Lakshmi VoIP

Seminar 2004 VoIP-Voice over Internet Protocol
Department of Electronics & Communication

Govt Engineering College Thrissur
VoIP
(Voice over Internet Protocol)
Submitted On Submitted by
15-10-04 Lakshmi Menon
S7 ECE
630
Co-ordinator:Muneera C R
Dept of ECE GEC,TRICHUR

ACKNOWLEDGEMENT
First,and foremost I thank God Almighty for making this
venture a success.
I extend my sincere gratitude to Prof. Indiradevi, Head of
Electronics and Communication Department, Govt
Engineering College, Thrissur for providing me with
necessary infrastructure. I would like to convey a deep
sense of gratitude to the seminar co-ordinator Mrs. C R
Muneera for the timely advices.
I also extend my sincere thanks to my friends for their
help.

ABSTRACT
The development of very fast, inexpensive microprocessors
and special-purpose switching chips, coupled with highly reliable
fibre-optic transmission systems, has made it possible to build
economical, ubiquitous, high speed packet-based data networks.
Similarly, the development of very fast, inexpensive digital signal
processors (DSPs) has made it practical to digitize and compress
voice and fax signals into data packets. The natural evolution of
these two developments is to combine digitized voice and fax
packets with packet data, creating integrated data-voice networks.
The voice-over-Internet protocol (VoIP) technology allows voice
information to pass over IP data networks. Primarily, the cost
savings that accrue from operating a single, shared network have
motivated this convergence of telecommunications and data
communications. VoIP allows you to make telephone calls using a
computer network, over a data network like the Internet. VoIP
converts the voice signal from your telephone into a digital signal
that travels over the Internet then converts it back at the other end
so you can speak to anyone with a regular phone number.

INDEX
PAGE
S.No TOPIC NO.
BASIC FLOW OF VoIP NETWORK
1
VOICE GATEWAY
2
A TYPICAL VoIP NETWORK
3
APPLICATIONS
4
IDENTIFICATION OF MAJOR SYSTEM COMPONENTS
• Gateways
• Gatekeepers
5
• IP Telephones
• PC Software Phones
VoIP PRODUCTS
• Hard Phones
6
• Soft Phones
VoIP QoS (Quality of Service) ISSUES
• Delay
7 • Lost Packet Compensation
• Echo Compensation
ADVANTAGES OF USING VoIP
8
TECHNICAL BARRIERS
9
FUTURE OF VoIP TELEPHONY
10
CONCLUSION
11
REFERENCES
12

BASIC FLOW OF VOIP NETWORK
The VoIP networks replace the traditional public-switched

telephone networks (PSTNs), as these can perform the same
functions as the PSTN networks. The functions performed include
signaling, databasing, call connect and disconnect, and coding-
decoding.
Signaling. Signaling in a VoIP network is accomplished by the

exchange of datagram messages between the components. The
format of these messages is covered by the standard datalink layer
protocols.
Database services. Database services are a way to locate an

endpoint and translate the addressing that two networks use; for
example, the PSTN uses phone numbers
to identify endpoints, while a VoIP network could use an IP
address and port numbers to identify an endpoint. A call
control database contains these mappings and translations.
Call connect and disconnect (bearer control). The connection of a

call is made by two endpoints opening communication sessions
between each other. In the PSTN,the public (or private) switch
connects logical channels through the network to complete the
calls. In a VoIP implementation, a multimedia stream (audio,
video, or both) is transported in real time. The connection path is
the bearer channel and represents the voice or video content being
delivered. When communication is complete, the IP sessions are
released and, optionally, network resources are freed.
CODEC operations. Voice communication is analogue, while data

networking is digital. Analogue waveforms are converted into
digital information by using a coder-decoder (CODEC).

VOICE GATEWAY
The VoIP network acts as a gateway to the existing PSTN
network. This gateway forms the interface for transportation of the
voice content over the IP network. Gateways are responsible for
call origination, call detection, analogue-to-digital conversion of
voice, and creation of voice packets (CODEC functions).
Voice(analogue and/or digital) compression, echo cancellation,
silence suppression, and statistics gathering are their optional
features. The gateways must also perform some of the database
services, such as phone number translations, host lookup, and
signaling. The extent of gateway functionalities is based on the
VoIP-enabling products used. Fig. 1 shows the architecture of a
typical gateway. The DSP in a gateway is responsible for signal
processing functions such as analogue- to-digital conversion of
voice signals, voice compression, echo cancellation, and voice-
activity detection. The functions like call origination, call
detection, signaling, and phone number translations are performed
by the microprocessor. Gateways exist in several forms; for
example, the gateway could be a dedicated telecommunication
equipment chassis, or even a generic PC running VoIP software.
FIG 1 ARCHITECTURE OF A TYPICAL GATEWAY

