VOIP Codecs Explained

A codec, which stands for coder-decoder, converts an audio signal into a compressed digital form for transmission and then back into an uncompressed audio signal for replay. This is the essence of VoIP. Digital-to-analog conversion is seen in everything from CD players to cell phones to video game consoles.

Codecs accomplish the conversion by sampling the audio signal several thousand times per second. For instance, a G.711 codec samples the audio 64,000 times a second. It converts each tiny sample into digitized data and compresses it for transmission. When the 64,000 samples are reassembled, the pieces of audio missing between each sample are so small that to the human ear, it sounds like one continuous second of audio signal. There are different sampling rates in VoIP depending on the codec being used:

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• 64,000 times per second      G.711 codec
• 32,000 times per second
• 8,000 times per second      G.729A codec
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A G.729A codec has a sampling rate of 8,000 times per second and is the most commonly used codec in VoIP. It is a compromised balance between sound quality and efficiency of bandwidth.

Codecs operate by using advanced algorithms that help them sample, sort, compress and packetize audio data. The CS-ACELP algorithm (CS-ACELP = conjugate-structure algebraic-code-excited linear prediction) is one of the most prevalent algorithms in VoIP. CS-ACELP helps to organize and streamline the available bandwidth. Annex B is an aspect of CS-ACELP that creates the transmission rule, which basically states "if no one is talking, don't send any data." As we learned before, the efficiency created by this rule is one of the greatest ways in which packet switching is superior to circuit switching. It is Annex B in the CS-ACELP algorithm that is responsible for that aspect of the VoIP call.

So the codec works with the algorithm to convert and sort everything out, but none of that is any good without knowing where to send the data. In VoIP, that task is handled by soft switches.