Beruflich Dokumente
Kultur Dokumente
SEMINAR REPORT
ON
“INTERNET PROTOCOL TELEPHONY”
CERTIFICATE
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.
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
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 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
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.
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.
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.
2.6.1 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.
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.
2.10 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
Chapter: 4
Adoption
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.
5.3 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
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
• “An Introduction to VoIP and VOCAL”, by Luan Dang, Cullen Jennings and
David G. Kelly, 10/11/2002
• “VoIP In A Nutshell“, Website, http://www.no vastars.com
• “Protocols.com”, Website, http://www.protocls.com
• “How IP Telephony Works”, Jeff Tyson, http://comp uter.howstuffworks.co
m/ip-telephony1.htm
• “The Growth of Internet Telephony: Legal and Policy Issues”, Emir A.
Mohammed,
• “VoIP”, http://members.tripod.com/nguyen225/page1.htm
• “Factors in the Success of Voice Quality in Converging Telephony and ip
Networks”, Stefan Pracht
• “Internet Telephony and Voice Compression”, Kelly Ann Smith and Daniel
Brushteyn,
• “Voice over Internet Protocol (VoIP)”, Balz Wyss, Microsoft Corporation,
March 2003
• “The Rise of Internet Telephony”, http://www.nortelnetworks.com
• “Feature Interaction in Internet Telephony”, Jonathan Lennox, Henning
Schulzrinne, Columbia University
• “Next-Gen VoIP Services and Applications Using SIP and Java”, Pingtel,
http://www.techguide.com
• “Voice over IP (VoIP)”,Jerry Ryan, Telogy Networks,
http://www.techguide.com
• “Next Generation Telephony: A Look at Session Initiation Protocol” Thomas
Doumas, May 1999, Hewlett Packard.
• IEEE Network Magazine, Special Issue on Internet Telephony, May 1999.
• IEEE Communications Magazine, Special Issue on Internet Telephony, April
2000.
• R. Cole and J.H. Rosenbluth, Voice over IP performance monitoring”, ACM
Computer Communications Review, Vol. 31, April 2001
• 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.
• 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
• Daniel Collins, Carrier Grade Voice Over IP, McGraw-Hill Professional Telecom
• P. Galiotos, T. Dagiuklas, “QoS Management for an Enhanced VoIP Platform
using R-factor and Network Load Estimation Functionality”, has been submitted for
presentation at HSNMC’02, January 2002.
APPENDIX
List of Figures