Tell me How does VoIP work?
Submitted by: AdministratorTo 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.
Submitted by: Administrator
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.
Submitted by: Administrator
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