Telephony VoIP IP

The basic principle of Voip is very simple. It's the same

technology you have probably used already to listen to

music over the Internet. Voice sounds are picked up by a

microphone and digitized by the sound card. The sounds are

then converted to a compressed form, compact enough to be

sent in real time over the Internet, using a software

driver called a codec. The term codec is short

for "encoder/decoder". The sounds are encoded at the

sending end, sent over the Internet and then decoded at the

receiving end, where they are played back over the

speakers. The only requirements are a connection between

the two computers of an adequate speed, and matching codecs

at each end.

To be usable, a Voip system also needs a method for

establishing and managing a connection, for example,

calling the other computer, finding out if they accept the

call, and closing the connection when a user hangs up.

Because Voip allows two way communication, and even

conference calls, it's a lot more complicated than simple

audio streaming. How calls are managed is the area in which

Voip systems fundamentally differ, and two Voip users must

be using the same system (or compatible ones) in order to

be able to call each other.

Because most Internet users don't have a permanent Internet

address (IP address, a number like 212.44.88.17 that

uniquely identifies that computer, at that moment), Voip

systems don't generally work by calling another computer

direct  although that may be an option for those who do

have a permanent address. Instead, each user of the service

registers with an intermediate server, which maintains a

record of their IP address all the time they are connected.

An example of a Voip application that works this way is

Picophone. The small size of the PicoPhone application file

(about 64Kb, barely larger than Windows Notepad)

demonstrates clearly that the basic principles of Voip are

not complicated to implement.

Another reason for using an intermediate server is that it

eases the problem of getting Voip to work through the

firewalls that everyone uses these days. Many firewalls

block any data from the Internet that is not sent in

response to a specific request. This makes it impossible to

call another computer direct. Because the called computer

did not request any data from the caller, the call request

would be blocked. By establishing a connection with a

server, the Voip software opens a channel of communication

through which other computers can call it. Communication

may continue using the server, or information may be passed

via the server that allows the two computers to open a

direct connection between them and continue using that.

The first thing all mean the same thing. Which is using IP

(Internet protocol) for voice services. Some voice networks

are only packet-switched and have no access outside of

their own VoIP network. Most VoIP networks have a Gateway

that connects to a circuit-switched external network which

gives them acces to external calling. One of the gateways

responsibilites is to convert G.711 Circuit-switched media

(typically a T1 provided by a telco company) to the 7.723

Packet-switched media that will traverse the companies VoIP

network. A device called a gatekeeper will then convert the

IP address (used by H.323 protocol) to a standard telephone

number (E.164 address) that can be used for external

calling.

A converged network is a network that passes both Voice and

Data over the same set of devices. Converged networks

generally implement QoS (Quality of service) on all actived

network devices to ensure the VoIP has priority over

standard data because of it's more rigid demands.

The big advantage is VoIP may save you money depending on

how much you are currently spending for local and long-

distance calls. What you will need to do is get the total

cost the phone company is charging and compare it against a

VoIP plan that interests you. With most plans, you get free

calls within the U.S. and Canada for a low flat rate.

International calls usually have very low rates with no

connection fees. For both residential customers and

businesses that make a lot of long distance and

international calls, the savings can be several hundred

dollars a year.

Another advantage is with the features available with VoIP.

Features such as caller ID, call waiting, call forwarding,

3 way conferencing and voice mail are usually included at

no extra cost. With the phone company, these services are

usually extra.

In addition, you can make free phone calls anywhere there

is a high speed Internet connection available. That means

you can be in another state or even in another country and

make calls as if you were back at your home or business.

You will just need to bring your phone adapter along with

you and possibly a phone in case one is not available

To understand how VoIP works, you will be taken through the

process of voice transmission from one end to the other.

The process starts with a person talking into the

mouthpiece on one end of a VoIP call.

This analog voice signal must first be sampled and

digitized. Voice sampling is usually done 8,000 times per

second (8KHz). In order to reduce bandwidth, a voice CODEC

is used. A voice CODEC is a compression/decompression

algorithm that is optimized for the voice frequency range.

The bit stream uncompressed is 64Kbps. By using an

available CODEC, the bit stream can be reduced to 8Kbps or

less.

In order for the compressed voice data to be sent over the

Internet, it must go through a process called

packetization. This is a packet consisting of a small

sample of the voice data (usually 10-30 milliseconds).

While being routed through the Internet, these packets can

get delayed or even lost. This can cause degradation in

voice quality. Simply put, there are various mechanisms in

place to compensate for these problems and help smooth out

the audio.

Once all the packets arrive on the listening end of the

call, they must be reassembled to their original state. The

packets are decompressed and converted from a digital to

analog voice signal.