Call Flow Specification for Contact

Center and CRM Integration

Call flows are important to specify and define early in the Contact Center and CRM Application
Integration project. The shown below is a basic call flow, followed by some additional call flow
details to consider.

Basic Call Flow

Contact centers use call or interaction ‘flows’ as the basic building blocks to defining customer
service processes. Interaction flows can also be mapped to the different systems involved in the
process to understand the system requirements.

A basic customer call flow is useful to explain the role of AMC Multi-Channel Integration Suite in
managing the integration between CRM and contact center systems.

This example is for an inbound customer phone call. For other channel interactions such as email
and chat, the process flow is very similar.

Customer Call Flow Example

1. An incoming customer call reaches the contact center’s PBX/ACD system, which prompts
the customer to input a customer account number.
2. A call routing decision is made by the PBX/ACD, or a contact center application, based
on the customer account number with the possible validation of the account number in
the CRM application.
3. PBX/ACD signals the inbound call event to the CTI server along with the customer
account number.
4. CTI server, through its standard communications API, passes the event data to the
integration software; for example, the AMC Multi-Channel Integration Suite (MCIS).
5. The integration software (AMC MCIS) communicates the event data to the CRM
application.
6. CRM application uses the customer account number to perform a look-up of the
customer information.
7. The agent’s desktop populates with the customer information, using the specific pre-
configured screen (i.e. service order) simultaneous with the arrival of the call to the
agent’s phone from the PBX.
8. Agent uses the CRM user interface to create or update the customer’s interaction
information and softphone call controls to manage the interaction.
9. A record and details of the customer interaction is automatically stored as part of the
customers interaction history.

Additional Call Flow Details

Agent Login and Logout

Agent can login and logout of queues through desktop controls. Login and logout is synchronized
with the agent’s hard phone. Agents can login for individual or multiple channels simultaneously
(i.e. phone, email and chat).
Agent Work Mode Settings

Agent can set work mode to specific work status: ready, not ready, or other work (AUX work)
modes. The work mode settings synchronize with the ACD to support “auto-in” and after call
work.

Desktop Controls

The integration supports the following standard telephony features:

 Answer call: an agent can answer an incoming call.

 Drop call: an agent can drop a connected call.
 Hold and Retrieve: an agent can place a call on hold and retrieve a call on hold.
 Make call: an agent can make or place a call.
 Blind transfer: an agent can transfer a call to another agent without consulting with the
agent.
 Warm transfer (with Reconnect option): an agent can transfer a call to another agent after
consulting with the agent. Optionally, the agent can reconnect to the call after the
consultation.
 Conference (with Reconnect option): an agent can conduct a conference with another
agent. Optionally, the agent can reconnect to a call after the conference.

Similar desktop controls are supported for email and web chat interaction management.

Call Attached Data

Transfer and conference functionality can be accompanied with relevant attached data including
ANI, DNIS and any custom data provided by the underlying switch, IVR or vendor CTI software.
(As well as additional data provided by the agent through the IC client [Ex. Activities])

Note: Information in this article has been copied with permission from an AMC Technology
reference document.

Since VoIP contact centers can cut a business's operating costs and increase sales, it’s no wonder that they
are cropping up everywhere. For starters, an IP-enabled contact center lets a company set up shop
anywhere in the world at record speed — and without costly infrastructure investments. What’s more, an IP
infrastructure allows for remote, at-home agents, which can enhance customer service while slashing
overhead costs. And by converging voice and data traffic, a company can reduce operating expenses and
simplify call-center-management processes.

But for all of a VoIP contact center's potential benefits, there are important steps that
companies need to take when setting one up. Here are the considerations that your
enterprise needs to bear in mind.
Calculate Bandwidth
According to Donna Fluss, president of DMG Consulting LLC, a firm specializing in call centers and real-
time analytics, “The biggest issue is that a company has to make sure it really has the proper amount of
bandwidth.” Before deploying a VoIP system, companies must carefully calculate the bandwidth required to
support such a network. That’s because voice communication is bulkier than conventional text and requires
greater connection speed. Ample bandwidth ensures that packets of data are transmitted at reasonable
speeds. Insufficient bandwidth, however, can give rise to performance issues such as latency, jitter and
echo.

Vendor Selection
Given the wide array of VoIP contact-center solutions that companies have to choose from, selecting a
vendor is no easy task. Said Fluss, “You have to go through an entire selection process to make sure that
you get the functionality you need and that you have the core infrastructure to support it.” For most
companies, the question comes down to whether it’s best to go with a hosted or on-premise solution.
Without a doubt, more and more businesses are handing over the VoIP contact-center reins to a third-party
provider. In fact, DMG Consulting predicts that by the end of 2011, 30 to 35 percent of all new contact-
center seats will be hosted. But despite cost and infrastructure benefits, a hosted environment isn’t
necessarily a contact-center cure-all. “You still have to make sure you have the bandwidth,” warned Fluss.
And as for in-house expertise, Fluss noted that, “Some hosted solution vendors have a tremendous amount
of expertise, but some of them don’t. In fact, if you’re a mature call center, you may have a lot more
internal expertise.”