A TYPICAL VOIP NETWORK

Fig. 2 shows a typical VoIP network. The IP network should
ensure smooth delivery of voice and signaling information to the
VoIP elements. Since the IP network is to carry both voice and
data, it must be able to prioritize the voice traffic. This
prioritization is required for real-time VoIP applications to ensure
that voice traffic is unaffected by other network traffic. Without
prioritization, the voice packets may be bogged down by heavy
data traffic like large file transfers using file transfer protocol
(FTP).The voice packets are encapsulated with real-time protocol
(RTP) and real-time control protocol (RTCP) for real-time transfer.
The resource reservation protocol (RSVP) is used at the
networking gateways (such as the routers) to reserve a particular
amount of bandwidth for real-time applications (VoIP, video
multicasting, etc).
FIG 2 A TYPICAL FULL SERVICE VOIP NETWORK

Unlike the PCM data streams in circuitswitched telephony,

VoIP data travels over the networks in packets. In VoIP digitized
voice is bundled into IP packets and sent out into the network for
delivery. Routers, switches, and other network equipment direct
the packets to their destination IP address. This mode is called
packetswitched telephony. The transport of voice packets is
affected by several factors, such as the amount of bandwidth
available in the network connection, the delay that the packet
experiences, and any packet loss or corruption that occurs. The
ability of the network to deliver the voice packets quickly and
consistently is referred to as Quality of Service (QoS).

APPLICATIONS
A wide variety of applications are enabled by the
transmission of VoIP networks. The first application, shown in
Figure 1, is a network configuration of an organization with many
branch offices (e.g., a bank) that wants to reduce costs and
combine traffic to provide voice and data access to the main office.
This is accomplished by using a packet network to provide
standard data transmission while at the same time enhancing it to
carry voice traffic along with the data. Typically, this network
configuration will benefit if the voice traffic is compressed.
Voice over packet provides the interworking function (IWF),

which is the physical implementation of the hardware and software
that allows the transmission of combined voice and data over the
packet network. The interfaces the IWF must support in this case
are analog interfaces, which directly connect to telephones or key
systems. The IWF must emulate the functions of both a private
branch exchange (PBX) for the telephony terminals at the
branches, as well as the functions of the telephony terminals for the
PBX at the home office.
FIGURE 1. BRANCH OFFICE APPLICATION

A second VoIP application, shown in Figure 2, is a trunking

application. In this scenario, an organization wishes to send voice
traffic between two locations over the packet network and replace
the tie trunks used to connect the PBXs at the locations. This
application usually requires the IWF to support a higher-capacity
digital channel than the branch application, such as a T1/E1
interface of 1.544 or 2.048 Mbps. The IWF emulates the signaling
functions of a PBX, resulting in significant savings to companies'
communications costs.
FIGURE 2. INTEROFFICE TRUNKING APPLICATION
A third application of VoIP software is interworking with

cellular networks, as shown in Figure 3. The voice data in a digital
cellular network is already compressed and packetized for
transmission over the air by the cellular phone. Packet networks
can then transmit the compressed cellular voice packet, saving a
tremendous amount of bandwidth. The IWF provides the

transcoding function required to convert the cellular voice data to

the format required by the public switched telephone network
(PSTN).
FIGURE 3. INTEROFFICE TRUNKING APPLICATION
Mainly the different types of communications that exist in a

VoIP are :
PC to PC connection.
PC to PHONE connection.
PHONE to PHONE connection

PC to PC Communication:
Need a PC with sound card

IP Telephony software: Cuseeme, Internet Phone, ...
Video optional
PC to Phone Communication:
Need a gateway that connects IP network to phone network

(Router to PBX)

``PC TO PHONE CONNECTION
Phone to Phone communication:
Need more gateways that connect IP network to phone

Networks.
The IP network could be dedicated intra-net or the Internet.
The phone networks could be intra-company PBXs or the
carrier switches.

Phone to Phone communication

IDENTIFICATION OF MAJOR SYSTEM

COMPONENTS
Gateways
The gateways are the devices that communicate between the
telephone signals and the IP endpoint. The IP endpoint usually
speaks H.323 for media stream and more recently Session
Initiation protocol (SIP). The gateways usually perform the
following 6 functions:
• Search function
When an IP gateway is used to place a call across an
IP network, it receives a called party phone number. It
converts it into the IP address of the far end gateway,
possibly through a table lookup in the originating gateway
or in a centralized directory server.
• Connection Function
The originating gateway establishes a connection to
the destination gateway, exchanges call setup,
compatibility information and performs any option
negotiation and security handshake.
• Digitizing function
Analog telephone signals coming into a trunk on the
gateway are digitized by the gateway into a format useful
to the gateway, usually 64 kbps PCM. This requires the
gateway to interface to a variety of Telephone-signaling
conventions.

