- Per call, each way
- Packet size and rate
- Total before headroom
- Based on
Figures apply to upload and download separately. Opus bitrate varies, so 24 kbps is an example setting.
How VoIP bandwidth is calculated
Each call sends audio packets at a steady rate. With a 20 ms interval that is 50 packets per second. The payload size comes from the codec bitrate, and every packet adds 40 bytes of IP, UDP and RTP headers. Add Ethernet framing if you are sizing a LAN link.
| Codec at 20 ms | Payload | IP layer per call | With Ethernet |
|---|---|---|---|
| G.711 / G.722 | 160 bytes | 80 kbps | 87.2 kbps |
| G.729 | 20 bytes | 24 kbps | 31.2 kbps |
How much VoIP bandwidth do I need?
Multiply the per-call figure by your peak simultaneous calls, then add headroom for signaling, jitter and other traffic. Use the Erlang calculator to estimate peak calls from call volume. Compare codecs in G.711 vs G.729.
VoIP bandwidth calculator FAQ
How much bandwidth does one VoIP call use?
About 80 kbps each way for G.711 at 20 ms packets at the IP layer, and about 24 kbps for G.729. Ethernet headers add a few more kbps.
Why is it more than the codec bitrate?
Every packet also carries IP, UDP and RTP headers (40 bytes), and 50 packets are sent each second at 20 ms.
Do I need bandwidth in both directions?
Yes. A call sends and receives audio at the same time, so size both upload and download for the number of simultaneous calls.
How much headroom should I add?
Many planners add 20 percent or more for signaling, jitter and other traffic. Increase it if your link is shared with heavy data use.
Does encryption change the numbers?
Secure RTP adds a small amount of overhead per packet. For planning, the headroom setting normally covers it.
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