This Presentation report on VoIP has been possible
only because of kind cooperation’s lent by many
I would like to express my sincere gratitude to my
parents, my elder sister, my teachers and my dear
I would like to give special thanks to Mr.Abhijit
Chatterjee and all the members of technical team of
TVT World Pvt Ltd., without whose guidance it
would not have been possible to bring me to this
At last I would like to thank all my batch mates for
their constant support and guidance.
b. Modes of operation…………………...05
c. Features of VoIP……………………...07
d. Standard bodies……………………….08
2. Traditional phone Vs VoIP………………..10
3. Application and Adoption…………………15
4. Technical details.
5. VoIP in India………………………………41
6. Legal issues……………………………….43
Q.What is Protocols?
A. In information technology, a protocol (from the
Greek protocollon, which was a leaf of paper glued
to a manuscript volume, describing its contents) is the
special set of rules that end points in a
telecommunication connection use when they
VoIP, an acronym for Voice over IP, is a technology
that allows one to make telephone calls using a
Broadband Internet connection instead of a regular
telephone line, thereby having phone service over the
Internet delivered through your Internet connection,
instead of from your local phone company.
Voice over Internet Protocol, is the routing of voice
conversations over the Internet (or through any other
IP-based network). Voice over IP traffic can be
utilized on any IP network, including those without a
connection to the rest of the Internet, such as LAN,
Traditionally, a phone conversation is converted into
electronic signals that travel along a network of
switches, in a dedicated circuit that lasts the length of
a call, as is with cellular providers, and long distance
The conversation is converted to packets of data that
travel over the Internet or private networks, just like
e-mails or Web pages. The packets get reassembled
and converted to sound on the other end of the call.
Modes of Operation:
VoIP can generally have four modes of operations,
which are as follows:
PC to PC
- PC to Telephone
- Telephone to PC
- Telephone to Telephone
VoIP can facilitate tasks and provide services that
may be more difficult to implement or expensive
using the more traditional PSTN. Examples include:
The ability to transmit more than one telephone
call down the same broadband-connected
telephone line. This can make VoIP a simple
way to add an extra telephone line to a home or
3-way calling call forwarding, automatic redial,
and caller ID; features that traditional
telecommunication companies (telcos) normally
charge extra for.
Secure calls using standardized protocols (such
as Secure Real-time Transport Protocol.) Most of
the difficulties of creating a secure phone over
traditional phone lines, like digitizing and digital
transmission are already in place with VoIP. It is
only necessary to encrypt and authenticate the
existing data stream.
Location independence. Only an internet
connection is needed to get a connection to a
VoIP provider. For instance, call center agents
using VoIP phones can work from anywhere
with a sufficiently fast and stable Internet
Integration with other services available over the
Internet, including video conversation, message
or data file exchange in parallel with the
conversation, audio conferencing, managing
address books, and passing information about
whether others (e.g. friends or colleagues) are
available online to interested parties.
Features of VoIP:
Caller ID – Any call will be linked with certain
Identity number which will be displayed when
call from that number is made.
Call waiting – This facility is active when there
already a call session going on.
Call transfer – This facility provides user an edge
to transfer his/her call to other ID. This is most
useful in call centers and corporate sectors.
Repeat dial – Repeat dial or redial option
provides the facility to automatically dial if a call
session is not connected.
Return call – This facility generally not seen in
common PSTN lines give the callers to set the
schedule to make a call.
Three-way calling – This is linked with
teleconferencing, where more than 2 callers can
join conversation with each others.
IETF (Internet Engineering Task Force)
The community of engineers that standardizes the
protocols that define how the Internet and Internet
Protocols work. http://www.ietf.org/.
ITU (International Telecommunications Union)
An international organization within the United
Nations System where governments and the private
sector coordinate global telecom networks and
Most common protocols:
An ITU Recommendation that defines ―Packet-based
multimedia communications systems‖. H.323 defines
a distributed architecture for creating multimedia
applications, including VoIP
Defined as IETF RFC 2543. SIP defines a
distributed architecture for creating multimedia
applications, including VoIP.