Establish Security Measures

Security is a factor that companies simply cannot afford to overlook when establishing a VoIP-supported
contact center. In fact, according to McAfee Inc., VoIP attacks are expected to increase by 50 percent in
2008. And more than twice the number of VoIP-related vulnerabilities were reported in 2007 as opposed to
2006. For this reason, a company needs to establish best practices for its agents, as well as measures for its
IT managers that cover everything from encryption and authentication procedures to the handling of
privacy breaches and DoS (denial-of-service) attacks.

Pull Together Expertise

Many companies that migrate from a traditional phone system to a VoIP contact center don't think about
the possibility of having to hire a whole new IT team. That’s a dangerous oversight, seeing as how
operating a TDM (Time-Division Multiplexing) -based contact center can differ greatly from one that relies
on IP telephony. Said Fluss, “If you’ve been supporting TDM environments, you have to make sure you
have the right knowledge to support a VoIP infrastructure.”

Set Up the Business Team

The single biggest reason that VoIP-based contact centers fail is that companies neglect to include business
process and operations into their planning. The new contact center needs to be properly integrated into how
your organization conducts business, or all the cost and time benefits will be wasted. Make sure that your
company has a team on the operations side that is involved in selecting and installing the solution so that
the new contact center will be effective.
The demand of VoIP is continuously increasing in the BPO firms and the business concerns as
well. Traditional phone calls cost more than the calls on VoIP. Thus, businesses utilize this
service to save on the expenses involved in telephone calls. The volume of outgoing calls is
quite huge in the case of the business houses and call centres. The entire business of the BPO
industry involves a huge volume of incoming and outgoing calls, thus cheaper calls are greatly
beneficial for them.

VoIP has remarkably improved in the recent past, which has resulted in its application in
business. Earlier, the VoIP was of a poor quality and so it could not be used for offering
professional business operations. Just like the other technologies, VoIP has also improved with
time. The VoIP providers are offering quality service that can be distinguished from the
conventional telephone calls.

VoIP acts as a cost effective platform for the businesses that use frequent telephone calls for
carrying out their operations. It not only helps them to save on the labor costs but also on the
cost of the outgoing and incoming telephone calls. International calls made over the
conventional telephones cost very high as compared to those made on VoIP. The use of this
technology helps in saving the extra expenses involved in the international phone calls.

Thus, the business owners can directly call up their customers without worrying about the costs
involved. In this way, they will be able to shape a good relationship with their customers. As
we know that, better customer relationship is one of the key factors in expanding the business.

Over the years, businesses have realized the potential of the VoIP service and that is why now
it is greatly in demand by the call centres and business organizations. Those organizations do
not have a provision of their own; they tend to hire the VoIP service from the BPO firms.

Technology has become so advanced that things have become easy for individuals as well as for
organizations. Everyone wants things to get done without wasting much of time, energy or
money. The VoIP technology is an effective solution to the execution of business operations
with efficiency.

The entire business of the call center industry is based on a large number of calls 24x7. The
employees need to be very alert so that not a single call goes unanswered or else it will have a
negative impact on the business of the concerned BPO agency. The use of VoIP has made work
simpler for the BPO workers through its voice recording operations. The agents can record their
sales pitches or answers relevant to the customer’s queries with the help of VoIP in order to
answer the customers in their absence. Thus, they are able to accomplish their task as well as
satisfy the customers, which is essential for their business. On the other hand, VoIP enables
them to save on the cost of phone calls and carry out their operations more efficiently.

Voip working-

If you've never heard of VoIP, get ready to change the way you think about long-distance phone calls.
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 the Internet.

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 service.

Above all else, VoIP is basically a clever "reinvention of the wheel." In this article, we'll explore the
principles behind VoIP, its applications and the potential of this emerging technology, which will more
than likely one day replace the traditional phone system entirely.

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:

• ATA -- The simplest and most common way is through the use of a device called
an ATA (analog telephone adaptor). The ATA allows you to connect a standard
phone to your computer or your Internet connection for use with VoIP. The ATA
is an analog-to-digital converter. It takes the analog signal from your traditional
phone and converts it into digital data for transmission over the Internet.
Providers like Vonage and AT&T CallVantage are bundling ATAs free with their
service. You simply crack the ATA out of the box, plug the cable from your
phone that would normally go in the wall socket into the ATA, and you're ready
to make VoIP calls. Some ATAs may ship with additional software that is loaded
onto the host computer to configure it; but in any case, it's a very straightforward
setup.
• IP Phones -- These specialized phones look just like normal phones with a
handset, cradle and buttons. But instead of having the standard RJ-11 phone
connectors, IP phones have an RJ-45 Ethernet connector. IP phones connect directly
to your router and have all the hardware and software necessary right onboard to
handle the IP call. Wi-Fi phones allow subscribing callers to make VoIP calls from
any Wi-Fi hot spot.
• Computer-to-computer -- This is certainly the easiest way to use VoIP. You
don't even have to pay for long-distance calls. There are several companies
offering free or very low-cost software that you can use for this type of VoIP. All
you need is the software, a microphone, speakers, a sound card and an Internet
connection, preferably a fast one like you would get through a cable or DSL modem.
Except for your normal monthly ISP fee, there is usually no charge for computer-
to-computer calls, no matter the distance.