• Demodulation functions
With some gateways the gateway trunk can accept
only a voice signal or a fax signal but not both. But
sophisticated gateways handle both. When the signal is a
fax,it is demodulated by the DSP back into the original
2.4-14.4 kbps digital format. This is then put into the IP
packets for transmission. The demodulated information is
remodulated back to the original analog fax signal by the
remote
• Compression functions
When the signal is determined to be voice, it is
usually compressed by a DSP from 64K PCM to a 5.3
Kbps signal, which is the G.723.1 standard.
• Decompression and Remodulation functions

At the same time that the gateway performs steps 1-
5, it is also receiving packets. Hence this function is
required
Gatekeepers
Terminals are the L AN client endpoints that provide real time

two-way communications. When an endpoint is switched on, it
performs a multicast discovery for a gatekeeper and registers with
it. Thus the gatekeeper knows how many users are connected and
where they are located. The collection of a gatekeeper and its
registered endpoints is called as a zone. A gatekeeper is required to
perform the following functions:
• Address translation
Translation of an alias address to a Transport
Address using a table updated via Registration messages.

• Admissions control
Authorization of LAN access, using Admissions
Requests or Confirm and Reject (ARQ/ARC/ARJ),
bandwidth or some other criteria.
• Bandwidth management
Support for Bandwidth Request, Confirm and Reject
messages, or a null function that accepts all requests for
bandwidth changes.
• Zone management
The Gatekeeper provides the above functions
forterminals, MCUs, and Gateways, which are registered in
its Zone of control.
FIG 1 ZONE MANAGEMENT IN A GATEKEEPER

FIG 2 INTERZONE GATEWAY GATEKEEPER COMMUNICATION
FIG 3 INTRAZONE GATEWAY GATEKEEPER COMMUNICATION
IP Telephones
These are devices, which replace the existing telephones by
providing enhanced services suited to VOIP. At the same time they
should retain the capabilities of the original phones to keep the
user comfortable.
PC Software phones
This arrangement consists of a microphone connected to a PC
interfaced by a card and running a software, which permits voice
and multimedia transfer over the Internet.
Microsoft NetMeeting is an example.

VOIP PRODUCTS
Hard Phones
• Broadband Hard Phones

A broadband hard phone is a self contained IP
telephone that looks just like a conventional phone but
instead of a conventional phone jack, it has an Ethernet port
through which it communicates directly with a VoIP server,
VoIP gateway or another VoIP phone. Since a broadband
hard phone communicates directly with a VoIP server, VoIP
gateway or another VoIP phone it does not require any
personal computer nor any software running on a personal
computer to make or receive VoIP phone calls. It can be used
independently, all that is required is an internet connection.
While PC based software solutions are cheaper, a hard phone
is the best solution for IP telephony.
• Dialup Hard Phones

A dialup hard phone is a hard phone with a built-in
modem instead of the Ethernet port. It will connect through
the modem via a dialup internet service to a remote VoIP
server and is therefore self contained. It does not require a
personal computer nor any software to be run on a personal
computer to make and receive VoIP phone calls. All that is
required is a phone line and a dialup internet account. Dialup
hard phones are popular in countries where there is very little
broadband infrastructure yet.

• WLAN or WiFi Phones

A WLAN or WiFi phone is a hard phone with a built-in
WiFi transceiver unit instead of an Ethernet port to connect
to a WiFi base station and from there to a remote VoIP
server. It does not require a personal computer nor any
software to be run on a personal computer to make and
receive VoIP phone calls. All that is required is access to a
WiFi base station.
Soft Phones
A soft phone is an IP telephone in software. It can be
installed on a personal computer and function as an IP phone. Soft
phones require appropriate audio hardware to be present on the
personal computer they run. This can either be a sound card with
speakers or earphones and a microphone, or, alternatively a USB
phone set. Soft phones are inferior to hard phones but cheaper to
obtain, many are available as a free download.

VOIP QOS(QUALITY OF SERVICE)

ISSUES
The advantages of reduced cost and bandwidth savings of
carrying voice-over-packet networks are associated with some
quality-of-service (QoS) issues unique to packet networks.
Delay
Delay causes two problems: echo and talker overlap. Echo is
caused by the signal reflections of the speaker's voice from the far-
end telephone equipment back into the speaker's ear. Echo
becomes a significant problem when the round-trip delay becomes
greater than 50 milliseconds. As echo is perceived as a significant
quality problem, voice-over-packet systems must address the need
for echo control and implement some means of echo cancellation.
Talker overlap (or the problem of one talker stepping on the other
talker's speech) becomes significant if the one-way delay becomes
greater than 250 milliseconds. The end-to-end delay budget is
therefore the major constraint and driving requirement for reducing
delay through a packet network.
The following are sources of delay in an end-to-end, voice-

over-packet call:
• Accumulation Delay (Sometimes Called Algorithmic

Delay)
This delay is caused by the need to collect a frame of

voice samples to be processed by the voice coder. It is related
to the type of voice coder used and varies from a single

sample time (.125 microseconds) to many milliseconds. A

representative list of standard voice coders and their frame
times follows:
• G.726 adaptive differential pulse-code modulation

