Estimate the IP bandwidth required for a VoIP system from the number of simultaneous calls, per-call network bitrate, and a planning headroom percentage. The per-call input should include the codec plus RTP/UDP/IP and link-layer overhead appropriate to the environment, so the result reflects network traffic rather than codec payload alone.
The estimate is useful for sizing WAN links, SIP trunks, site connectivity, and QoS reservations. Because a voice call is full duplex, the calculated requirement applies to each direction of a symmetric link when the same number of calls is active. Signaling traffic is usually small compared with media traffic and is not separately modeled here.
1. Enter concurrent calls
Use the highest realistic number of simultaneous active calls, not the number of extensions.
2. Enter per-call bitrate
Use the on-wire bitrate per call in kbps, including protocol overhead for the codec and packetization you expect.
3. Add headroom
Reserve capacity for bursts, call growth, and other small overheads.
4. Review the result
The main result is the estimated Mbps needed in each direction for the stated concurrent call load.
The per-call bitrate is an engineering input because codec, packetization interval, tunneling, and link framing can change the on-wire value.
The result is a planning bandwidth target for the voice media load in each direction of the link.
If calls traverse encrypted tunnels or additional encapsulation, use a per-call bitrate that reflects that overhead or increase headroom.
Given: 80 concurrent calls, 100 kbps per call, and 20% headroom.
Calculation: Base = 80 × 100 = 8,000 kbps = 8 Mbps. Required = 8 × 1.20 = 9.6 Mbps.
Result: Plan for about 9.6 Mbps of voice capacity in each direction.
RTP, UDP, IP, Ethernet, tunneling, and packetization add overhead. The on-wire bitrate can therefore be materially higher than the codec payload rate.
For a symmetric link, plan the displayed amount in each direction. Do not simply double it unless you specifically need one combined two-direction traffic figure.
No. Use expected simultaneous active calls. Extension count is usually higher than concurrency.
Not separately. SIP or similar signaling is generally small compared with continuous RTP media, but it can be covered by headroom.
Shorter packets increase the number of headers sent per second, which raises overhead. Use a per-call bitrate appropriate to your packetization setting.