If you're interested in trying VoIP, then you should check out some of the free VoIP software available on
the Internet. You should be able to download and set it up in about three to five minutes. Get a friend to
download the software, too, and you can start tinkering with VoIP to get a feel for how it works.

Next, we'll look at exactly how VoIP is used.

Using VoIP
VoIP phone users can make calls from anywhere there's a broadband connection.
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.

With VoIP, you can make a call from anywhere you have broadband connectivity. Since the IP phones or
ATAs broadcast their info over the Internet, they can be administered by the provider anywhere there's a
connection. So business travelers can take their phones or ATAs with them on trips and always have access
to their home phone. Another alternative is the softphone. A softphone is client software that loads the
VoIP service onto your desktop or laptop. The Vonage softphone has an interface on your screen that looks
like a traditional telephone. As long as you have a headset/microphone, you can place calls from your
laptop anywhere in the broadband-connected world.

Most VoIP companies are offering minute-rate plans structured like cell phone bills for as little as $30 per
month. On the higher end, some offer unlimited plans for $79. With the elimination of unregulated charges
and the suite of free features that are included with these plans, it can be quite a savings.

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
between.

Let's say 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. 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.

On the next page, we'll talk about packet switching.

VoIP: Packet Switching

VoIP phone users can make calls using their Internet connection.

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 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?

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 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.

Next, we'll look at the advantages of using VoIP.

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.

Let's say that you and your friend both have service through a VoIP provider. You both have your analog
phones hooked up to the service-provided ATAs. Let's take another look at that typical telephone call, but
this time using VoIP over a packet-switched network:

1. You pick up the receiver, which sends a signal to the ATA.

2. The ATA receives the signal and sends a dial tone. This lets you know that you
have a connection to the Internet.
3. You dial the phone number of the party you wish to talk to. The tones are
converted by the ATA into digital data and temporarily stored.
4. The phone number data is sent in the form of a request to your VoIP company's
call processor. The call processor checks it to ensure that it's in a valid format.
5. The call processor determines to whom to map the phone number. In mapping,
the phone number is translated to an IP address (more on this later). The soft switch
connects the two devices on either end of the call. On the other end, a signal is
sent to your friend's ATA, telling it to ask the connected phone to ring.
6. Once your friend picks up the phone, a session is established between your
computer and your friend's computer. This means that each system knows to
expect packets of data from the other system. In the middle, the normal Internet
infrastructure handles the call as if it were e-mail or a Web page. Each system must use
the same protocol to communicate. The systems implement two channels, one for
each direction, as part of the session.
7. You talk for a period of time. During the conversation, your system and your
friend's system transmit packets back and forth when there is data to be sent. The
ATAs at each end translate these packets as they are received and convert them to
the analog audio signal that you hear. Your ATA also keeps the circuit open
between itself and your analog phone while it forwards packets to and from the IP
host at the other end.
8. You finish talking and hang up the receiver.
9. When you hang up, the circuit is closed between your phone and the ATA.
 10. The ATA sends a signal to the soft switch connecting the call, terminating the
session.

VoIP Terms
The central call processor is a piece of hardware running a specialized database/mapping program called a
soft switch. See the "Soft Switches" section to learn more.

Probably one of the most compelling advantages of packet switching is that data networks already
understand the technology. By migrating to this technology, telephone networks immediately gain the
ability to communicate the way computers do.

It will still be at least a decade before communications companies can make the full switch over to VoIP.
As with all emerging technologies, there are certain hurdles that have to be overcome. We'll look at those in
the next section.

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 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.

• First of all, VoIP is dependant on wall power. Your current phone runs on
phantom power that is provided over the line from the central office. Even if your
power goes out, your phone (unless it is a cordless) still works. With VoIP, no
power means no phone. A stable power source must be created for VoIP.
• Another consideration is that many other systems in your home may be
integrated into the phone line. Digital video recorders, digital subscription TV
services and home security systems all use a standard phone line to do their thing.
There's currently no way to integrate these products with VoIP. The related
industries are going to have to get together to make this work.
• Emergency 911 calls also become a challenge with VoIP. As stated before, VoIP
uses IP-addressed phone numbers, not NANP phone numbers. There's no way to
associate a geographic location with an IP address. So if the caller can't tell the
911 operator where he is located, then there's no way to know which call center to
route the emergency call to and which EMS should respond. To fix this, perhaps
geographical information could somehow be integrated into the packets.

Testing, Testing...
Wondering if your broadband connection could support VoIP service? Brix
Network offers a way to test your Internet connection to see how well it works.
 • Because VoIP uses an Internet connection, it's susceptible to all the hiccups
normally associated with home broadband services. All of these factors affect
call quality:
 Latency
 Jitter
 Packet loss

Phone conversations can become distorted, garbled or lost because of transmission errors. Some
kind of stability in Internet data transfer needs to be guaranteed before VoIP could truly replace
traditional phones.