(ADPCM) (16, 24, 32, 40 kbps)—0.125 microseconds
• G.728 LD–code excited linear prediction (CELP)(16
kbps)—2.5 milliseconds
• G.729 CS–ACELP (8 kbps)—10 milliseconds
• G.723.1 Multirate Coder (5.3, 6.3 kbps)—30
milliseconds
• Processing Delay
This delay is caused by the actual process of

encoding and collecting the encoded samples into a
packet for transmission over the packet network. The
encoding delay is a function of both the processor
execution time and the type of algorithm used. Often,
multiple voice-coder frames will be collected in a single
packet to reduce the packet network overhead. For
example, three frames of G.729 code words, equaling
30 milliseconds of speech, may be collected and packed
into a single packet.
• Network Delay
This delay is caused by the physical medium and

protocols used to transmit the voice data and by the
buffers used to remove packet jitter on the receive side.
Network delay is a function of the capacity of the links
in the network and the processing that occurs as the

packets transit the network. The jitter buffers add delay,

which is used to remove the packet-delay variation to
which each packet is subjected as it transits the packet
network. This delay can be a significant part of the
overall delay, as packet-delay variations can be as high
as 70 to 100 milliseconds in some frame-relay and IP
networks.
• Jitter
The delay problem is compounded by the need to

remove jitter, a variable interpacket timing caused by
the network a packet traverses. Removing jitter requires
collecting packets and holding them long enough to
allow the slowest packets to arrive in time to be played
in the correct sequence. This causes additional delay.
The two conflicting goals of minimizing delay

and removing jitter have engendered various schemes
to adapt the jitter buffer size to match the time-varying
requirements of network jitter removal. This adaptation
has the explicit goal of minimizing the size and delay of
the jitter buffer, while at the same time preventing
buffer underflow caused by jitter.
Two approaches to adapting the jitter buffer size

are detailed below. The approach selected will depend
on the type of network the packets are traversing.
The first approach is to measure the variation of

packet level in the jitter buffer over a period of time and
incrementally adapt the buffer size to match the
calculated jitter. This approach works best with
networks that provide a consistent jitter performance
over time, such as ATM networks.

The second approach is to count the number of

packets that arrive late and create a ratio of these
packets to the number of packets that are successfully
processed. This ratio is then used to adjust the jitter
buffer to target a predetermined, allowable late-packet
ratio. This approach works best with the networks with
highly variable packet-interarrival intervals—such as IP
networks.
In addition to the techniques described, the network must be

configured and managed to provide minimal delay and jitter,
enabling a consistent QoS.
Lost-Packet Compensation
Lost packets can be an even more severe problem, depending
on the type of packet network that is being used. Because IP
networks do not guarantee service, they will usually exhibit a
much higher incidence of lost voice packets than ATM networks.
In current IP networks, all voice frames are treated like data. Under
peak loads and congestion, voice frames will be dropped equally
with data frames. The data frames, however, are not time sensitive,
and dropped packets can be appropriately corrected through the
process of retransmission. Lost voice packets, however, cannot be
dealt with in this manner.
Some schemes used by voice-over-packet software to address

the problem of lost frames are as follows:
• interpolate for lost speech packets by replaying the last

packet received during the interval when the lost packet was
supposed to be played out; this scheme is a simple method
that fills the time between noncontiguous speech frames; it
works well when the incidence of lost frames is infrequent; it

does not work well if there are a number of lost packets in a

row or a burst of lost packets
• send redundant information at the expense of bandwidth
utilization; this basic approach replicates and sends the nth
packet of voice information along with the (n+1)th packet;
this method has the advantage of being able to correct for the
lost packet exactly; however, this approach uses more

bandwidth and also creates greater delay
• use a hybrid approach with a much lower bandwidth voice

coder to provide redundant information carried along in the
(n+1)th packet; this reduces the problem of the extra
bandwidth required but fails to solve the problem of delay.
Echo Compensation
Echo in a telephone network is caused by signal reflections
generated by the hybrid circuit that converts between a four-wire
circuit (a separate transmit and receive pair) and a two-wire circuit
(a single transmit and receive pair). These reflections of the
speaker's voice are heard in the speaker's ear. Echo is present even
in a conventional circuit-switched telephone network. However, it
is acceptable because the round-trip delays through the network are
smaller than 50 milliseconds and the echo is masked by the normal
side tone every telephone generates.
Echo becomes a problem in voice-over-packet networks

because the round-trip delay through the network is almost always
greater than 50 milliseconds. Thus, echo-cancellation techniques
are always used. ITU standard G.165 defines performance
requirements that are currently required for echo cancellers. The
ITU is defining much more stringent performance requirements in
the G.IEC specification. Echo is generated toward the packet

network from the telephone network. The echo canceller compares

the voice data received from the packet network with voice data
being transmitted to the packet network.

