SureTel
VoIP resource guide
VoIP Codecs Explained: G.711, G.729 and Opus
Understand how VoIP codecs affect business call quality, bandwidth and compatibility.
- • South African business VoIP provider
- • Cloud PBX and SIP trunking support
- • Network and voice configuration guidance
Educational resource · Not a quote · Licensed SA ISP · ICASA licence 0009/CECS/AUG/09
Answer first
VoIP codecs, in one paragraph
A VoIP codec is the audio format that converts speech into data for internet calling. The right codec depends on call quality, bandwidth, PBX or SIP-trunk compatibility, softphones, WebRTC use and available connectivity. SureTel helps South African businesses choose and configure practical codec settings for VoIP, Cloud PBX and SIP trunking. Request VoIP Pricing.
Summary
- • Explains G.711, G.729 and Opus in plain English.
- • Shows full-packet bandwidth examples, not payload-only figures.
- • Positions G.729 as the common practical standard in South Africa.
- • Covers codec mismatch, transcoding and one-way-audio risks.
- • Routes commercial readers to SureTel VoIP pricing.
For pricing and fit, speak to SureTel — see business VoIP services.
Problems this solves
When codec choice may be part of the problem
Codec issues are often hidden behind vague complaints like “bad call quality” or “VoIP is unstable.” This section should help readers recognise when codec choice or compatibility may be part of the problem.
Calls sound compressed or unclear
Wrong codec, heavy compression or weak network conditions can affect perceived quality.
Calls fail between systems
A SIP trunk, PBX and endpoint must support a compatible codec path.
One-way audio appears after changes
Codec negotiation is only one possible cause; NAT, firewall and routing issues can also contribute.
Bandwidth estimates are wrong
Payload-only numbers understate real usage because IP/UDP/RTP and link-layer overhead must be included.
Softphones behave differently from desk phones
Browser/WebRTC, mobile apps and hardware SIP phones may support different codec sets.
Settings are changed without testing
Forcing codecs can cause failed calls, transcoding, quality loss or compatibility problems.
When it fits
Who this codec guide is for
This article is for buyers who need enough codec knowledge to make sensible VoIP decisions, not for readers trying to become telecoms engineers.
| Reader situation | Why this guide helps | Next step |
|---|---|---|
| Considering VoIP or Cloud PBX | Explains how codecs affect quality and bandwidth | Business VoIP services |
| Planning a SIP trunk | Highlights compatibility and codec negotiation risks | SIP trunking |
| Experiencing poor call quality | Shows why “internet speed” is not the only factor | VoIP bandwidth requirements |
| Using softphones or WebRTC | Explains why Opus may matter where supported | Softphones vs desk phones |
| Running multiple branches | Clarifies why standardised settings and support matter | Multi-branch businesses |
Codec comparison
G.711, G.729, Opus and G.722 compared
The page should compare codecs by practical business fit. Avoid making a universal “best codec” claim because codec choice depends on the PBX, SIP trunk, endpoint, softphone and connection.
| Codec | Practical role | Voice quality | Bandwidth profile | SureTel positioning |
|---|---|---|---|---|
| G.711 | Quality-first legacy telephony codec | Highest of the three in typical narrowband business VoIP use | Higher bandwidth | Technically high quality, but rarely used as the default in South Africa unless a customer, PBX or provider specifically supports/requires it |
| G.729 | Compressed business voice codec | Lower than G.711, but practical for business calling | Much lower bandwidth | De facto standard in South African business VoIP environments in SureTel’s practical experience |
| Opus | Modern flexible codec for softphones/WebRTC-style use | Can be very good where supported and configured correctly | Variable | Useful for modern softphones, browser/WebRTC and adaptive environments, but support depends on the platform |
| G.722 | Wideband HD voice codec | Can offer wider audio than narrowband codecs | Similar planning class to G.711 in common examples | Mention briefly only; do not make it the main focus |
The best codec is the one that gives acceptable quality, works across the full call path and fits the available network capacity.
Business impact
How codec choice affects business calls
Codec settings should be evaluated for the real deployment, not chosen from a generic article. Codec choice affects perceived voice quality, per-call bandwidth and whether the PBX, SIP trunk, desk phones, softphones and WebRTC endpoints can all talk to each other cleanly. In SureTel’s practical experience, G.729 is the de facto codec commonly used in South African business VoIP because it balances acceptable quality against lower per-call bandwidth. G.711 is technically higher quality but is rarely default locally unless the provider, PBX or endpoint requires or only supports it. Opus is useful for modern softphones and browser/WebRTC calling where the platform supports it.
For simultaneous-call bandwidth planning, see VoIP bandwidth requirements. For a plain-English intro to VoIP, see what is VoIP?
Full-packet bandwidth
Full-packet bandwidth examples
Codec bitrate is not the same as real network usage. A VoIP packet also carries IP, UDP, RTP and link-layer overhead, so planning must use full-packet bandwidth rather than payload-only figures.