• VoIP is susceptible to worms, viruses and hacking, although this is very rare and
VoIP developers are working on VoIP encryption to counter this.
• Another issue associated with VoIP is having a phone system dependant on
individual PCs of varying specifications and power. A call can be affected by
processor drain. Let's say you are chatting away on your softphone, and you
decide to open a program that saps your processor. Quality loss will become
immediately evident. In a worst case scenario, your system could crash in the
middle of an important call. In VoIP, all phone calls are subject to the limitations
of normal computer issues.

One of the hurdles that was overcome some time ago was the conversion of the analog audio signal your
phone receives into packets of data. How it is that analog audio is turned into packets for VoIP
transmission? The answer is codecs.

VoIP: Codecs

©2007 HowStuffWorks
VoIP software processes and routes the calls.

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 VoIP.

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.

Codecs use advanced algorithms to help sample, sort, compress and packetize audio data. The CS-ACELP
algorithm (CS-ACELP = conjugate-structure algebraic-code-excited linear prediction) is one of the most
prevalent algorithms in VoIP. CS-ACELP organizes and streamlines the available bandwidth. Annex B is
an aspect of CS-ACELP that creates the transmission rule, which basically states "if no one is talking, don't
send any data." The efficiency created by this rule is one of the greatest ways in which packet switching is
superior to circuit switching. It's Annex B in the CS-ACELP algorithm that's responsible for that aspect of
the VoIP call.

The codec works with the algorithm to convert and sort everything out, but it's not any good without
knowing where to send the data. In VoIP, that task is handled by soft switches.

E.164 is the name given to the standard for the North American Numbering Plan (NANP). This is the
numbering system that phone networks use to know where to route a call based on the dialed numbers. A
phone number is like an address:

(313) 555-1212

313 = State
555 = City
1212 = Street address

The switches use "313" to route the phone call to the area code's region. The "555" prefix sends the call to a
central office, and the network routes the call using the last four digits, which are associated with a specific
location. Based on that system, no matter where you're in the world, the number combination "(313) 555"
always puts you in the same central office, which has a switch that knows which phone is associated with
"1212."

The challenge with VoIP is that IP-based networks don't read phone numbers based on NANP. They look
for IP addresses, which look like this:

192.158.10.7

IP addresses correspond to a particular device on the network like a computer, a router, a switch, a gateway
or a telephone. However, IP addresses are not always static. They're assigned by a DHCP server on the
network and change with each new connection. VoIP's challenge is translating NANP phone numbers to IP
addresses and then finding out the current IP address of the requested number. This mapping process is
handled by a central call processor running a soft switch.

The central call processor is hardware that runs a specialized database/mapping program called a soft
switch. Think of the user and the phone or computer as one package -- man and machine. That package is
called the endpoint. The soft switch connects endpoints.
Soft switches know:

• Where the network's endpoint is

• What phone number is associated with that endpoint
• The endpoint's current IP address

VoIP: Soft Switches and Protocols

Customer call centers like this hotline require consistent call quality and many rely on VoIP
technology.

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.

Protocols
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
G.723.1 T.125 H.245
G.728 T.126 H.450.1
G.729 T.127 H.450.2
 H.450.3
RTP
X.224.0

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 VoIP.

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.

One of the challenges facing the worldwide use of VoIP is that these three protocols are not always
compatible. VoIP calls going between several networks may run into a snag if they hit conflicting
protocols. Since VoIP is a relatively new technology, this compatibility issue will continue to be a problem
until a governing body creates a standard universal protocol for VoIP.

VoIP is a vast improvement over the current phone system in efficiency, cost and flexibility. Like any
emerging technology, VoIP has some challenges to overcome, but it's clear that developers will keep
refining this technology until it eventually replaces the current phone system.

VoIP Call Monitoring

Newer Skype services are equipped to handle VoIP protocol.

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. 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. An MOS of 3.5 or above is considered a "good call".

To come up with the MOS, call monitoring hardware and software analyzes several different call quality
parameters, the most common being:

• Latency -- This is the time delay between two ends of a VoIP phone
conversation. It can be measured either one-way or round trip. Round-trip latency
contributes to the "talk-over effect" experienced during bad VoIP calls, where
people end up talking over each other because they think the other person has
stopped speaking. A round-trip latency of over 300 millisecond is considered poor
• Jitter -- Jitter is latency caused by packets arriving late or in the wrong order.
Most VoIP networks try to get rid of jitter with something called a jitter buffer
that collects packets in small groups, puts them in the right order and delivers
them to the end user all at once. VoIP callers will notice a jitter of 50 msec or
greater.
• Packet loss -- Part of the problem with a jitter buffer is that sometimes it gets
overloaded and late-arriving packets get "dropped" or lost. Sometimes the packets
will get lost sporadically throughout a conversation (random loss) and sometimes
whole sentences will get dropped (bursty loss). Packet loss is measured as a
percentage of lost packets to received packets.

There are two different types of call monitoring: active and passive. Active (or subjective) call monitoring
happens before a company deploys its VoIP network. Active monitoring is often done by equipment
manufacturers and network specialists who use a company's VoIP network exclusively for testing purposes.
Active testing can't occur once a VoIP network is deployed and employees are already using the system.

Passive call monitoring analyzes VoIP calls in real-time while they're being made by actual users. Passive
call monitoring can detect network traffic problems, buffer overloads and other glitches that network
administrators can fix in network down time.

