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VoIP: Talk To Me!

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DQC Bureau
New Update

The ad campaign for a particular toothpaste brand claims that all the

beautiful girls in this world will get attracted to you and will say "Talk

to me!" if you use this toothpaste. Now, there is another technology coming

your way that will soon make everyone say "Talk to me!". That is Voice

over Internet Protocol (VoIP).

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VoIP, as the name suggests, carries voice signals over the

Internet Protocol, which was initially designed to carry only digital data. It

can be seen as the ability to make telephone calls over IP-based data networks

with a suitable quality of service (QoS) and a much superior cost/benefit.

Equipment manufacturers are looking at VoIP as a new opportunity to innovate and

compete.

Telecom equipment manufacturers will try to build VoIP-enabled

equipment. Networking equipment manufacturers will try to build new equipment,

which can convert existing data networks into integrated voice and data

networks. The race can be won by quickly developing new VoIP-enabled equipment

that will do the job with minimum of rewiring. Successfully delivering voice

over data networks presents a tremendous opportunity.

However, implementing the products is not as straightforward

a task as it may first appear. The entire solution involves setting up the

infrastructure, software, and systems that will be necessary to realize VoIP on

a large scale. Product development challenges such as ensuring interoperability,

scalability, and cost/effectiveness are very important.

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Critical to success is the ability to deploy value-added and

high-margin services. For example, you could deploy a unified messaging system

that would voice synthesize e-mails over a phone to the subscriber.

In the future, there will be telephony-aware applications

that are integrated into other business applications. One example of a

telephony-aware application is "click to call". The tools and the

communication devices we use will change when IP telephony is fully deployed.

Though VoIP is still new to India, packet-based telephony is

becoming more advanced all over the world. Voice protocols have further

developed to offer a richer set of features, scalability and standardization

than what was available a few years ago.

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By driving voice and data traffic through the same pipes,

annual telephone bills can be dramatically lowered, by as much as 90 percent.

Payback time, depending on call volumes and network configurations, can be as

short as six months. From then on, savings can continue long into the future.

VoIP applications

There are basically three types of Voice over IP calls–PC

to PC, PC to Phone, Phone to Phone. Take for example, a company that already has

an IP network in place at locations in Bombay and Los Angeles. By placing a

voice gateway between the network routers and EPBXs at each site, voice traffic

can piggyback the WAN link between these cities.

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As a result, expensive international calls can now be made

almost for free. Alternatively, if no WAN link exists, router-based Internet

connectivity can be used to make the voice calls over the Internet.

TCP/IP (transmission control protocol/Internet protocol) is

the basic communication protocol of the Internet. It can also be used as a

communications protocol in private networks like intranets and extranets. When

you set up a direct access to the Internet, communications take place over

TCP/IP. TCP/IP is a two-layered program.

The higher layer, TCP, manages the assembling of a message or

file into smaller packets that are transmitted over the Internet and received by

a TCP layer that reassembles the packets into the original message. The lower

layer, IP, handles the address part of each packet so that it gets to the right

destination.

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Each router/gateway on the network checks this address to see

where to forward the message. Even though some packets from the same message are

routed differently than others, they will be reassembled at the destination. The

same process can carry packetized voice.

Voice quality

Voice quality is the first thing to get affected in VoIP.

However a delay between two end-points in one direction of less than 150

millisecond is considered an excellent quality level for voice. Delay can create

two potential problems.

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The first is that long delays in conversation cause the talk

from two parties to overlap. The receiver may start talking, believing that the

sender has stopped talking, when in fact he has not.

The second problem is echo, or the reflection of the original

signal back to the sender. Echo is common in all voice transmission, but is

mostly unnoticed under low delay conditions. It becomes more noticeable when the

delay becomes too large. Echo chancellors are necessary to remove echo from such

conversations.

The most important standard specification for VoIP is the

H.323 specification, approved in 1996 by the International Telecommunications

Union (ITU). H.323 has been the de facto standard for interoperability between

different products for real-time communication over IP. The real-time

communication can be voice, video and other multimedia services.

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VoIP components

Terminal is a PC or a standalone device running an H.323

protocol and the multimedia applications. A terminal supports audio

communications and can optionally support video or data communications.

The primary goal of H.323 is to interwork with other

multimedia terminals. Because the basic service provided by an H.323 terminal is

audio communications, a H.323 terminal plays a key role in IP-telephony

services.

VoIP gateway is a product that delivers data and voice over

an IP network. Gateway takes traditional telephony traffic, compresses it, and

then places the compressed information into an IP packet and "routes"

this into the IP network. Gateways adapt traditional telephony to the Internet.

Data packets are not sensitive to delays but voice packets are.

The routers must recognize the voice packets and handle them

on priority, so that they arrive at the receiver at a consistent rate. Further,

the sender and receiver must have VoIP equipment that are standardized and

compatible with each other.

A gateway connects two dissimilar networks. An H.323 gateway

provides connectivity between an H.323 network and a non-H.323 network. This

connectivity of dissimilar networks is achieved by translating protocols for

call-setup and release, converting media formats between different networks, and

transferring information between different networks connected by the gateway. A

gateway is not required however for communication between two terminals on an

H.323 network.

The gatekeeper can be considered the brain of H.323 network.

It is the focal point for all calls within the H.323 network. Although they are

not mandatory, they provide important services like address translation,

admission control, bandwidth management, zone-management and call-routing

services.

Multipoint control units (MCU) provides support for

conferences between three or more H.323 terminals. All terminals participating

in the conference establish a connection with the MCU. The MCU manages

conference resources, negotiates between terminals for the purpose of

determining the audio or video coder/decoder to use, and may handle the media.

The impact of VoIP

The impact of VoIP can be felt most in area of teleworking.

As the WTC attacks of September 11 in the US have highlighted, teleworking can

be an effective answer to avoid concentrated presence of highly specialized

manpower in a few high-rise buildings.

People are the biggest and most important asset a business

has, and teleworking can maximize the return on this investment. An IP-based

teleworking infrastructure with VoIP capabilities can let a team of individuals

work together effectively irrespective of their geographical location.

Ashok Dongre

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