Voip

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1.INTRODUCTION

The public telephone network and the equipment that makes it possible are taken
for granted in most parts of the world. Availability of a telephone and access to a

low cost ,high quality worldwide network is considered to be essential in modern
society .Anything that would jeopardize this is usually since more and more
communication is in digital format and transported via packet networks such as

IP,ATM cells etc. Since data traffic is growing much faster than telephone traffic,

there has been considerable interest in transporting voice over data networks.

Support for voice communication using the Internet Protocol (IP), which is

usually just called “voice over IP” or VoIP, has been become especially attractive
given the cost, flat rate pricing of the public internet. In fact, toll quality telephone

over IP has now become one of the key steps leading to the convergence of the

voice, video, and data communication industries. The feasibility carrying voice
and call signaling messages over the Internet has already been demonstrated but

delivering high quality commercial products, establishing public services, and
convincing users to buy in to the vision are just beginning.

VoIP can be defined as the ability make telephone calls and to send facsimiles or
IP based data networks with suitable quality of service (QoS) and much superior

cost / benefit. Equipments producers see VoIP as a new opportunity to innovate

compete. The challenge for them is turning this vision in to reality by quickly

developing new VoIP enabled equipments. For Internet service providers the


possibility of introducing usage based pricing and increasing their traffic volumes

is very attractive. Users are seeking new types of integrated voice / data
application as well as cost benefits.

Success fully delivering voice over packet networks precedence a tremendous
opportunity; however, implementing the products is not as straight forward a
task as it may first appear.

2. INTERGRATION OF IP & PSTN
Using VoIP we can communicate from PC to any other device – PC, phone, internet

phone, PDA, etc.

1.PC-PC: we can community from one PC to another PC using VoIP i.e. voice mail
or voice chat. For this PC with headphone, a microphone and sound card are

required.

2.PC- Phone: Allows PC user to establish a call with conventional phone. This
facility enables us to make long distance a call with rate cheaper than

conventional telephone call.
3.Phone – Internet Phone: Extension of PC to phone architecture. The IP phone

connects to the Internet through a cable or DSL modem using an RJ-45 Ethernet

jack.
3. PSTN vs. INTERNET.

To understand Internet telephony or VoIP, it is necessary to be familiar with the

fundamental principles behind the Internet and how it compares to PSTN.


Although the Internet shares some characteristics of the PSTN, it also has

significant differences.

Computer to computer data communication was far cheaper than voice

communication through Internet. One of the main reasons for these is the basic
difference in the technologies used for voice and data communication. Voice
networks use circuit switching while data network use packet switching. What

this means is that for a voice conversation to happen, one complete and

continuous circuit has to be held open between the two parties during the full
duration of conversation, while for a data transmission such a continuous connect

is unnecessary. In data transmission, the data is split into independent packets

that can take alternate rules to the intended destination, where they are
reassembled. In voice communication, circuit switching is used because it is able

to handle data in real time. Packet switching, on the other hand, can lead to

delays, which can lead to considerable degradation in the quality of the
conversation.

So the networks when their separate ways till the Real time Transport Protocol

(RTP) was developed. RTP provides support for streaming audio and video over

computer networks. Unlike what its name seems to suggest, RTP does not ensure
real time delivery of data. Instead it provides mechanisms for times stamping the

packets and methods for synchronizing data streams with time properties. Thus

it became technically possible to send voice over the net.

PSTN INTERNET


1.circuit switching 1. packet switching

2.dedicated path between calling and 2. There is no dedicated Called
party. path between Sender and.
receiver

3.Has a guaranteed QoS . 3. Cannot guarantee QoS
4. Reserve required bandwidth in advance 4.It acquires and releases
bandwidth, as it is needed.

5. Cost is based on distance and time. 5.Cost mechanism for

Internet telephony is not
dependent on time and

distance.

4. HOW VoIP WORKS?

In VoIP, for transmitting voice, it is pocketsize and then transmitted through

packet switching networks. This process is described in following steps.
1. Our voice is converted into analog electrical signal using micro phone

2. These electrical signals are digitized using PCM-Pulse Code Modulation PCM
samples analog signal at a rate of 8000 samples/sec and coded into 64

kb/sec. Each sample there fore represents 125 microseconds of a voice

stream, and is 8 bits, or 1byte long.
3. These digital voice samples are buffered on IP gate way- it converts PCM data

into IP packets using DSP. DSP s(Digital Signal Processors ) are responsible

for converting from analog to digital as well as compression. This has

following steps.


a. Remove line echoes: Echo becomes a problem when the round trip delay

is more than 50milli seconds. Since round trip delay for VoIP is always
greater than 50 ms, echo cancellation is a requirement. Therefore line
echoes are cancelled using a digital filter.

b. Silence cancellation using Voice Activity Detector (VAD):It checks the
speaks for all the moments of silence .The length and beginning of the
pauses is noted, while the remaining silence is removed from data set.

