A codec (Coder/Decoder) is technology that compresses (encodes) audio data for transmission and decompresses (decodes) it on the receiving end. In telephone communications, the codec is a critical factor that directly determines the tradeoff between call quality and required bandwidth, and the choice of codec directly affects how a call sounds.
Representative voice codecs, organized by bandwidth and quality, include the following. G.711 is the standard landline codec, using 64 kbps of bandwidth to cover the 300-3,400 Hz voice band. Quality is good but bandwidth consumption is high, suited for environments with ample capacity. G.729 is widely used for IP phones, requiring only 8 kbps - one-eighth of G.711 - excellent for bandwidth savings but with slightly lower quality. AMR-WB (G.722.2) is the standard VoLTE codec, delivering wideband audio (HD Voice) at 50-7,000 Hz, providing significantly clearer sound than traditional telephony.
Next-generation codecs are gaining traction. EVS (Enhanced Voice Services), developed by 3GPP, supports super-wideband audio (Super HD Voice) at 50-14,400 Hz, capable of transmitting music and ambient sounds in high quality. It is being adopted for 5G VoNR. Opus, developed by the IETF as an open-source codec, operates across a wide range of 6 kbps to 510 kbps, covering everything from voice to full-band music in a single codec. It is the standard codec for WebRTC (browser-based calling).
Codec choice also affects latency. G.711 has simple encoding, adding minimal delay (about 0.125 ms), while G.729's complex compression adds about 15 ms. In VoIP environments, G.711 is typically chosen when bandwidth is ample, and G.729 when bandwidth is limited. Combined with QoS settings, codec selection determines call quality.