Traditional Phone Vs VoIP:
existing phone systems are driven by a very reliable
but somewhat inefficient method for connecting calls
called circuit switching. Circuit switching is a very
basic concept that has been used by telephone
networks for more than 100 years. When a call is
made between two parties, the connection is
maintained for the duration of the call. Because you
are connecting two points in both directions, the
connection is called a circuit. This is the foundation
of the Public Switched Telephone Network
Here's how a typical telephone call works:
1. You pick up the receiver and listen for a dial
tone. This lets you know that you have a
connection to the local office of your telephone
2. You dial the number of the party you wish to talk
3. The call is routed through the switch at your
local carrier to the party you are calling.
4. A connection is made between your telephone
and the other party's line using several
interconnected switches along the way.
5. The phone at the other end rings, and someone
answers the call.
6. The connection opens the circuit.
7. You talk for a period of time and then hang up
8. When you hang up, the circuit is closed, freeing
your line and all the lines in between.
Let's say that you talk for 10 minutes. During this
time, the circuit is continuously open between the
two phones. In the early phone system, up until 1960
or so, every call had to have a dedicated wire
stretching from one end of the call to the other for the
duration of the call. So if you were in New York and
you wanted to call Los Angeles, the switches
between New York and Los Angeles would connect
pieces of copper wire all the way across the United
States. You would use all those pieces of wire just for
your call for the full 10 minutes. You paid a lot for
the call, because you actually owned a 3,000-mile-
long copper wire for 10 minutes.
Telephone conversations over today's traditional
phone network are somewhat more efficient and they
cost a lot less. Your voice is digitized, and your
voice along with thousands of others can be
combined onto a single fiber optic cable for much of
the journey (there's still a dedicated piece of copper
wire going into your house, though). These calls are
transmitted at a fixed rate of 64 kilobits per second
(Kbps) in each direction, for a total transmission rate
of 128 Kbps. Since there are 8 kilobits (Kb) in a
kilobyte (KB), this translates to a transmission of 16
KB each second the circuit is open, and 960 KB
every minute it's open. So in a 10-minute
conversation, the total transmission is 9,600 KB,
which is roughly equal to 10 megabytes (check out
How Bits and Bytes Work to learn about these
conversions). If you look at a typical phone
conversation, much of this transmitted data is wasted.
While you are talking, the other party is listening,
which means that only half of the connection is in use
at any given time. Based on that, we can surmise that
we could cut the file in half, down to about 4.7 MB,
for efficiency. Plus, a significant amount of the time
in most conversations is dead air -- for seconds at a
time, neither party is talking. If we could remove
these silent intervals, the file would be even smaller.
Then, instead of sending a continuous stream of bytes
(both silent and noisy), what if we sent just the
packets of noisy bytes when you created them? That
is the basis of a packet-switched phone network, the
alternative to circuit switching.
Data networks do not use circuit switching. Your
Internet connection would be a lot slower if it
maintained a constant connection to the Web page
you were viewing at any given time. Instead, data
networks simply send and retrieve data as you need
it. And, instead of routing the data over a dedicated
line, the data packets flow through a chaotic network
along thousands of possible paths. This is called
While circuit switching keeps the connection open
and constant, packet switching opens a brief
connection -- just long enough to send a small chunk
of data, called a packet, from one system to another.
It works like this:
The sending computer chops data into small
packets, with an address on each one telling the
network devices where to send them.
Inside of each packet is a payload. The payload
is a piece of the e-mail, a music file or whatever
type of file is being transmitted inside the packet.
The sending computer sends the packet to a
nearby router and forgets about it. The nearby
router send the packet to another router that is
closer to the recipient computer. That router
sends the packet along to another, even closer
router, and so on.
When the receiving computer finally gets the
packets (which may have all taken completely
different paths to get there), it uses instructions
contained within the packets to reassemble the
data into its original state.
