VoIP Bandwidth Calculator
Work out how much bandwidth your calls actually need. Pick a codec, set your concurrent calls, and get a sizing figure with headroom included. No sign-up, no email required.
symmetric (same up and down)
Figures include IP, UDP and RTP header overhead at 20ms packetisation. Voice is symmetric, so you need the stated capacity in both directions.
How much bandwidth does a VoIP call use?
On the Opus codec, a call uses roughly 50 kbps in each direction once protocol overhead is counted. The codec payload itself is around 32 kbps; the rest is IP, UDP and RTP headers, which add about 16 kbps at 50 packets per second.
| Codec | Payload | With overhead | Notes |
|---|---|---|---|
| Opus | 32 kbps | 50 kbps | Default on Salamtalk. Variable bitrate, best quality per kbps. |
| G.711 | 64 kbps | 87 kbps | Uncompressed PCM. Highest quality, highest bandwidth. Common on legacy PBX. |
| G.722 | 64 kbps | 87 kbps | HD wideband audio at the same bitrate as G.711. |
| G.729 | 8 kbps | 31 kbps | Heavily compressed. Use when bandwidth is genuinely constrained. |
Size by concurrent calls, not headcount
The most common sizing mistake is multiplying total staff by the per-call figure. What matters is how many calls are live at the same moment. A 50-person office where a third are on the phone at peak needs capacity for about 17 concurrent calls — under 1 Mbps on Opus, which almost any business connection already has.
Contact centers are the exception. If most seats are on calls for most of the day, size closer to one concurrent call per agent. See call center software for how queueing changes the picture.
Bandwidth is rarely the real problem
If calls sound bad, insufficient bandwidth is usually not the cause. Voice is sensitive to jitter (variation in packet arrival time) and packet loss, not raw throughput. A link with plenty of spare capacity can still deliver poor audio if voice packets queue behind a large file upload.
The fix is prioritisation, not a bigger circuit. Apply QoS so voice traffic is handled ahead of bulk data. Our migration guide covers this as part of the cutover checklist.
Frequently asked questions
How much bandwidth does one VoIP call use?
On the Opus codec, roughly 50 kbps in each direction once IP, UDP and RTP overhead are included. G.711 uses about 87 kbps and G.729 about 31 kbps. The raw codec bitrate is lower, but overhead is real and should be budgeted for.
Why is the total higher than the codec bitrate?
Every voice packet carries IP, UDP and RTP headers totalling 40 bytes. At 50 packets per second (20ms packetisation) that adds roughly 16 kbps on top of the payload, regardless of codec.
How many concurrent calls will my team actually make?
Size by concurrent calls, not headcount. A typical office runs 20 to 33 percent of staff on calls at peak. Sales and support teams run far higher, often 70 to 90 percent, so count those separately.
Is bandwidth usually the problem with VoIP quality?
Rarely. Most quality issues are caused by jitter and packet loss, not insufficient bandwidth. A 50-person office needs under 1 Mbps for voice, which almost any modern connection can supply. Prioritising voice traffic with QoS fixes more problems than buying a larger circuit.
Should I add headroom to the result?
Yes. Plan for at least 30 percent above your calculated peak so that call quality does not degrade when traffic spikes or when voice competes with video and file transfers on the same link.
Use this calculator on your own site
You are welcome to reference or link to this tool. If it is useful to your readers, a link back is appreciated:
<a href="https://www.salamtalk.com/tools/voip-bandwidth-calculator">VoIP Bandwidth Calculator</a> Need the bandwidth figures for a real deployment?
Our team will size your connection and run a quality test on your actual network before you commit.