Another method for call monitoring is recording VoIP phone calls for later analysis. This type of analysis is
limited, however, to what can be heard during the call, not what's happening on the actual network. This
type of monitoring is usually done by human beings, not computers, and is called quality assurance.
VoIP Cell Phones

HotSpot@Home lets you make cell phone calls over your home WiFi network. Learn how this useful
technology integrates seamlessly between two networks.

VoIP-enabled cell phones are just entering the consumer market. In the United States, only T-Mobile's
HotSpot@Home service allows customers to make cell phone calls over a VoIP network. HotSpot@Home
relies on a device called a dual-mode cell phone.

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.

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.

In England, a company called Hutchinson 3G has partnered with the popular VoIP service Skype to
introduce the 3 Skypephone. The Skypephone allows users to make free cell phone calls to other Skype
users. The phone can also make regular cell-phone calls to non-Skype users for the normal fees. Here's how
it works:

1. To make a Skype call using the 3 Skypephone, you have to be on 3's cellular
network.
2. To initiate a Skype call, find a Skype user in your phone's address book and press
the big "Skype" button.
3. The call first goes over 3's cellular GSM network to a fixed Internet line, which
then connects the call to Skype.
4. From your 3 Skypephone, you can make free VoIP calls to other Skype users
whether they have a Skypephone or not. You can talk to Skype users on their PCs
or using other Skype VoIP products.

Use of VoIP in Amateur Radio

Amateur or ham radio operators can use VoIP technology to set up temporary stations such as this
one used by the Red Cross following Sept. 11.

Think of amateur radio, or ham radio, as an early version of the Internet. Using a worldwide network of
radio towers, antennas and transceivers, amateur radio enthusiasts are able to communicate with fellow
hobbyists around the globe, sometimes by voice and sometimes by Morse code.

Amateur radio is limited by the distance that radio waves can travel. To send a signal to the other side of
the world requires calculated timing and more than a little bit of luck. Every 11 years, for example, there's a
peak in the number of sunspots produced by the sun, which increases the intensity of something called
ionospheric propagation [source: International Solar Terrestrial Physics Program]. By bouncing radio signals
high into the ionosphere, ham radio users can send long-distance messages. During off-peak years it's much
more difficult.

Now amateur radio fans are using VoIP technology to link users around the globe. Here's how it works.
Ham radio has always relied on FM repeaters, large radio towers that act as base stations for accessing the
radio network from home. By attaching an Internet-connected PC to these repeater stations, people can
communicate with the repeater using VoIP.
Several amateur radio fans have developed special software that helps connect home radio transceivers to
the Internet. Users can connect their ham radio transceivers to their PC sound card and use the computer
software to search for available repeater stations across the world [source: ARRL]. No longer are ham radio
fans limited to the closest repeater station. If you live in Indiana, you can call into a repeater station in
Mozambique and chat with local amateur radio aficionados instantly.

There are also software programs that allow you to communicate with other amateur radio users directly
from your PC, without having an actual ham radio [source: ARRL]. Some ham radio purists wouldn't call
this amateur radio, while others hope that this new technology will draw more young people into the hobby.

For more information about VoIP, amateur radio and related topics, check out the links on the next page.

How VoIP has helped CRM in Cost Management?

The most recent trend in the industry insinuates that VoIP has been proficiently used in the industry. A
number of business enterprises and companies have successfully executed VoIP in their applications. . In
recent years, the VoIP call center industry has seen a gigantic growth. Telephony is an important part of call
center industry. VoIP call center applications play an important role.

VoIP call centers fill many needs of the business enterprise. They provide traditional customer service as
well as support to sales and marketing. Gainfulness is the key objective of call center solution VoIP, which
aids them to meet the necessary requirements. The severe competition among the call centers compelled
them to aim for the correct and perfect solutions. Delivering perfection and efficiency has become the need
of the hour and will bring more business for call center solution VoIP.

New call center VoIP technology enables the call centers to acquire some of the advantages of smaller
costs while preserving employment levels in the U.S. It is difficult to move the call centers to another
physical site. Therefore, many operators are currently moving several of the operations directly to the
Internet. C all center VoIP Internet is being used as a communication medium. Using this advance
technology, distributors can afford the most costly components of call center solution VoIP to be shared.

Telephony infrastructure within call center VoIP has been improved vastly, mostly due to the use of VoIP.
Broadband connection has made the telephone infrastructure in VoIP call centers available anywhere. It
also helps them go to a place where skills and talents are plentiful rather than going to places with high labor
rates and too expensive for VoIP call centers in the marketplace.

Especially since one physical infrastructure for voice and data is much more proficient than dual networks.
What merchants don't constantly mention is that achieving incredible ROI requires budgeting for likely
hidden costs in the VoIP marketplace.

Transferring a call center's infrastructure to the Internet is beneficial for many types of organizations. It will
resolve the trials of managing people who stay at a far away place. It also provides call center VoIP tools
and techniques to manage a disseminated workforce with proficient management. VoIP call centers have
made this all possible.