Packet switching is very efficient. It lets the network
route the packets along the least congested and
cheapest lines. It also frees up the two computers
communicating with each other so that they can
accept information from other computers, as well.
Application And Adoption:
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
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 "Softphone"
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
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-
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 the U.S. emergency services number 9-1-1
may not automatically be routed to the nearest local
emergency dispatch center, and would be of no use
for subscribers outside the U.S. This is potentially
true for users who select a number with an area code
outside their area. Some VoIP providers offer users
the ability to register their address so that 9-1-1
services work as expected.
Another challenge for these services is the proper
handling of outgoing calls from fax machines,
TiVo/ReplayTV 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 website 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 network.
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
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) allow 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 Softphone on their PC.
Internal VoIP phone networks allow outbound and
inbound calling on standard PSTN lines through the
use of FXO adapters.
Use 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 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
It is a cornerstone technology for transmission of
real-time audio, video, data communication over
packet based network. It is part of ITU-T family
recommendations called H.32x that provides
multimedia communication services over variety of
It turns out that everything you do on the Internet
involves packets. For example, every Web page that
you receive comes as a series of packets, and every e-
mail you send leaves as a series of packets. Networks
that ship data around in small packets are called
packet switched networks.
On the Internet, the network breaks an e-mail
message into parts of a certain size in bytes. These
are the packets. Each packet carries the information
that will help it get to its destination -- the sender's IP
address, the intended receiver's IP address, something
that tells the network how many packets this e-mail
message has been broken into and the number of this
particular packet. The packets carry the data in the
protocols that the Internet uses: Transmission Control
Protocol/Internet Protocol (TCP/IP). Each packet
contains part of the body of your message. A typical
packet contains perhaps 1,000 or 1,500 bytes.
Each packet is then sent off to its destination by the
best available route -- a route that might be taken by
all the other packets in the message or by none of the
other packets in the message. This makes the network
more efficient. First, the network can balance the
load across various pieces of equipment on a
millisecond-by-millisecond basis. Second, if there is
a problem with one piece of equipment in the
network while a message is being transferred, packets
can be routed around the problem, ensuring the
delivery of the entire message.
Depending on the type of network, packets may be
referred to by another name:
Most packets are split into three parts:
header - The header contains instructions
about the data carried by the packet. These
instructions may include:
Length of packet (some networks
have fixed-length packets, while
others rely on the header to contain
Synchronization (a few bits that
help the packet match up to the
Packet number (which packet this
is in a sequence of packets)
Protocol (on networks that carry
multiple types of information, the
protocol defines what type of
packet is being transmitted: e-
mail, Web page, streaming video)
Destination address (where the
packet is going)
Originating address (where the
packet came from)
payload - Also called the body or data of a
packet. This is the actual data that the
packet is delivering to the destination. If a
packet is fixed-length, then the payload
may be padded with blank information to
make it the right size.
trailer - The trailer, sometimes called the
footer, typically contains a couple of bits
that tell the receiving device that it has
reached the end of the packet. It may also
have some type of error checking. The
most common error checking used in
packets is Cyclic Redundancy Check
(CRC). CRC is pretty neat. Here is how it
works in certain computer networks: It
takes the sum of all the 1s in the payload
and adds them together. The result is stored
as a hexadecimal value in the trailer. The
receiving device adds up the 1s in the
payload and compares the result to the
value stored in the trailer. If the values
match, the packet is good. But if the values
do not match, the receiving device sends a
request to the originating device to resend
As an example, let's look at how an e-mail message
might get broken into packets. Let's say that you send
an e-mail to a friend. The e-mail is about 3,500 bits
(3.5 kilobits) in size. The network you send it over
uses fixed-length packets of 1,024 bits (1 kilobit).
