--- title: "VoIP Bandwidth Calculator: Upload Speed per Call" description: "Free VoIP bandwidth calculator: the upload speed your calls need by codec, concurrent calls, Wi-Fi or Ethernet, VPN and headroom, plus a verdict on your line." url: https://skipcall.io/en/tools/voip-bandwidth-calculator language: en --- [Free tools](https://skipcall.io/en/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. Concurrent calls 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 Headroom Editable assumption % For traffic spikes and the unexpected. Your internet connection Upload is the direction that carries your voice. Connection type Your upload speed 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](https://www.arcep.fr/la-regulation/grands-dossiers-reseaux-fixes/le-haut-debit-fixe-cable-adsl-vdsl.html) Don’t know your upload speed? [Measure it in 30 seconds](https://skipcall.io/en/tools/voip-speed-test) **Everything else on the lineOptional. What shares the connection with your calls.** Video meetings at once Upload per meeting kbps Screen shares Upload per share kbps Computers online Upload per computer 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.** Packetization ms Audio per packet. 20 ms is the norm. IP version Voice encryption VPN RTP header extensions 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 5% / 10 Mbps Upload in use 5% / 10 Mbps Voice 420 kbps Headroom 126 kbps Free 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) - 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. | Codec | Per call | Upload needed | Max calls | | - | - | - | - | | | 52 kbps | 338 kbps | 147 | | Your pick | 84 kbps | 546 kbps | 91 | | | 84 kbps | 546 kbps | 91 | | | 28 kbps | 182 kbps | 274 | ## 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](https://skipcall.io/en/tools/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 VoIP](https://skipcall.io/en/business-voip) [Test my connection](https://skipcall.io/en/tools/voip-speed-test) Everything runs in your browser: no data leaves it. How it’s calculated ## 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. | Codec | Voice only | Over the internet (IP) | On Ethernet | With 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 network228 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 type | Starting value | What 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. - 720p group video meeting (Zoom) 2.6 Mbps up, 1.8 Mbps down [Zoom system requirements, checked November 6, 2026](https://support.zoom.com/hc/en/article?id=zm_kb\&sysparm_article=KB0060748) - One-to-one screen sharing (Microsoft Teams, recommended) 1.5 Mbps each way [Microsoft Learn, checked November 6, 2026](https://learn.microsoft.com/en-us/microsoftteams/prepare-network) - Teams audio (recommended) 58 kbps each way [Microsoft Learn, checked November 6, 2026](https://learn.microsoft.com/en-us/microsoftteams/prepare-network) - Work computer (browsing, email, sync) 0.2 Mbps up, 1 Mbps down Skipcall assumption, adjust to your usage ## 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. - [VoIP speed test Measure your upload speed, latency and jitter in 30 seconds.](https://skipcall.io/en/tools/voip-speed-test) - [Mic test A perfect connection won’t save a bad headset: check your audio.](https://skipcall.io/en/tools/mic-test) - [Business VoIP phone service Real costs, network requirements and an 8-point rollout checklist.](https://skipcall.io/en/blog/business-voip-phone-service) - [SIP trunking explained How many channels to plan and what the trunk changes on your network.](https://skipcall.io/en/blog/sip-trunking) - [What is a softphone? Calling from your computer: codecs, headset, network.](https://skipcall.io/en/blog/what-is-a-softphone) - [VoIP over Starlink Satellite handovers, a narrow upload lane and the router settings that fix them.](https://skipcall.io/en/blog/voip-over-starlink) - [POTS line retirement Still on copper? What the phase-out means for your business lines.](https://skipcall.io/en/blog/pots-line-retirement-business) - [All free tools More than thirty tools for teams that live on the phone.](https://skipcall.io/en/tools) ## Sources Standards and documents consulted, checked November 6, 2026. - [IETF RFC 3550: RTP, 12-byte header](https://www.rfc-editor.org/rfc/rfc3550) - [IETF RFC 768 (UDP), RFC 791 (IPv4), RFC 8200 (IPv6)](https://www.rfc-editor.org/rfc/rfc8200) - [IETF RFC 3711: SRTP, 80-bit authentication tag by default](https://www.rfc-editor.org/rfc/rfc3711) - [IETF RFC 7714: AES-GCM for SRTP, 16-byte tag](https://www.rfc-editor.org/rfc/rfc7714) - [IETF RFC 6716: Opus codec, 6 to 510 kbps, 28 to 40 kbps for fullband speech](https://www.rfc-editor.org/rfc/rfc6716) - [ITU-T G.711, G.722 and G.729](https://www.itu.int/rec/T-REC-G.711) - [Cisco: Voice Over IP, Per Call Bandwidth Consumption](https://www.cisco.com/c/en/us/support/docs/voice/voice-quality/7934-bwidth-consume.html) - [WireGuard protocol (16-byte header, 16-byte tag, padding to 16)](https://www.wireguard.com/protocol/) - [IETF RFC 4303 (ESP), RFC 4106 (AES-GCM in ESP), RFC 3948 (NAT-T)](https://www.rfc-editor.org/rfc/rfc4303) - [IETF RFC 4594: DSCP EF marking for telephony](https://www.rfc-editor.org/rfc/rfc4594) - [IETF RFC 2508: IP/UDP/RTP header compression for low-speed serial links, hop by hop](https://www.rfc-editor.org/rfc/rfc2508) - [Arcep: mobile service quality, equivalent 4G and 5G upload (Nov 19, 2021)](https://www.arcep.fr/actualites/les-communiques-de-presse/detail/n/qualite-des-services-mobiles-191121.html) - [Arcep: fixed broadband (cable, DSL, VDSL), archived file](https://www.arcep.fr/la-regulation/grands-dossiers-reseaux-fixes/le-haut-debit-fixe-cable-adsl-vdsl.html) - [Telecompetitor, citing Ookla: Starlink US median upload of 16.91 Mbps (2H 2025)](https://www.telecompetitor.com/starlink-nearing-full-fcc-broadband-compliance-ookla-report/) - [Starlink Residential customer information sheet, 20 to 40 Mbps upload (March 2025)](https://starlink.com/public-files/CIS_Residential_March2025.pdf) ## 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. ## A calendar full of meetings starts with a call. [Get started now](https://app.skipcall.io/signup) [Book a demo](https://skipcall.io/en/demo) **No commitment** · Number live in 2 minutes ## Frequently asked questions **How much bandwidth does a VoIP call need?** 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. **How many concurrent calls fit in 1 Mbps of upload?** 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. **How much bandwidth for 10 or 20 VoIP users?** 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. **Do silence suppression or header compression lower the need?** 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. **Does upload or download speed matter for VoIP?** 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. **Does Wi-Fi use more bandwidth than Ethernet?** 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. **Does a VPN increase the bandwidth I need?** 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. **Which codec should I pick?** 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. **Does this replace a speed test?** 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. **Are my settings saved?** No. Everything is calculated in your browser. The share link only contains the calculator settings.