Calculator inputs
Use the responsive grid below. Large screens show three columns, medium screens show two, and mobile screens show one.
Example data table
These sample cases use the same formula engine as the calculator. Values help you compare likely bandwidth ranges across different voice deployments.
| Scenario | Codec | Calls | Per call duplex | Total duplex voice | Recommended link |
|---|---|---|---|---|---|
| Branch office default voice | G.711 | 15 | 193.60 kbps | 2.904 Mbps | 4.675 Mbps |
| WAN SIP trunk with compression | G.729 | 40 | 56.00 kbps | 2.240 Mbps | 3.606 Mbps |
| Secure remote workers over VPN | Opus Wideband | 30 | 193.60 kbps | 5.808 Mbps | 10.753 Mbps |
| High-density LAN deployment | G.722 | 80 | 193.60 kbps | 15.488 Mbps | 24.936 Mbps |
Formula used
1) Voice payload bytes per packet
Payload bytes = (codec bitrate in kbps × packetization in ms) ÷ 8
2) Total packet bytes
Total packet bytes = payload bytes + transport headers + Layer 2 overhead + security bytes + tunnel bytes
3) Packets per second
Packets per second = 1000 ÷ packetization interval in ms
4) One-way media bandwidth
One-way bandwidth = total packet bytes × 8 × packets per second ÷ 1000
5) Adjusted one-way bandwidth
Adjusted one-way bandwidth = one-way bandwidth × voice activity factor × FEC multiplier
6) Duplex and planning bandwidth
Total duplex bandwidth = adjusted one-way bandwidth × 2 × concurrent calls
Planned traffic = total duplex bandwidth + signaling reserve + planning headroom
Recommended link = planned traffic ÷ utilization target
This model is intended for media path sizing and capacity planning. Real deployments may also require QoS policy review, access speed checks, and device processing limits.
How to use this calculator
- Select a codec preset or choose a custom bitrate.
- Enter the packetization interval used by your phones or gateway.
- Set IP version, VLAN tags, MPLS labels, and any tunnel or security overhead.
- Enter peak concurrent calls for your busy period.
- Adjust activity factor if silence suppression lowers average media traffic.
- Add FEC, signaling reserve, and planning headroom if your design requires them.
- Enter your current link speed to compare planned usage against available capacity.
- Press calculate to show results above the form, review the graph, then export CSV or PDF if needed.
FAQs
1) Why is the calculated bandwidth much higher than codec bitrate?
The codec bitrate only covers voice payload. Real VoIP traffic also carries RTP, UDP, IP, Ethernet, VLAN, security, and sometimes tunnel overhead. Smaller packetization intervals increase packets per second, so overhead consumes a larger share of the link.
2) Should I use one-way or duplex bandwidth for planning?
Use duplex bandwidth when sizing a full bidirectional link for active conversations. One-way bandwidth is helpful when analyzing a single traffic direction, such as uplink congestion or directional bottlenecks through a WAN edge device.
3) What packetization interval is best?
Twenty milliseconds is common because it balances efficiency and delay. Shorter intervals can improve loss recovery and latency but increase overhead. Longer intervals reduce overhead but can increase delay and make packet loss more noticeable.
4) When should I enable compressed RTP headers?
Use it when your WAN design truly supports header compression and your devices are configured for it. It is especially helpful on slow links. Do not enable it merely for estimation unless that feature is active in production.
5) Does silence suppression always reduce required bandwidth?
It lowers average traffic when users are not speaking, but peak demand can still be high. Conservative capacity plans often keep activity factor at 100 percent for worst-case sizing during many simultaneous active calls.
6) Why add signaling reserve and headroom separately?
Signaling reserve covers call setup, control, and related traffic. Headroom covers growth, bursts, policy overhead, and design margin. Using both gives a more realistic planning figure than sizing only for bare RTP media.
7) How do VPN and SRTP affect VoIP bandwidth?
SRTP adds security bytes to each packet, while VPNs or tunnels add encapsulation overhead. These additions can materially increase required bandwidth, especially for small payloads and short packetization intervals.
8) Is this enough to guarantee call quality?
No. Bandwidth planning is essential, but call quality also depends on latency, jitter, packet loss, queue management, codec behavior, echo control, and QoS policies. Use this calculator together with end-to-end network validation.