Codec
Software that compresses and decompresses audio data for transmission over the internet — determines VoIP call quality and bandwidth usage.
A codec (coder-decoder) is an algorithm that converts analog audio — your voice — into compressed digital data for transmission over an IP network, and then decompresses it back into audio at the receiving end. In VoIP, the codec chosen for a call directly determines two things: the audio quality the caller hears, and how much internet bandwidth the call consumes.
Different codecs make different tradeoffs between quality and compression. G.711 (also called PCMU or PCMA) is the gold standard for audio quality — it's the same codec used in traditional phone networks — but uses the most bandwidth (64 Kbps per call). G.729 compresses more aggressively, using only 8 Kbps, but introduces some audio quality reduction. Opus, the modern standard used by WebRTC, dynamically adjusts quality and compression based on available bandwidth.
For most businesses on modern broadband connections, codec choice is handled automatically by the VoIP platform and doesn't require manual configuration. However, in bandwidth-constrained environments — remote offices with slow connections, or deployments with many simultaneous calls — selecting a more efficient codec can prevent call quality degradation. Zonitel automatically negotiates the best codec for each call based on network conditions.