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.