Free tools

VoIP bandwidth calculator

Concurrent calls, codec, Wi-Fi or Ethernet, headroom: the upload speed you need, worked out layer by layer, and a straight answer on whether your connection can take it.

Upload is the bottleneck

Your setup

Every change recalculates everything.

Start from a typical setup

One click fills everything in, then adjust.

Your calls

How many calls at once, at the busiest moment of the day.

calls

For a sales team calling in blocks, count one call per rep on the phones.

Codec

It sets the size of the voice stream. Not sure? Keep G.711, the heaviest one.

Your network

The link between the computer and the router, and your headroom.

Link between computer and router

18 bytes per packet

Editable assumption
%

For traffic spikes and the unexpected.

Your internet connection

Upload is the direction that carries your voice.

Editable assumption
Mbps

Replace the default with your measured speed or your plan’s upload.

Conservative assumption. On cable, capacity is shared by the subscribers in the same area, so upload can dip at busy hours. Arcep, fixed broadband file

Don’t know your upload speed? Measure it in 30 seconds

Everything else on the lineOptional. What shares the connection with your calls.
kbps
kbps
Editable assumption
kbps

Per-unit values: 720p group video per Zoom, screen sharing per Microsoft Teams, computer per a Skipcall assumption.

Advanced settingsPacketization, IPv6, encryption, VPN. The defaults fit most setups.
ms

Audio per packet. 20 ms is the norm.

IP version
bytes

Browsers (WebRTC) often add a few bytes per packet. Leave 0 if unsure.

Upload needed: 546 kbps. Your connection has plenty of room.

What you need

Your connection has plenty of room

Upload speed needed

546 kbps

5 × 84 kbps per call, including 30% headroom

Download: 546 kbps

Upload in use

5% / 10 Mbps

Voice 420 kbpsHeadroom 126 kbpsFree 9.45 Mbps

With 10 Mbps of upload, your needs, headroom included, use only 5% of it. Bandwidth won’t be the problem: watch stability instead (latency, jitter).

Your connection can carry up to 91 concurrent calls with these settings.

One voice packet, layer by layer

50 packets per second in each direction (one every 20 ms). Block width: bytes in the packet.

228 bytes

  • Voice (codec) 160
  • RTP 12
  • SRTP 10
  • UDP 8
  • IP 20
  • Ethernet / Wi-Fi 18

  1. 1. Voice only

    64 kbps

    per call, each direction

  2. 2. Over the internet (what the router sees)

    84 kbps

    per call, each direction

  3. 3. On your local network

    91.2 kbps

    per call, each direction

76% of what you send over the internet is voice. The rest is headers.

All calls together on the local network (Ethernet cable): 456 kbps.

What about another codec?

Same settings, same other traffic, same headroom.

CodecPer callUpload neededMax calls
52 kbps338 kbps147
84 kbps546 kbps91
84 kbps546 kbps91
28 kbps182 kbps274

What we recommend

  1. 1

    Prioritize voice on the router

    Turn on your router’s Quality of Service (QoS) so voice packets go first. The standard marking for telephony is DSCP EF (46), per RFC 4594.

  2. 2

    Measure your real connection

    This calculator tells you what you need. Our VoIP speed test measures what you have: upload speed, latency and jitter.

    Run the VoIP speed test
How it’s calculatedThe formulas, with your numbers.
  • Payload = 64 kbps × 20 ms ÷ 8 = 160 bytes
  • IP packet = payload + RTP 12 + SRTP 10 + UDP 8 + IP 20 = 210 bytes
  • Packets per second = 1,000 ÷ 20 ms = 50
  • Bandwidth per call, each direction = 210 bytes × 8 × 50 = 84 kbps
  • Local network: + 18 bytes, for 228 bytes per packet and 91.2 kbps per call
  • Upload needed = (5 × 84 kbps + 0 kbps) × (1 + 30%) = 546 kbps
  • Utilization = 546 kbps ÷ 10 Mbps = 5.5%
  • Max calls = ⌊(10 Mbps ÷ (1 + 30%) − 0 kbps) ÷ 84 kbps⌋ = 91

Verdict thresholds (Skipcall assumptions): up to 50% utilization, plenty of room; up to 80%, fine; up to 100%, tight; above, not enough.

Save or share this calculation

The link reopens the page with your settings. Nothing is stored on our side.

Connection ready? Skipcall’s business VoIP runs in the browser or the mobile app, with no hardware to install: a connection like the one you just sized is all it takes.