Similarly redundant data is also removed, making the data set more

compact so that it takes up much less band width.
c. This digitized voice signals are compressed and framed on the basis of

ITU standards G.729.

d. This voice frame is converted into IP packets. First voice frame is
converted into RTP packet by adding 12 byte header, for sequencing the

data packet. Then 8- byte UDP header with source and destination

number added. Then20-byte IP header containing the gateway IP
addresses added.

4. Voice packet is sending on the Internet, it finds its way to the destination just
like any other data packet. It passes through various routers and switches to

reach the destination gate way.

5. At receiver VoIP system reverses process for voice play back.
System extracts

IP packet- UDP packet – RTP packet-compressed voice frame-analog form of

voice.


Figure showing 5 steps of Internet Telephony


5. VoIP: COMPONENTS

Sender’s end:

The PCM Interface, which receives samples from the telephony interface and

forward s them to the VoIP software module for processing (and vice versa).

The Echo Cancellation Unit, which performs echo cancellation on sampled, full

duplex voice port signals in accordance with the ITU G.165 or G.168 standards.
The Voice Activity Detector, which monitors the input device for device so that the

process of sending it over the network occurs only in case of voice activity at the

mouthpiece.
The Tone Detector, which detects the reception of DTMF tones and discriminate

between voice and facsimile signals

Voice Encoder, which encoded the voice signals from the PCM Interface and sent
over to the host interface that is the gateway to the carrier like Internet Protocol


Comfort Noise Encoder, which measures idle noise level and reported to the

destination so that comfort noise can be inserted into the call (so that the listener
does not get dead air on their telephone).
Voice fax classifier, which separates voice and fax signals, if we are using the

same set up for fax also.
Modem /Fax Demodulator, an optional element for processing fax data.
Receiver’s End:

Comfort Noise Generator, which generates a comfort noise at the receivers end, as

per the data received from the comfort noise encoder at the sender’s end.
Voice Decoder, which decodes the encoded signal.

Tone /DTMF Generator, which generates DTMF tones and call progress tone

under command of the operating system.


Bad Frame Handler, which detects the corrupted packet and may ask for it to be

retransmitted.
6. IMPLEMENTING VoIP.
Most of the VoIP implementation follows the ITU H.323 standard. The H.323

standard was originally a multimedia standard meant for transferring audio and
video data over network.

 Basic element of H.323 architecture

1. The terminals.
2. Gate way

3. Gate keepers

4. Multipoint control unit (MCU)
Out of these, the first two are key elements, while the other two are optional

components.

Terminals- is an end user device. These could be PCs with headphones and a

microphone, or, IP telephone or telephone.


Gateway-is an intermediate device to provide inter operation between H.323

device and non-H.323 device. The gateway can be designed inside a PBX (private
branch exchange) or stand alone device such as a router. One side of this gateway
will be the PSTN or a company’s IP WAN, while the other side would have the

company’s internal network.
Gate keepers – is a piece of software controlling gateways. Simply speaking, a
gatekeeper acts as a routine device for voice calls. They do address translations to

determine the calling terminals. They can also control access given to the other

VOIP devices such as terminals, gateways etc. They determine how much
bandwidth is required for a voice call, and can perform other functions like call

authorization, bandwidth management, call management, and a directory control.

Multipoint control unit (MCU) – This is meant to provide conferencing facilities
between three or more H.323 terminals or gateways.

7. STANDARDS
One important consideration of VoIP software is interoperability. Many Internet

telephony products require all communicating parties to use the same application.
All the vendors begin to support an international can implement this

telecommunications union (ITU-T) standard for multimedia communications

systems known as H.323.It is important standard set of procedures that provide a
specification for voice, data and video communications over packet switched

network (or TCP/IP networks). Using H.323, a device or application from one

vendor can place a call to another. It tries to provide real time audio, video and

data communications by making use of Real time Transport Protocol (RTP) which


was proposed by IETF (Internet Engineering Task Force) as a standards for real

time data transmission over internet .The H.323 recommendation is independent
of network topology.

H.323 offers reduced bandwidth requirement for conversions from 64Kbps/each
to under 10Kbps /each. This is nearly an order of magnitude improvement .net
meeting and Vocal Tec Internet Phone supports H.323. The most widely accepted

standard now is the H.323 set of protocol. However recently Session Limitation

Protocol (SIP) is coming up an alternative for H.323 signaling

8. HOW VOICE GOES OVER IP?

When we talk of sending voice over Internet connections, the first question that
pops up is “how can you do that, when the connections are not even broad enough
to handle regular data?”. Traditionally, voice is a lot of data and if sent as a

regular analog data, it would simply clog the networks and won’t be able to get
through. Instead the voice is sampled, compressed and packetized to send it over
an IP network. This may need much less bandwidth.