ADVANTAGES OF USING VOIP

When you are using PSTN line, you typically pay for time
used to a PSTN line manager company: more time you stay at
phone and more you'll pay. In addition you couldn't talk with other
that one person at a time.
In opposite with VoIP mechanism you can talk all the time
with every person you want (the needed is that other person is also
connected to Internet at the same time), as far as you want (money
independent) and, in addition, you can talk with many people at the
same time.
If you're still not persuaded you can consider that, at the same
time, you can exchange data with people are you talking with,
sending images, graphs and videos.
• Integration of Voice and Data

The integration of voice and data traffic will be demanded by
multi application software. The inevitable evolution will be web
servers capable of interacting with voice, data and images.
• Simplification
An integrated infra structure that supports all forms of
communication allows more standardization and lesser equipment
management. The result is a fault tolerant design.
• Network Efficiency
The integration of voice and data effectively fills up the data
communication channels efficiently, thus providing bandwidth
consolidation. The idea is to move away from the TDM scheme
wherein the user is given bandwidth when he is not talking. Data
networks do not do this. It is a big saving when one considers the

statistics that 50% of a conversation is silence. The network

efficiency can be further boosted, by removing the redundancy in
certain speech patterns.
• Cost reduction
The Public Switched Telephone Networks' toll services can
be bypassed using the Internet backbone, which means slash in
prices of the long distance calls. However these reductions may
slightly decrease when the Federal communications Commission
(FCC) removes the Enhanced Service Provider (ESP) status
granted to Internet service providers (ISPs) by which they do not
have to pay the local access fees to use the telephone company
(TELCO) local access facilities.Access fees form a significant part
of all long distance calls. But in spite of this, the circuit switched
telephony would be expensive because of lack of bandwidth
consolidation and speech compression techniques.

TECHNICAL BARRIERS
The ultimate objective of Internet telephony is, of course,

reliable, high-quality voice service, the kind that users expect from
the PSTN. At the moment, however, that level of reliability and
sound quality is not available on the Internet, primarily because of
bandwidth limitations that lead to packet loss. In voice
communications, packet loss shows up in the form of gaps or
periods of silence in the conversation, leading to a clipped-speech
effect that is unsatisfactory for most users and unacceptable in
business communications.
The Internet, a collection of more than 130,000 networks, is

gaining in popularity as millions of new users sign on every month.
The increasingly heavy use of the Internet's limited bandwidth
often results in congestion, which, in turn, can cause delays in
packet transmission. Such network delays mean packets are lost or
discarded.
In addition, because the Internet is a packet-switched or

connectionless network, the individual packets of each voice signal
travel over separate network paths for reassembly in the proper
sequence at their ultimate destination. While this makes for a more
efficient use of network resources than the circuit-switched PSTN,
which routes a call over a single path, it also increases the chances
for packet loss.
Network reliability and sound quality also are functions of

the voice-encoding techniques and associated voice-processing
functions of the gateway servers. To date, most developers of
Internet-telephony software, as well as vendors of gateway servers,
have been using a variety of speech-compression protocols. The
use of various speech-coding algorithms—with their different bit
rates and mechanisms for reconstructing voice packets and

handling delays—produces varying levels of intelligibility and

fidelity in sound transmitted over the Internet. The lack of
standardized protocols also means that many Internet-telephony
products do not interoperate with each other or with the PSTN.

FUTURE OF VOICE-OVER-INTERNET
PROTOCOL (VOIP) TELEPHONY
Several factors will influence future developments in VoIP

products and services. Currently, the most promising areas for
VoIP are corporate intranets and commercial extranets. Their IP–
based infrastructures enable operators to control who can—and
cannot—use the network.
Another influential element in the ongoing Internet-telephony

evolution is the VoIP gateway. As these gateways evolve from
PC–based platforms to robust embedded systems, each will be able
to handle hundreds of simultaneous calls. Consequently,
corporations will deploy large numbers of them in an effort to
reduce the expenses associated with high-volume voice, fax, and
videoconferencing traffic. The economics of placing all traffic—
data, voice, and video—over an IP–based network will pull
companies in this direction, simply because IP will act as a
unifying agent, regardless of the underlying architecture (i.e.,
leased lines, frame relay, or ATM) of an organization's network.
Commercial extranets, based on conservatively engineered IP

networks, will deliver VoIP and facsimile over Internet protocol
(FAXoIP) services to the general public. By guaranteeing specific
parameters, such as packet delay, packet jitter, and service
interoperability, these extranets will ensure reliable network
support for such applications.
VoIP products and services transported via the public