The header of each packet is 96 bits long and the
trailer is 32 bits long, leaving 896 bits for the
payload. To break the 3,500 bits of message into
packets, you will need four packets (divide 3,500 by
896). Three packets will contain 896 bits of payload
and the fourth will have 812 bits. Here is what one of
the four packets would contain:
Each packet's header will contain the proper
protocols, the originating address (the IP address of
your computer), the destination address (the IP
address of the computer where you are sending the e-
mail) and the packet number (1, 2, 3 or 4 since there
are 4 packets). Routers in the network will look at the
destination address in the header and compare it to
their lookup table to find out where to send the
packet. Once the packet arrives at its destination,
your friend's computer will strip the header and
trailer off each packet and reassemble the e-mail
based on the numbered sequence of the packets.
Versions of H.323
It was accepted on October 1996, with heavy
stress on multimedia communication over packet
Accepted in January 1998 and included following
The H.235 standard addresses four general issues
when dealing with security, Authentication, Integrity,
Privacy, and non-Repudiation. Authentication is a
mechanism to make sure that the endpoints
participating in the conference are really who they
say they are. Integrity provides a means to validate
that the data within a packet is indeed an unchanged
representation of the data. Privacy/Confidentiality is
provided by encryption and decryption mechanisms
that hide the data from eavesdroppers so that if it is
intercepted, it cannot be viewed. Non-Repudiation is
a means of protection against someone denying that
they participated in a conference when you know
they were there. Hooks for each of these security
features are specified in H.323 Version 2. The proper
usage of these hooks is specified in H.235.
Fast Connect (a.k.a. Fast Start)
Fast Connect is a new method of call setup that
bypasses some usual steps in order to make it faster.
In addition to the speed improvement, Fast Connect
allows the media channels to be operational before
the CONNECT message is sent, which is a
requirement for certain billing procedures.
Supplementary Services for H.323, namely Call
Transfer and Call Diversion, have been defined by
the H.450 series. H.450.1 defines the signaling
protocol between H.323 endpoints for the control of
supplementary services. H.450.2 defines Call
Transfer and H.450.3 Call Diversion. Call Transfer
allows a call established between endpoint A and
endpoint B to be transformed into a new call between
endpoint B and a third endpoint, endpoint C. Call
Diversion provides the supplementary services Call
Forwarding Unconditional, Call Forwarding Busy,
Call Forwarding No Reply and Call Deflection. The
hooks for Supplementary Services are specified in
H.323 Version 2. The proper usage of these hooks is
specified in H.450.x.
The Call Reference Value (CRV) method of
identifying a call in Version 1 was not sufficiently
unique when going through a gatekeeper. The
Version 2 Call Identifier is a globally unique ID, so
you can always correctly identify which call a packet
is referencing, even when going through a
Arbitration of V.Chat
V.Chat is a chat protocol. H.245 now allows for the
capability exchange of V.Chat.
Accepted on 30 September 1999 it carried out
Maintaining and Reusing Connections
In order to provide better performance and preserve
system resources, version 3 introduces the ability for
an endpoint to specify whether it has the ability to
"reuse" a call signaling connection and whether it can
support using the same call signaling channel for
multiple calls. This is particularly important for
gateways that may have thousands of calls running
simultaneously. By utilizing these two features, a
gateway may maintain a single TCP connection
between itself and the gatekeeper in order to perform
all call signaling.
Conference out of Consultation
Suppose you place a call to someone and a
receptionist answers the phone. Typically, he will put
your call on hold while he calls the person you are
trying to reach. The receptionist may then connect
you to the other party, leaving only you and the
person you were calling in a call together. This
feature, called "conference out of consultation", is
introduced in version 3.
H.323 now supports the feature of "Caller ID" that
one finds in the traditional telephone network,
including the ability of the caller to request that name
and address information be withheld from the callee
and the ability of the network equipment (e.g., the
gatekeeper) to screen caller information.