The compression technique used in VoIP is similar to the compression concept of
MP3s.Regular music or audio is compressed by checking out the noise, silence and
certain in audible frequencies, so that it takes up much less space.

There was a speech codecs, also called voice coders or ‘vocoders’ are speech
compression algorithms that let us drastically reduce the amount of data that goes

into the network, while still preserving voice quality. The devices that send and

receive audio and video work on the H.323 protocol Vocoders are used
depending on the network conditions. When we talk of sending voice over IP, the

overall realism depends on sound quality and latency. Latency is the delay

between the times; the voice is spoken at the originating end and the time it’s
perceived at the receiving end. There needs to be a trade-off between the two,

which means that if we go in for higher speech quality (less compression), we

might have more latency. So VoIP solutions come with standard vocoders.

The amount of network traffic is not the same at all times. So, VoIP systems are
generally dynamic, that is, they support multiple vocoders with the codec being

automatically switched to match the network conditions, that is, if the network

traffic increases the system switches to a high compression rate vocoders and vice
versa. This however increases the system complexity and resource requirements.

9. LIMITATIONS AND SOLUTIONS

There are various problems associated with VoIP.


1. Packet-delay –The voice packets may get delayed in reaching the

destination. If a router is free, it sends the packet quickly and if it is
under heavy load of traffic the voice packet will get delayed, leading
to latency. The route taken by a packet can also take delay. If it takes

a short path, it will reach quickly. And if it reaches the destination
after the multiple hops the delay would be longer.
2. Packet error & Packet loss. If a voice packet encounters a bad router,

it might get corrupted or get lost all together. If it is the former, the

packet that reaches the destination is of no value. If it gets lost, then
the router may ask for it to be re-transmitted.

Solutions:

1. Traffic priorization-Routers can be configured to give preference to
certain type of packets over others. So voice packet can be given higher

priority over normal data packets.

2 Weighted fair queuing-Here; a minimum amount of bandwidth is
allocated to certain traffic, in this case voice. This can be done using the

Resource Reservation Voice Protocol (RSVP).

10. ADVANTAGES:

1. Cost reduction-Long distance voice telephony is where telcos make most
of their money. But using VoIP we can able to make long distance calls

with cheaper rate.

2. Simplification-an integrated infrastructure that support all form of

communication allows more standardization and reduces the total


equipment complement. This combined infrastructure can support

dynamic bandwidth optimization and fault tolerant design. These
differences between the patterns of voice and data offer further
opportunities for significant efficiency improvement.

3 Consolidation-since people are among the most significant cost elements
in an network, any opportunity to combine operations, to eliminate points
of failure, to consolidate accounting system would be beneficial.

4. Advanced applications-even though basic telephony and facsimile are the

initial applications for VoIP, longer benefits are expected to be derived
from multimedia and multi-service applications.

11. THE DEATH OF DISTANCE
Long distance voice telephony is where telcos make most of their money. The

further away our call is to the more we are charged. This means that we think

twice before making that long distance call, and even when we make the call we
keep the conversation short. Traditional economics says that if the cost of the call

were reduced, the more people would make the call, and if properly done, the
voice service provider could end up making more money than before. So telcos

have extended the distance we can call for the same amount.

When we send mail over the internet we do not pay different rates for different

destinations of the mail .we do not pay more for our internet connection when we

are downloading software from an FTP server in Iceland as against when we are

accessing a web page on a sever in Mumbai. The fact that we can do both


simultaneously, from different windows of our browsing makes a switch over to a

distance-based model impossibly complex. We do not even pay for the amount of
data transferred. We just pay for the bandwidth we opted for and the time we
stayed connected.

so when voice traffic moves over completely to IP based networks, it just possible
that we would pay the same for an international callas we would for a local call . in
short , the concept of long distance calls just disappear .

12. APPLICATON AND SERVICES OF INTERNET TELEPHONY.
1. Integration of data voice and fax.

3. Sound grading.

4. Video telephony.
5. Unified messaging.

6. A virtual second line.

7. Web based call centers.
8. Low cost voice calls.

9. Real time billing.
10. Remote Teleporting.

11. Enhanced teleconferencing.

13. CONCLUSION

With the advances in computer technology, phenomenal growth in Internet use

and declining cost of computer hardware, there has been growing interest in

recent years in developing real time voice communication software for Internet


telephony. In spite of the numerous Internet telephony systems available today,

these are still at infancy and far from being substitutes of conventional telephony
systems. While the Internet telephony may have difficulty matching the reliability
of switched voice networks, it is dirt-cheap and is growing rapidly. Moreover with

advances in compression algorithms, standards, network technology and higher
bandwidth in future, these problems will diminish overtime.

Voip

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