Internet will be niche markets that can tolerate the varying
performance levels of that transport medium. Telecommunications
carriers most likely will rely on the public Internet to provide
telephone service between/among geographic locations that today

are high-tariff areas. It is unlikely that the public Internet's

performance characteristics will improve sufficiently within the
next two years to stimulate significant growth in VoIP for that
medium.
However, the public Internet will be able to handle voice and

video services quite reliably within the next three to five years,
once two critical changes take place:
• an increase by several orders of magnitude in backbone

bandwidth and access speeds, stemming from the deployment
of IP/ATM/synchronous optical network (SONET) and
ISDN, cable modems, and x digital subscriber line (xDSL)
technologies, respectively
• the tiering of the public Internet, in which users will be
required to pay for the specific service levels they require
On the other hand, FAXoIP products and services via the

public Internet will become economically viable more quickly than
voice and video, primarily because the technical roadblocks are
less challenging. Within two years, corporations will take their fax
traffic off the PSTN and move it quickly to the public Internet and
corporate Intranet, first through FAXoIP gateways and then via IP–
capable fax machines. Standards for IP–based fax transmission
will be in place by the end of this year.
Throughout the remainder of this decade, videoconferencing

(H.323) with data collaboration (T.120) will become the normal
method of corporate communications, as network performance and
interoperability increase and business organizations appreciate the
economics of telecommuting. Soon, the video camera will be a
standard piece of computer hardware, for full-featured multimedia
systems, as well as for the less-than-$500 network-computer
appliances now starting to appear in the market. The latter in
particular should stimulate the residential demand and bring VoIP

services to the mass market—including the roughly 60 percent of

American households that still do not have a PC.
The migration to VoIP networks is being driven by a number

of factors; the key concept is that of advanced services. These
services will be able to provide all the existing AIN features and
add new ones based on IP services. A key aspect of the new VoIP
infrastructure is that there is no need to build circuit switched
connections between the devices, which reduces the cost of
providing the services, and simplifies deployment. All systems,
except the media gateway itself, require only IP interfaces. The IP
interfaces of these devices can provide the telephony signaling as
well as the media interfaces. This provides for simpler distributed
signaling and processing capability, reduces the cost of
components, and speeds up application development and
deployment. The new IP based applications can be delivered in a
variety of methods depending upon their complexity. For simple
applications the Media Gateway Controller (MGC) can provide the
application intelligence in a very distributed fashion. The MGC
provides call control to the user via the Media Gateway with a
client/server protocol called MGCP. The MGC controls all
routing and call control to the devices within its’ MGCP domain.
This functions very similarly to a Class-5 End Office switch and
provides the same features one would expect on a standard POTS
line. However it also has the ability to play tone and
announcements to a caller, as well as gather digits from the caller.
This provides capabilities similar to an SCP or Announcement
Server for simple applications. These features are provided by the
capabilities defined in MGCP and H.248. An example
application and call flow is shown below. In this application a
caller incorrectly dials a number and receives a message such as:
“Your call cannot be completed as dialed”.

Advanced Intelligent Network applications
Today’s Advanced Intelligent Network (AIN) provides many

useful features common to callers today. These features include
Caller ID, Voice Mail, Call Waiting, Pre-and-post paid calling
cards, 911, Call Blocking, and Auto Call-back to name a few.
These features represent years of development and investment by
vendors and service providers, and are delivered via a proven
circuit switched infrastructure. The network architecture is shown
in general format below:
.
A subscriber in this network is provided primary dial tone

and feature set by the Class-5 End Office switch. This system
provides the basic call control features such as Dial Tone, Call
Waiting, forwarding to Voice Mail, and billing. It passes calls to
the Class-4 Tandem switch as needed based on the dial plan in the
region. For messaging, a local mail server is usually employed for
the subscriber base in the region. The calls requiring other
advanced features such as 911, Local Number Portability, or 800
service are forwarded to a TCAP Service Control Point (SCP) for
servicing when the proper Trigger Detection Points are met. A
connection is established from the subscriber to an Intelligent

Peripheral when media output and input analysis is required. Once

the information is processed, with the help of a backend Database,
the call can then be rerouted and the subscriber connected to the
proper destination with the proper billing. Although this system is
proven and widely deployed it is inefficient in terms of port and
equipment utilization. A connection must be established between
the End Office Switch and the Tandem switch as well as a
connection from the Tandem to the SCP, or the Tandem to the IP.
Numerous devices and paths could be required for any given call.
It also mandates that the Intelligent Peripherals, Voice Mail Server,
and some SCP have telephony interfaces as well as database
access. This creates an expensive system, which has potential
capacity constraints.
VoIP Advanced Applications
The migration to VoIP networks is being driven by a number