- • IP/UDP/RTP headers: 40 bytes.
- • Ethernet Layer 2 header including FCS/CRC: 18 bytes.
- • Default 20 ms voice payload for G.711 and G.729 in the examples.
- • 50 packets per second at 20 ms packetisation.
| Codec | Raw codec bitrate | 20 ms voice payload | Header + Ethernet overhead | Full packet size | Approx. Ethernet bandwidth per call | Notes |
|---|---|---|---|---|---|---|
| G.711 | 64 kbps | 160 bytes | 58 bytes | 218 bytes | 87.2 kbps | Highest-quality narrowband example; more bandwidth than G.729 |
| G.729 | 8 kbps | 20 bytes | 58 bytes | 78 bytes | 31.2 kbps | Common compressed business VoIP example; de facto standard in South Africa from SureTel experience |
| G.722 64k | 64 kbps | 160 bytes | 58 bytes | 218 bytes | 87.2 kbps | Mention as a wideband/HD voice codec where supported |
| Opus WB speech example | 16–20 kbps | approx. 40–50 bytes | 58 bytes | approx. 98–108 bytes | approx. 39.2–43.2 kbps | Example only; Opus is variable and platform-dependent |
| Opus FB speech example | 28–40 kbps | approx. 70–100 bytes | 58 bytes | approx. 128–158 bytes | approx. 51.2–63.2 kbps | Example only; actual WebRTC usage may differ |
- • Full packet size = payload bytes + IP/UDP/RTP header bytes + link-layer overhead.
- • Approximate bandwidth = full packet bytes × 8 × packets per second.
- • With 20 ms packets, there are 50 packets per second.
- • Values are planning examples, not guarantees.
- • Packetisation interval, SRTP/encryption, VPNs, VLANs, PPPoE, Wi-Fi overhead, WebRTC/ICE/TURN paths and provider configuration can change real usage.
- • Opus is variable by design, so it should be planned from the platform’s actual configured bitrate and environment.
- • For simultaneous-call planning, see VoIP bandwidth requirements.
Scenarios
Which codec fits which scenario?
Codec selection should follow the business use case and the systems involved. Do not encourage readers to change codec settings without testing the full call path.
Most South African business VoIP calling
Likely: G.729 where supported
Why: Practical compressed voice, lower bandwidth, common compatibility in local business VoIP contexts.
Caveat: Must still be compatible with provider, PBX and endpoints.
Quality-first or legacy interconnect requirement
Likely: G.711
Why: Highest quality of the main narrowband codecs covered here.
Caveat: Rarely default in South Africa unless the customer/provider/system requires or only supports it.
Browser/WebRTC or modern softphone use
Likely: Opus where supported
Why: Flexible bitrate and good modern speech support.
Caveat: Not every desk phone, PBX, SIP trunk or interconnect path supports Opus.
Mixed phone estate
Likely: Choose the codec set supported end-to-end
Why: Desk phones, DECT, softphones and SIP trunks may not all support the same options.
Caveat: Transcoding can add complexity and quality loss.
Bandwidth-constrained branch or backup link
Likely: Compressed codec where supported
Why: Reduces per-call bandwidth.
Caveat: Call quality also depends on latency, jitter, packet loss and contention.
Configuration support
What SureTel helps configure
SureTel can help choose and configure codec settings based on the real deployment rather than a generic internet article.
Cloud PBX codec settings
Codec order, extension/device compatibility, softphone support and WebRTC considerations.
SIP trunk compatibility
Align codec support between the PBX, provider and endpoint path.
Desk phones and DECT phones
Check codec support against the selected Yealink devices and deployment model.
Softphones and mobile apps
Configure supported codec options where appropriate, including Opus/WebRTC considerations when the platform allows.
Network readiness
Consider bandwidth, QoS, latency, jitter, packet loss, firewall/NAT and backup connectivity.
Troubleshooting
Investigate failed calls, poor quality, one-way audio symptoms and codec negotiation problems without assuming codec is the only cause.
Decision support: do not force codecs blindly
Codec settings should be changed with care. A codec that looks better on paper can create problems if one part of the call path cannot support it.
| Mistake | What can happen | Safer approach |
|---|---|---|
| Forcing one codec on every device | Calls may fail when another system does not support it | Test endpoint, PBX and SIP trunk compatibility |
| Choosing by bitrate only | Calls may sound poor despite low bandwidth use | Balance quality, network and compatibility |
| Ignoring packet overhead | Bandwidth planning is too low | Use full-packet estimates |
| Assuming Opus is universal | Some desk phones/PBX/SIP paths may not support it | Confirm platform support first |
| Overusing transcoding | CPU load, delay or quality loss may increase | Prefer end-to-end compatible codec paths |
| Blaming codec for every audio issue | NAT, firewall, packet loss, jitter or routing may be the real issue | Diagnose the full voice path |
Related services: Cloud PBX, SIP trunking and business VoIP.
