1. IP Telephony & VoIP
Introduction to IP telephony & Voice-over-
Prepared by :
Md. Minhajul Hoque
ID # 071-849556
Ete ~ 605
Section ~ 02
Prepared for :
Dr. mashiur Rahman
Assistant Professor, Department of Engineering
North South University
Department of Engineering
North South University
2. IP Telephony & VoIP
What is Internet Telephony?
Technologies that use the Internet and Internet protocol (“IP”) networks to deliver voice
communications have the potential to reduce costs, support innovation, and improve access to
communications services within developing countries and around the world.
Internet telephony (IPT) is transport of telephone calls over the Internet, no matter whether
traditional telephony devices, multimedia PCs or dedicated terminals take part in the calls and no
matter whether the calls are entirely or only partially transmitted over the Internet.
IP telephony (Internet Protocol telephony) is a general term for the technologies that use the
Internet Protocol's packet-switched connections to exchange voice, fax, and other forms of
information that have traditionally been carried over the dedicated circuit-switched connections of
the public switched telephone network (PSTN). Using the Internet, calls travel as packets of data on
shared lines, avoiding the tolls of the PSTN. The challenge in IP telephony is to deliver the voice,
fax, or video packets in a dependable flow to the user. Much of IP telephony focuses on that
IP telephony service providers include or soon will include local telephone companies, long
distance providers such as AT&T, cable TV companies, Internet service providers (ISPs), and fixed
service wireless operators. IP telephony services also affect vendors of traditional handheld
devices. Currently, unlike traditional phone service, IP telephony service is relatively unregulated
by government. In the United States, the Federal Communications Commission (FCC) regulates
phone-to-phone connections, but says they do not plan to regulate connections between a phone
user and an IP telephony service provider.
VoIP is an organized effort to standardize IP telephony. IP telephony is an important part of the
convergence of computers, telephones, and television into a single integrated information
environment. Also see another general term, computer-telephony integration (CTI), which
describes technologies for using computers to manage telephone calls.
VoIP, Internet Telephony, Voice-over-the-Internet: What are they?
The terms Voice-over-Internet Protocol (“VoIP”), IP telephony, Internet telephony, and
Voice-over-the-Internet (“VoN”) are given different meanings by different commentators and in
fact have no universally agreed-upon meaning. There are, however, distinctions to be kept in mind,
for IP can be used in various ways for the transmission of voice. As used in this memo, –
VoIP is a generic term that refers to all types of voice communication using Internet protocol
(IP) technology instead of traditional circuit switched technology. This includes use of packet
technologies by telecommunications companies to carry voice at the core of their networks in ways
that are not controlled by and not apparent to end users.
Transmission Of Voice Using IP Networks: How Does It Work?
Here is how a VoIP transmission is completed:
Step 1: Because all transmissions must be digital, the caller’s voice is digitized. This can be done
by the telephone company (which is how carriers use IP in their networks), by an Internet service
provider (ISP), or by a PC on your desk.
Step 2: Next using complex algorithms the digital voice is compressed and then separated into
packets; and using the Internet protocol, the packets are addressed and sent across the network to be
3. IP Telephony & VoIP
reassembled in the proper order at the destination. Again, this reassembly can be done by a carrier,
and ISP, or by one’s PC.
Step 3: During transmission on the Internet, packets may be lost or delayed, or errors may damage
the packets. Conventional error correction techniques would request retransmission of unusable or
lost packets, but if the transmission is a real-time voice communication that technique obviously
would not work, so sophisticated error detection and correction systems are used to create sound to
fill in the gaps. (This process stores a portion of the incoming speaker’s voice, and uses a complex
algorithm to “guess” the contents of the missing packets and create new sound information to
enhance the communication.)
Step 4: After the packets are transmitted and arrive at the destination, the transmission is assembled
and decompressed to restore the data to an approximation of the original form. As this explanation
suggests, technology that works fine for sending data may be less than perfect for voice
transmissions. The technology is improving, but still the quality of a voice transmission using
packet technology is inferior to a circuit-switched connection, and that difference in quality would
normally be obvious to any listener. As IP technology improves, the quality advantage for voice
communication enjoyed by the circuit-switched will decrease, but most experts see parity in quality
as still a distant prospect.
