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The word codec is a portmanteau of two terms: Compression and Decompression. A VoIP codec is a technology that determines the audio quality, bandwidth, and compression of VoIP phone calls. Codec’s aren’t just used for VoIP, they are used in all sorts of activities such as image capture (eg GIF / JPEG / PNG), or music (eg MP3). Without them files would be a lot larger in raw form. In the case of a VoIP codec, it converts analog voice signals into digital packets or a compressed digital form for transmission, and then back into an uncompressed audio signal. The success of a codec depends on its compatibility with different devices. With this in mind, it may be necessary to try different codec to find the one that meets your circumstances.

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Terminology

It’s worthwhile explaining some of the terminology associated with audio quality: Bitrate
Refers to the audio quality of the stream. It is measured in Kilobits Per Sec(kbps or k). Bitrate is the number of bits (data) encoded per second or the number of bits transmitted or received per second. Higher the bit rate with more sampling rate, requires high bandwidth and produces good audio quality. Low bit rates refer to smaller file size and less bandwidth with a drop in audio quality. For good quality music usually 64-128kbps(96kbps+ recommended) bit rate is preferred. Sample Rate
Also known as sample frequency, it refers to the audio samples taken per second. Every individual sample will tell you the signal waveform’s total amplitude value over a specific period. The higher the sample rate, the better the audio quality. The human hearing bandwidth is 20Hz-20kHz. Bandwidth
Is the speed that you can send data or receive the data. It depends upon the bit rate at which the data is sent or received. The higher the bitrate the higher the bandwidth consumed. For VoIP its important to consider the bandwidth being consumed when simultaneous calls are taking place alongside other demands for bandwidth. It may be necessary to ring-fence minimum bandwidth allowances for VoIP or improving bandwidth availability. Phone Call Quality
Phone audio typically involves two bands: narrowband and wideband. Narrowband covers audio frequencies ranging between 300Hz and 3400Hz. For audio between 50Hz to 7000Hz, those are considered wideband. The Codec you use impacts the frequency range which is picked up and encoded, you’ll be able to hear an expanded range of pitches, most closely resembling an in-person conversation. Using G.722 as opposed to G.711 widens the frequency range and improves the quality of the call. Codecs which have a wider signal range are sometimes referred to as HD Voice.

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Opus Codec - What makes it special?

  • It is adaptive meaning it can span low-bandwidth call applications through to CD quality. It is a super wide band signal. Whilst G.722 is considered a HD call, if bandwidth falls, it can result in a poor quality call.
  • If G.722 was comparable with HD television, think of Opus at the high-end as 4k/UltraHD.
  • It’s not without its drawbacks, the processing power required to handle Opus can be high, rendering it’s ability to function correctly restricted to your choice of handset.