.e. a network, which has been dimensioned in such a
way as to enable voice to be carried with an acceptable quality of service.
Use of adapter boxes: A number of companies market boxes, which resemble modems
and are installed between the user's telephone set and his connection to the PSTN. The
calling party initiates his call in the same way as in a conventional telecommunication
network. The first phase of the call is set-up on that network, however, immediately after
this the boxes exchange the information required for the second phase. Data they have
exchanged and the pre-established parameters, establish a connection between each of the
two correspondents and their respective ISP.

Fig.3 – Phone To Phone over Gateway

Once the call has been established, the boxes locally convert the voice signals into IP
packets to be transported over the Internet

7
Fig.4 – Phone To Phone using Adapter Box

2.2.3 PC-to-Phone:
When the computerized user wishes to call a correspondent on the latter's telephone set,
he must begin by connecting to the Internet in the traditional manner via the network of
his ISP. Once connected, he uses the services of an Internet telephony service provider
(ITSP) operating a gateway, which ensures access to the point that, is closest to the
telephone exchange of the called subscriber. It is this gateway that will handle the calling
party's call and all of the signaling relating to the telephone call at the called party end.

2.2.4 Phone-to-PC:
The calling party is the telephony user and the called party is the PC user.

Fig.5 – PC To Phone
8
2.3. What are the building blocks of IP Solution

 Stations or telephone sets that allow user to access the system’s capabilities.
 A quality of service gateway that converts the voice signal into data packets and places
these packets into external data network.
 A data network – a virtual private network or a private network.
 A wide area interface such as a Qos router or Switch.
 A voice switch or system that allows user regular access to the PSTN for calls that cannot
be carried over the private network.
 Qos LAN infrastructure and cabling to connect all the places together.
 A secure firewall behind any public access point.

2.4. What issues do I need to be aware of before I buy?

 Management:
Ongoing management and monitoring of your IP telephony solution will be a key
operational requirement. Decide if the management will be performed by someone in your
organization or out-sourced. Carefully consider the choices different vendors offer, and
determine how they meet your critical business requirements.
 Open VS Proprietary:
Ensure that your solution is based on open standards that will allow you the flexibility you’ll
want when selecting vendors, or adding new components or capabilities to your IP telephony
solution.
 Vendor Selection:
Get more than one proposal from more than one provider to validate the various
offers
 Reliability:
You should ensure that the voice quality of your IP Telephony solution meets your
requirements. Ensure your unified network can prioritize voice traffic and can deal with
high traffic conditions. Should Quality deteriorate below your defined minimum level your
system should be equipped to support a backup capability
 Quality:
Poor voice quality will make your IP Telephony solution difficult and unpopular to use.
Ensure your vendor understands and designs your solution to your voice and data traffic
demands, and has a proven track record of successful integration and ongoing support

9
 Ease of Use:
With a proper solution an IP Telephony call should be as easy to make as a regular PSTN call.
The solution should have the intelligence to route all calls correctly and make cost saving
decisions while respecting quality demands. In addition, you should not feel the need to give
up standard voice features to gain the benefits of IP Telephony – today’s solutions should
demonstrate expert capabilities in both voice services and data.

CHAPTER 3
How do IP phone systems work?

10
The “IP” in IP phone system refers to Voice over IP, or having your phone calls routed over
the internet or your local network (LAN). This is great for many reasons. First of all, you
don’t have to use the telephone network of your telephony service provider for making calls,
which will reduce your costs for phone calls. At the same time you are gaining many
technical advantages by using IP technology for your telephony. Users of an VoIP phone
system simply plug their IP phone into the nearest LAN port. Then, the IP phone registers
automatically at the VoIP phone system. The IP phone always keeps its number, and behaves
exactly the same way, no matter where you plug it in – on your desk, in the office next door
or on a tropical island.

