Glossary
Codec
Audio compression format used by VoIP phone systems to package voice for transmission over a network.
also known as: Audio codec, Voice codec
A codec (short for “coder-decoder”) is the audio compression format that VoIP phone systems use to package voice for transmission over a network.
The main codecs in business VoIP
G.711 - uncompressed PCM audio. Highest quality (MOS up to 4.4) but uses ~64 Kbps per call. Standard for high-quality calls within an office where bandwidth isn’t a constraint.
G.729 - compressed at ~8 Kbps per call. Audible degradation vs G.711 but acceptable (MOS ~3.9). Useful for low-bandwidth connections or where many concurrent calls must fit on a constrained link.
Opus - modern adaptive codec used by 3CX, most cloud PBXs, and WebRTC. Scales from ~6 Kbps (G.729-like) to ~510 Kbps (better than uncompressed). Adapts dynamically to network conditions. Default for 3CX in modern deployments.
iLBC - niche, designed for lossy mobile networks. Rare in 2026.
G.722 - wideband audio (HD voice). Uses ~64 Kbps but with double the audio bandwidth of G.711. Common for premium calls within an office.
How codec choice happens
Two phone-system endpoints negotiate the best mutually-supported codec at call setup:
- Caller’s phone offers a list of codecs it supports (typically Opus, G.711, G.729).
- Receiver’s phone responds with its preferred codec from that list.
- The call proceeds using the negotiated codec.
If the network degrades mid-call, modern codecs (Opus) can adapt without dropping the call.
Codec and call quality (MOS)
Codec choice is one of the inputs to [[mos|MOS scoring]]. G.711 has the highest ceiling; G.729 lower; Opus widest range.
For business VoIP on modern Australian broadband, Opus or G.711 is standard. G.729 is rare except for very low-bandwidth scenarios.
Codec and bandwidth planning
A rough calculation for SIP-trunk bandwidth:
- G.711: ~64 Kbps payload + headers = ~85 Kbps per concurrent call
- G.729: ~8 Kbps payload + headers = ~32 Kbps per concurrent call
- Opus (default): ~25–40 Kbps per concurrent call
A 16-channel SIP trunk on Opus uses up to ~640 Kbps peak. Comfortable on any modern NBN connection.
See also
- [[sip]] - signalling protocol that negotiates codec
- [[rtp]] - protocol that carries the codec audio
- [[mos]] - quality score affected by codec choice
- [[sip-trunking]] - codec choice matters for trunk bandwidth
