What is VOIP:
used by a single call through out the
Voice over IP (VoIP) is a transmission
technology for transfer of voice
communications over IP networks such as
the Internet or other packet-switched
networks.
Its a communication service including voice,
voice-messaging(voice mail) applications
that are transported via the Internet, rather
than the public switched telephone
network (PSTN) that are typical analog
telephone lines. Interconnected VoIP
services also allow you to make and receive
calls to and from traditional landline
numbers.
The basic steps involved in originating an
Internet telephone call are conversion of
the analog voice signal to digital format and
placing them into Internet protocol (IP)
packets for transmission over the Internet;
the process is reversed at the receiving end.
Why take all this trouble:
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.
Comparison of circuit switched
phone system and VOIP:
circuit switching
circuit switching takes a lot of dedicated
resources that are mostly wasted. first is
the overhead of establishing a dedicated
path: involving dedicated channels on every
medium in that path and creating a
connection state in every intermediate
switching node being traversed. the
dedicated bandwidth is never shared and is
duration of a conversation that makes CS
very inefficient. dedicated resources makes
CS very expensive. Next there are
limitations of using the same signaling rate
on both ends. in terms of resilience
complex algorithms are needed if the
dedicated path breaks, that might result in
call being dropped. in the current digital
infrastructure merging analog CS with the
digital equipment also poses a great
challenge. since the circuit is continuously
open resources are wasted since most of
the time there is nothing to send
VoIP
VOIP uses packet switching: instead of
continuous circuit data is transmitted when
required. No channels or other resources
are dedicated. Overhead of establishing a
circuit and switch configuration is
eliminated switches just forward the
packets. The data medium is shared
between packets from various applications.
So bandwidth of a single CS call can be
utilized by many VoIP calls. The network
routes the packets along the least
congested and cheapest lines and in terms
of resilience nodes or link failures will cause
packets to take alternative routes and the
call might be interrupted but will not be
dropped.
How VoIP works:
VoIP systems employ session control
protocols to control the set-up and tear-
down of calls as well as audio codecs which
encode speech allowing transmission over
an IP network.
Codecs
Codecs normally work at the application
layer.voip softwares normally employ
codecs.
Codecs use advanced algorithms to help
sample, sort, compress and packetize audio
data
Codecs simply code/digitize the analog
voice signals using PCM etc,that might
further be compressed depending on the
voice quality required and sampling
frequency can be varied accordingly the
PCM data is then framed and sent in an ip
packet.on the reciever’s side PCM samples
are torn off the ip packets and
reconstructed into analog voice again.
Codec used is varied between different
implementations of VoIP , some
implementations rely on narrowband and
compressed speech, while others support
high quality stereo codecs.use of codecs
largely depends on the connection the
available bandwidth and the quality of the
line.
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.
G.711 codec samples the audio at 64,000
times a second
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.
There are three variations of VoIP
service in common use today:
ATA -- (analog telephone adaptor). The
ATA is a device 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 calls.
IP Phones -- These special phones look
just like normal phones. 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.
Computer-to-computer -- This is
certainly the easiest way to use VoIP. There
are several companies like
Skype/Google/yahoo 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.
The main attraction is that we are charged
for the internet connection, not for the time
or the distance of our call.
Making a VOIP Call
if both end users have service through a
VoIP provider. Both should have their
analog phones hooked up to the service-
provided ATAs. Let's take a look at that
typical telephone call, using VoIP over a
packet-switched network:
You pick up the receiver, which sends a
signal to the ATA.
The ATA receives the signal and sends a dial
tone. This lets you know that you have a
connection to the “Internet”.
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.
The phone number data is sent in the form
of a request to your VoIP Company’s call
processor(softswitch).
The call processor determines to whom to
map the phone number. In mapping, the
phone number is translated to an “IP
address”.
The soft switch connects the two devices
on either ends.
On the other end, a signal is sent to your
friend's ATA, telling it to ask the connected
phone to ring.
Once your friend picks up the phone, a
session is established between your
computer and your friend's computer.
In the middle, the normal Internet
infrastructure routes the voice carrying
packets. Each system must use the same
protocol to communicate. The systems
implement two virtual channels, one for
each direction, as part of the session.
When you hang up, the circuit is closed
between your phone and the ATA.
The ATA sends a signal to the soft switch
connecting the call, terminating the session.
Routing in VOIP
Softswitch and Central Call
Processor
The central call processor is a piece of
hardware running a specialized
database/mapping program called a soft
switch. The soft switch connects voip users.
the routing issues
in voip each user has a phone number that
is fixed but since users connect through
internet they at any given time have
random ips. call forwarding/routing is area
dependant(country /city/ area codes) but
current ip addresses of a voip users is hard
to locate.
The challenge with VoIP is that IP-based
networks don't understand phone numbers.
They look for IP addresses.
Soft switches know:
Where the network's endpoint is
What phone number is associated with that
endpoint
The endpoint's current IP address .
Once the soft switch needs to find out the
current ip address of a dialed user. then
packet is left at the mercy of ip network to
route the packet to the desired host .
