VoIP codec guide: G.711, Opus, and G.729 explained

Who this guide is for

Audience: Network engineers and IT admins optimising VoIP call quality.

Understand how VoIP calling works — SIP, PBX, codecs, trunking, DID numbers, STIR/SHAKEN, and the protocols behind business phone systems.

Use this guide when you want the setup to be correct the first time and easy for another admin, manager, or supervisor to verify later.

What this workflow helps you accomplish

A VoIP codec (coder-decoder) converts analogue voice into a digital bitstream for transmission over IP networks and decodes incoming streams back into audio. The codec choice directly affects call quality, bandwidth usage, and CPU load on endpoints and PBX servers.

This workflow matters because numbers, routing, access, and reporting in CallOrbit are connected. Skipping one setup detail usually creates avoidable support work later.

• Step 1: Compare the three most common VoIP codecs by bandwidth: G.711 uses 64 Kbps per call with uncompressed toll-grade audio (the PSTN standard). Opus uses 6-510 Kbps with adaptive bitrate — typically 32-64 Kbps for excellent quality. G.729 uses 8 Kbps with compression that saves bandwidth at the cost of some audio fidelity.
• Step 2: Choose G.711 when bandwidth is sufficient: G.711 offers the best audio quality and lowest latency because it does not compress the audio stream. It is the default codec on most SIP trunks and PBX platforms. Use G.711 if your network has at least 100 Kbps available per concurrent call.
• Step 3: Choose Opus for adaptive networks: Opus automatically adjusts its bitrate based on network conditions. On a good connection it delivers near-G.711 quality at half the bandwidth. On a congested link it drops bitrate gracefully rather than dropping packets. Opus is ideal for softphone and mobile app calling.
• Step 4: Choose G.729 for bandwidth-constrained links: G.729 compresses audio to 8 Kbps with tolerable quality for voice. Use G.729 when you have limited WAN links, many concurrent calls need to share a small pipe, or when connecting remote offices with DSL or LTE backup links.
• Step 5: Configure codec negotiation order in your PBX or SIP endpoint: set the most preferred codec first in the allowed list. Endpoints negotiate to the first mutually supported codec during the SIP handshake. Prioritise G.711 or Opus for local calls and G.729 for WAN or high-density situations.

Setup checklist

• Compare the three most common VoIP codecs by bandwidth: G.711 uses 64 Kbps per call with uncompressed toll-grade audio (the PSTN standard). Opus uses 6-510 Kbps with adaptive bitrate — typically 32-64 Kbps for excellent quality. G.729 uses 8 Kbps with compression that saves bandwidth at the cost of some audio fidelity.
• Choose G.711 when bandwidth is sufficient: G.711 offers the best audio quality and lowest latency because it does not compress the audio stream. It is the default codec on most SIP trunks and PBX platforms. Use G.711 if your network has at least 100 Kbps available per concurrent call.
• Choose Opus for adaptive networks: Opus automatically adjusts its bitrate based on network conditions. On a good connection it delivers near-G.711 quality at half the bandwidth. On a congested link it drops bitrate gracefully rather than dropping packets. Opus is ideal for softphone and mobile app calling.
• Choose G.729 for bandwidth-constrained links: G.729 compresses audio to 8 Kbps with tolerable quality for voice. Use G.729 when you have limited WAN links, many concurrent calls need to share a small pipe, or when connecting remote offices with DSL or LTE backup links.
• Configure codec negotiation order in your PBX or SIP endpoint: set the most preferred codec first in the allowed list. Endpoints negotiate to the first mutually supported codec during the SIP handshake. Prioritise G.711 or Opus for local calls and G.729 for WAN or high-density situations.

Operational follow-up

After you complete this flow, confirm the live experience from both the agent and customer side so ownership, routing, permissions, and reporting all match what the workspace expects.

If your team is rolling this out across multiple users, queues, or phone numbers, pair this article with the broader knowledge base and the relevant routing or numbers guides to keep deployment consistent.

• What is the CallOrbit Knowledge Base for? — It is the public help hub for how CallOrbit works, covering numbers, webphone setup, SIP, extensions, routing, users, roles, and billing basics.
• Can customers read this without signing in? — Yes. The Knowledge Base now lives on a public route so customers can read setup guidance before or after they enter the portal.
• Does the portal still have its own Knowledge Base page? — No. The signed-in portal navigation no longer carries a separate Knowledge Base page, and the old portal path now redirects to this public version.
• What is VoIP and how does it work? — VoIP (Voice over Internet Protocol) converts analogue voice signals into digital packets and transmits them over IP networks. Unlike traditional PSTN phone lines that require dedicated copper wiring per line, VoIP calls use your existing internet connection, which makes them cheaper, more flexible, and easier to scale.
• What is SIP trunking? — SIP trunking is a virtual connection that replaces traditional analogue phone lines or PRI circuits. A SIP trunk carries multiple concurrent voice channels over a single IP connection to your PBX or phone system, eliminating per-line hardware costs and monthly line rental fees.
• What is the difference between hosted PBX and cloud PBX? — Hosted PBX runs on dedicated virtual infrastructure managed by a provider, while cloud PBX uses shared multi-tenant cloud infrastructure. Hosted PBX suits organisations needing custom configuration and predictable pricing. Cloud PBX is better for instant scalability and per-user monthly billing.
• What is a DID number? — A DID (Direct Inward Dialling) number is a virtual phone number that routes directly to a specific extension, IVR menu, queue, or user within a phone system without an operator. DIDs decouple the phone number from the physical phone line, so you can have hundreds of numbers routed through a single SIP trunk.
• What are G.711, Opus, and G.729 codecs used for? — These are VoIP codecs that convert voice into digital data. G.711 uses 64 Kbps for toll-grade quality and is the PSTN standard. Opus uses 6-510 Kbps and adjusts to network conditions. G.729 uses 8 Kbps for bandwidth-constrained links. The right codec depends on your available bandwidth and call quality requirements.

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