With version 3, a caller has the ability to specify a
language preference. This information may be
utilized by call centers to help route calls to operator
who can speak the caller's language. It may also be
utilized by interactive voice response (IVR) systems
and announcement servers so that they can provide
audio streams to the caller in the caller's preferred
Remote Device Control
H.323 now has this ability, through the use of H.282,
to perform remote device control. This feature will
allow a user to control such devices as cameras from
Several new supplementary service documents have
been added to the H.323 series, including call hold,
call park and pickup, message waiting indication, and
Accepted on 17 November 2000 its feature was:
One of the most important aspects of any telephony
system is "uptime". Customers do not want to be
without phone service and service providers do not
want a loss in revenue. Gatekeeper failure often
results in missed calls, lost revenue, or both. Fields
were introduced into H.323v2 to provide for
Gatekeeper redundancy, but the usage of those fields
was never fully explained. Version 4 introduces a
new section that details the procedure that endpoints
may follow in order to provide some robustness to
In addition to procedural text, a new field was added
to allow an endpoint to indicate whether it supports
the Alternate Gatekeeper procedures. This allows the
Gatekeeper to make intelligent decisions about
redirecting an endpoint to provide for some level of
load balancing across Gatekeepers.
Prior to H.323 Version 4, and endpoint could request
much more bandwidth than it actually needed and,
thus, cause network resources to go unutilized. With
Version 4, it is now mandatory that an endpoint make
bandwidth requests with a lower value if, indeed, the
endpoint is using less bandwidth than it had initially
indicated in the ARQ.
In addition, managing bandwidth for multicast
sessions has been nearly impossible since, unless the
Gatekeeper routed the H.245 signalling and carefully
monitored the media channels that were opened, it
could not determine whether two endpoints that
request bandwidth are actually requesting bandwidth
for a multicast session or unicast session. This
becomes a much bigger issue when many people are
participating in a multipoint multicast conference.
With Version 4, specific details about the media
channels are conveyed to the Gatekeeper in IRR
messages (if the Gatekeeper requests them), so that
the Gatekeeper can better control bandwidth
Quality of Service is very important in any VoIP
network. As a first step in improving QoS in H.323
systems, new procedures are defined in H.323 to
allow for RSVP when not using Fast Connect.
Obviously, work is continuing in this area in both the
ITU and the IETF.
Call Credit-Related Capabilities
An extremely popular service which utilizes IP
telephony today is to allow users to dial a Gateway to
place a call (with the anticipation that the call will be
much lower than a traditional PSTN call), which is
then charged against a pre-paid calling card or to a
user's account. Until now, there has been no standard
means of communicating available funds or for the
Gateway to control early call termination based on
available funds. H.323v4 adds these features to the
Accepted on July 2003 its main feature was giving
more edge to already established standards instead
of introducing new functions.
Accepted on June 2006 , it is the latest version of
all and has all the features of its above versions.
Elements of H.323 System
An H.323 terminal is an endpoint in the LAN that
participates in real-time, two-way communications
with another H.323 terminal, gateway, or multipoint
control unit (MCU). A terminal must support audio
communication and can also support audio with
video, audio with data, or a combination of all three.
H.323 terminals must support the following standards
• H.245—An ITU standard used by the terminal to
negotiate its use usage of the channel. The H.245
control channel provides in-band reliable transport
for capabilities exchange, mode preference from
the receiving end, logical channel signaling, and
control and indication. Part of the capabilities
exchange includes specifying which coder-decoders
(CODECs) are available. Recommended audio
CODECs include G.711, G.722, G.723, G.723.1,
G.728, and G.729. Recommended video CODECs
include H.261 and H.263.
• H.225.0—An ITU standard that uses a variant of
Q.931 to set up the connection between two H.323
• RAS—(Registration Admission Status) A protocol
used to communicate with the H.323 gatekeeper.
• RTP and RTCP—(Real-Time Transport Protocol
and Real-Time Control Protocol) Protocols used to
sequence the audio and video packets. The RTP
header contains a time stamp and sequence number,
allowing the receiving device to buffer as much as
necessary to remove jitter and latency by
synchronizing the packets to play back a continuous
stream of sound. RTCP controls RTP and gathers
reliability information and periodically passes this
information onto session participants.
