Voice codec
What is Voice codec?
A voice codec (coder-decoder) is the algorithm that compresses analog voice into a digital format for transmission over a network and decompresses it back to audio at the other end, trading off audio quality, bandwidth usage, and processing overhead differently depending on which codec is used. G.711 is the classic, uncompressed toll-quality codec used across the PSTN and most SIP trunks; Opus and G.722 are newer wideband codecs that deliver noticeably clearer audio at similar or lower bandwidth.
More detail
G.711 (PCM) uses no compression and needs about 64 kbps per call, making it the safe, universally compatible default for interop with the PSTN and legacy carriers.
Wideband codecs like Opus and G.722 capture a broader frequency range than G.711, which is what makes calls sound noticeably clearer and more natural — often called HD voice.
Frequently asked
- Why do some calls sound clearer than others on the same network?
- Audio clarity depends heavily on which voice codec both endpoints negotiate — a wideband codec like Opus or G.722 captures a broader frequency range and sounds noticeably clearer than the narrowband G.711 codec still used across most of the PSTN.
- Does a better codec always mean more bandwidth used?
- Not necessarily — Opus, for example, can deliver clearer audio than G.711 while using similar or even less bandwidth, since modern codecs compress more efficiently rather than simply using a higher bitrate.
See also
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