Fig.6 – Working Of IP telephony

All of this works because of the SIP protocol. It is a standard widely used by ISPs, VoIP
phone systems and VoIP phones world-wide. It makes expensive proprietary phones obsolete,
and helps that all devices can talk to each other. IP phone systems are usually built on

11
standard PC or embedded hardware which are more cost-effective and powerful than the
hardware of the traditional phone manufacturers. At the same time, IP phone systems are
scalable, as they are not limited to a certain number of physical phone ports. That means you
don’t need to replace your phone system when your company grows. A great example for an
IP phone system is Askozia PBX.

3.1. VoIP
Since the late 90’s that terms like Voice over Internet Protocol (VoIP), Internet Telephony and
IP Telephony invades the way we communicate over the Internet. These terms may seem
equal at first glance, but they encompass some differences .The first term, VoIP, translates
into passing phone calls over a packet data network, without bounds to network type or
topology. As a result, VoIP allow phone calls from over Local Area Networks (LAN) upto the
global Internet. The term Internet Telephony comes from passing calls across the Internet,
and it may use VoIP, but also specific proprietary hardware or even computer software.
Finally, the term IP Telephony denotes converged data and voice, and uses VoIP technology.
Such integration can lead, for instance, to seamless use of both voice and data messaging.
VoIP is basically the means to grab audio and video in digital form, divide it in small chunks
that can be transferred through the network as packets. After, it reassembles the chunks on the
other side in a convenient way so that people having a phone conversation have the idea of a
circuit switching ordinary call, as shown in Figure 1.VoIP presents a shift on corporate voice
communications where the traditional Private Branch exchange (PBX) based systems were
used to provide internal cost-free communications and sharing of external telephone lines.
With VoIP, the PBX gives place to a gateway router and a server on the computer network
that controls all calls.
If we analyze the beginning of the telephony service back in 1876 when Alexander Graham
Bell achieved the first phone call with his assistant, up to now there is not much difference
for the end user in terms of voice communication. The main advances are in the way such
information is transmitted, the quality of speech, introduction of wireless communication and
smaller, faster and service-rich terminals. The first attempt to transmit voice over a packet
network came with the Network Voice Protocol (NVP) in the ARPANET. In 1995 a company
named Volcatec (still working today on VoIP technologies) introduced the first Internet phone
software called “Internet Phone”. The phone used a home computer (an Intel® 486 processor
at 33MHz) with a sound card, speakers, microphone and a modem. Among the limitations

12
were the need for both users to use the same software and hardware, and sound quality was
nowhere near the conventional equipment at that time.
In 1996 United States (US) telecommunication companies ask the US congress to ban
Internet Phone technology. By 1998 VoIP has reached some potential, with some gateways
that allowed computer-to-phone and phone-to-phone IP solutions. Nowadays, for house-
holds, broadband Internet access is available in the majority of developed and in-
development countries, empowering the use of VoIP significantly. Also, instant messaging
systems that combine voice, video and messaging are very popular. The most important
operating systems provide support for VoIP communications and Microsoft®, Apple® and
Linux® distributions all provide client software and realize the potential of instant messaging
communication. IP telephony brings many advantages over the traditional Public Switched
Telephone Network (PSTN) system, namely in terms of cost. Internet access is not billed
according to user location, as a PSTN system. So, although the billing parameter is still call
time for connections between VoIP and PSTN systems, distance is no longer a primary billing
factor .Another great advantage is flexibility. For PSTN to roll out a new service, it needs to
reprogram the entire network of smart circuit-switched systems, while anew service in IP
telephony may be as easy to set as the development of a new software feature. One of the
most important parts of a telephone call is the establishment of the call itself. In a packet
switched network this is accomplished by a protocol that performs signaling. This paper
addresses such protocols, namely the well spread H.323 and Session Initiation Protocol (SIP).
However, these protocols are over a decade old and a new approach is also referred here: the
Advanced Multimedia System (AMS) project.
IP telephony communication services depend on the use of signaling protocol stacks to set up
and tear down calls, to negotiate capabilities, and to carry information required to locate users
.Here we discusses the following standard protocols: H.323, Session Initiation Protocol
(SIP),Media Gateway Control Protocol (MGCP), and H.248.

