In VoIP, a codec which stands for coder‑decoder,
is a software or hardware component that encodes and decodes a
digital data stream or signal. In the context of VoIP, a codec is
used to compress and decompress the audio data that is transmitted
over the internet to reduce the amount of bandwidth required to
transmit the audio signal.
A codec is like a secret code that allows computers to talk
and understand each other over the internet.
It's just like you and your bestie have a secret language
that only the two of you understand, and here, it's a secret
language between two computers.
Different codecs have different ways of compressing the data,
and some are better than others at preserving the quality of
the sound or video while making the file size smaller There are
many different codecs available, each with its own set of tradeoffs.
Some codecs offer high compression ratio and low latency,
but, with a lower audio quality.
Other codecs offer high audio quality,
but with a lower compression ratio and higher latency.
The choice of codecs depend on the specific requirements of
the VoIP application that you're using, such as the desired
audio quality by the users,
the available bandwidth in your network, and the latency requirements.
It is important to note that the codec used in VoIP application must be
supported by both the sending and receiving devices.
This ensures that the audio data can be transmitted
and received without any issues.
Some of the popular codecs are G.711.
This codec is widely used for voice communication over the internet.
It provides high‑quality voice compression and is the standard for
most web communication systems today. G.729, this codec is designed
for low‑bandwidth communication and provides overall good voice call
quality, even at low data rates.
It is commonly used in situations where bandwidth is limited. Then comes G.722.
This codec provides high‑quality wideband audio and is commonly
used for high‑fidelity voice communication.
Opus, this is a new,
highly versatile codec that provides excellent voice quality
and can adapt to varying network conditions.
It is widely used today in modern VoIP systems. SILK, this codec is
designed specifically for use in low‑bandwidth communication,
such as internet‑based voice calls. It provides good call
quality, even in challenging network conditions.
Each codec has its own strengths and weaknesses,
and the choice of codec can have a significant impact on
the quality of voice communication.
It's important to choose the right codec for a particular VoIP application to ensure the best possible voice call quality.