VoIP Audio Codecs Explained: G.711, G.722 and Opus

What Are VoIP Audio Codecs?

A codec (coder-decoder) is a technology that encodes and decodes audio for transmission over a network. When you make a VoIP call, the codec on your phone converts your voice into digital data packets, sends them across the internet, and the codec on the receiving end converts them back into audible sound.

The codec you use directly affects two critical factors: call quality and bandwidth consumption. Choosing the right codec for your business ensures crystal-clear calls without overwhelming your internet connection.

Common VoIP Codecs

G.711 — The Standard Codec

G.711 is the most widely used VoIP codec and the default on most SIP trunks and IP phones. It comes in two variants:

G.711a (A-law) — used in the UK, Europe and most of the world
G.711u (μ-law) — used in North America and Japan

Key characteristics:

Bandwidth: 64 kbps per call (plus overhead, approximately 87 kbps total)
Quality: Toll-quality audio, equivalent to a traditional phone line
Latency: Very low processing delay
Compatibility: Supported by virtually every VoIP device and provider

G.711 is the safest choice for most businesses. It offers reliable, consistent audio quality and is universally compatible. The trade-off is higher bandwidth usage compared to compressed codecs.

G.722 — HD Voice

G.722 is a wideband codec that delivers HD (High Definition) voice. It samples audio at a higher frequency range (50–7000 Hz compared to G.711's 300–3400 Hz), resulting in noticeably richer, clearer audio.

Bandwidth: 64 kbps per call (same as G.711)
Quality: HD wideband audio — significantly clearer than G.711
Latency: Low processing delay
Compatibility: Supported by most modern IP phones and SIP providers

G.722 is an excellent choice when both endpoints support it. You get HD voice quality at the same bandwidth as G.711. Most modern IP phones from Yealink, Poly and Cisco support G.722 natively.

G.729 — Low Bandwidth

G.729 is a compressed codec designed for environments where bandwidth is limited. It achieves dramatic bandwidth savings by using advanced compression algorithms.

Bandwidth: 8 kbps per call (approximately 31 kbps with overhead)
Quality: Good but noticeably lower than G.711 — slightly compressed sound
Latency: Higher processing delay due to compression
Compatibility: Widely supported, but some providers charge extra for G.729 channels

G.729 is ideal when bandwidth is severely constrained, such as over satellite links or very slow broadband connections. For most UK businesses with standard broadband, G.711 or G.722 is a better choice.

Opus — The Modern Adaptive Codec

Opus is a modern, open-source codec that can dynamically adapt its bitrate based on network conditions. It supports both narrowband and wideband audio.

Bandwidth: 6–510 kbps (adaptive)
Quality: Excellent — can match or exceed G.722 quality
Latency: Very low, designed for real-time communication
Compatibility: Growing support, used by WebRTC and modern softphones

Opus is increasingly popular in modern VoIP platforms and WebRTC applications. Its ability to adapt to changing network conditions makes it particularly resilient on variable-quality connections.

Codec Comparison Table

Here is a quick comparison of bandwidth requirements per concurrent call:

G.711 — 87 kbps per call — toll quality — universal compatibility
G.722 — 87 kbps per call — HD quality — wide compatibility
G.729 — 31 kbps per call — good quality — wide compatibility
Opus — 10–120 kbps per call — excellent quality — growing compatibility

Which Codec Should You Choose?

For most UK businesses, we recommend the following codec priority order:

Priority 1: G.722 — use HD voice when both endpoints support it
Priority 2: G.711a — fall back to standard quality for maximum compatibility
Priority 3: G.729 — only if bandwidth is severely limited

Choosing by Scenario

Standard office with good broadband — G.722 primary, G.711a secondary
Remote workers on home broadband — G.722 primary, G.711a secondary
Limited bandwidth or satellite connection — G.729 primary, G.711a secondary
WebRTC or softphone users — Opus primary, G.722 secondary
Call centre with many concurrent calls — G.729 to conserve bandwidth, or G.711a for quality

Bandwidth Planning

When planning your network capacity, calculate the bandwidth required based on your maximum concurrent calls:

10 concurrent calls on G.711 — approximately 870 kbps (under 1 Mbps)
10 concurrent calls on G.729 — approximately 310 kbps
50 concurrent calls on G.711 — approximately 4.35 Mbps

Always ensure you have at least 20% headroom above your calculated voice bandwidth requirement, and configure QoS settings to prioritise voice traffic.

Configuring Codec Priority

Most IP phones and PBX systems allow you to set a codec priority list. This determines which codec is used when a call is established — the two endpoints negotiate and agree on the highest-priority codec they both support.

Configure your codec priority in your phone system's admin panel. If you need assistance optimising your codec settings, contact our team for expert advice.