13
Fig.7 - Basic VoIP functional architecture

3.1.1. H.323:
H.323 is the first international multimedia communications protocol standard. It was
published by the ITU Telecommunication Standardization Sector (ITU-T) in February 1996
and its current version H.323v6 was approved on June 2006 .H.323 allows the convergence
of voice, video, and data on packet networks. It features World Wide Web and Internet
integration, together with PSTN interfacing. Furthermore, it provides diverse applications
such as wholesale transit of voice, prepaid calling card services, residential/enterprise voice
and video services. Remote users can perform a video call and simultaneously edit a
document in real time over the Internet. H.323 goes beyond, allowing phone or phone
services customization, user location, call transfer, or other tasks taking advantage of the
HTTP interface between the client/server on the network. The H.323 architecture is based on
the following elements Terminals, Multipoint Control Units (MCUs), Gateways, Gatekeepers,
Peer and Border Elements [3, 4]. The Terminals represent the end device of the connection
and can be telephones, video phones, IVR devices, voicemail systems, or soft phones. The
Multipoint Control Units are used for multiparty conferencing. The Gateways interface the
H.323 network to other voice and video networks such as PSTN or H.320. The Gatekeepers
are an optional component that is essentially used for call admission and address resolution. A
Gatekeeper can allow a call to be placed directly between endpoints (Terminals, MCUs or
Gateways) or it may route the call signaling through itself. Peer Elements exchange
addressing information and participate in call authorization inside administrative domains and
between them. They can be co-located with a Gatekeeper, and may aggregate address
information reducing the volume of routing information. Finally, the Border Elements exist
between two administrative domains and can assist in call authentication or authorization.
They are a particular type of Peer Element.

14
Fig.8 - H.323 architecture

H.323 protocol stack is composed of many different protocols (ITU-T standards). H.225.0
defines call signaling. H.225.0/RAS is used for registration, admission and status. H.245 is
the control protocol for multimedia communication. T.120 is a protocol suite for data
conferencing. G.7xx is a series of audio processing protocols. H.26x is a series of video
processing protocols. H.235 provides security. H.450.x is a series of supplementary service
protocols. H.460.x is a series of version-independent extensions to the base H.323 protocol.
Real-Time Transport Protocol (RTP) and RTP Control Protocol (RTCP) are used for media
transportation.

Fig.9 - H.323 protocol stack

3.1.2. SIP

15
The Session Initiation Protocol (SIP) is an Internet Engineering Task Force (IETF) standard
designed for initiating, maintaining and terminating interactive communication sessions
between users. These sessions may include voice, video, instant messaging, interactive
games, and virtual reality. SIP.’s first draft was published in 1996 and the first recognized
standard was published later in 1999. The most recent specification was published in RFC
multicast sessions. The basic functionality and operation of SIP does not depend on any of
these protocols. 3261 on June 2002. SIP functions allow user location, user availability,
endpoint capabilities, and session setup and session management. It enables voicemail and
unified messaging, context-aware communications and location services, integration of
communications and applications, and Internet conferencing and collaboration. The SIP
architecture is based on the following elements: User Agent (UA), User Agent Client (UAC),
User Agent Server (UAS), Proxy Server, Redirect Server and Location Server. The UA is a
SIP network terminal (SIP telephone or gateway to other networks) and contains the UAC
and the UAS. The UAC is the client in the terminal that initiates SIP signaling, while the
UAS is the server in the terminal that responds to the SIP signaling from the UAC. The Proxy
Server receives connection requests from the UA and transfers them to another proxy server
if the particular station is not in its administration. The Redirect Server receives SIP
connection requests and sends them back to the requester including destination data instead
of sending them to the calling party. Finally, the Location Server receives registration
requests from the UA updating the terminal database with the information about UAC
location. It also provides this information to Proxy and Redirect servers. SIP should be used
in conjunction with other protocols to build the complete multimedia architecture The Real-
Time Transport (RTP) is used for transporting real-time data. The Session Description
Protocol (SDP) is used to describe multimedia session. The Session Announcement Protocol
(SAP) is used to advertise

16
Fig.10 - SIP architecture

Fig.11 - SIP protocol stack

3.1.3. MGCP
The Media Gateway Control Protocol (MGCP) is an IETF VoIP protocol destined for
residential gateways, IP phones and large-scale trunk gateways. MGCP latest specification
was published in RFC 3661 on December 2003. MGCP is used between elements of a

17
decomposed Multimedia Gateway, which consists of a Media Gateway Controller (the Call
Agent) that contains the Call control "intelligence", and a Media Gateway that contains the
media functions. It supports a centralized call control model. MGCP is not a peer-to-peer
protocol, since Call Agents are located at the edge of a network and communicate using a call
control protocol such as H.323 or SIP, it is an edge

Fig.12 - MGCP architecture

18
CHAPTER 4
ADVANTAGES AND DISADVANTAGES

The IP telephony or call access via VoIP is not without its own set of disadvantages. These
include:
 Limited or no use in the absence of a dedicated internet access. The system does not
accommodate calls beyond the Local area network or LAN, unless there is an integrated,
compatible PBX system in place.

 Total dependency on separate electric connectivity. Unlike the PSTN phones, IP Phones
and routers connect only via mains electricity. The system is not empowered to work via
power generated from telephone exchanges.
 Easy congestion. These networks, particularly residential internet connectivity, easily
succumb to congestion. The result is poor voice quality or a complete call-drop, in the
midst of an emergency.
 Lapse in connectivity when exposed to high-latency connectivity. The technology does
not empower internet-call connectivity when exposed to latency induced by protocol
overhead. The system also fails to function effectively when exposed to satellite internet
integration.
 Failure when integrated alongside other digital equipment. This technology becomes
redundant when other digital systems are integrated into the adopted phone line.
Equipment like digital video recorders and home security systems do not integrate with
VoIP.
 Challenging emergency calls. The technology becomes a challenge to surpass an
emergency. VoIP uses special IP-addressed phone numbers and not regular public-service
NANP phone numbers. Hence, it becomes difficult for a 911 operator to identify the exact
geographic location of the given IP address.
 Distorted facilitation when challenged by latency and packet-loss. The Internet
connection used by this technology makes the integrated system susceptible to broadband
latency, jitter and packet-loss. The result is distorted and garbled communication due to
transmission error.

19
 Exposure to Denial of Service attacks ( DoS attack ). IP telephony, like other internet
integrated networks, is subjected to Denial of Service break-downs if the address used in
a public IP id.
The overall advantage of IP telephony comes from treating voice as another form of data.
While claims that the PSTN is dying are premature and unfounded, the advantages presented
by IP telephony are clearly visible today:
 More Bandwidth: One advantage of IP telephony is that it dramatically improves
efficiency of bandwidth for real-time voice transmission, in many cases by a factor of 6 or
more. This increase in efficiency is a real long-term driver for the evolution from circuit-
switched to packet-switched technology.
 New Services: Another advantage IP telephony has over the PSTN is that it enables the
creation of a new class of service that combines the best characteristics of real-time voice
communications and data processing, such as web-enabled call centres, collaborative
white-boarding, multimedia, telecommuting, and distance learning. This combination of
human interaction and the power and efficiency of computers is opening up an entirely
 Progressive Deployment: The final advantage of IP telephony is that it is additive to
today's communications networks. IP telephony can be used in conjunction with existing
PSTN switches, leased and dial-up lines, PBXs and other customer premise equipment
(CPE), enterprise LANs, and Internet connections. IP telephony applications can be
implemented through dedicated gateways, which in turn can be based on open standards
platforms for reliability and scalability.

20
Chapter 5
The Future Scope

5.1. Continued Growth


IP telephony enables the integration of voice, data and video streams on the Internet. As such,
we expect it to continue to grow rapidly. It is a primarily 'bits' business and will consequently
continue to be impacted as the Internet grows and evolves further.
5.2. Bifurcation
The market is likely to undergo a bifurcation (parallel to what happened in the first years of
Internet Brokerage into a low end and a high end IP telephony service segment catering to
different segments.
 Cheap lower end IP telephony Services targeted at specific price-sensitive niches who
will care less about lower quality (international calling for example. The prevalent
business model for this segment is likely to be ad-revenue based with cheap (or free) long
distance calling.
 High End Services sub-market that will appeal both to the mass consumer market as well
as the corporate sector. In addition to higher quality and reliable IP Telephony, providers
will supply value added services such as conference facilities, Voice Mail, speech
recognition and E-commerce solutions (such as Click2Talk options on e-trailers, etc.. The
revenue model is likely to be either subscription-based or pay-per-use. Among the
competitors, partnerships and revenue sharing are likely to prevail, as a network of
services will need to be provided to acquire customers and build a customer base.
5.3. Development of the Competitive Landscape
As the industry becomes more mature, we expect big network companies (like cable
companies, utilities) to become increasingly involved in IP telephony, joining the big telecom
players, which have already started to acquire stakes in the business. Both possess a good
understanding of networking technologies and have the necessary funds to provide large-
scale leverage.

21
CONCLUSION

IP telephony represents an efficient and inexpensive technology that will make a difference in
the way that companies approach their clients. Telecommunications operators worldwide
have been expanding the global telephone network based on government regulations aimed at
supplying basic social needs. European Union Countries, the United States and few others
have completed the first stage, represented in very acceptable phone penetration rates.
However, numerous countries worldwide have not accomplished such a task. Around forty
WTO countries still have different kinds of market limitations in order to allow the
telecommunications operators to be profitable and at the same time continue expanding the
basic network where it is not naturally lucrative.
The fact that IP telephony emerges in a liberalizing environment provides this new
technology with a promising development. Yet, IP telephony will not be able to provide any
social good, or expand where there are no outlets to connect to the Internet. In this sense,
PSTN is a necessary and good social prerequisite to the benefits of IP telephony. Over time,
some forms of IP telephony, particularly those that originate and terminate on a user's
telephone set which is connected to the PSTN are more closely approximating conventional
voice telephony, not only in terms of quality and reliability, but also in their use of local
PSTN access facilities. As a result, even those regulators that have maintained an ambiguous
approach towards IP telephony are starting to re-consider the application of telephone subsidy
charges. The Universal Service Report published by the FCC signals a preparedness to levy
local access and universal service fund charges on IP telephony services.
The EU has signaled a similar willingness to reconsider the application of universal service
charges and other regulatory burdens on IP telephony services once their quality and
reliability develop to a level where they are similar to conventional "real time" voice
telephony. Canada has moved to impose traditional regulatory subsidies on IP telephony
faster than the US or EU, since the CRTC has ruled that local access contribution charges are
payable on Internet access lines used for voice telephony purposes. Several other countries,
mostly in the developing world, have gone further than trying to "level the playing field"
between IP and conventional voice telephony. These countries have moved to prohibit IP
telephony services outright. A few other countries, which are in the minority so far, have gone
22
the other way, and have assertively authorized the provision of IP telephony services subject
to relatively light-handed forms of regulation
.This is the case in Japan and in most industrialized countries, providers of IP telephony
services are starting to face a somewhat more regulated and costly operating environment
than in the early days of their services. The initial advantages accruing from the avoidance of
subsidy payments and other regulatory obligations will likely diminish in those markets. IP
telephony services will then have to rely on other factors to compete effectively with
conventional voice telephony services. These factors may include cost advantages inherent in
packet switched services, service innovations such as multimedia voice applications, Internet
"voice-buttons" and other value added features, and the quality, functionality and ubiquity of
the international networks of individual service providers. However, in the longer run, the
regulatory burdens and restrictions imposed upon all international and domestic voice service
providers (including IP telephony providers) will inevitably continue to decrease. It will
never be possible for regulators to identify, restrict and tax all IP telephony providers, any
more than they have been able to do so with call-back providers, refillers, switched-hubbing
providers, or other operators that bypass current accounting rates. Consequently, IP telephony
will add to the pressures to deregulate and simplify international and domestic regulation of
the telecommunications sector. The major challenge to developing countries' regulators will
be to continue the solid expansion of the basic telecommunications infrastructure while
allowing new technologies like IP telephony to develop in a competitive environment.

23

You might also like