The Future of Voice over Internet Protocol

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26 Οκτ 2013 (πριν από 3 χρόνια και 7 μήνες)

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The Future of Voice o
ver Internet Protocol

Scott T. Martin

Department of Computer Science

and Software Engineering

University Of Wisconsin


Advances in the telephone communication industry have led to the implementation of
oice over

. VoIP makes it possible

to route phone calls over
the I

which has greatly decre

the cost and the amount of data being se
between call locations. Power concerns, network consistency, virus concerns, and
emergency calls are all issues that are still being dealt wit
h that continue to hold back its

success. In this paper, I plan
on discussing how VoIP works
. I will also cover the

of switching over to VoIP
and issues that

are currently preventing VoIP from
becoming the dominant method of telephony.





When Alexander Graham Bell made the

first telephone call to his assistant Thomas
Watson in 1876, his invention was a huge breakthrough in communications. However, in
the 100 years that followed that monumental call, there were no massive advances in the
telephone communication industry. I
n the late 20

researchers wan
ted to know
if it would be feasi
ble to send video and voice over IP networks. This technology would
be especially useful for sending information across corp
orate intranets and across the
nternet. The research done

by these ind
ividuals has led to the creation



Voice over Internet Protocol’s roots can be traced back to
one of the first computer
network protocols for transporting human speech known as the Network Voice Protoc
NVP was first implemented in December of 1973
nternet researcher Danny Cohen of
the Information Sciences Institute at the University of Southern California. The project
had several goals. One was to develop and demonstrate the feasibility of sec
ure, high
quality, low
bandwidth, real
time, full
duplex digital voice communications over packet
switched comp
uter communications networks. A second goal was to
supply digitized
speech which can be secured by existing encryption devices. T
his research
ject hoped
to creat
e a digital high
quality, low
bandwidth, secu
re voice handling capability as part of
the general military requirement for worldwide secure voice communication.


The pro
tocol had two separate

: control protocols and a data transp
ort protocol. The
control protocol part handles basic features such as origin and destination information,
ring tones, and call termination. The data transport proto
col handled the encoded speech.

Circuit Switching

Current phone systems are based on
circuit switching. It’s a very reliable system that is
somewhat inefficient.
In the early phone systems,
for two parties t
o have a conversation,
they had to

make a connection
between the two locations that stayed open for the
duration of the call. The p
roblem is that the call has to have a dedicated line specifically
for that call, which was rather expensive since the line was not available for other use. In
today’s phone systems, the call would be less expensive due to the high bandwidth of the
fiber o
ptic cable system that calls are transmitted over.




Figure 1: Example of a Circuit Switched Connection

However, current phone calls are very inefficient. Each phone call today is transmitted
over fiber optic lines at a fixed rate of 64 k
in each direction. So for
every second that a conversation lasts,
128 kbps (or 16 kilobytes) are being transmitted.
If a call lasts 10 minutes, this means that roughly 10 megabytes of data is being sent back
and forth. For the most part, when

one person is talking, the other is listening, meaning
that the 10 megabytes of data could be cut in half to be a better use of resources. Better
yet, this could be cut down even more if dead air were able to be removed, moments
when neither party is tal
king. If data was only transmitted when one party was saying
something, it would be much more effective.


Packet Switching

Packet switching
accomplishes the same as c
ircuit switching

but works in a completely
different way
. Instead of opening a con
stant connection between the two parties, packet
switching opens up a short connection, just long enough to send a small chunk of data
called a packet. The sending computer breaks the data into small pieces, addressing each
packet with an address as to wh
ere its destination is. Each packet contains a payload
which contains a piece of the original message. The sending computer sends each packet
to the nearest router and then forgets about it. Each packet can be sent along any one of
thousands of paths to

the destination computer. Once it arrives at its destination, the
packets are reassembled into the original message.

Figure 2: Example of a Packet Switched Connection




The benefit to this system is that the network is free to route the packets along t
he least
congested and cheapest paths. The computers that sent the information are not tied up on
one connection, allowing for the sending and receiving of other information from other

The theory and implementation of packet switching is the m
ajor idea behind
Quality of Service (QoS).