What is Internet Telephony Good For?
The most significant benefit of IPT and driver of its evolution is money-saving and easy
implementation of innovative services:
• In the future, Internet Telephony Service Providers (ITSP) may use a single infrastructure
for providing both, Internet access and Internet telephony. Only data-oriented switches
could be deployed for switching data as well as packetized voice. Multiplexing data and
voice could also result in better bandwidth utilization than in today's over-engineered voice-
or-nothing links. Not only the providers, but also their clients will profit of lower costs
• Now, customers may take advantage of flat Internet rating vs. hierarchical PSTN rating and
save money while letting their long-distance calls be routed over Internet. This is especially
true in Europe, where the prices of long-distance calls are still higher than in US. But:
according to some estimations, the prices of the traditional and the Internet telephony will
equalize together with the convergence of quality of services provided by them.
• The IPT users may also profit of its software-oriented nature: software solutions may be
easily extended and integrated with other services and applications, e.g. whiteboarding,
electronic calendar, or WWW. Deployment of new IP telephony services requires
significantly lower investment in terms of time and money than in the traditional PSTN
Internet Telephony Scenarios
The IPT usage scenarios are commonly classified by the type of devices terminating an Internet
call. Because there may be either a PSTN device or a data-oriented terminal on each side of a call,
there are 4 generic classes. Note, that although "PC" is a well established term, any device capable
of transmitting voice over data network may apply in this context. See for example the dedicated
Notes Costs Paid By Caller
4. IP Telephony & VoIP
This class is attractive especially
Costs of ownership and
for private users who already
maintenance of the hardware (PC
have an Internet access and an
with modem and sound or a
audio-capable PC. Necessary
dedicated device) and software
software is available for free .
PC PC (IPT software is often provided for
This pure-IP scenario is likely to
take advantage of integration with
Costs of Internet access (incl.
other Internet services, such as
the local call).
WWW, instant messaging, E-
Costs of ownership and
maintenance of the hardware (PC
This is an extension of the
with modem/dedicated device) and
previous class in that the PC-
software (IPT software is often
callers may reach also the PSTN
provided for free).
callees. A gateway converting the
Costs of Internet access (incl.
telephone Internet call into a PSTN call has
the local call).
PC (POTS/ISDN/ to be used and located as near to
Costs charged by the gateway
GSM...) the callee as possible to minimize
operator. (~ 5-12 cents per minute
the price for the gateway-to-callee
to the U.S. in August 98) The costs
connection. This scenario is
charged by the operator are
commercially provided by
determined mainly by the costs of
gateway operators like iConnect.
the call placed from the gateway to
This class is attractive for those
who want to save on long-
distance call and do not
Costs charged by both gateway
have/want to use a PC. For
operators.(~ 7-17 cents per minute
example, mobile phone users
to the U.S. in August 98) The costs
telephone telephone certainly prefer to carry only the
charged by the destination gateway
(POTS/ISDN/ (POTS/ISDN/ mobile phone without any
are determined mainly by the costs
GSM...) GSM...) additional boxes. The call has to
of the call placed from the gateway
pass two gateways: GSTN-to-
to the callee.
Internet and Internet-to-GSTN.
Local Call Costs
This solution has been comercialy
provided by gateway operators
like Access Power, DeltaThree.
This class is useful for those who
want to reach Internet users with
telephone Costs charged by a gateway
an ordinary telephone. Telenor
(POTS/ISDN/ PC operator.
provides this service
GSM...) Local Call Costs
commercially in Norway under
the name "Interfon".
Architecture: the Internet telephony systems are composed of these elements:
• end devices; these may be either traditional telephones (analog/GSM/ISDN/...), audio-
equipped personal computers, or single use appliances
5. IP Telephony & VoIP
• gateways; if a traditional telephone is used at either calling side the call (i.e. its transmission
format, signaling procedures, audio codecs) has to be translated to/from the format for
transport over Internet; this is the task of the gateways
• gatekeepers/proxies; the gatekeepers/proxies provide centralized call management
functions; they may provide call admission control, bandwidth management, address
translation, authentication, user location, etc.