See business VoIPTest my connection

Everything runs in your browser: no data leaves it. How it’s calculated

546 kbps upload

Plenty of room · 5%

How much bandwidth does a VoIP call use?

Not much: 24 to 88 kbps in each direction for the common codecs, headers included. What surprises people is the gap between the codec’s bitrate and what actually crosses the wire. G.711 encodes voice at 64 kbps, but every 20 ms packet carries 40 bytes of headers (RTP, UDP, IP) for 160 bytes of audio: 80 kbps over the internet, 87.2 kbps on an Ethernet cable.

The table below gives those numbers for the four most common codecs at reference settings (20 ms, IPv4, unencrypted). The calculator adds everything else: SRTP encryption, IPv6, VPN, Wi-Fi, call count, other traffic and headroom.

Bandwidth per call, each direction, 20 ms packets, IPv4, plain RTP
CodecVoice onlyOver the internet (IP)On EthernetWith SRTP (IP)
Opus 32 kbps 48 kbps 55.2 kbps 52 kbps
G.711 64 kbps 80 kbps 87.2 kbps 84 kbps
G.722 64 kbps 80 kbps 87.2 kbps 84 kbps
G.729 8 kbps 24 kbps 31.2 kbps 28 kbps

Computed with the same model as the tool. The Ethernet figures match Cisco’s reference table (87.2 kbps for G.711, 31.2 kbps for G.729). Opus counted at 32 kbps.

How the math works, layer by layer

We follow one voice packet from the microphone to the cable, then multiply by packets per second. Every header size comes from the standard that defines it.

Over the internet (what the router sees) 210 bytes × 50 packets/s × 8 = 84 kbps

On your local network 228 bytes × 50 packets/s × 8 = 91.2 kbps

One voice packet, layer by layer : G.711, 20 ms, SRTP, Ethernet cable. Sizes in bytes.
  1. 1

    Voice: codec bitrate × packet duration

    The codec produces a steady stream, cut into packets, usually every 20 ms. G.711 at 64 kbps: 64,000 × 0.020 ÷ 8 = 160 bytes of audio per packet. Opus is variable, so we use its target bitrate as a ceiling.

  2. 2

    Transport headers: RTP, UDP, IP

    Each packet gets 12 bytes of RTP (sequence number, timestamp), 8 bytes of UDP and 20 bytes of IPv4, or 40 with IPv6. That is 40 bytes on IPv4 no matter how small the audio is, which is why light codecs save less than you’d expect.

  3. 3

    SRTP encryption

    Encrypted audio keeps its size, but each packet carries an authentication tag: 10 bytes with the most common profile (80-bit HMAC-SHA1), 4 with the 32-bit variant, 16 with AES-GCM.

  4. 4

    The VPN, if any

    A VPN wraps each packet in another one. WireGuard adds 60 bytes on IPv4 and pads the inner packet to a multiple of 16 bytes. IPsec tunnel mode with NAT traversal adds about 62 to 65 bytes. On 200-byte packets, that is a third more bandwidth.

  5. 5

    The local link: Ethernet or Wi-Fi

    On a cable, the Ethernet frame adds 18 bytes. On Wi-Fi, the header and WPA encryption add about 54, and each frame holds the air far longer. This layer never leaves your office, so it doesn’t count against your router’s upload.

  6. 6

    From packet to bandwidth to verdict

    Bandwidth per call = packet bytes × 8 × packets per second (50 at 20 ms). Multiply by concurrent calls, add other traffic, then headroom. The verdict compares that to your connection’s upload speed.

Why upload speed decides everything

A conversation is symmetrical: you send as much audio as you receive. Most internet plans aren’t, with upload far slower than download. So the upload side almost always fills up first, especially since it also carries your file uploads, backups and your camera in video meetings.

Advertised speeds are theoretical peaks. For this calculation, use the upload you measure during business hours, or the committed rate in your contract if you have a business plan. The table below lists the tool’s starting values: conservative assumptions unless a source is named.

Connection typeStarting valueWhat to know
DSL (ADSL) 0.8 Mbps Conservative assumption. DSL has the slowest upload of any connection type, and it drops with line length: measure yours.
VDSL 5 Mbps Conservative assumption. VDSL only delivers a real gain on short lines close to the exchange.
Cable (DOCSIS) 10 Mbps Conservative assumption. On cable, capacity is shared by the subscribers in the same area, so upload can dip at busy hours.
Fiber 100 Mbps Conservative assumption: fiber upload depends on your plan. Check your contract or measure it.
Dedicated business fiber 100 Mbps Enter the committed upload rate in your contract.
4G home internet 10 Mbps Conservative assumption. Speed depends on where the router sits and how loaded the cell is.
5G home internet 10 Mbps Conservative assumption. France’s regulator Arcep measured equivalent upload speeds on 4G and 5G in 2021.
Starlink 17 Mbps Median measured by Ookla in the US in the second half of 2025: 16.91 Mbps upload.