VoIP Implementation

P uses this
internet technology of packet switching to provide phone service. Packet
switching allows for multiple calls to occupy the space of one call on a circuit switching
rk. In the
previously used example, if the
MB call that was made on a circuit
switching network had been on a packet switching network, it would be compacted
enough to allow four or five calls in place of the one call.

A VoIP call
from home
works si
milarly to a regular call. A call is initiated by picking up
a receiver, which sends a signal to an analog telephone adapter. The ATA sends back a
dial tone alerting you that you have a working
nternet connection and that you are ready
to make a call.
The sender then dials a phone number which is sent to a VoIP call
This call processor is called a soft switch, which will be covered more in
depth a little later in this paper.

call processor translates the number entered into a
n IP address. Once this address is known, a signal is sent to the receiver’s
phone, alerting it to start ringing. Once the connection is made, a session is created
between the two computers. This alerts the two computers to expect packets from each
r. The data transfers during the call are handled by the regular infrastructure of the

While the call is going on, the ATAs on each end convert the data back and forth between
digital and analog formats.
When the call is completed, one party

hangs up
, terminating
the signal between the phone and the ATA. The ATA then sends a signal along to the
call processor ending the session. One of the best advantages about this technology is
that it uses technology already in
common use. This technolo
gy will allow telephone
networks to communicate the way the computers do, making the transition from the old
phone system to VoIP considerably easier.



decoders (or codecs) are used to convert the analog audio signal into a digital
l for transmission over the internet. After the transmission is complete, a codec is
again used to reassemble the digital signal into an uncompressed
audio signal.
This audio signal is what is played and is what the receiver of the call hears.

odecs create the digital signal by sampling the original analog audio signal several
thousand times per second. Each tiny sample is converted and compressed into digital
data for transmission. When the tiny data samples are received, they are reassembled




nearly the original analog signal. Because there are so many pieces, the gaps between the
reassembled pieces are inaudible to the human ear so it sounds like one continuous piece
of audio.

The most commonly used codec in VoIP is the G.729A codec wh
ich has a sampling
speed of 8,000 times per s
econd. While it is one of the lower sampling rates
, it is the best
compromise of sound quality and transmission speed. Sampling rates of 64,000 and
32,000 times per second are both

commonly used in VoIP techn

Codecs are controlled by advanced algorithms that sort, sample, compress, and pack the
data for transmission. The most commonly used algorithm in VoIP is the CS
structure algebraic
excited linear prediction) algorithm. This

organizes the data packets and streamlines the available bandwidth for the call. This
algorithm is responsible for the transmission rule that was previously mentioned in this
paper that stated “if no one is talking, don’t send any data.” This
rule creates the large
gap in performance for packet switching compared to circuit switching that makes VoIP


Soft Switches

The central call processor that was mentioned earlier is a specially designed database
mapping program known in the
industry as a soft switch. This switch connects the caller
to the destination phone. The caller and the hardware making the call are referred to as
endpoints. The soft switch uses information about the endpoint to connect the call. The
location of the
endpoint, the phone number attached to the endpoint, and the IP address of
the endpoint are all needed to make the connection.

When a call is made via VoIP, the sending endpoint sends a request to the soft switch.
This request is trying to locate the I
P address the call needs to be routed to. If the soft
switch doesn’t contain the information about the needed endpoint, the soft switch passed
the request off to other soft switches until one can find the number needed. When the
correct endpoint is locat
ed, the information about the destination endpoint is sent back to
the sender. Finally, a connection can be made and data can be exchanged between the
two locations.

This soft switch is what allows all the different d
evices that currently support VoIP to

talk to each other. By forcing all VoIP connection to communicate the same way, all
phones, computers, and WiFi phones can work together interchangeably.



Many different protocols exist to manage the way the software and

hardware for VoIP
work together. The most common and most widely used protocol is H.323. This
standard was
created by the International Telecommunication Union (ITU)




use in video conferencing. It provides specifications for real
time, in
videoconferencing, data sharing and audio applications. The H.323 protocol is actually a
collection of previously created protocols designed for other systems.