Gatekeepers are optional nodes that manage other
nodes in an H.323 network. Other nodes
with the gatekeeper using the RAS protocol.
A gatekeeper is not required in an H.323 network, but
itmust be used if one is present. The H.323 nodes
attempt to register with a gatekeeper on startup.
When an H.323 node wants to
communicate with another endpoint, it requests
admission to the call, using a symbolic alias for the
endpoint name such as an E.164 (ITU-T
recommendation for international telecommunication
numbering) address or an e-mail ID. If the gatekeeper
decides the call can proceed, it returns a
destination IP address to the originating H.323 node.
This IP address can be the actual address of the
target endpoint or it can be an intermediate address.
Finally, a gatekeeper and its registered endpoint
exchange status information.
An H.323 gateway can provide an interface between
H.323 and the Public Switched Telephone Network
(PSTN), H.320 terminals, V.70 terminals, H.324
terminals, and other speech terminals. It provides
standard interfaces to the PSTN, processes the voice
and fax signals using CODECs to convert between
circuit-switched and packet formats, and works with
the gatekeeper through the RAS protocol to route
calls through the network. Gateways provide
translation between transmission formats, such as
H.245 and H.242
4. Multi point control unit(MCU)
An MCU is an endpoint on the LAN that provides the
capability for three or more terminals and
gateways to participate in a multipoint conference. It
controls and mixes video, audio, and data from
terminals to create a robust video conference. An
MCU can also connect two terminals in a point-to-
point conference that can later develop into a
H.323 Interworking with SCN
The Session Initiation Protocol (SIP) is an Internet
Engineering Task Force (IETF) standard protocol for
establishing, manipulating, and tearing down an
interactive user session that involves multimedia
elements such as audio, video, instant messaging, or
other real-time data communications. Even though
H.323 was the first protocol to introduce VoIP,
analysts estimate that SIP will play a major role in
the coming years and will replace H.323 in VoIP
applications. SIP is a request-response protocol that
works in the Application layer of the Open Systems
Interconnection (OSI) communications model, and
provides the capability to:
• Determine the location of the target end point —
SIP supports address resolution, name mapping, and
• Determine the media capabilities of the target end
point — Via Session Description Protocol (SDP); SIP
determines the ―lowest level‖ of common services
between the end points. Conferences are established
using only the media capabilities that can be
supported by all end points.
• Determine the availability of the target end point —
If a call cannot be completed because the target end
point is unavailable, SIP determines whether the
called party is already on the phone or did not answer
in the allotted number of rings. It then returns a
message indicating why the target end point was
In SIP, a User Agent (UA) is the endpoint entity.
User Agents initiate and terminate sessions by
exchanging requests and responses. RFC 2543
defines the User Agent as an application, which
contains both a User Agent client and User
Agent server, as follows:
User Agent Client (UAC)—a client application that
initiates SIP requests.
User Agent Server (UAS)—a server application that
contacts the user when a SIP request is received and
that returns a response on behalf of the user.
A Proxy Server is an intermediary entity that acts as
both a server and a client for the purpose of making
requests on behalf of other clients. Requests are
serviced either internally or by passing them on,
possibly after translation, to other servers. A Proxy
interprets, and, if necessary, rewrites a request
messagebefore forwarding it.
A Redirect Server is a server that accepts a SIP
request, maps the SIP address of the called party into
zero (if there is no known address) or more new
addresses and returns them to the client. Unlike
Proxy servers, Redirect Servers do not pass the
request on to other servers.
A Registrar is a server that accepts REGISTER
requests for the purpose of updating a location
database with the contact information of the user
specified inthe request.
Call Establishment/Termination Model
CALL FLOW(SESSION ESTABLISHMENT)
1. The calling User Agent Client sends an INVITE
message to Bob’s SIP address: sip:email@example.com.
This message also contains an SDP packet describing
the media capabilities of the calling terminal.
2. The UAS receives the request and immediately
responds with a 100-Trying response message.
