Audio Compression and Codec Selection in FreePBX
by Provider Adminlast verified 2026-08-13
Overview
Learn how audio compression works in VoIP and how codec selection impacts call quality, bandwidth consumption, and server performance in FreePBX. This guide explains the differences between narrowband, wideband, and adaptive codecs like G.711, G.729, G.722, and Opus. You will also learn how to configure codec priorities and minimize transcoding CPU overhead on your system.
Key takeaways
- Audio compression reduces network bandwidth usage at the cost of increased CPU usage for transcoding.
- Uncompressed narrowband codecs like G.711 offer low latency and low CPU overhead but require ~87 kbps per call including overhead.
- Wideband codecs like G.722 provide HD audio quality without increasing bandwidth beyond standard G.711 rates.
- Opus is an adaptive codec ideal for mobile and remote endpoints over unpredictable network connections.
- Matching codec preferences across endpoints and SIP trunks prevents resource-heavy Asterisk transcoding.
Prerequisites
- Administrative access to the FreePBX Web GUI
- Basic understanding of SIP signaling and RTP media streams
- Knowledge of local network bandwidth limits and WAN constraints
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