In short
VoIP requirements for a business come down to seven things: an internet connection with enough upload speed for your busiest hour, bandwidth for each simultaneous call, low latency, jitter and packet loss, a LAN that powers and prioritises phones, firewall settings with SIP ALG off, backup power, and compatible devices.
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The VoIP requirements checklist
Use this as a quick audit before you choose or switch a VoIP phone system. Each line is explained further down the page, and the detailed set-up for prioritising voice and configuring routers has its own guide.
| Requirement | What to check | Why it matters |
|---|---|---|
| Internet connection | Upload speed, contention at busy times, a second connection for backup | Every call sends audio both ways, so upload runs out first on most broadband |
| Bandwidth per call | Simultaneous calls multiplied by the codec's bandwidth, plus headroom | Too little bandwidth causes choppy, robotic audio when the line is busy |
| Latency, jitter and loss | One-way delay, variation in delay and lost packets, tested at busy times | Delay makes people talk over each other; jitter and loss break up speech |
| Office network (LAN) | PoE switch ports, a voice VLAN, QoS markings, Wi-Fi tuned for voice | Phones need power and a clear, prioritised path to the router |
| Firewall and router | SIP ALG off, the platform's ports and addresses allowed, stable NAT | Wrong settings cause one-way audio, failed registrations and dropped calls |
| Power and backup | UPS for the router, switches and key phones; a backup connection | Digital phone lines need power to work |
| Devices | Supported desk phones, headsets, apps and analogue adapters | Each platform supports its own list of devices |
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Internet connection: upload speed and contention
Most broadband is asymmetric: much more download than upload. VoIP is symmetric, because each call sends your voice up while it brings the other party's voice down, so the upload figure is the one to plan around.
Contention matters as much as the headline speed. Broadband is shared with other users of the network, so throughput can dip at busy times. A leased line is a dedicated, uncontended connection with the same speed in both directions, which is why larger offices and contact centres often choose one. Our sister sites explain broadband for VoIP and choosing a leased line for VoIP.
Plan for the busiest hour, not an average day: everyone on calls while files sync to the cloud and someone joins a video meeting. If a single connection would carry everything, a second connection from a different route gives calls somewhere to go if the first one fails. See choosing a router for VoIP for dual WAN and 4G or 5G failover.
Bandwidth per call: VoIP bandwidth requirements by codec
A codec turns speech into data. Cisco publishes the bandwidth each common codec uses per call over Ethernet at the standard 20 millisecond packet size, including packet overheads:
| Codec | Typical use | Bandwidth per call (Cisco, Ethernet, 20 ms) |
|---|---|---|
| G.711 | Standard narrowband audio on SIP trunks and desk phones | About 87.2 kbps |
| G.722 | Wideband (HD) audio on desk phones | About 87.2 kbps |
| G.729 | Compressed audio where bandwidth is tight | About 31.2 kbps |
| Opus | Apps and softphones on many platforms | Adapts its rate to the network |
The sum is simple: multiply your peak number of simultaneous calls by the per-call figure, then add headroom for everything else on the line. Ten simultaneous G.711 calls need roughly 872 kbps in each direction; the same ten calls on G.729 need about 312 kbps. Because audio flows both ways, check the result against your upload speed.
Microsoft publishes its own figures for Teams audio: a recommended 58 kbps up and down per endpoint, 76 kbps for best performance and a 10 kbps minimum, and Teams gives audio priority over video when bandwidth runs short. Microsoft's Network Planner in the Teams admin centre can work the numbers out per site. For the number of simultaneous calls you need in the first place, see how many SIP channels on our sister site and our guide to a multi-line VoIP phone system.
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VoIP latency requirements: delay, jitter and packet loss
Bandwidth is not the only measure of a good voice connection. Three others decide how a call sounds:
- Latency (delay): the time audio takes to travel one way. Cisco summarises the ITU-T G.114 recommendation as up to 150 ms one-way being acceptable for most user applications. Beyond that, people start to talk over each other
- Jitter: variation in delay from packet to packet. Phones hold audio in a de-jitter buffer to smooth it out, but too much jitter still breaks up speech
- Packet loss: packets that do not arrive. Small amounts are concealed by the codec; more is heard as clipped words and robotic audio
All three rise when a connection is congested, which is why tests should run at your busiest times, not first thing in the morning. Our VoIP call quality guide explains how to test them with each platform's own tools.
Microsoft also advises keeping real-time calls off VPNs where possible, because a VPN adds overhead and delay. A split-tunnel VPN sends Teams traffic straight to Microsoft while the rest stays in the tunnel.
Office network: PoE, voice VLANs, QoS and Wi-Fi
Inside the building, the network has three jobs: power the phones, keep voice separate and give it priority.