How many concurrent calls should you count?

It’s the input that matters most, and the one most often guessed wrong. In a general office, not everyone is on the phone at once. On a sales floor it’s the opposite: during a call block, every rep is on a call almost nonstop. Count one call per rep on the phones, at the moment the whole team is dialing.

For a mixed office, add both: reps in a call block, plus a cautious estimate for everyone else. The calculator also shows the maximum number of calls your connection can carry, which answers the real question: can we add three more reps?

The other traffic sharing your line

Voice rarely has the line to itself. These vendor-published figures are the tool’s defaults, and you can edit them.

Common mistakes

  • Looking at download speed

    A “1 Gig” plan may upload only a fraction of that, and a DSL line sold on its download speed may upload under 1 Mbps. Voice fills the upload side long before the download side.

  • Forgetting the headers

    Budgeting 64 kbps for a G.711 call underestimates the need by 25% over the internet, and by more than a third on Wi-Fi or behind a VPN.

  • Counting employees instead of calls

    The right number is calls at the same time at the busiest moment, not headcount. For a sales team, the two get close.

  • Assuming more bandwidth fixes everything

    Past what you need, more bandwidth doesn’t improve voice. Stability does: latency, jitter and packet loss, which only a test can measure.

  • Sending everything through the VPN

    Voice already encrypted with SRTP doesn’t need the tunnel: the VPN adds bytes, latency and sometimes a detour through headquarters.

  • Measuring once, on a Sunday

    Usable throughput on 4G, cable or satellite changes with the time of day. Measure when your team is actually calling.

Keep going

This calculator tells you what you need. These pages help you check what you have and roll it out.

Sources

Standards and documents consulted, checked November 6, 2026.

How this calculator handles your data

Everything runs in your browser: nothing is sent, nothing is stored, no sign-up. The share link only carries your settings (call count, codec, speed), never personal information.

The model is an engineering estimate: it counts the exact bytes of every layer, but not Wi-Fi airtime, retransmissions or encapsulation specific to some connections (ATM cells on DSL, for instance). Headroom is there to cover the gap.

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Frequently asked questions

About 80 kbps in each direction over the internet with G.711 (the classic codec: 64 kbps of audio plus headers), 84 kbps if the audio is encrypted with SRTP, and 87.2 kbps on an Ethernet cable. Opus at 32 kbps comes to about 52 kbps encrypted; G.729 to 24 kbps unencrypted.
With encrypted G.711 (84 kbps per call) and 30% headroom: 9 calls. Without headroom, 11. With Opus at 32 kbps, about 14 at the same headroom. The calculator runs this for your own connection and settings.
Count concurrent calls, not employees. Ten calls at once on encrypted G.711 with 30% headroom need about 1.1 Mbps of upload (and the same down); twenty calls, about 2.2 Mbps. Then add the video meetings and computers sharing the line: they are usually what tips the verdict.
Yes, but the calculator leaves them out on purpose. Silence suppression (VAD) sends almost nothing while nobody talks, but it isn’t always on and saves nothing when everyone talks at once. Header compression (cRTP, RFC 2508) works hop by hop between two routers, not end to end across the internet. Sizing without them means sizing for your busiest moment.
Upload. A call uses the same amount both ways, but upload is the narrow side on DSL, cable, 4G and satellite, so it fills up first.
A few more bytes (about 54 per packet versus 18), and much more airtime, shared with every nearby device. Wi-Fi’s real problem is jitter, not bandwidth: wire up the computers that call the most.
Yes. With encrypted G.711, WireGuard grows each packet from 210 to 284 bytes: 113.6 kbps per call instead of 84, or 35% more. Route voice outside the tunnel when you can.
Usually whichever your VoIP provider negotiates. G.711 is universal and delivers classic phone quality with no voice compression; Opus gives HD voice for less bandwidth and adapts to the network; G.729 is the lightest but sounds more metallic. When in doubt, size for G.711: it’s the heaviest common case.
No, it complements one. The calculator tells you what you need; the VoIP speed test measures what you have (speed, latency, jitter). Enough bandwidth with high jitter still makes for bad calls.
No. Everything is calculated in your browser. The share link only contains the calculator settings.