This c
ollection can
be seen at the bottom of the previous page
. The major flaw with

the H.323 protoc
ol is
that it was not specifically designed for use with VoIP.

Figure 3: Protocols used in the H.323 Protocol Suite

Another major protocol in use in VoIP systems is the Session Initiation Protocol (SIP).
SIP is a smaller, more efficien
t protocol when compared to H.323 as it was created for
use with VoIP.

SIP handles five major parts of the connection process for VoIP: user
location, user capabilities, user availability, call setup, and call handling.

Other protocols
in use today inclu
de Megaco H.248, Media Gateway Control Protocol, Remote Voice
Protocol over IP Specification, Session

Protocol, Simple Gateway
Control Protocol, and Skinny Client Control Protocol.


VoIP Service Types

Voice over Internet Protocol can be

used in several different ways, via

home phone
with an

adapter, a specially designed I
nternet phone, or through


In order to convert home phone service to VoIP, an analog telephone adapter (ATA) is
needed. The ATA changes sounds from an
analog signal to a digital signal. After the
signal is converted, it is sent out over the
nternet to a routing station. Most VoIP service
providers include an ATA free of charge when subscribing to package deals.

IP phones come in two varieties: Ether
net phones and WiFi phones. Ethernet phones
look just like the phones in homes today. These phones eliminate the need for an ATA
by incorporating all the hardware and software the ATA prov
ides allowing

the phone to
plug directly into

router bypassing t
he computer completely. WiFi phones work in a
similar fashion to cell phones, allowing the owner to make a call from anywhere in the
world that has a WiFi hotspot.




Figure 4: Connection of multiple types of VoIP services

computer calls are
the simplest way to make a VoIP call. All it takes is a
microphone, a

speaker, a sound card, and an I
nternet connection, all things most
computers already have. Cable and DSL connections work the best for VoIP calls. The
cost is one of the most appealin
g things about computer
computer calls. There is the
cost for the necessary software for your computer, which is usually very small or
onexistent. Also, the cost of I
nternet provider service is still there. There is no cost for
calls made however, n
o matter the distance, making this service extremely appealing to
those making lots of long distance or international calls. [4]



has several very appealing benefits to

its use. Primarily, the

cost of using VoIP
compared to conventional p
hone systems is considerably lower.
lls from computer to
computer are

basically free a
most long distance calls could be made at almost no cost.
VoIP would have extremely high or no limits on call volumes, making it ideal for use in
the corporate wo

International calls are not free but
compared to regular
international phone companies, the rate is consi
derably better, as shown above.




Table 1: Cost Compassion of Calls Made From the United States

Additionally, most VoIP companies bu
ndle many services together in one low
package. These bundles usually cost less than basic phone service and include several
services that would cost you extra through a regular phone company, such as Soft Phone,
Caller ID, Voicemail, Call Wait

Call Forwarding, and
many others. As an added
benefit, many of these services can be managed online. Soft Phone is a feature that
allows for the owner to turn a PC into a working phone. With this service, calls can be
made or received and the user can
access their voicemail from their computer, making a
laptop into a phone that works anywhere
in the world where there is an I


Finally, with VoIP making connections through the internet, it allows for considerably
higher bandwidth d
ata transfers. This will allow VoIP service providers to offer their
customers streaming movies, televisions, and high access speeds than anything a cellular
phone service provider could offer.


As with any new techno
logy, there are still man
y flaws

in the system that need to be
worked out for VoIP to dominate the phone service market. One of the biggest
advantages that cell phones and home phones have over VoIP continues to be the
availability of 911 emergency services. 911 calls are
fied by the number being
called from which makes it possible for emergency services to locate the source of the
call if the caller cannot give their location. This service is vitally important, but early
VoIP services had no way to locate the caller. Sin
ce VoIP calls are routed and identified
by IP number, it was extremely difficult to route calls to the correct emergency call
center. In
2005, the FCC sent a warning to VoIP providers requiring them to fix the
problem; E911 is now available through most V
oIP service centers but it still experiences
problems routing emergency calls to the correct location.