of factors; the key concept is that of advanced services. These
services will be able to provide all the existing AIN features and
add new ones based on IP services. A key aspect of the new VoIP
infrastructure is that there is no need to build circuit switched
connections between the devices, which reduces the cost of
providing the services, and simplifies deployment. All systems,
except the media gateway itself, require only IP interfaces. The IP
interfaces of these devices can provide the telephony signaling as
well as the media interfaces. This provides for simpler distributed
signaling and processing capability, reduces the cost of
components, and speeds up application development and
deployment. The new IP based applications can be delivered in a
variety of methods depending upon their complexity. For simple
applications the Media Gateway Controller (MGC) can provide the
application intelligence in a very distributed fashion. The MGC
provides call control to the user via the Media Gateway with a
client/server protocol called MGCP. The MGC controls all routing

and call control to the devices within its’ MGCP domain. This
functions very similarly to a Class-5 End Office switch and
provides the same features one would expect on a standard POTS
line. However it also has the ability to play tone and
announcements to a caller, as well as gather digits from the caller.
This provides capabilities similar to an SCP or Announcement
Server for simple applications. These features are provided by the
capabilities defined in MGCP and H.248. An example application
and call flow is shown below. In this application a caller
incorrectly dials a number and receives a message such as: “Your
call cannot be completed as dialed”.
EXAMPLE 1. MEDIA GATEWAY PROVIDED PROMPTS

The above scenario is fairly simple and saves bandwidth on

the network. Since the announcement is stored on the MG there is
no bandwidth required except the MGCP signaling shown. This is
effective for small networks with minimal prompting requirements.
However due to the need for coordination of the file systems of the
MGs and the messaging of the MGCs it can create administrative
overhead. When networks grow to a certain size it becomes much
easier to administer a separate Media or Announcement Server.
This centralizes the configuration/maintenance for the audio files
and allows all gateways in the domain to use the same source files.
The calls are connected as though the Media Server was simply
another endpoint, but in this case the endpoint is instructed to play
a file using the same messages used in the above example. A
network topology using a Media Server is shown below.

The approach shown above permits many MGs to share the

file information stored in the Announcement Server, so their local
files are not used. In general this will be made available to all MGs
within the direct control of the MGC. This is called the MGC’s
domain, and it controls all ports and all calls within its domain.
Nonetheless this approach does require the MGC to be aware of
the Announcement Server and know at what point to refer the
caller to which prompt. This logic is fairly simple in the case of a
misdialed number but is more difficult when the application
requires more complexity and interaction with the caller. In the
case of an Interactive Voice Response (IVR) server providing
calling card service the logic for the interaction is much greater
and is not feasibly controlled by an MGCP endpoint. In this case
the Announcement Server has the ability to play announcements
and retrieve DTMF tones, but due to the nature of MGCP, cannot
easily control a call. This would be analogous to a POTs subscriber
attempting to transfer a call on a Class-5 End Office switch, while
maintaining control of the call. In the case of a Calling Card
application the MGC can intercept the callers DTMF input, but the
logic must be tightly coupled with the prompts of the

Announcement Server. In practice the coordination of this is

beyond the capability of most MGCs and the function is moved
onto a more specialized platform. This is usually done with a SIP
Application Server.
Scaling Networks and Adding Intelligence
As the VoIP networks grow they require a centralized control

system for the specialized applications that become incorporated
into the network. SIP provides for a lightweight and flexible
protocol and architecture for this type of application server. In the
case of a SIP Application Server the MGC passes the call to the
appropriate server, which handles the call logic required for the
call. In the case of misdialed call for instance the MGC can route
the call to the server with a specific URI on the server and tells the
server to play the appropriate message. All routing is handled
similarly by the MGC, reducing the logic and processing
requirements. Since the MGC can load level between a number of
SIP Servers it is very beneficial to minimize the loading on the
MGC and maximize the loading on the SIP Servers. This provides
a more scalable End Office environment with a single MGC
routing calls to many redundant SIP Servers. In addition the
Application Server is available to take calls from any MGCP
Domain in the network and can provide a centralized point for
database access when used in advanced applications such as IVRs.
The database activity can also be run in a distributed fashion
depending upon the back end database selected. An example of the
misdialed call is shown below.

EXAMPLE 3. SIP APPLICATION SERVER
The application here is very simple, however in more

advanced applications the requirement for full call control becomes
more critical. In some applications the call must be returned to the

Application Server due to lack of pre-paid funds and the caller

given the chance to “recharge” their calling card. This requires a
well-written script application to intelligently handle the
unpredictable range of responses, as well as database read/write
capability. It requires the IVR to have complete call control for re-
establishing calls in progress, and providing prompts based on
database information. It is important to note that the RTP media
stream can be originated or terminated by the Application Server.
A unified messaging server for instance can receive RTP media
traffic from a gateway and perform many different operations on
this media. The media can be stored as speech, and it can be
retrieved as speech as in normal voice mail. It could also be
converted to an email message or the Application Server could
even perform speech recognition on the inbound media. Designs
have been put in place for an entire speech driven web application;
SIP will be used to access these very intelligent voice assistants
and voice browsers. The call flow for such a call is shown below
but it utilizes the same fundamental and simple equipment listed in
the above application. Whatever control logic is used, such as
XML or CPL, the only thing that is changed is the complexity of
the call flow and the application. In this way a single SIP
Application Server (or Server Farm) can manage to perform many
application within a network, and have those application easily and
simply changed-out or upgraded.