3. The UAS starts ―ringing‖ to inform Bob of the new
call. Simultaneously a 180 (Ringing) message is sent
to the UAC.
4. The UAS sends a 182 (Queued) call status
message to report that the call is behind two other
calls in the queue.
5. The UAS sends a 182 (Queued) call status
message to report that the call is behind one other call
in the queue.
6. Bob picks up the call and the UAS sends a 200
(OK) message to the calling UA. This message also
contains an SDP packet describing the media
capabilities of Bob’s terminal.
7. The calling UAC sends an ACK request to confirm
the 200 (OK) response was received.
CALL FLOW(SESSION TERMINATION)
1. The caller decides to end the call and ―hangs-up‖.
This results in a BYE request being sent to Bob’s
UAS at SIP address sip:firstname.lastname@example.org
2. Bob’s UAS responds with 200 (OK) message and
notifies Bob that the conversation has ended
VoIP in INDIA
World Phone Internet Services, India's leading
Category "A" ISP, has announced its plans to
provide post-paid VoIP for the first time in the
country. The monthly plans are targeted towards
Indian companies, and will enable them to make
calls to the USA, UK, Australia and Canada at
Rs 995 onwards, effectively bringing the price
down to less than 50 paisa per minute. This would
be the cheapest international call rate in India on
any legal Internet telephony network. World Phone
has many leading Indian companies among its
clients, including HCL Technologies, NIIT
Technologies, Perot Systems, NDTV, Fidelity
Information Systems and TVS InfoTech. Currently,
World Phone negotiates over seven million
international call minutes through its Internet
telephony network every month, placing it amongst
the frontrunners in the Internet telephony industry
Mahanagar Telephone Nigam Ltd (MTNL),
India’s state-controlled telco, has entered the
VoIP market, becoming the first landline carrier
in India to provide the service. The company is
offering competitive international long distance
rates, some of which are cheaper than those
charged by Skype.The new service, which
MTNL provides in partnership with Verso
Technologies, Inc and Aksh Optifibre Ltd, is
available to 30 million citizens in New Delhi and
Mumbai. Subscribers to MTNL’s broadband
connection can now convert their fixed line
phone to a VoIP phone using an Analog Telephone
Adaptor which is available to rent from MTNL.A
fixed-line phone call from India to the US costs
around 14.5 cents per minute, but with the new VoIP
service, the cost falls by over 80% to 2.5 cents per
Voice over Internet Protocol, the technology which
allows a call from a PC to a landline at a cheaper rate
than landline to landline, is still considered illegal in
the country of India.
The UAE Telecommunications Regulatory Authority
(TRA) denied reports that VOIP will be liberalized.
There had been reports in the Press suggesting other
The TRA’s official position is that VOIP is illegal
until it is legalized. The announcement came from the
TRA’s manager, administration and public relations,
Adnan Al Bahar. The TRA further pointed out that
until there is a firm regulation system in place,
legalization will no occur.
Putting even more rain on some techies’ parade, The
Director-General of the TRA, Mohamed Al Ghanim
also announced that when VOIP does become
available, it will only be offered by the countries two
licensed operators, etisalat and du. It had been hoped
that the market would be opened to all companies
wanting to offer the service.
The TRA was stressing the importance of regulations
and careful review after noting problems that other
countries are having. They pointed out that the US,
leaders in VOIP technology, has already run into
situations with providers going out of business. The
care taken now before entering the VOIP market
place will help limit those kinds of problems in India
in the future.
The TRA’s officials did make a distinction between
the new technology of VOIP and the current
communication between PC’s. This kind of
communication is legal.
Frequently Asked Questions
What Kind of Equipment Do I Need?
A broadband (high speed Internet) connection is
required. This can be through a cable modem, or
high speed services such as DSL or a local area
network. A computer, adaptor, or specialized
phone is required. Some VoIP services only work
over your computer or a special VoIP phone, while
other services allow you to use a traditional phone
connected to a VoIP adapter. If you use your
computer, you will need some software and an
inexpensive microphone. Special VoIP phones plug
directly into your broadband connection and operate
largely like a traditional telephone. If you use a
telephone with a VoIP adapter, you'll be able to dial
just as you always have, and the service provider may
also provide a dial tone.