Protocols it.for example one can control his computer
Voice over IP has been implemented in and other devices remotely via voice
various ways using both proprietary and commands that are interpreted at the other
open protocols and standards. Examples of end in a digital form.
technologies used to implement Voice over
Most VoIP companies provide the features
IP include:
that normal phone companies charge extra
H.323
for when they are added to your service
IP Multimedia Subsystem (IMS)
plan. VoIP includes:
Media Gateway Control Protocol (MGCP)
Session Initiation Protocol (SIP)
Caller ID
In voice, the protocol performs basic call-
Call waiting
control tasks such as session set up and tear
Call transfer
down, or the signaling for call initiation, dial
Repeat dial
tone and termination. SIP also controls
Return call
other signaling for features such as hold,
Three-way calling
caller ID and call transferring. Its functions
call filtering options available.
are similar to the Signaling System 7
Forward the call to a particular number
protocol in standard telephony andH.323 or
Send the call directly to voice mail
Media Gateway Control Protocol in IP
Give the caller a busy signal
telephony.
Play a "not-in-service" message
Real-time Transport Protocol (RTP) audio conferencing:
The Session Initiation Protocol has gained
The voip service can be merged with video
widespread VoIP market penetration, while
conferencing to allow the people tele
H.323 deployments are increasingly limited
presence,with people talking face to face.
to carrying existing long-haul network
traffic. [Citation needed]
A notable proprietary implementation is the
Skype protocol. Requirements and Issues:
The issues concerning Packet switched
Features networks also affect call quality:
There is a lot of flexibility in using voip not Voip requires a constant flow of packets so
like normal phone calls where we are that the reconstructed voice signls are
restricted to just talk and exchange voice replica of sampled original voice signals.
signals,in voip we are playing with digital
information we can use it the way we want Bandwidth
voip is heavily integrated with other digital
VOIP needs a certain amount of bandwidth
services we can even invent our own
to be usable. VOIP needs bandwidth so that
features for example a person can make an
voice conversations are intelligible by both
encryption program that secures PCM data
parties.that means we cant run BW hungry
.we can mould the calls the way we want
apps while using VOIP, BW variations can
also cause distortion of the call.
codecs
improper sampling can cause Voice
cutoffs,echoing and can reduce intelligibility
of voice.
Latency
the greater the distance between calling
parties, the greater the latency. That
means the time one speaks to the time
other listens is dependant on the time it
takes for packets to go from one endpoint
to other. Greater the latency less is
understandability and quality of
conversation. Latency can also be caused
due to heavy traffic congestion in the
network
Jitter
jitter can also affect speech quality; jitter
is the variable latency between packets.
Jitter is more common in IP-based speech
because the path for voice packets across
the network may not always follow the
same route. The buffers commonly used
in IP networks can also increase packet-
induced jitter
Packet loss
Another contributing factor can result from
packet loss or discard somewhere
between the calling parties. packet discard
can "throw away" a "lot of speech" as
opposed to an uncompressed sound
wave.The more highly compressed the
voice packet, the greater the amount of
conversation lost when a packet is
discarded
Phone conversations can become distorted,
garbled or lost because of transmission
errors.
Threats
While VoIP vulnerabilities are typically
similar to the ones users face on the
internet, new threats, scams, and attacks
unique to IP telephony are now emerging.
VoIP is also susceptible to worms, viruses
and hacking
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.
PSTN VOIP Integration:
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.
On a user level a PSTN user’s call can be
routed to an ip gateway through circuit
switching where devices such as Soft
switches can route the call to the intended
VOIP through a packet switched network by
digitizing the Circuit switched analog data.
Disadvantages of Using VoIP
PSTN may lack efficiency but it is more
reliabile. 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 be one of the
major flaws in VoIP: reliability.
power problem
First of all, VoIP is dependant on wall
power. phone runs on phantom power that
is provided over the line from the central
office. Even if your power goes out, phone
still works. With VoIP, no power means no
phone. A stable power source must be
created for VoIP.
Emergency Calls
Emergency 911 calls also become a
challenge with VoIP. VoIP uses IP-
addressed 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.
Lack of redundancy
With the current separation of the Internet
and the PSTN, a certain amount of
redundancy is provided. An Internet outage
does not necessarily mean that a voice
communication outage will occur
simultaneously, allowing individuals to call
for emergency services and many
businesses to continue to operate normally.
In situations where telephone services
become completely reliant on the Internet
infrastructure, a single-point failure can
isolate communities from all
communication, including Enhanced 911
and equivalent services in other locales.
Benefits
Perhaps the biggest draws to VoIP for the
home users that are making the switch are
price and flexibility.
Operational cost
VoIP can be a benefit for reducing
communication and infrastructure costs.
Examples include:
one network:Routing phone calls over
existing data networks to avoid the need for
separate voice and data networks.
Costs are lower, mainly because of the way
Internet access is billed compared to
regular telephone calls. While regular
telephone calls are billed by the minute or
second, VoIP calls are billed per megabyte
(MB). In other words, VoIP calls are billed
per amount of information (data) sent over
the Internet and not according to the time
connected to the telephone network. In
Flexibility
VoIP can facilitate tasks and provide
services that may be more difficult to
implement using the PSTN. Examples
include:
Multiplexing(TDM)The ability to transmit
more than one telephone call over a single
broadband connection without the need to
add extra lines.
Secure calls using security protocols . Data
can be encrypted and authenticated.
Location independence. Only a sufficiently
fast and stable Internet connection is
needed to get a connection from anywhere
to a VoIP provider.

In the future as internet becomes a more

reliable means of data communication
voipo may take over the PSTN networks
entirely.