CONCLUSION
VoIP technology offers broadband services and the
integration of voice and data at all levels. One key factor that is
driving the VoIP application development and deployment is
reduced voice service charges. In addition to cost advantages,
VoIP services have compelling technical advantages over circuit
switching.Moreover,VoIP is suitable for computer telephony
integration and other next generation applications. A VoIP network
can centralize much of its intelligence so the management of the
network can also be centralized and there is no longer a need to
have diverse PBXs located around the country and around the
world. A service provider can centralize the network operations
and billing. Enterprises can also deploy an IP telephony system
next to the existing PBX integrating it into a new and expanding IP
telephony network. This allows them to leverage the existing PBX
and all the expansion will be on the new IP Telephony system,
while still maintaining a standard dial plan and seamless voice
network. Branch offices can be swapped over to the IP telephony
system when the existing PBXs are ready to be removed. VoIP will
one day make voice communications an integrated element of rich
communication experience that includes video and data. VoIP will
no longer be a service but a technology that is utilized in an
application that may run on a computer, a PDA or other
information and communication appliance. Voice communication
will be initiated via something like an e-mail address, hyper link or
initiated within an application the application itself.VOIP is
growing fast. The very knowledge of the applications of this
technology is enough for users and manufacturers to flock towards
it. It is ideal for computer based communications and at the same
time bringing down the cost of multimedia transfer.

REFERENCES
Internet Telephony Conference &EXPO.
IEC:Voice over Internet Protocol.
“Voice over IP issues and challenges”,by Prof Raj

Jain,Ohio State University.
“VoIP for next generation economical telephony” -

INFOTECH
“Advanced VoIP applications”,Glen Gerhard

Lakshmi VoIP

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Seminar 2004 VoIP-Voice over Internet Protocol

Department of Electronics & Communication

Dept of ECE GEC,TRICHUR

First,and foremost I thank God Almighty for making this

I extend my sincere gratitude to Prof. Indiradevi, Head of

Electronics and Communication Department, Govt

Engineering College, Thrissur for providing me with

necessary infrastructure. I would like to convey a deep

sense of gratitude to the seminar co-ordinator Mrs. C R

Muneera for the timely advices.

I also extend my sincere thanks to my friends for their

Dept of ECE GEC,TRICHUR

Dept of ECE GEC,TRICHUR

Dept of ECE GEC,TRICHUR

BASIC FLOW OF VOIP NETWORK

The VoIP networks replace the traditional public-switched

Signaling. Signaling in a VoIP network is accomplished by the

Database services. Database services are a way to locate an

Call connect and disconnect (bearer control). The connection of a

CODEC operations. Voice communication is analogue, while data

Dept of ECE GEC,TRICHUR

FIG 1 ARCHITECTURE OF A TYPICAL GATEWAY

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A TYPICAL VOIP NETWORK

FIG 2 A TYPICAL FULL SERVICE VOIP NETWORK

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Unlike the PCM data streams in circuitswitched telephony,

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Voice over packet provides the interworking function (IWF),

FIGURE 1. BRANCH OFFICE APPLICATION

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A second VoIP application, shown in Figure 2, is a trunking

FIGURE 2. INTEROFFICE TRUNKING APPLICATION

A third application of VoIP software is interworking with

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transcoding function required to convert the cellular voice data to

FIGURE 3. INTEROFFICE TRUNKING APPLICATION

Mainly the different types of communications that exist in a

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 Need a PC with sound card

 Need a gateway that connects IP network to phone network

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``PC TO PHONE CONNECTION

Phone to Phone communication:

 Need more gateways that connect IP network to phone

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Phone to Phone communication

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IDENTIFICATION OF MAJOR SYSTEM

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• Decompression and Remodulation functions

Terminals are the L AN client endpoints that provide real time

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FIG 1 ZONE MANAGEMENT IN A GATEKEEPER

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FIG 2 INTERZONE GATEWAY GATEKEEPER COMMUNICATION

FIG 3 INTRAZONE GATEWAY GATEKEEPER COMMUNICATION

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• Broadband Hard Phones

• Dialup Hard Phones

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• WLAN or WiFi Phones

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Need a PC with sound card

Need a gateway that connects IP network to phone network

Need more gateways that connect IP network to phone