Voice Codec Bandwidth
Calculate bandwidth consumption per call for common VoIP codecs including overhead.
About Voice Codec Bandwidth
Each VoIP call's actual network bandwidth is higher than its raw codec bitrate because every RTP packet is wrapped in UDP and IP headers, and voice is typically packetized every 20ms, meaning small packets are sent frequently, amplifying the relative header overhead. G.711 (64 kbps raw) becomes about 87.2 kbps on the wire; G.729 (8 kbps raw, using compression) becomes about 31.2 kbps — proportionally much more overhead-heavy because its payload is so small per packet. Total required bandwidth is simply the per-call bandwidth multiplied by the number of concurrent calls.
Formula
Total bandwidth required
total_bandwidth = calls × codec_bandwidth_per_call
- n
- Number of concurrent calls
- b
- Bandwidth per call including IP/UDP/RTP overhead
Worked example
With the default values below, Voice Codec Bandwidth returns Total Bandwidth Required of 4.360. Change any input above and every figure here is recalculated in your browser.
Inputs used
- Voice Codecone of G.711 (64 kbps, 87.2 kbps with IP overhead), G.729 (8 kbps, 31.2 kbps with IP overhead), G.722 (64 kbps HD voice, 87.2 kbps with IP overhead), Opus (variable, ~24-40 kbps typical, ~55 kbps with IP overhead)
- G.711 (64 kbps, 87.2 kbps with IP overhead)
- Number of Concurrent Callsminimum 1
- 50
Results
- Total Bandwidth Required
- 4.360 Mbps
- Bandwidth per Call (with IP overhead)
- 87.2 kbps
- Total Bandwidth
- 4,360.0 kbps
- Codec Payload Only (no IP overhead)
- 3,200.0 kbps
Step by step
- 1Values used
- 2Total bandwidth required
- 3Total Bandwidth Required= 4.360 Mbps
- 4Bandwidth per Call (with IP overhead)= 87.2 kbps
- 5Total Bandwidth= 4,360.0 kbps
- 6Codec Payload Only (no IP overhead)= 3,200.0 kbps
Assumptions and limits
- The result is derived only from the 2 inputs listed above. Anything not among them is not part of the calculation.
- In particular, this calculator does not take inflation, tax, fees or charges or compounding frequency into account.
- Figures are rounded for display to 3 decimal places, so totals built from several rounded parts can differ slightly from the unrounded arithmetic.
How these calculators are built and tested is described on the Methodology page. If a figure looks wrong, please report it.
Frequently Asked Questions
Why is G.729's overhead-inclusive bandwidth so much higher than its 8 kbps raw rate?
G.729 compresses voice down to just 8 kbps of actual payload, but the IP+UDP+RTP headers (typically 40 bytes) are the same fixed size regardless of payload, so on a small 20ms packet, header bytes make up a much larger fraction of the total — pushing effective bandwidth to about 31.2 kbps, nearly 4x the raw codec rate.
Why choose G.729 over G.711 if it needs proportionally more overhead?
Even with that overhead, G.729's total ~31.2 kbps per call is still far less than G.711's ~87.2 kbps, so it remains the better choice for bandwidth-constrained links like WAN trunks — the tradeoff is somewhat lower audio quality and the CPU cost of compression.
What is Opus and why is it increasingly popular?
Opus is a modern, royalty-free codec that adapts its bitrate (roughly 6-510 kbps) based on content and network conditions, offering better quality per kbps than older codecs. It's now the default in WebRTC and many modern VoIP/video platforms.
How can I reduce total VoIP bandwidth on a constrained WAN link?
Use a lower-bitrate codec like G.729, enable RTP header compression (cRTP) to shrink the per-packet overhead, or use a larger packetization interval (e.g. 30ms instead of 20ms) to amortize header overhead across more payload per packet — at the cost of slightly higher latency.