Integration of Telecom and Computer Services

The Enterprise solutions provided by CRM VoIP call center technology are designed to subtract the costly
replacement of existing equipment and needless retraining of technical personnel to learn how to use a wide
telephone system at a new company. By incorporating the partners' infrastructure solutions into your
accessible telecom and data networks, VoIP call centers can supply you with the reliability and investment
return that today's business standards usually demand.

With the combination of multi-vendor interoperability using CRM VoIP call center technology, carriers and
service providers are capable of reducing their termination costs while revenue is received for every
telephone call they lapse. The Call Center CRM technology VoIP has absolute relationships with these
distributors, and can supply carriers and service providers with a blueprint to building a profitable Call Center
CRM technology VoIP.

Service providers and carriers no longer need to train and pay a team of staff to develop and maintain
relationships with multiple carriers. CRM VoIP call center technology determines rate agreements and
interconnectivity with carriers throughout the globe through its diversity of its carrier alliances.

VoIP is quickly evolving in building or expanding a Call Center CRM VoIP technology network, which is
expensive and time-consuming as the carriers can't afford to squander. Expert technicians on board can
build or expand a network which is very crucial. CRM VoIP call center technology helps carriers expand or
build all or even parts of their VoIP network, while they spend time and investments on retaining and
increasing the customer base.

Call Center CRM technology VoIP provides onsite technical expertise through its multiple staff augmentation
option s. The team of expert engineers has extensive VoIP and TDM field operations backgrounds, which
include the implementation, design and management of the largest VoIP networks on the map.

Call Center and VoIP

CRM VoIP call center technology enables service-like custom e-business with core information systems for
companies around the world. CRM VoIP call center technology operates under a high quality onsite and
offshore model of high value that affords better, more cost effective and faster development and deployment
of prestigious systems across a wide range of transaction-demanding business needs. These time-tested
processes guarantee critical cost savings.

The employees are devoted to partnerships that uphold long-term, confirmed value in order to achieve in the
current global marketplace. Influencing years of experience, VoIP call centers have developed a rich
portfolio of tried-and-true modular frameworks across several cutting-edge technologies.

Step 1—Get High Speed Internet

In order to use VoIP, you need a broadband connection. If the person you are calling will be receiving the
call on a computer, they need a broadband connection too. This means a high-speed Internet connection
usually provided by a cable or DSL modem.

Broadband modems are usually used to connect computers to the Internet, but if you make VOIP calls from
a phone, a computer is not necessary. The broadband phone connects either directly to a VOIP phone or to
a regular phone via an Analog Terminal Adaptor (see Step 2 below).

Step 2 Make Calls

Next, decide what type of call you want to make: computer to computer, computer to phone, or phone to
phone.
Computer to Computer

The simplest form of VOIP is a computer-to-computer voice connection. All that is required is a computer
with sound card, a headset consisting of earphones and microphone, and some VOIP software. Most
software packages are free and allow you to connect to any computer running the same software. There is
no charge for this type of connection and calls can be made to anywhere in the world. is an example
of free VOIP software. Read more about Skype...

Computer to Phone

To make a computer to phone connection, you also need VOIP software, headset, earphones and
microphone, but the party on the other end can use any regular land-line phone and sometimes cell phone—
in other words, a phone not connected directly to the Internet. This type of call is not usually free but the
cost is quite a bit lower than what your telephone company charges.

The only time that both parties need a particular VoIP software package is when they are making computer-
to-computer calls. Parties receiving land-line or cellular calls do not need any extra equipment or software.

Phone to Phone

To make a phone to phone connection, there are two options. (1) Use a regular phone plugged into an ATA
adaptor, which in turn plugs into your broadband modem. (2) Use an IP phone (also called a SIP phone or
VOIP phone) that plugs directly into your broadband connection. Voip.com is a popular Internet phone
company that provides this kind of service. Read more about Voip.com.
Step 3—Turn Voice into Data
VOIP is based on digital data transmission. So, when you make a VOIP call the first thing that happens is
the analog signal of the your voice is converted to digital data. This voice-to-data conversion is done with
an Analog-Terminal Adaptor (ATA).

The ATA can be a separate box that plugs into your phone and then into your broadband modem, or it can
be built into a special IP phone (sometimes called VOIP phone or SIP phone). If you are using a computer
to make calls, the ATA function is built into the VOIP software. Service providers, such as Voip.com, will
provide the ATA or VOIP phone to you as part of their VOIP package.

The ATA divides the analog voice signal into chunks—or packets—and then compresses the chunks which
significantly reduces the amount of digital data to be transmitted. Voice compression is handled by a
CODEC (Coder/Decoder).

Now that your voice has been converted to digital and compressed, it can be sent over the Internet. The data
stream must be divided into packets which, besides containing the voice data, also has information
concerning its destination and their sequence in the data stream. The transmission of voice packets over the
internet are handled by various protocols.

Among other things, protocols contain information about the sequence of the data packets so they can be
reconstructed in the correct order at their destination. In traveling from A to B, some packets may be
dropped. However, there is usually enough information to make the conversation legible. The number of
packets that are dropped depends on the speed of your Internet connection in the distance between the two
parties.