Another major problem that VoIP has is its dependence on wall power. Current home
phones (not cordless sets) will
still work if the power goes out; VoI
P needs its own stable
power source. With no power, VoIP phones cannot connect to the
nternet and become
better paperweights than phones.




As VoIP works through the internet, it only works as well as the network it is connected
works. Latency issues,
jitter, and packet loss are all issues that can cause phone calls to
become distorted, garbled or lost because of transmission errors. Stability in Internet data
transfers need to be guaranteed before VoIP could replace traditional phones systems
y. Additionally, VoIP is vulnerable to viruses,
worms, and hackers. These
attacks are rare, but the threat still exists. VoIP companies are continually developing
more advanced encryption to counteract these risks.

Many homes have other services that u
se the current phone system to work properly.
Systems such as digital video recorders, digital subscription television services, and home
security systems depend on standard phone lines to provide their services. Currently,
VoIP cannot handle these devic
es. Until VoIP providers figure out a way to provide
accessibility for these products, it may have difficulty getting into many homes.

Other issues include problems with se
nding faxes over VoIP and its

dependence on a
completely separate servic
e (Interne
t Service Providers).

VoIP has no fall back plan if an

goes down and ca
nnot provide service. With no I
nternet service, voice over IP cannot
be used which brings us to the final major flaw in the technology. Currently, the service
will only work on a
high speed connection. There are a lot of places in the world that
don’t have high speed access making VoIP worthless in those regions. This is a prob
that the WiFi


are experiencing

as well since there are still a relatively small
number of hot
spots available for use to make calls.


Current and Projected Use

Currently, VoIP use is increasing rapidly with the additions of new features and the
continual improvement in overall service quality. The New Paradigm Group estimated
that in a stud
y released in 2006 that there were roughly 6 million VoIP users in the
United States. The study stated that it thought the number of VoIP users would climb to
9 million by the
end of 2006 and to 24 million users by the end of 2008. VoIP use
appears to ve
ry difficult to estimate as a user could be using VoIP services in several
different ways or just one way, whether that be just a computer connection or a IP phone.

This increase can largely be attributed to the increased number of service providers
ing VoIP. Additionally, in the last couple of years, AOL, Microsoft, Yahoo!, and
Google have all come out with voice
enabled instant messaging services. The report
listed the above 4 companies in addition to 51 others that were all providing VoIP service

at the end of 2006. The providers listed included traditional phone company
powerhouses like AT&T, Sprint Nextel,
Time Warner,
Verizon, and Charter
Communications in ad
dition to VoIP specific corporations like Skype, Vonage, and VoIP
Inc. With so many op
it’s easily conceivable to see VoIP replacing conventional
phone systems in the years to come. However, one thing that will be holding back its
development is the speed at which high speed
nternet and wireless hotspots are advanced
in coverage are
a. As more and more outlying areas get high speed
nternet, the
possibilities for VoIP will increase as well.






The technology of Voice over Internet Protocol continues to advance at an incredible
speed. It’s something that will obviously
be a huge factor in the 21

century. However,
with the limitations currently slowing it down and the time it will take to convert from
traditional phone systems to VoIP, it will be at least another decade before it may be the
dominate method of communica

Still, if VoIP continues to narrow the gap by
offering better packages at lower prices, that predication could come true in a
considerably shorter amount of time.





] Wikipedia. “Network Voice Protocol”. March 2007


Valdes, Robert. HowStufWor
ks. “How VoIP Works”. May 2001


] “Voice Over IP Protocols”. March 2007

[4] Bell, Michael. “Ways to Use VoIP”. 2007 (

[5] 2007 (

] AT& “AT&T International Rate Finde
r”. 2006

] “International Calling Plan Rates”.

] Qwest. “Qwest OneFlex International Rates”. 2007

] Verizon. “Country Code and International Rate Finder”. 2007

[10] FCC Consumer & Governmental Affairs

Bureau. “Voice Over Internet Protocol”.
June 2006 (

] Wikipedia. “Voice Over IP”. March 2007 (


New Paradigm

Resources Group. ISP
- “Executive Summary, VoIP
Report 3

Edition”. December 2006 (