Audio encoding is the process of converting analog audio signals, like the sound of your voice, into a digital format that can be transmitted over an IP network. This process involves the use of a codec (short for coder-decoder) to compress the audio data, reducing its size to save bandwidth while aiming to preserve call quality. When the data reaches its destination, the same codec is used to decompress and decode it back into an audible signal.

The choice of codec represents a trade-off between audio fidelity (call quality), bandwidth consumption, and the processing power required. Different codecs are optimized for different scenarios, from high-definition voice calls on a stable network to highly compressed audio for low-bandwidth environments.

While many codecs exist, several have become industry standards, particularly those standardized by the International Telecommunication Union (ITU). These codecs are compatible with most modern communication protocols, including SIP and H.323.

Common Audio Codecs Include:

• G.711: This codec is the standard for digital voice transmission over traditional telephone networks (PCM). It offers high-fidelity audio (3 kHz) at a higher bandwidth of 64 kbps, making it a great choice when network capacity is not a concern.
• G.722: Often referred to as "HD Voice," G.722 captures a wider range of audio frequencies (7 kHz), resulting in richer, clearer conversations. It operates at 48, 56, or 64 kbps. Lower-bandwidth variations also exist, such as G.722.1 and G.722.2.
• G.729: A highly efficient codec that provides good voice quality using only 8 kbps of bandwidth. Its combination of low bandwidth usage and reliable quality makes it one of the most popular codecs for VoIP.
• G.723.1 & G.728: These are very-low-bitrate codecs designed for environments where bandwidth is extremely limited. They use advanced compression techniques to operate at bitrates as low as 5.3 kbps (G.723.1) and 16 kbps (G.728).
• Open-Source Codecs: Unlike the patented G-series codecs, open-source options are freely available.
  • Opus: A modern, highly versatile codec that can dynamically adjust its bitrate from 6 kbps to 510 kbps, delivering everything from highly compressed speech to high-fidelity stereo music. It is now the standard for most real-time web communication (WebRTC).
  • Speex: An older, patent-free voice codec designed specifically for VoIP, offering variable bitrate encoding. While still in use, it has largely been succeeded by Opus.
  • GSM 6.10: The codec originally used for GSM mobile phone networks, it encodes at 13.3 kbps and is supported by many software applications.