Codec

Coder-decoder algorithm that compresses and decompresses voice data for VoIP transmission

What is a codec?

A codec (coder-decoder) is an algorithm that compresses voice or video data for transmission over a network and decompresses it at the receiving end. In VoIP, the choice of codec directly impacts call quality, bandwidth consumption, and latency.

Common VoIP codecs

Popular codecs include G.711 (uncompressed, toll quality), G.729 (compressed, low bandwidth), and Opus (adaptive, used in WebRTC). Each trades off between bandwidth efficiency and audio fidelity.

Codec negotiation in softphones

During call setup, SIP endpoints negotiate which codec to use via SDP (Session Description Protocol). Modern softphone platforms support multiple codecs and select the optimal one based on network conditions.

What codecs should I support in a business VoIP app?

Core codecs (must-have):

Opus — best all-around. Adaptive bitrate, wideband, handles poor networks well.
G.711 (μ-law/a-law) — universal compatibility, no licensing cost. Use as fallback.
G.722 — HD voice over LAN/Wi-Fi, ideal for office environments.

Optional but common:

G.729 — low bandwidth, useful on mobile data. In Acrobits apps it requires licensing (in-app purchase or post-paid per active user).
iLBC — avoid prioritizing. It can be selected unexpectedly on mobile data even with good signal. Remove it or push it last.

Configuration tips:

Set separate codec priorities for Wi-Fi vs. mobile data.
Opus first on both, G.711 as fallback.
G.729 only when bandwidth is genuinely constrained.
Avoid iLBC unless your PBX requires it.

One thing operators often miss: codec order in provisioning is your offered SDP priority. If your PBX only supports G.729 and it is not in the app's SDP, calls will fail with 488 Unacceptable Here. Align both sides.