Educational
What is a VoIP codec?
A VoIP codec is the method used to encode speech into digital packets and decode it back into audio at the other end. It affects how much bandwidth a call uses, what quality the user hears and which devices or systems can talk to each other.
Codec vs VoIP service
A codec is not the same thing as VoIP service.
Part of the call path
The codec is one part of the total call path.
SIP negotiation
SIP endpoints usually negotiate a mutually supported codec.
Quality vs bandwidth
Some codecs prioritise quality; others prioritise lower bandwidth.
Support varies
Codec support can vary by phone, PBX, softphone, browser and provider.
For a plain-English VoIP introduction, see what is VoIP? For device and network hardware context, see the VoIP equipment guide; for softphone vs desk phone choices, see softphones vs desk phones.
Why SureTel
Why ask SureTel about VoIP codecs?
Codec decisions are easier when voice, PBX and connectivity are considered together. SureTel supports business VoIP, Cloud PBX, SIP trunking and network readiness so customers are not left guessing between suppliers.
South African business VoIP
South African business VoIP and communications provider.
Cloud PBX and SIP trunking
Cloud PBX and SIP trunking support.
Practical experience
Practical network and voice troubleshooting experience.
Endpoint support
Support for desk phones, softphones, WebRTC and SIP endpoints where suitable.
Connectivity options
Business connectivity options available where feasible.
Clear scope
Clear scope: configuration support, not unsupported guarantees.
Do not claim SureTel controls third-party networks, all customer devices or every provider interconnect. Frame support as assessment, configuration and troubleshooting within the supplied solution and agreed scope. Related services: business VoIP, Cloud PBX and SIP trunking.
Process
How codec planning works
For business deployments, codec planning should happen before or during VoIP setup, not only after users complain about call quality.
- Step 1
Confirm the voice environment
Cloud PBX, SIP trunk, onsite PBX, softphones, desk phones, DECT or mixed setup.
- Step 2
Check endpoint and provider support
Confirm which codecs are supported by the PBX, trunk, phones and apps.
- Step 3
Assess network conditions
Review bandwidth, latency, jitter, packet loss, firewall/NAT and internet failover.
- Step 4
Choose practical codec preferences
Balance quality, bandwidth and compatibility.
- Step 5
Configure and test
Validate live calling, inbound/outbound paths, transfers, recordings, WebRTC and branch calls where relevant.
- Step 6
Monitor and adjust
Revisit settings when devices, connectivity or usage patterns change.
Not sure which codec fits your VoIP setup?
Request VoIP Pricing and tell SureTel about your phones, PBX, SIP trunk, softphones, branches or call-quality issue.
FAQs
VoIP codec FAQs
What is a VoIP codec?
A VoIP codec is the audio format used to encode speech into data packets and decode it back into sound. It affects call quality, bandwidth usage and compatibility between phones, softphones, PBX systems and SIP trunks.
Which VoIP codec is best for business calls?
There is no single best codec for every business. G.711 is high quality but uses more bandwidth, G.729 is bandwidth-efficient and commonly used in South African business VoIP, and Opus can work well for modern softphones or WebRTC where supported.
Is G.711 better than G.729?
G.711 usually offers higher voice quality, but it uses more bandwidth. G.729 is more compressed and is often more practical where many calls share the same internet connection, branch link or backup connection. Compatibility and network quality still matter.
Why is G.729 common in South African VoIP?
In SureTel’s practical experience, G.729 is commonly used because it offers a practical balance between acceptable voice quality and lower per-call bandwidth. It is still subject to provider, PBX, endpoint and licensing or support compatibility.
What is Opus used for in VoIP?
Opus is a modern flexible codec often used in softphones, browser-based calling and WebRTC-style applications. It can adapt across different bitrate and quality levels, but it must be supported by the platform, PBX, app and call path.
Why do codec bandwidth figures differ online?
Some figures show only the raw codec payload, while others include packet overhead such as IP, UDP, RTP and Ethernet headers. For business planning, full-packet usage is more useful because it better reflects real network traffic.
Can wrong codec settings cause failed calls or one-way audio?
Codec mismatch can contribute to failed calls or audio problems, but it is not the only cause. NAT, firewall rules, routing, packet loss, jitter and SIP configuration can also create symptoms such as one-way audio or poor call quality.
Should I change codec settings myself?
Avoid forcing codec settings without understanding the full call path. A PBX, SIP trunk, desk phone, softphone or WebRTC client may support different codecs. A tested configuration is safer than changing settings blindly.
Next step
Need help choosing the right VoIP setup?
SureTel can help assess your VoIP environment, codec compatibility, Cloud PBX setup, SIP trunking, devices and connectivity so your business has a practical voice solution.
Educational resource · Not a quote · Codec choice depends on the PBX, SIP trunk, endpoint, softphone and connection. Bandwidth figures on this page are planning examples, not guarantees. Request VoIP Pricing for a scoped recommendation.