In short
VoIP call quality depends on bandwidth, delay, jitter and packet loss on the path between callers, plus the firewall, Wi-Fi and devices at each end. Match the symptom to its likely cause, test at busy times with your platform's own tools, such as Teams Call Quality Dashboard or Webex Control Hub, then fix the weak link.
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What good business VoIP call quality depends on
A VoIP call is a stream of small audio packets, typically one every 20 milliseconds in each direction. It sounds good when those packets arrive quickly, evenly and all together. Four network measures decide that:
- Bandwidth: enough capacity, especially upload, for every simultaneous call plus everything else on the line
- Latency: the one-way delay. Cisco summarises ITU-T G.114 as up to 150 ms one-way being acceptable for most user applications
- Jitter: variation in delay. Cisco explains that phones use a de-jitter buffer to turn variable delay into a fixed delay, but a buffer can only absorb so much
- Packet loss: packets that do not arrive, heard as clipped words or robotic audio
The rest is local: the firewall and router settings, Wi-Fi, headsets and the phones themselves. On a well-set-up connection, internet VoIP call quality can match a traditional line, and wideband (HD) codecs such as G.722 carry a wider range of speech frequencies than a narrowband landline. The full checklist is in VoIP requirements.
Symptoms and their likely causes
| What you hear | