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A

SEMINAR REPORT
ON
“INTERNET PROTOCOL TELEPHONY”

SUBMITTED IN PARTIAL FULFILLMENT OF THE


REQUIREMENT FOR THE AWARD OF THE DEGREE
OF
BACHELOR OF ENGINEERING
IN
ELECTRONICS AND COMMUNICATION ENGINEERING

GUIDED BY: - SUBMITTED BY: -

MR. TOUSIF KAMAAL NARENDRA BAGORIA


LECTURER (E.C.E DEPTT.) FINAL YEAR (E.C.E.)

DEPARTMENT OF ELECTRONICS & COMMUNICATION ENGINEERING

SOBHASARIA ENGINEERING COLLEGE, SIKAR


UNIVERSITY OF RAJASTHAN
2007-2008
SOBHASARIA ENGINEERING COLLEGE, SIKAR
DEPARTMENT OF ELECTRONICS & COMMUNICATION ENGINEERING

CERTIFICATE

THIS IS TO CERTIFY THAT THE WORK, WHICH IS BEING PRESENTED IN THE


SEMINAR ENTITLED “INTERNET PROTOCOL TELEPHONY” SUBMITTED
BY MR. NARENDRA BAGORIA, A STUDENT OF FINAL YEAR B.E. IN
ELECTRONICS & COMMUNICATION ENGINEERING AS A PARTIAL
FULFILLMENT FOR THE AWARD OF DEGREE OF BACHELOR OF
ENGINEERING IS A RECORD OF STUDENT’S WORK CARRIED OUT UNDER
MY GUIDANCE AND SUPERVISION.
THIS WORK HAS NOT BEEN SUBMITTED ELSEWHERE FOR THE AWARD OF
ANY OTHER DEGREE.

DATE: (MR TOUSIF KAMAAL)


PLACE: S.E.C., SIKAR (SEMINAR GUIDE)

(MR PRADEEP SHARMA) (PROF. K. B.SINGH)


(SEMINAR INCHARGE) (H.O.D OF E.C.E. DEPTT.)
CANDIDATE’S DECLARATION

THIS IS TO CERTIFY THAT WORK, WHICH IS BEING PRESENTED IN THE


SEMINAR ENTITLED “INTERNET PROTOCOL TELEPHONY” SUBMITTED
BY UNDERSIGNED STUDENT OF FINAL YEAR B.E. IN ELECTRONICS &
COMMUNICATION ENGINEERING IN PARTIAL FULFILLMENT FOR AWARD
OF DEGREE OF BACHELOR OF ENGINEERING IS A RECORD OF MY OWN
WORK CARRIED OUT UNDER THE GUIDANCE AND SUPERVISION OF
MR.TOUSIF KAMAAL, LECTURER, DEPARTMENT OF ELECTRONICS &
COMMUNICATION ENGINEERING.
THIS WORK HAS NOT BEEN SUBMITTED ELSEWHERE FOR THE AWARD OF
ANY OTHER DEGREE.

DATE:- / /2008 NARENDRA BAGORIA


PLACE: S.E.C.,SIKAR FINAL YEAR, E.C.E.
ACKNOWLEDGEMENT

The work written in this report is an outcome of the precious guidance


cooperation of some persons. It is moment of great pleasure to acknowledge their
help and encouragement.

The work has been performed under the guidance of Mr. Tousif Kamaal,
Lecturer(E.C.E. deptt.). Words are incapable to formulate my deep sense of
gratitude to him for his keen interest and encouragement. He was more willing to
share his treasure of knowledge with me. I appreciably acknowledge his helpful
comment for the improvement of the work.

I am cordially thankful to Mr. Pradeep Sharma for providing me the


opportunity to present my seminar on a topic of my area of interest.

Last but not the least I am also grateful to Prof. K.B.Singh, HOD, E.C.E.
deptt., for his guidance and kind support throughout.

NARENDRA BAGORIA
Final year,E.C.E.
ABSTRACT

A major development starting in 2004 has been the introduction of mass-market


VoIP services over broadband Internet access services, in which subscribers make and
receive calls as they would over the PSTN. Full phone service VoIP phone companies
provide inbound and outbound calling with Direct Inbound Dialing. Many offer unlimited
calling to the U.S. and some to Canada or selected countries in Europe or Asia as well,
for a flat monthly fee.
IP telephony (Internet Protocol telephony) is a general term for the technologies
that use the Internet Protocol's packet-switched connections to exchange voice, fax, and
other forms of information that have traditionally been carried over the dedicated circuit-
switched connections of the public switched telephone network (PSTN). Using the
Internet, calls travel as packets of data on shared lines, avoiding the tolls of the PSTN.
The challenge in IP telephony is to deliver the voice, fax, or video packets in a
dependable flow to the user. Much of IP telephony focuses on that challenge.

IP telephony service providers include or soon will include local telephone


companies, long distance providers such as AT&T, cable TV companies, Internet service
providers (ISPs), and fixed service wireless operators. IP telephony services also affect
vendors of traditional handheld devices.
INDEX
Page No.

Chapter 1 Introduction 1
1.1 Internet Protocol Telephony 1
1.1.1 Principles of IP telephony 2
1.2 VoIP 3
1.2.1 Resources from around the Web 4

Chapter 2 Implementation 7
2.1 Implementation 7
2.2 Reliability 8
2.3 Quality of service 8
2.4 Difficulty with sending faxes 9
2.5 Emergency calls 10
2.6 IP telephony scenarios 10
2.6.1 PC to PC 11
2.6.2 PC to Telephone 11
2.6.3 Telephone to PC 12
2.6.4 Telephone to Telephone 12
2.7 Integration into global telephone number system 13
2.8 VoIP phone accessibility and portability 13
2.9 Mobile phones & Hand held Devices 14
2.10 Security 15

Chapter 3 H.323 Protocols 16


3.1 H.323 Protocols 16
3.2 H.323 Architecture components 17

Chapter 4 Adoption 19
4.1 Mass-market telephony 19
4.2 Corporate and Telco use 20
4.3 Uses in Amateur Radio 21
4.4 Click to call 22

Chapter 5 Legal issues in different countries 23


5.1 IP telephony in Japan 24
5.2 Technical details 25
5.3 Applications 27

Conclusion 29
References 30
Appendix: List of figures 32
Chapter: 1
Introduction
1.1 Internet Protocol Telephony
Telephony is the technology associated with the electronic transmission of voice,
fax, or other information between distant parties using systems historically associated
with the telephone, a handheld device containing both a speaker or transmitter and a
receiver. With the arrival of computers and the transmittal of digital information over
telephone systems and the use of radio to transmit telephone signals, the distinction
between telephony and telecommunication has become difficult to make.
In the other words the Internet Protocol (IP) is the method or protocol by which
data is sent from one computer to another on the Internet. Each computer (known as a
host) on the Internet has at least one IP address that uniquely identifies it from all other
computers on the Internet. When you send or receive data (for example, an e-mail note or
a Web page), the message gets divided into little chunks called packets. Each of these
packets contains both the sender's Internet address and the receiver's address. Any packet
is sent first to a gateway computer that understands a small part of the Internet. The
gateway computer reads the destination address and forwards the packet to an adjacent
gateway that in turn reads the destination address and so forth across the Internet until
one gateway recognizes the packet as belonging to a computer within its immediate
neighborhood or domain. That gateway then forwards the packet directly to the computer
whose address is specified.
On the Internet, three new services are now or will soon be available:
• The ability to make a normal voice phone call (whether or not the person
called is immediately available; that is, the phone will ring at the location of
the person called) through the Internet at the price of a local call.
• The ability to send fax transmissions at very low cost (at local call prices)
through a gateway point on the Internet in major cities.
• The ability to send voice messages along with text e-mail.
Some companies that make products that provide or plan to provide these
capabilities include: IDT Corporation (Net2Phone), Netspeak, NetXchange, Rockwell
International, VocalTec, and Voxspeak. Among uses planned for Internet phone services
are phone calls to customer service people while viewing a product catalog online at a
Web site.

Fig.1.1 An overview of IP Telephony

A Telephony API (TAPI) is available from Microsoft and Intel that allows
Windows client applications to access voice services on a server and that interconnects
PC and phone systems. Both Microsoft and Netscape provide or plan to provide support
for voice e-mail.
1.1.1 Principle of IP telephony
To understand IP telephony, it’s necessary to be familiar with the fundamental
characteristics behind the Internet and how it compares to the Public Switched Telephone
Network (PSTN). The most important of these characteristics is the data transport mode,
also known as data connection type which is either a circuit switched or packet switched
as explained below:
Circuit Switched Connection: A device using a circuit switched connection only
connects when data is to be sent. The connection is dedicated exclusively to the sending
and receiving nodes for the entire duration of the call. Because the two points are
connected in both the directions, the connection is called a circuit. The connection is
only present when you need it and, since bandwidth remains constant, you only pay for
the duration of the connection. While connected on a circuit switched network you have
exclusive use of the established connection and data can be sent continuously. This type
of data transaction is typically routed through the PSTN. Although the circuit switched
network pro vides a very reliable connection for voice transmissions, it makes very
inefficient use of the available bandwidth.
Packet Switched Connection: While circuit switched connection is open and constant
for the entire duration of the call, packet switched connection opens just long enough to
send a small chunk of data, called a packet, from one system to another. A packet
switched connection keeps you connected all the time but you only pay for the amount
of data transferred. In this case, the data is divided into small packets and each packet
contains a source and a destination address. Packets of data are sent from source to
destination using the quickest route available. The network bandwidth is shared and
multiple simultaneous users are allowed to access multiple locations across a network.
This provides for much more efficient use of available bandwidth but can create
problems for voice traffic, which is very sensitive to delay.
1.2VoIP
VoIP (voice over IP) is an IP telephony term for a set of facilities used to manage
the delivery of voice information over the Internet.VoIP involves sending voice
information in digital form in discrete packets rather than by using the traditional circuit-
committed protocols of the public switched telephone network (PSTN). A major
advantage of VoIP and Internet telephony is that it avoids the tolls charged by ordinary
telephone service.
VoIP derives from the VoIP Forum, an effort by major equipment providers,
including Cisco, Vocal Tec, 3Com, and Netspeak to promote the use of ITU-T H.323, the
standard for sending voice (audio) and video using IP on the public Internet and within an
intranet. The Forum also promotes the user of directory service standards so that users
can locate other users and the use of touch-tone signals for automatic call distribution and
voice mail.
In addition to IP, VoIP uses the real-time protocol (RTP) to help ensure that
packets get delivered in a timely way. Using public networks, it is currently difficult to
guarantee Quality of Service (QoS). Better service is possible with private networks
managed by an enterprise or by an Internet telephony service provider (ITSP).
A technique used by at least one equipment manufacturer, Adir Technologies
(formerly Netspeak), to help ensure faster packet delivery is to use ping to contact all
possible network gateway computers that have access to the public network and choose
the fastest path before establishing a Transmission Control Protocol (TCP) sockets
connection with the other end

Fig.1.2 A typical analog telephone adapter for connecting an ordinary phone to a VoIP network
Using VoIP, an enterprise positions a "VoIP device" at a gateway. The gateway
receives packetized voice transmissions from users within the company and then routes
them to other parts of its intranet (local area or wide area network) or, using a T-carrier
system or E-carrier interface, sends them over the public switched telephone network.

1.1.1 Resources from around the Web


Companies providing VoIP service are commonly referred to as providers, and
protocols which are used to carry voice signals over the IP network are commonly
referred to as Voice over IP or VoIP protocols. They may be viewed as commercial
realizations of the experimental Network Voice Protocol (1973) invented for the
ARPANET providers. Some cost savings are due to utilizing a single network – see
attached image - to carry voice and data, especially where users have existing
underutilized network capacity that can carry VoIP at no additional cost. VoIP to VoIP
phone calls are sometimes free, while VoIP to PSTN may have a cost that's borne by the
VoIP user.
There are two types of PSTN to VoIP services: DID (Direct Inward Dialing) and
access numbers. DID will connect the caller directly to the VoIP user while access
numbers require the caller to input the extension number of the VoIP user. Access
numbers are usually charged as a local call to the caller and free to the VoIP user while
DID usually has a monthly fee. There are also DIDs that are free to the VoIP user but
chargeable to the caller.
A major development starting in 2004 has been the introduction of mass-market
VoIP services over broadband Internet access services, in which subscribers make and
receive calls as they would over the PSTN. Full phone service VoIP phone companies
provide inbound and outbound calling with Direct Inbound Dialing. Many offer unlimited
calling to the U.S. and some to Canada or selected countries in Europe or Asia as well,
for a flat monthly fee.

Fig.1.3 A complete VoIP solution


These services take a wide variety of forms which can be more or less similar to
traditional POTS. At one extreme, an analog telephone adapter (ATA) may be connected
to the broadband Internet connection and an existing telephone jack in order to provide
service nearly indistinguishable from POTS on all the other jacks in the residence. This
type of service, which is fixed to one location, is generally offered by broadband Internet
providers such as cable companies and telephone companies as a cheaper flat-rate
traditional phone service. Often the phrase "VoIP" is not used in selling these services,
but instead the industry has marketed the phrase "Internet Phone" or "Digital Phone"
which is aimed at typical phone users who are not necessarily tech-savvy. Typically, the
provider touts the advantage of being able to keep one's existing phone number.
At the other extreme are services like Gizmo Project and Skype which rely on a
software client on the computer in order to place a call over the network, where one user
ID can be used on many different computers or in different locations on a laptop. In the
middle lie services which also provide a telephone adapter for connecting to the
broadband connection similar to the services offered by broadband providers (and in
some cases also allow direct connections of SIP phones) but which are aimed at a more
tech-savvy user and allow portability from location to location. One advantage of these
two types of services is the ability to make and receive calls as one would at home,
anywhere in the world, at no extra cost.
Corporate customer telephone support often use IP telephony exclusively to take
advantage of the data abstraction. The benefit of using this technology is the need for
only one class of circuit connection and better bandwidth use. Companies can acquire
their own gateways to eliminate third-party costs, which is worthwhile in some situations.
VoIP is widely employed by carriers, especially for international telephone calls. It is
commonly used to route traffic starting and ending at conventional PSTN telephones.

Chapter: 2
Implementation
2.1 Implementation
Because UDP does not provide a mechanism to ensure that data packets are
delivered in sequential order, or provide Quality of Service (QoS) guarantees, VoIP
implementations face problems dealing with latency and jitter. This is especially true
when satellite circuits are involved, due to long round-trip propagation delay (400–600
milliseconds for links through geostationary satellites). The receiving node must
restructure IP packets that may be out of order, delayed or missing, while ensuring that
the audio stream maintains a proper time consistency. This function is usually
accomplished by means of a jitter buffer in the voice engine.
Another challenge is routing VoIP traffic through firewalls and address
translators. Private Session Border Controllers are used along with firewalls to enable
VoIP calls to and from protected networks. Skype uses a proprietary protocol to route
calls through other Skype peers on the network, allowing it to traverse symmetric NATs
and firewalls. Other methods to traverse firewalls involve using protocols such as STUN
or ICE.
VoIP challenges:
• Available bandwidth
• Network Latency
• Packet loss
• Jitter
• Echo
• Security
• Reliability
• In rare cases, decoding of pulse dialing
Many VoIP providers do not decode pulse dialing from older phones. The VoIP
user may use a pulse-to-tone converter, if needed.
The principal cause of packet loss is congestion, which can sometimes be
managed or avoided. Carrier VoIP networks avoid congestion by means of teletraffic
engineering. Variation in delay is called jitter. The effects of jitter can be mitigated by
storing voice packets in a jitter buffer upon arrival and before producing audio, although
this increases delay. This avoids a condition known as buffer under run, in which the
voice engine is missing audio since the next voice packet has not yet arrived.
Common causes of echo include impedance mismatches in analog circuitry and
acoustic coupling of the transmit and receive signal at the receiving end.

2.2 Reliability
Conventional phones are connected directly to telephone company phone lines,
which in the event of a power failure are kept functioning by back-up generators or
batteries located at the telephone exchange. However, household VoIP hardware uses
broadband modems and other equipment powered by household electricity, which may be
subject to outages in the absence of an uninterruptible power supply or generator. Early
adopters of VoIP may also be users of other phone equipment, such as PBX and cordless
phone bases that rely on power not provided by the telephone company. Even with local
power still available, the broadband carrier itself may experience outages as well. While
the PSTN has been matured over decades and is typically reliable, most broadband
networks are less than 10 years old, and even the best are still subject to intermittent
outages. Furthermore, consumer network technologies such as cable and DSL often are
not subject to the same restoration service levels as the PSTN or business technologies
such as T-1 connection.

2.3 Quality of Service


Proved call quality calculation and a variety of other applications. Some
broadband connections may have less than desirable quality. Where IP packets are lost or
delayed at any point in the network between VoIP users, there will be a momentary drop-
out of voice. This is more noticeable in highly congested networks and/or where there are
long distances and/or interworking between end points. Technology has improved the
reliability and voice quality over time and will continue to improve VoIP performance as
time goes on.
It has been suggested to rely on the packetized nature of media in VoIP
communications and transmit the stream of packets from the source phone to the
destination phone simultaneously across different routes (multi-path routing). In such a
way, temporary failures have less impact on the communication quality. In capillary
routing it has been suggested to use at the packet level Fountain codes or particularly
raptor codes for transmitting extra redundant packets making the communication more
reliable.
A number of protocols have been defined to support the reporting of QoS/QoE for
VoIP calls. These include RTCP XR (RFC3611), SIP RTCP Summary Reports, H.460.9
Annex B (for H.323), H.248.30 and MGCP extensions. The RFC3611 VoIP Metrics
block is generated by an IP phone or gateway during a live call and contains information
on packet loss rate, packet discard rate (due to jitter), packet loss/discard burst metrics
(burst length/density, gap length/density), network delay, end system delay, signal / noise
/ echo level, MOS scores and R factors and configuration information related to the jitter
buffer.
RFC3611 VoIP metrics reports are exchanged between IP endpoints on an
occasional basis during a call, and an end of call message sent via SIP RTCP Summary
Report or one of the other signaling protocol extensions. RFC3611 VoIP metrics reports
are intended to support real time feedback related to QoS problems, the exchange of
information between the endpoints for improved call quality calculation and a variety of
other applications.

2.4 Difficulty with Sending Faxes


The support of sending faxes over VoIP is still limited. The existing voice codecs
are not designed for fax transmission. (They are designed to digitize an analog
representation of a human voice efficiently, but the inefficiency of digitizing an analog
representation (modem signal) of a digital representation (a document image) of analog
data (an original document) more than negates any bandwidth advantage of VoIP. In
other words, the fax “sounds” simply doesn’t fit in the VoIP channel.) An effort is
underway to remedy this by defining an alternate IP-based solution for delivering fax-
over-IP, namely the T.38 protocol. Another possible solution to overcome the drawback
is to treat the fax system as a message switching system, which does not need a real-time
data transmission—such as sending a fax as an email attachment (see Fax) or remote
printout (see Internet Printing Protocol). The end system can completely buffer the
incoming fax data before displaying or printing the fax image.

2.5 Emergency Calls


The nature of IP makes it difficult to locate network users geographically.
Emergency calls, therefore, cannot easily be routed to a nearby call center, and are
impossible on some VoIP systems. Sometimes, VoIP systems may route emergency calls
to a non-emergency phone line at the intended department. In the US, at least one major
police department has strongly objected to this practice as potentially endangering the
public.
Moreover, in the event that the caller is unable to give an address, emergency
services may be unable to locate them in any other way. Following the lead of mobile
phone operators, several VoIP carriers are already implementing a technical work-
around. For instance, one large VoIP carrier requires the registration of the physical
address where the VoIP line will be used. When you dial the emergency number for your
country, they will route it to the appropriate local system. They also maintain their own
emergency call center that will take non-routable emergency calls (made, for example,
from a software-based service that is not tied to any particular physical location) and then
will manually route your call after learning your physical location.
E911 is another method by which VoIP providers in the US are able to support
emergency services. The e911 emergency-calling system automatically associates a
physical address with the calling party's telephone number as required by the Wireless
Communications and Public Safety Act of 1999 and is being successfully used by many
VoIP providers to provide physical address information to emergency service operators.

2.6 IP Telephony Scenarios


The IP telephony usage scenarios, as shown in Figure 1, are commonly classified
by the type of devices terminating an IP call. Because there may be either a PSTN
device (e.g. telephone) or a data-oriented terminal (e.g. personal computer) on each side
of a call, there are four generic classes as below:

2.6.1 PC-To-PC
PC-to-PC communication can be provided for multimedia PCs (i.e. Personal
Computers with a microphone, speaker and a sound card) operating over an IP-based
network without connecting to the PSTN. PC applications (and IP-enabled telephones)
can communicate using point-to-point or multipoint sessions. This set up requires that
parties be equipped to talk at the time of the call. This class is attractive especially for
private users who already have an Internet access and a multimedia PC. Necessary
software is available from several comp anises for free or at a very low cost. There is
usually no charge for PC-to-PC calls, except for the cost of Internet access. The user
doesn’t even have to pay for long-distance calls. This pure-IP scenario can also take
advantage of integration with other Internet services, such as instant messaging, video
conferencing, etc.

Fig.2.1 PC to PC telephone call

2.6.2 PC-To-Telephone
The PC-to-Telephone method allows a user to call any ordinary telephone on a
PSTN from his computer. In this case, a gateway converting the IP call into a PSTN call
has to be used. The gateway is required to be located as near to the called party as
possible, to minimize the price for the gateway-to-called party connection. The call is
converted to a PSTN call at the gateway and is then sent over PSTN to its destination.
Like PC-to-PC calling; this scenario requires a software client. The software is usually
free, but the caller may have to pay a small per-minute charge to a gateway operator.
The cost charged by the operator is determined mainly by the cost of the call placed
from the gateway to the called party. This solution is commercially available from
Net2Phone, Phone serve and many other companies.

2.6.3 Telephone -To-PC


The Telephone-to-PC method allows a user to call from any ordinary telephone
on a PSTN to a PC connected to the Internet. In this case, a gateway converting the
PSTN call into an IP call has to be used, and the gateway is required to be located as
near to the caller as possible. The call is converted to an IP call at the gateway. The
voice data then
“Hops on” the Internet and finds the PC on the other end by using the unique IP
address.
A few companies are providing special numbers or calling cards that allow a
standard telephone user to initiate a call to a computer user. The caveat is that the
computer user must have the vendor's software installed and running on his computer.
This solution may require user to pay local call charges, in addition to small per-minute
charge to a gateway operator.

2.6.4 Telephone -To- Telephone


The Telephone-to-Telephone communication appears like a normal telephone to
the caller but may actually consist of various forms of voice over packet network, all
Interconnected to the PSTN. In this scenario, a caller dials into a gateway using a regular
telephone. The call is converted to an IP call at the gateway and the voice data “hops on”
the Internet. At the end point the voice data hits another gateway and “hops off” the
Internet. The voice data is converted back to PSTN format and sent over the PSTN to its
destination. This class is attractive for those who want to save on long-distance call and
do not want to use PC. Since the call has to pass through two gateways – PSTN-to-
Internet and Internet-to-PSTN, the cost is charged by both the gateway operators. In
addition, the user may have to pay local call charges. This solution is commercially
available from many companies, offering discounted rates for long distance IP telephony
calls.

Fig.2.2 Phone to Phone call via IP

2.7 Integration into Global Telephone Number System


While the wired public switched telephone network (PSTN) and mobile phone
networks share a common global standard (E.164) which allocates and identifies any
specific telephone line, there is no widely adopted similar standard for VoIP networks.
Some allocate an E.164 number which can be used for VoIP as well as incoming and
external calls. However, there are often different, incompatible schemes when calling
between VoIP providers which use provider-specific short codes.

2.8 VoIP Phone Accessibility and Portability


If using a software based soft-phone, calls can only be placed from the computer
on which the soft-phone software resides. Thus with a soft-phone the caller is typically
limited to a single point of calling. When using a hardware based VoIP phone-device/
phone-adapter it is possible to connect traditional analog phones directly to a VoIP
phone-adapter without the need to operate a computer. The converted analog phone
signal can then be connected to multiple house phones or extensions, just as any
traditional phone company signal can be connected. A second VoIP hardware
configuration option involves the use of a specially designed VoIP telephone which
incorporates a VoIP phone adapter directly into the phone itself, and which also does not
require the use of a computer. A third VoIP hardware configuration option involves the
use of a WiFi router and a WiFi SIP phone which can extend a service range throughout a
home or office. WiFi SIP phones can also be used at any location where an
"unauthenticated" open hotspot Wi-Fi signal is available. However, note that many
hotspots require browser-based authentication, which most SIP phones do not support.

2.9 Mobile Phones & Hand Held Devices


Telcos and consumers have invested billions of dollars in mobile phone
equipment. In developed countries, mobile phones have achieved nearly complete market
penetration, and many people are giving up landlines and using mobiles exclusively.
Given this situation, it is not entirely clear whether there would be a significant higher
demand for VoIP among consumers until either public or community wireless networks
have similar geographical coverage to cellular networks (thereby enabling mobile VoIP
phones, so called WiFi phones or VoWLAN) or VoIP is implemented over 3G networks.
However, "dual mode" telephone sets, which allow for the seamless handover between a
cellular network and a WiFi network, are expected to help VoIP become more popular.
Phones like the NEC N900iL, and later many of the Nokia Eseries and several
WiFi enabled mobile phones have SIP clients hardcoded into the firmware. Such clients
operate independently of the mobile phone network unless a network operator decides to
remove the client in the firmware of a heavily branded handset. Some operators such as
Vodafone actively try to block VoIP traffic from their network and therefore most VoIP
calls from such devices are done over WiFi.
Several WiFi only IP hardphones exist, most of them supporting either Skype or
the SIP protocol. These phones are intended as a replacement for PSTN based cordless
phones but can be used anywhere where WiFi internet access is available.
Another addition to hand held devices are ruggedized bar code type devices that
are used in warehouses and retail environments. These types of devices rely on "inside
the 4 walls" type of VoIP services that do not connect to the outside world and are solely
to be used from employee to employee communications.

2.10 Security
Many consumer VoIP solutions do not support encryption yet, although having a
secure phone is much easier to implement with VoIP than traditional phone lines. As a
result, it is relatively easy to eavesdrop on VoIP calls and even change their content. [9]
There are several open source solutions that facilitate sniffing of VoIP conversations. A
modicum of security is afforded due to patented audio codecs that are not easily available
for open source applications, however such security through obscurity has not proven
effective in the long run in other fields. Some vendors also use compression to make
eavesdropping more difficult. However, real security requires encryption and
cryptographic authentication which are not widely available at a consumer level. The
existing secure standard SRTP and the new ZRTP protocol are available on Analog
Telephone Adapters (ATAs) as well as various soft phones. It is possible to use IPsec to
secure P2P VoIP by using opportunistic encryption. Skype does not use SRTP, but uses
encryption which is transparent to the Skype provider.
Chapter: 3
H.323 Protocols

3.1 H.323 Protocols


The H.323 is an umbrella recommendation from the Telecommunication
Standardization Sector of the International Telecommunications Union (ITU-T) that
specifies the components, protocols and procedures that provide multimedia
communication services (real-time voice, video, chat, whiteboard, file sharing, etc.) over
packet–based networks, including IP. By conforming to H.323 standards, multimedia
products and applications from different vendors can interoperate across IP based
networks, including the Internet. H.323 is part of a family of ITU-T recommendations
called H.32x that provides multimedia communication services over a variety of
networks. H.323 covers both protected and unprotected connections. Control and data
information requires a protected transmission to prevent packets from being lost or not
received in the right order. In IP-based networks, TCP protocol guarantees an error-free
transmission in the right order but causes delays and has a lower throughput. Therefore,
unprotected connections are used for audio and video transmissions, which are more
efficient.
H.323 covers both protected and unprotected connections. Control and data
information requires a protected transmission to prevent packets from being lost or not
received in the right order. In IP-based networks, TCP protocol guarantees an error-free
transmission in the right order but causes delays and has a lower throughput. Therefore,
unprotected connections are used for audio and video transmissions, which are more
efficient.
The H.323 standard's mandatory components are transmission of audio,
connection control according to Q.931, communication with the gatekeeper over the
RAS protocol, and use of the H.245 signaling protocol; the rest of the text, including
coverage o f the ability to transmit video and data, is optional. Although H.323 uses TCP
to carry the signaling channels, the real-time media streams are transported on
RTP/RTCP (discussed earlier). RTP carries the actual media and RTCP carries status
and control information.
Being the first widely available VoIP protocol, H.323 enjoyed a head start as
developers implemented it as toll-bypass systems as well as PC-to-phone and video-
conferencing applications. The best-known H.323 application was Microsoft
NetMeeting.

3.2 H.323 Architecture Components


The H.323 standard specifies a number of components (entities), which, when
networked together, provide the point-to-point and point-to-multipoint multimedia
communication services. Some components are mandatory, while others are optional.
Fig.3.1 H.323 VoIP gateway architecture

The five most important components are listed below:

Terminal: An H.323 terminal is an endpoint on a network which provides two-way


communications with another terminal, gateway or a Multipoint Control Unit (MCU). An
H.323 terminal can either be a personal computer or a stand-alone device such as IP
telephone running H.323 and the multimedia applications. It supports audio
communications and can optionally support video or data communications.
Gatekeeper: A gatekeeper provides basic admission control onto a network by allowing
or refusing communications between other H.323 entities within its zone of control. They
also provide call-control services for H.323 endpoints, such as address translation (to use
name instead of IP address), authentication, accounting and bandwidth management.
Gatekeepers in H.323 networks are optional.
Multipoint Control Unit (MCU): An MCU provide services that allow three or more
endpoints to take part in a conference call. All terminals participating in the conference
establish a connection with the MCU. The MCU manages conference resources,
negotiates between terminals for the purpose of determining the audio or video
coder/decoder (codec) to use, and may handle the media stream.
H.245 Control Signaling: H.245 control signaling is used to exchange end -to-end
control messages governing the operation of the H.323 endpoint. The messages carried
include messages to exchange capabilities of terminals and to open and close logical
channels. The H.245 control messages are carried over H.245 control channels.
RTP: Real-Time Transport Protocol (RTP) provides end-to-end network transport
functions for applications transmitting real-time data over IP networks. It provides
services such as payload type identification, sequence numbering, time-stamping, and
delivery monitoring to real-time applications. RTP can also be used with other transport
protocols.

Chapter: 4
Adoption

4.1 Mass-Market Telephony


A major development starting in 2004 has been the introduction of mass-market
VoIP services over broadband Internet access services, in which subscribers make and
receive calls as they would over the PSTN. Full phone service VoIP phone companies
provide inbound and outbound calling with Direct Inbound Dialing. Many offer unlimited
calling to the U.S. and some to Canada or selected countries in Europe or Asia as well,
for a flat monthly fee.
These services take a wide variety of forms which can be more or less similar to
traditional POTS. At one extreme, an analog telephone adapter (ATA) may be connected
to the broadband Internet connection and an existing telephone jack in order to provide
service nearly indistinguishable from POTS on all the other jacks in the residence. This
type of service, which is fixed to one location, is generally offered by broadband Internet
providers such as cable companies and telephone companies as a cheaper flat-rate
traditional phone service. Often the phrase "VoIP" is not used in selling these services,
but instead the industry has marketed the phrases "Internet Phone", "Digital Phone" or
"Soft phone" which is aimed at typical phone users who are not necessarily tech-savvy.
Typically, the provider touts the advantage of being able to keep one's existing phone
number.
At the other extreme are services like Gizmo Project and Skype which rely on a
software client on the computer in order to place a call over the network, where one user
ID can be used on many different computers or in different locations on a laptop. In the
middle lie services which also provide a telephone adapter for connecting to the
broadband connection similar to the services offered by broadband providers (and in
some cases also allow direct connections of SIP phones) but which are aimed at a more
tech-savvy user and allow portability from location to location. One advantage of these
two types of services is the ability to make and receive calls as one would at home,
anywhere in the world, at no extra cost. No additional charges are incurred, as call
diversion via the PSTN would, and the called party does not have to pay for the call. For
example, if a subscriber with a home phone number in the U.S. or Canada calls someone
else within his local calling area, it will be treated as a local call regardless of where that
person is in the world. Often the user may elect to use someone else's area code as his
own to minimize phone costs to a frequently called long-distance number.
For some users, the broadband phone complements, rather than replaces, a PSTN
line, due to a number of inconveniences compared to traditional services. VoIP requires a
broadband Internet connection and, if a telephone adapter is used, a power adapter is
usually needed. In the case of a power failure, VoIP services will generally not function.
Additionally, a call to an emergency services number may not automatically be routed to
the nearest local emergency dispatch center. Some VoIP providers only handle
emergency call for one country. Some VoIP providers offer users the ability to register
their address so that emergency services work as expected.
Another challenge for these services is the proper handling of outgoing calls from
fax machines, DVR boxes, satellite television receivers, alarm systems, conventional
modems or FAXmodems, and other similar devices that depend on access to a voice-
grade telephone line for some or all of their functionality. At present, these types of calls
sometimes go through without any problems, but in other cases they will not go through
at all. And in some cases, this equipment can be made to work over a VoIP connection if
the sending speed can be changed to a lower bits per second rate. If VoIP and cellular
substitution becomes very popular, some ancillary equipment makers may be forced to
redesign equipment, because it would no longer be possible to assume a conventional
voice-grade telephone line would be available in almost all homes in North America and
Western Europe. The TestYourVoIP Web site offers a free service to test the quality of or
diagnose an Internet connection by placing simulated VoIP calls from any Java-enabled
Web browser, or from any phone or VoIP device capable of calling the PSTN.

4.2 Corporate and Telco use


Although few office environments and even fewer homes use a pure VoIP
infrastructure, telecommunications providers routinely use IP telephony, often over a
dedicated IP network, to connect switching stations, converting voice signals to IP
packets and back. The result is a data-abstracted digital network which the provider can
easily upgrade and use for multiple purposes.
Corporate customer telephone support often use IP telephony exclusively to take
advantage of the data abstraction. The benefit of using this technology is the need for
only one class of circuit connection and better bandwidth use. Companies can acquire
their own gateways to eliminate third-party costs, which is worthwhile in some situations.
VoIP is widely employed by carriers, especially for international telephone calls.
It is commonly used to route traffic starting and ending at conventional PSTN telephones.
Many telecommunications companies are looking at the IP Multimedia
Subsystem (IMS) which will merge Internet technologies with the mobile world, using a
pure VoIP infrastructure. It will enable them to upgrade their existing systems while
embracing Internet technologies such as the Web, email, instant messaging, presence, and
video conferencing. It will also allow existing VoIP systems to interface with the
conventional PSTN and mobile phones.
Electronic Numbering (ENUM) uses standard phone numbers (E.164), but allows
connections entirely over the Internet. If the other party uses ENUM, the only expense is
the Internet connection. Virtual PBX (or IP PBX) allows companies to control their
internal phone network over an existing LAN and server without needing to wire a
separate telephone network. Users within this environment can then use standard
telephones coupled with an FXS, IP Phones connected to a data port or a Soft phone on
their PC. Internal VoIP phone networks allow outbound and inbound calling on standard
PSTN lines through the use of FXO adapters

4.3 Uses in Amateur Radio


Sometimes called Radio Over Internet Protocol or RoIP, Amateur radio has
adopted VoIP by linking repeaters and users with Echolink, IRLP, D-STAR, Dingotel
and EQSO. In fact, Echolink allows users to connect to repeaters via their computer (over
the Internet) rather than by using a radio. By using VoIP Amateur Radio operators are
able to create large repeater networks with repeaters all over the world where operators
can access the system with actual ham radios.
Ham Radio operators using radios are able to tune to repeaters with VoIP
capabilities and use DTMF signals to command the repeater to connect to various other
repeaters, thus allowing them to talk to people all around the world, even with "line of
sight" VHF radios.

4.4 Click to Call


Click-to-call is a service which lets users click a button and immediately speak
with a customer service representative. The call can either be carried over VoIP, or the
customer may request an immediate call back by entering their phone number. One
significant benefit to click-to-call providers is that it allows companies to monitor when
online visitors change from the website to a phone sales channel.
Chapter: 5
Legal issues in different countries

As the popularity of VoIP grows, and PSTN users switch to VoIP in increasing
numbers, governments are becoming more interested in regulating VoIP in a manner
similar to PSTN services, especially with the encouragement of the state-mandated
telephone monopolies/oligopolies in a given country, who see this as a way to stifle the
new competition.
In the U.S., the Federal Communications Commission now requires all
interconnected VoIP service providers to comply with requirements comparable to those
for traditional telecommunications service providers. VoIP operators in the U.S. are
required to support local number portability; make service accessible to people with
disabilities; pay regulatory fees, universal service contributions, and other mandated
payments; and enable law enforcement authorities to conduct surveillance pursuant to the
Communications Assistance for Law Enforcement Act(CALEA). VoIP operators also
must provide Enhanced 911 service, disclose any limitations on their E-911 functionality
to their consumers, and obtain affirmative acknowledgements of these disclosures from
all consumers. VoIP operators also receive the benefit of certain U.S. telecommunications
regulations, including an entitlement to interconnection and exchange of traffic with
incumbent local exchange carriers via wholesale carriers. Providers of "nomadic" VoIP
service -- those who are unable to determine the location of their users -- are exempt from
state telecommunications regulation.
Throughout the developing world, countries where regulation is weak or captured
by the dominant operator, restrictions on the use of VoIP are imposed, including in
Panama where VoIP is taxed, Guyana where VoIP is prohibited and India where its retail
commercial sales is allowed but only for long distance service. In Ethiopia, where the
government is monopolizing telecommunication service, it is a criminal offense to offer
services using VoIP. The country has installed firewalls to prevent international calls
being made using VoIP. These measures were taken after popularity in VoIP reduced the
income generated by the state owned telecommunication company.
In the European Union, the treatment of VoIP service providers is a decision for
each Member State's national telecoms regulator, which must use competition law to
define relevant national markets and then determine whether any service provider on
those national markets has "significant market power" (and so should be subject to certain
obligations). A general distinction is usually made between VoIP services that function
over managed networks (via broadband connections) and VoIP services that function
over unmanaged networks (essentially, the Internet).
VoIP services that function over unmanaged networks are often considered to be
too poor in quality to be a viable substitute for PSTN services; as a result, they may be
provided without any specific obligations, even if a service provider has "significant
market power".
The relevant EU Directive is not clearly drafted concerning obligations which can
exist independently of market power (e.g., the obligation to offer access to emergency
calls), and it is impossible to say definitively whether VoIP service providers of either
type are bound by them. A review of the EU Directive is under way and should be
complete by 2007.
In India, it is legal to use VoIP, but it is illegal to have VoIP gateways inside
India. This effectively means that people who have PCs can use them to make a VoIP call
to any number, but if the remote side is a normal phone, the gateway that converts the
VoIP call to a POTS call should not be inside India.
In the UAE, it is illegal to use any form of VoIP, to the extent that websites of
Skype and Gizmo Project don't work.

5.1 IP Telephony in Japan


In Japan, IP telephony (IP 電 話 IP Denwa) is regarded as a service applied by
VoIP technology to whole or a part of the telephone line. As of 2003, IP telephony
services have been assigned telephone numbers. IP telephony services also often include
videophone/video conferencing services. According to the Telecommunication Business
Law, the service category for IP telephony also implies the service provided via Internet,
which is not assigned any telephone number. IP telephony is basically regulated by
Ministry of Internal Affairs and Communications (MIC) as a telecommunication service.
The operators have to disclose necessary information on its quality, etc., prior to
making contracts with customers, and have an obligation to respond to their complaints
cordially. Many Japanese Internet service providers (ISP) are including IP telephony
services. An ISP who also provides IP telephony service is known as a "ITSP (Internet
Telephony Service Provider)". Recently, the competition among ITSPs has been
activated, by option or set sales, in connection with ADSL or FTTH services.
The tariff system normally applied to Japanese IP telephony is described below;
• A call between IP telephony subscribers, limited to the same group, is usually free
of charge.
• A call from IP telephony subscribers to a fixed line or PHS is usually a uniformly
fixed rate all over the country.
Between ITSPs, the interconnection is mostly maintained at VoIP level.
• Where the IP telephony is assigned normal telephone number (0AB-J), the
condition for its interconnection is considered same as normal telephony.
• Where the IP telephony is assigned specific telephone number (050), the
condition for its interconnection is described below.
Interconnection is sometimes charged. (Sometimes, it's free of charge.) In case of
free-of-charge, mostly, communication traffic is exchanged via a P2P connection
with the same VoIP standard. Otherwise, certain conversions are needed at the
point of the VoIP gateway which incurs operating costs.
5.2 Technical Details
The two major competing standards for VoIP are the ITU standard H.323 and the
IETF standard SIP. Initially H.323 was the most popular protocol, though in the "local
loop" it has since been surpassed by SIP. This was primarily due to the latter's better
traversal of NAT and firewalls, although recent changes introduced for H.323 have
removed this advantage.
However, in backbone voice networks where everything is under the control of
the network operator or Telco, H.323 is the protocol of choice. Many of the largest
carriers use H.323 in their core backbones, and the vast majority of callers have little or
no idea that their POTS calls are being carried over VoIP.
User will log on to an IP telephone like they log on to a computer. Phone charging
could then be made on per user instead on per phone basis, which would greatly enhance
user mobility. Many user could log on to the same IP telephone to accept incoming calls
without needing to worry about forwarding calls when leaving their office.
Logging on to an IP telephone with a smart card will make the logging on
procedure very simple and, will furthermore offer a range of new possibilities, from
online shopping and payment to accessing a bank account. The preferred messaging
system for users would be one universal (in and out) box for voice, fax, SMS and e-mail
message. The convergence of these services will enable users to manage fax, e-mail and
voicemail from a single application on a dedicated VoIP terminal.
The survey is the best way to establish which services, functions and applications
are important to users. The question posed to a group of CT (Computer Telephony) users
was:” How useful is each of the following CT features? (Answer rate on a scale of 1 to 10
where 10 is extremely useful a 1 is not at all useful)”. As we can see call switching and
automated routing were identified as the most important services, followed by CLIP and
screen-pops. Name dialing, VoIP and FoIP fared marginally. This example tells us that
VoIP without supplementary services, enhanced functionally and new applications will
not be sufficient to persuade users to migrate to the technology.
Fig.5.1 Useful of CT features

5.3 Applications
IP telephony enables a whole new generation of applications which are
impossible with other telephony architectures. Some examples of the applications that
are likely to be useful are as follows:
1. Advanced Intelligent Network Features: Use advanced intelligent network
(AIN) features such as Caller ID, voice mail, call waiting, pre-and-post paid calling
cards, call blocking, and auto call-back in IP telephony.
2. Voice Calls from Mobile Laptop PCs: Call office or home, fro m hotel,
airport, etc. using multimedia laptop PCs with wireless connection to Internet. This
could be ideal for submitting or retrieving voice messages.
3. Airlines Reservations: Use a Java applet to visually display interactive voice
response options rather than forcing users to wait through very long recorded
instructions and go through multi-level menus requiring the use of a telephone keypad.
4. Internet-aware Telephones: Use enhanced ordinary telephone (wired or
wireless) as an Internet access device as well as for normal telephony. Directory
services, for example, could be accessed over the Internet by submitting a name and
receiving a voice (or text) reply.
5. Voice Annotated Documents: Send voice messages and voice annotated
documents to integrated voice/data mailboxes. Voice annotated documents and
multimedia files can easily become standard within office suites in the near future.
6. Internet Call Center Access: Access customer service agents online over an
Internet call center.
7. Virtual Call Centers: Support the integrated voice and data requirements of
call center agents working from their homes.
8. Live Auction Websites: Create live audio auction websites for excess
inventory. Use Java applets on the phone to manage the bidding process and to track
who “raised a hand” to bid first, etc.
9. Presence and Instant Messaging: Use instant messenger service to determine
when geographically distributed colleagues are available for a quick conference call with
a customer.
10. Electronic Business Cards: Send an enriched electronic virtual business card
(vCard) including photo and audio file automatically with every call as caller ID
information (or selectively during the middle of call).
11. Integrated Voice and Data Information: Integrate voice and data
information collected during a call with sales force automation applications.
12. Personalized Music On-hold: Play personalized announcements or music
from a favorite MP3 recording or Internet radio station while callers are on hold.
Conclusion

Throughout the developing world, countries where regulation is weak or captured


by the dominant operator, restrictions on the use of VoIP are imposed, including in
Panama where VoIP is taxed, Guyana where VoIP is prohibited and India where its retail
commercial sales is allowed but only for long distance service. In Ethiopia, where the
government is monopolizing telecommunication service, it is a criminal offense to offer
services using VoIP. The country has installed firewalls to prevent international calls
being made using VoIP. These measures were taken after popularity in VoIP reduced the
income generated by the state owned telecommunication company.

User will log on to an IP telephone like they log on to a computer. Phone charging
could then be made on per user instead on per phone basis, which would greatly enhance
user mobility. Many user could log on to the same IP telephone to accept incoming calls
without needing to worry about forwarding calls when leaving their office. The preferred
messaging system for users would be one universal (in and out) box for voice, fax, SMS
and e-mail message. The convergence of these services will enable users to manage fax,
e-mail and voicemail from a single application on a dedicated VoIP terminal.

References
• IEEE Network Magazine, Special Issue on Internet Telephony, May 1999.
• IEEE Communications Magazine, Special Issue on Internet Telephony, April
2000.
• H. Liu and P. Mouchtaris, “VoIP Signaling: H.323 and Beyond”, IEEE
Communications Magazine, October 2000.
• ITU-T Recommendation G.107, “The E-Model, a computation model for use in
transmission planning”, December 1998.
• ITU-T Recommendation P.861, “Objective quality measurement of telephone-
band speech codecs”, August 1996.
• IETF RFC 2960, “Stream Control Transmission Protocol”, October 2000
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David G. Kelly, 10/11/2002
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Mohammed,
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Brushteyn,
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• ITU-T Recommendation G.107, “The E-Model, a computation model for use in
transmission planning”, December 1998.
• ITU-T Recommendation P.861, “Objective quality measurement of telephone
band speech codecs”, August 1996.
• IETF RFC 2960, “Stream Control Transmission Protocol”, October 2000.
• ITU-T Recommendation H.225.0, “Call Signaling protocols and media stream
packetization for packet based multimedia communication systems”, September 1999.
• IETF RFC 2705, “Media Gateway Control Protocol (MGCP), Version 1.0”,
October 1999
• IETF RFC 2508, “Compressing IP/UDP/RTP Headers for Low-Speed Serial
Links”, February 1999
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using R-factor and Network Load Estimation Functionality”, has been submitted for
presentation at HSNMC’02, January 2002.

APPENDIX
List of Figures

Title Fig. No. Page No.

1. An overview of IP Telephony Fig.1.1 2

2. Analog telephone adaptor Fig.1.2 4

3. Complete VoIP solution Fig.1.3 5

4. PC to PC telephone call Fig.2.1 11

5. Phone to phone call via IP Fig.2.2 13

6. H.323 VoIP gateway architecture Fig.3.1 17

7. Useful of CT features Fig.5.1 26

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