• multipoint conference units; these manage multiparty conferences
The components may be implemented as hardware or software and may be integrated into single
They communicate with each other over signaling and voice-transporting protocols. To ensure
interoperability between products of different vendors, standardization bodies have elaborated
standards for both classes of protocols. See the section "Players and ..." for more details.
How VoIP works :
VoIP, or Voice over Internet Protocol, is a method for taking analog audio signals, like the kind
you hear when you talk on the phone, and turning them into digital data that can be transmitted over
How is this useful? VoIP can turn a standard Internet connection into a way to place free phone
calls. The practical upshot of this is that by using some of the free VoIP software that is available to
make Internet phone calls, you're bypassing the phone company (and its charges) entirely.
VoIP is a revolutionary technology that has the potential to completely rework the world's phone
systems. VoIP providers like Vonage have already been around for a while and are growing
steadily. Major carriers like AT&T are already setting up VoIP calling plans in several markets
around the United States, and the FCC is looking seriously at the potential ramifications of VoIP
The interesting thing about VoIP is that there is not just one way to place a call. There are three
different "flavors" of VoIP service in common use today:
2. IP PHONE
3. COMPUTER TO COMPUTER
Using VoIP :
Chances are good you're already making VoIP calls any time you place a long-distance call. Phone
companies use VoIP to streamline their networks. By routing thousands of phone calls through a
circuit switch and into an IP gateway, they can seriously reduce the bandwidth they're using for the
long haul. Once the call is received by a gateway on the other side of the call, it's decompressed,
reassembled and routed to a local circuit switch.
Although it will take some time, you can be sure that eventually all of the current circuit-switched
networks will be replaced with packet-switching technology (more on packet switching and
circuit switching later). IP telephony just makes sense, in terms of both economics and
infrastructure requirements. More and more businesses are installing VoIP systems, and the
technology will continue to grow in popularity as it makes its way into our homes. Perhaps the
biggest draws to VoIP for the home users that are making the switch are price and flexibility.
6. IP Telephony & VoIP
Most VoIP companies provide the features that normal phone companies charge extra for when
they are added to your service plan. VoIP includes:
• Caller ID
• Call waiting
• Call transfer
• Repeat dial
• Return call
• Three-way calling
There are also advanced call-filtering options available from some carriers. These features use
caller ID information to allow you make a choice about how calls from a particular number are
handled. You can:
• Forward the call to a particular number
• Send the call directly to voice mail
• Give the caller a busy signal
• Play a "not-in-service" message
• Send the caller to a funny rejection hotline
With many VoIP services, you can also check voice mail via the Web or attach messages to an e-
mail that is sent to your computer or handheld. Not all VoIP services offer all of the features above.
Prices and services vary, so if you're interested, it's best to do a little shopping.
Now that we've looked at VoIP in a general sense, let's look more closely at the components that
make the system work. To understand how VoIP really works and why it's an improvement over
the traditional phone system, it helps to first understand how a traditional phone system works.
VoIP: Circuit switching
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're connecting two points in both directions, the connection is called a
circuit. This is the foundation of the Public Switched Telephone Network (PSTN).
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 carrier.
2. You dial the number of the party you wish to talk to.
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 the receiver.
8. When you hang up, the circuit is closed, freeing your line and all the lines in
VoIP: Packet switching
A packet-switched phone network is the alternative to circuit switching. It works like this: While
you're 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
7. IP Telephony & VoIP
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.
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 packet switching.
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
• 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.
Advantages of Using VoIP
VoIP technology uses the Internet's packet-switching capabilities to provide phone service. VoIP
has several advantages over circuit switching. For example, packet switching allows several
telephone calls to occupy the amount of space occupied by only one in a circuit-switched network.
Using PSTN, that 10-minute phone call we talked about earlier consumed 10 full minutes of
transmission time at a cost of 128 Kbps. With VoIP, that same call may have occupied only 3.5
minutes of transmission time at a cost of 64 Kbps, leaving another 64 Kbps free for that 3.5
minutes, plus an additional 128 Kbps for the remaining 6.5 minutes. Based on this simple estimate,
another three or four calls could easily fit into the space used by a single call under the
conventional system. And this example doesn't even factor in the use of data compression, which
further reduces the size of each call.
