VoIP Codecs for Call Centers: G.711, G.729 and Opus Compared

The common codecs (general VoIP)

• G.711 (PCMU/PCMA) — uncompressed 64 kbit/s audio. Widely supported on PSTN interconnects, good quality, no licensing, highest bandwidth. Often preferred for AI voice and speech recognition because nothing is lost to compression.
• G.729 — compressed 8 kbit/s audio payload. Lower bandwidth, slightly lower quality, historically licensed; less common on modern wholesale routes.
• Opus — modern variable-bitrate codec, excellent quality at low bitrates, common in WebRTC and some AI platforms; frequently transcoded to G.711 at the PSTN edge.

Bandwidth per call

Codec payload is only part of the bandwidth. With 20 ms packets, IP/UDP/RTP headers add roughly 16 kbit/s per direction, so G.711 uses about 80–90 kbit/s per direction per call on Ethernet and G.729 about 25–35 kbit/s. Use the SIP channels and bandwidth calculator to size your connection with headroom.

Transcoding

If your dialer offers a codec the trunk does not, the call fails or someone must transcode. Transcoding adds CPU load and latency and can degrade audio. Offering G.711 as a common baseline avoids most problems.

At CallCenterBoost

Codec support is confirmed per route during route-test review rather than published as a blanket figure. If you have a specific codec requirement (for example for an AI voice platform), state it in the route-test request.