Is there a difference between making a Local Call
and a Long Distance Call?
Some VoIP providers offer their services for free,
normally only for calls to other subscribers to the
service. Your VoIP provider may permit you to select
an area code different from the area in which you
live. It also means that people who call you may
incur long distance charges depending on their area
code and service.
Some VoIP providers charge for a long distance call
to a number outside your calling area, similar to
existing, traditional wireline telephone service. Other
VoIP providers permit you to call anywhere at a flat
rate for a fixed number of minutes.
If I have VoIP service, who can I call?
Depending upon your service, you might be limited
only to other subscribers to the service, or you may
be able to call anyone who has a telephone number -
including local, long distance, mobile, and
international numbers. If you are calling someone
who has a regular analog phone, that person does not
need any special equipment to talk to you. Some
VoIP services may allow you to speak with more
than one person at a time.
What Are Some Advantages of VoIP?
Some VoIP services offer features and services that
are not available with a traditional phone, or are
available but only for an additional fee. You may also
be able to avoid paying for both a broadband
connection and a traditional telephone line.
What Are Some disadvantages of VoIP?
If you're considering replacing your traditional
telephone service with VoIP, there are some possible
Some VoIP services don't work during power
outages and the service provider may not offer
Not all VoIP services connect directly to
emergency services through 9-1-1. For additional
information, see www.voip911.gov.
VoIP providers may or may not offer directory
assistance/white page listings.
Can I use my Computer While I talk on the Phone?
In most cases, yes.
Can I Take My Phone Adapter with me When I
Some VoIP service providers offer services that can
be used wherever a high speed Internet connection
available. Using a VoIP service from a new location
may impact your ability to connect directly to
emergency services through 9-1-1. For additional
information, see www.voip911.gov.
Does my Computer Have to be Turned on?
Only if your service requires you to make calls using
your computer. All VoIP services require your
broadband Internet connection to be active.
How Do I Know If I have a VoIP phone Call?
If you have a special VoIP phone or a regular
telephone connected to a VoIP adapter, the phone
will ring like a traditional telephone. If your VoIP
service requires you to make calls using your
computer, the software supplied by your service
provider will alert you when you have an incoming
Does the FCC Regulate VoIP?
In June 2005 the FCC imposed 911 obligations on
providers of ―interconnected‖ VoIP services – VoIP
services that allow users generally to make calls to
and receive calls from the regular telephone network.
You should know, however, that 911 calls using
VoIP are handled differently than 911 calls using
your regular telephone service. Please see our
consumer fact sheet on VoIP and 911 services at
www.voip911.gov for complete information on these
In addition, the FCC requires interconnected VoIP
providers to comply with the Communications
Assistance for Law Enforcement Act of 1994
(CALEA) and to contribute to the Universal Service
Fund, which supports communications services in
high-cost areas and for income-eligible telephone
2. Understanding Voice over IP Protocols by Cisco
Systems—Service Provider Solutions Engineering
3. IETF (Internet Engineering Task Force)
4. ITU (International Telecommunications Union)
5. SIP guide – XIXIA
6. QoS for VoIP- Dr.Peter J Walcher.
7. H.323 Protocol Overview- Paul E Jones.
8. VoIP and Amateur Radio- Steve Ford.
9. VoIP News – www.voip-news.com
10. Voice over IP- David Feiner.
11. VoIP-Internet Telephony- Rainer Oechsle.
12.VoIP for next generation- B.V.Harsha and
DR.B.R AMBEDKAR NATIONAL INSTITUTE OF
TECHNOLOGY JALANDHAR, PUNJAB
Department of Electronics and Communication
Voice over Internet Protocol (VoIP)
03104003, 6th Semester