Disadvantages of Using VoIP
The current Public Switched Telephone Network is a robust and fairly bulletproof system for
delivering phone calls. Phones just work, and we've all come to depend on that. On the other hand,
computers, e-mail and other related devices are still kind of flaky. Let's face it -- few people really
panic when their e-mail goes down for 30 minutes. It's expected from time to time. On the other
hand, a half hour of no dial tone can easily send people into a panic. So what the PSTN may lack in
efficiency it more than makes up for in reliability. But the network that makes up the Internet is far
8. IP Telephony & VoIP
more complex and therefore functions within a far greater margin of error. What this all adds up to
is one of the major flaws in VoIP: reliability.
A codec, which stands for coder-decoder, converts an audio signal into compressed digital form
for transmission and then back into an uncompressed audio signal for replay. It's the essence of
Codecs accomplish the conversion by sampling the audio signal several thousand times per second.
For instance, a G.711 codec samples the audio at 64,000 times a second. It converts each tiny
sample into digitized data and compresses it for transmission. When the 64,000 samples are
reassembled, the pieces of audio missing between each sample are so small that to the human ear, it
sounds like one continuous second of audio signal. There are different sampling rates in VoIP
depending on the codec being used:
• 64,000 times per second
• 32,000 times per second
• 8,000 times per second
A G.729A codec has a sampling rate of 8,000 times per second and is the most commonly used
codec in VoIP.
VoIP: Soft Switches and Protocols
The soft switch contains a database of users and phone numbers. If it doesn't have the information it
needs, it hands off the request downstream to other soft switches until it finds one that can answer
the request. Once it finds the user, it locates the current IP address of the device associated with that
user in a similar series of requests. It sends back all the relevant information to the softphone or IP
phone, allowing the exchange of data between the two endpoints.
Soft switches work in tandem with network devices to make VoIP possible. For all these devices to
work together, they must communicate in the same way. This communication is one of the most
important aspects that will have to be refined for VoIP to take off.
As we've seen, on each end of a VoIP call we can have any combination of an analog, soft or IP
phone as acting as a user interface, ATAs or client software working with a codec to handle the
digital-to-analog conversion, and soft switches mapping the calls. How do you get all of these
completely different pieces of hardware and software to communicate efficiently to pull all of this
off? The answer is protocols.
There are several protocols currently used for VoIP. These protocols define ways in which devices
like codecs connect to each other and to the network using VoIP. They also include specifications
for audio codecs. The most widely used protocol is H.323, a standard created by the International
Telecommunication Union (ITU). H.323 is a comprehensive and very complex protocol that was
originally designed for video conferencing. It provides specifications for real-time, interactive
videoconferencing, data sharing and audio applications such as VoIP. Actually a suite of protocols,
H.323 incorporates many individual protocols that have been developed for specific applications.
H.323 Protocol Suite
Video Audio Data Transport
H.261 G.711 T.122 H.225
H.263 G.722 T.124 H.235
9. IP Telephony & VoIP
G.723.1 T.125 H.245
G.728 T.126 H.450.1
G.729 T.127 H.450.2
As you can see, H.323 is a large collection of protocols and specifications. That's what allows it to
be used for so many applications. The problem with H.323 is that it's not specifically tailored to
An alternative to H.323 emerged with the development of Session Initiation Protocol (SIP). SIP is a
more streamlined protocol, developed specifically for VoIP applications. Smaller and more
efficient than H.323, SIP takes advantage of existing protocols to handle certain parts of the
process. Media Gateway Control Protocol (MGCP) is a third commonly used VoIP protocol that
focuses on endpoint control. MGCP is geared toward features like call waiting. You can learn more
about the architecture of these protocols at Protocols.com: Voice Over IP.
VoIP Call Monitoring
VoIP has its distinct advantages and disadvantages. The greatest advantage of VoIP is price and the
greatest disadvantage is call quality. For businesses who deploy VoIP phone networks --
particularly those who operate busy call centers (customer service, tech support, telemarketing, et
cetera) -- call quality issues are both inevitable and unacceptable. To analyze and fix call quality
issues, most of these businesses use a technique called VoIP call monitoring.
VoIP call monitoring, also known as quality monitoring (QM), uses hardware and software
solutions to test, analyze and rate the overall quality of calls made over a VoIP phone network
[source: ManageEngine]. Call monitoring is a key component of a business's overall quality of
service (QoS) plan.