Step 4—Turn Data Back into Voice

Before voice data packets can be received by a listener, they must be reassembled in the correct order and
converted back to analog. If you are making a computer to computer call, the voice data stream is
reassembled at the receiving party's computer by the VOIP software. The entire conversation never leaves
the IP world--the Internet.

If you are calling someone on a landline, the voice data stream jumps off the internet and onto the regular
phone lines at a VOIP gateway. The gateway is usually located in the city or region where the receiving
party is located. This is the reason why long distance rates using VOIP can be so cheap. VOIP service
providers basically have their VOIP gateways set up in major centers all over the world. When you call
someone in one of these cities, all you are really paying for is the cost of a local call from the VOIP
gateway to the receiving party.

Step 5—Talk!
Now that your phone call has been routed through the Internet, and possibly shunted through a VOIP
gateway onto the PSTN, and has made it to the receiving party (all in a matter of seconds), you can get
talking!

In many cases the person you are calling won't even know that this is a VOIP call and the voice quality will
be the same as on a regular landline call. However, if too many voice packets were dropped along the way,
or the Internet connection is slow or poor, you may notice any of the following:

• You or the other party's voice is breaking up, or words seem to drop out
(packet loss)
 • A time delay between the speaking and hearing (packet delay or lag)

• Noisy lines or background noise, and voice echo (traditional phone line
problems if the call has been integrated with the PSTN)

If VOIP is such a great thing, why isn’t everybody using it? Well, for starters the bandwagon is crowded—
there are lots of VOIP broadband phone providers out there. The other thing is there are some drawbacks to
VOIP.

For instance, during power outages, you'll have no Internet connection, and therefore, no phone service. In
other words, no power, no phone.

So if the downsides are so well known, the real question is, what are all these thousands of VOIP phone
companies doing about them?

Compare internet phone plans

VOIP Advantages
First, we have an expectation that VOIP has many advantages over regular phone service.

Low cost. If you have a broadband Internet connection (DSL or cable), you can make PC-to-PC phone
calls anywhere in the world for free. If you wish to make a PC-to-phone connection, there's usually a
charge for this but probably much cheaper than your regular phone service.

You can pay as you go or you can sign up with a VOIP service provider and pay a monthly fee in return for
unlimited calls within a certain geographic area. For example, some VOIP services in the United States
allow you to call anywhere in North America at no extra charge. Overseas calls are charged at a relatively
small rate.

Portability. You can take your cheap phone service with you and make and receive phone calls wherever
there is a broadband connection simply by signing in to your VOIP account. This makes VOIP as
convenient as e-mail.

If you are traveling, simply pack a headset and use your laptop plugged into the Internet, or plug a VOIP
phone directly into the Internet connection and you can talk to your family or business associates for almost
nothing. If you don't have a VOIP phone, use an analog terminal adaptor (ATA) to connect a regular phone
to the Internet. Today you can find ATAs that small, portable and inexpensive. A great idea for frequent
travelers.

Phone-to-phone VOIP is also portable. When you sign up with a VoIP service provider the Internet phone
or adaptor that is used with that service is assigned a unique number. This 'phone number' remains valid
even if your VoIP service is in Cleveland and you are connected to the Internet in Bangkok. An Internet
phone is small and light enough to take with you anywhere. Simply plug it into a broadband connection
anywhere in the world and you can make and receive calls just as though you were in your own home or
office.

Features. Unlike regular phone service which usually charges more for extra features, VOIP comes with a
host of advanced communication features. For example, call forwarding, call waiting, voicemail, caller ID
and three-way calling are some of the many services included with VOIP telephone service at no extra
charge. You can also send data such as pictures and documents at the same time you are talking on the
phone.

VOIP Disadvantages
If VOIP is starting to sound really good to you, make sure you understand the following downsides as well.

No service during a power outage. During a blackout a regular phone is kept in service by the current
supplied through the phone line. This is not possible with IP phones, so when the power goes out, there is
no VOIP phone service. One solution to this problem is to use battery backups or power generators to
provide electricity.

If you decide to continue subscribing to a regular phone line as an emergency backup, consider that
monthly cost cuts into your overall VOIP savings. However, VOIP would still make sense in this case if
your home or business made significant long distance calls.

Emergency 911 calls. Another major concern with VOIP involves emergency 911 calls. Traditional phone
equipment can trace your location. Emergency calls are diverted to the nearest call center where the
operator can see your location in case you can't talk. However, because a voice-over-IP call is essentially a
transfer of data between two IP addresses, not physical addresses, with VOIP there is currently no way to
determine where your VOIP phone call is originating from.

To solve this issue, the E911 standard by law mandates that VOIP service providers pass name and address
information to the nearest Public Safety Access Point (PSAP) when 911 is dialed. Not all PSAPs in the
country support E911 yet, so you should ask when deciding on a VOIP service provider if E911 is
supported in your area.

Quality and Reliability. Because VOIP relies on an Internet connection, your VOIP service will be
affected by the quality and reliability of your broadband Internet service and sometimes by the limitations
of your PC. Poor Internet connections and congestion can result in garbled or distorted voice quality. If you
are using your computer at the same time as making a computer VOIP call, you may find that voice quality
deteriorates dramatically.