- PoE switches: Power over Ethernet runs desk phones from the switch port, so there is no power adapter at each desk and a UPS on the switch keeps phones up in a power cut
- Voice VLAN: a separate virtual network for phones keeps voice away from bulk data and makes it easy to prioritise. Phones can learn the voice VLAN automatically through LLDP-MED or DHCP
- QoS markings: voice is marked DSCP EF (46) so switches and routers put it at the front of the queue. Our guide to VoIP QoS settings explains the markings each platform uses
- Wi-Fi for voice: softphones and Wi-Fi handsets need good coverage where people take calls. Microsoft recommends WMM or QoS on Wi-Fi, the 5 GHz band, band steering and non-overlapping channels
- Cabling: a phone with a gigabit pass-through port can share one cable with a PC; see VoIP desk phones for the features to look for
Firewall, router and SIP ALG
Many call problems that are not about bandwidth trace back to the firewall or router. The settings that matter are:
- SIP ALG off: 3CX advises a router or firewall without SIP ALG, or one where it can be disabled, and Cisco recommends turning it off for Webex Calling
- The right ports and addresses: each platform publishes what to allow. An on-premise 3CX system, for example, needs SIP on port 5060 and RTP audio on ports 9000 to 10999, two ports per call
- Stable NAT for UDP: Microsoft asks that the firewall does not change NAT addresses or ports for UDP mid-session
- Session timers: firewall timeouts that keep phone registrations and long calls open
Our VoIP router guide covers each setting, and VoIP security covers locking the system down at the same time.
Power and backup
Openreach notes that digital phone lines need power to work. On a VoIP phone system that means the router, the switches and the phones themselves.
- A UPS on the router and PoE switch keeps desk phones working through a short power cut
- Mobile apps carry on over mobile data when the office is dark
- Call forwarding and out-of-hours rules on a cloud platform can send calls to mobiles or another site if the office connection drops
- Alarms, lift lines and other special lines need their own plan; see VoIP adapters
If anyone relies on a landline, for example for a telecare device, Openreach advises telling the provider, who can offer battery backup.
Devices: VoIP hardware requirements
The last requirement is the equipment people use. Each platform supports its own list: 3CX publishes supported IP phones, DECT systems and adapters; Microsoft certifies Teams phones and supports compatible SIP phones through SIP Gateway; Webex Calling works with Cisco devices and the Webex App.
- VoIP handsets for business: desk, conference, cordless and reception devices
- VoIP headsets: wired, USB, Bluetooth and DECT
- Softphone vs desk phone: which users need hardware at all
- VoIP adapters: keeping analogue phones, door entry and franking machines working
Desk and DECT handsets can be bought outright or, in some cases, included in your monthly rental, with the options confirmed at quote.
How we help
Forever Systems can check these requirements as part of your VoIP project. We look at your connection, your network and the devices you rely on, recommend the platform and settings that suit them, and help put any gaps right before go-live, much of it remotely. Our VoIP phone system installation guide shows where the requirements check fits in the wider project.
Contact us with your users, sites and current connection, and we can tell you what your network needs for business VoIP.
Explore VoIP requirements
How Forever Systems helps you choose a VoIP phone system
We supply and support several VoIP platforms, so we can recommend the one that fits how your team takes calls rather than fitting your business around one product.
- Platform advice across hosted 3CX, Microsoft Teams Phone, Webex Calling, Forever UC and Forever Teams, with SIP trunks for phone systems you already own
- A requirements check of your broadband, network and routers before anything goes live
- Numbers and porting: we can help you keep your existing business numbers and plan the move away from analogue and ISDN lines
- Handsets and headsets that suit each platform, bought outright or, in some cases, included in your monthly rental, with the options confirmed at quote
- Set-up and changes: call flows, users and devices configured for you, and help with moves, adds and changes afterwards
- For BT Cloud Voice, see our sister site cloudvoice.cloud, and for hosted and cloud platforms in depth, cloudphone.cloud
We work with businesses, schools, charities and public sector organisations.
Frequently asked questions
What internet speed do I need for VoIP?
Work from upload speed rather than download. Cisco puts a G.711 or G.722 call at about 87.2 kbps per call over Ethernet and a G.729 call at about 31.2 kbps, so ten simultaneous G.711 calls need roughly 872 kbps each way, plus headroom for everything else on the line at your busiest time.
How much bandwidth does a VoIP call use?
It depends on the codec. Cisco's figures, including packet overhead at a 20 ms packet size, are about 87.2 kbps for G.711 and G.722 and about 31.2 kbps for G.729. Microsoft recommends 58 kbps up and down per endpoint for Teams audio. Opus adapts its rate to the network.
What latency is acceptable for VoIP?
Cisco summarises the ITU-T G.114 recommendation as up to 150 ms of one-way delay being acceptable for most user applications. Jitter and packet loss matter too, so test all three at your busiest time of day.
What are the requirements for VoIP?
A business VoIP phone system needs an internet connection with enough upload for your peak simultaneous calls, low latency, jitter and packet loss, a network that powers and prioritises phones, firewall and router settings that allow the platform's traffic with SIP ALG off, backup power, and devices the platform supports.
Can VoIP run over Wi-Fi?
Yes. Softphones, mobile apps and Wi-Fi handsets work over Wi-Fi, provided coverage is good where people take calls. Microsoft recommends WMM or QoS on the wireless network, the 5 GHz band, band steering and non-overlapping channels.
Is broadband good enough for a VoIP phone system?
For many small offices, yes, provided the upload speed covers your peak calls with headroom. Larger offices and call-heavy teams often choose a leased line, which is uncontended and symmetric, with broadband or 4G as backup.
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