Call monitoring hardware and software uses various mathematical algorithms to measure the
quality of a VoIP call and generate a score. The most common score is called the mean opinion
score (MOS). The MOS is measured on a scale of one to five, although 4.4 is technically the
highest score possible on a VoIP network [source: TestYourVoIP.com]. An MOS of 3.5 or above is
considered a "good call" [source: ManageEngine].
VoIP Cell Phones
Dual-mode cell phones contain both a regular cellular radio and a Wi-Fi (802.11 b/g) radio. The
Wi-Fi radio enables the cell phone to connect to a wireless Internet network through a wireless
router. If you have a wireless Internet router in your home, or if you're sitting at a Starbucks with
wireless Internet access, you can use your cell phone to make VoIP calls. Here's how it works:
1. When the cell phone is in range of a wireless Internet network, the phone
automatically recognizes and connects to the network.
2. Any calls you initiate on the wireless network are routed through the Internet as
VoIP calls. With HotSpot@Home, all VoIP calls are free.
3. If the phone is out of range of a wireless Internet signal, it automatically switches
over to the regular cellular network and calls are charged as normal.
4. Dual-mode phones can hand off seamlessly from Wi-Fi to cellular (and vice versa)
in the middle of a call as you enter and exit Wi-Fi networks.
10. IP Telephony & VoIP
Similar to dual-mode cell phones are Wi-Fi phones. Wi-Fi phones aren't technically cell phones
because they only have a Wi-Fi radio, not a cellular radio. Wi-Fi phones look like cell phones
(small, lightweight handsets), but can only make calls when connected to a wireless Internet
network. That means all Wi-Fi phone calls are VoIP calls.
Wi-Fi phones are useful in large companies and offices with their own extensive wireless networks.
And could prove to be the next big thing, with the expanding market for municipal Wi-Fi. [source:
Dr. Dobb's Portal]. Imagine that your entire city was covered by a high-speed wireless network.
That means cheap (if not free) VoIP calls wherever you go.
The Future Of VoIP Goes Wireless
Voice over Internet Protocol (VoIP) is one of the hottest and most hyped technologies in the
communications industry. Businesses and consumers are already taking advantage of the cost
savings and new features of making calls over a converged voice-data network, and the logical next
step is to take those advantages to the wireless world. The potential impact of wireless VoIP on the
communications market is enormous — market research firm ABI Research has forecasted that
dual mode cellular/voice over WiFi enabled handsets will surpass 50 million by 2009 —
accounting for seven percent of the overall handset market. Early adopters are already touting the
benefits of using WiFi to make inexpensive phone calls, but is this truly a technology that will take
off any time soon?
The Wireless VoIP Opportunity
Wireless VoIP theoretically has many advantages, including reduced cost for calls and higher-
bandwidth data transfers versus a traditional cellular connection. WiFi networks cost a fraction of
what traditional cell tower technology costs to deploy, and can be rolled out quickly without the
detailed site reviews required to install radio towers. More importantly, wireless VoIP can actually
dramatically improve call quality — especially in residential areas or office towers where
traditional mobile network coverage is spotty. What does this mean for the average user? As the
workforce moves to a flexible, non-static environment, wireless VoIP will allow employees to roam
from mobile networks to WiFi-based home and office networks — using a single device to manage
communications that currently traverses mobile, home, and office handsets. Within the United
States, nearly five million homes already have a WiFi network installed — imagine if you could
use the same handset both in your home for cheap IP-based calls, and then switch to cellular when
you leave the building? Clearly, this technology has a lot to offer, from the enterprise to the average
Wireless VoIP offers potential savings by allowing companies to change the way they manage their
phone systems. For example, instead of having voicemail, caller ID and e-mail separately, wireless
VoIP will allow customers to retrieve all of their messages in one place, alleviating the pain of
having different operators for different services and ultimately dealing with several bills at a time.
Employees can also download software applications, enabling them to turn their phones into “mini-
computers” and track inventory, or log onto the company’s intranet.
From a consumer perspective, the increased bandwidth from using a dual handset will potentially
allow us to download video and movies, watch TV shows via on-demand technology and even
videoconference with friends and family — all at speeds faster than cellular networks.