Bad or unreliable internet connections can result in poor voice quality, such as clipping, voice delay, or
dropped calls. And emergency 911 service may not work as you expect.

VoIP Tutorial - Common Terms

Asynchronous Communication - data communications method in which bits are sent without using a clock signal
for synchronization as opposed to synchronous communication where blocks of data are transmitted using a
synchronizing clock.

Broadband Connection - a high-speed Internet connection using DSL, cable, wireless, fiber optic or satellite
means of transmitting data. The terms broadband usually refers to a high-speed network connection with data
speeds in excess of 128 kilobits per second.

Conference Bridge - is used to connect multiple people over a phone line or broadband Internet connection such
as for a conference call.

Digital Subscriber Line (DSL) - high-speed broadband service used to connect subscribers over existing phone
lines or interfacing with other broadband connections, such as cable, as well.

Frame Relay - is a packet switching method that uses available bandwidth only when it is needed and is efficient
enough to transmit voice data.
Full Duplex - ability for both ends of a communication to simultaneously send and receive data without
degradation in quality.

Internet Telephony - another term for "VoIP" and means a way to turn analog voice data into digital data for
transmission over the Internet.

Internet Service Provider (ISP) - is a business that provides subscriber-based access to the Internet for both
businesses and individuals. With VoIP sometimes two service providers are needed: a VoIP service provider and a
broadband service provider.

Packet - a logically group unit of data that includes payload, originator, destination, and synchronization
information. VoIP is based on the ability to send and receive packets of data.

Private Branch Exchange (PBX) - functions as a small telephone company switchboard, using computer
telephony platforms and automated attendants inside a company's premises to handle calls.

Public Switched Telephone Network (PSTN) - public circuit-switched telephone networks including analog and
digital systems along with mobile and fixed telephones. The traditional local and long distance phone companies
have PSTN networks.

T1 Line - provides a 1.544 Mbps link which is divided into 24 discrete, 64 kpbs voice-grade channels.

VoIP - VoIP stands for Voice Over Internet Protocol. Another term for VoIP is Voice Over IP or Internet telephony.
In general, VoIP is converting analog voice data into digital packets and sending those packets over the Internet to
a receiver at the other end.

IP Telephony Vs VoIP

IP telephony has to do mainly with digital telephony systems (LAN based IP PBX
systems) which use the IP protocol entirely for voice communication. All components of
the IP telephony system use digitized voice which is transferred as IP packets through an
IP network (usually the LAN network). The telephone handsets (VoIP phones) translate
the analogue voice signal into digital voice (binary voice) which is transferred as IP
packets from one phone to another. The call control system is usually a software based
(softswitch) server which handles all call signaling, call routing, IP phone management
etc, again using IP protocol for transport. So think about IP telephony as a bigger
concept.

VoIP on the other hand is a subset of IP Telephony. Basically, VoIP is the technology
which is used by IP Telephony as the vehicle to transport phone calls. VoIP is the
technology in which the analogue voice signal is digitized (analog to digital conversion)
and becomes binary numbers in order to be transferred by the IP protocol. VoIP is the
basis for the implementation and functionality of an IP Telephony system. VoIP can also
be used by legacy TDM based PBX systems to transport voice calls over an IP WAN
network or even over the Internet. Special voice gateways are used to connect to the
legacy PBX telephone system on one end and to the IP network on the other end in order
to translate the TDM voice stream into IP voice packets.

So to summarize, IP Telephony is the overall concept of the modern form of voice

communication which harnesses the power and features of VoIP technology in order to
offer the overall experience of communicating effectively and with lots of extra features.
Now that we described the difference between IP Telephony and VoIP, let’s see more
details about the two concepts:

1. More details about Voice over IP

The term VoIP or Voice over IP refers to the transfer of voice packets over networks
based on Internet technology and, more specifically, the IP Protocol. The IP protocol on
which the whole Internet is based on was created to implement the transmission of data in
the form of data packets. This means that when a data document is transferred over the
Internet is cut into small IP packets and sent over the network. When the document
reaches its destination, the packets are joined again thus recreating the original document.
The same logic applies if the data transferred corresponds to a voice conversation. The
voice is digitized, chopped into packets of data transferred over the network via the IP
protocol. At the destination the packets are rejoined to recreate the voice stream. Here we
should make clear that VoIP refers to the transfer of voice over any IP network. Such a
network is the Internet of course, but when considering VoIP it does not necessarily mean
that we carry voice over the Internet only. It can be any IP-based network (such as a
private corporate WAN network).

2. Packet based (IP Telephony) Vs Circuit Switched Telephone Systems

IP Telephony systems are those using entirely IP packets for voice communication, as
explained before. In contrast to packet switched telephone systems (those based on IP
protocol), conventional telephone systems apply the logic of direct connection between
the two communicating voice parties through a dedicated circuit reserved exclusively for
each contact. Thus the term Circuit switched telephone systems. In packet switched
systems, however, the same communication line can be used to simultaneously pass
different kinds of packets. Thus, the voice packets of one or more conversations may
travel through the same route as other packets transferring data, video etc. This is the
main difference between traditional telephony which is implemented to the public
switched telephone network (PSTN) and telephony implementation on IP networks (or
more generally to packet switched networks).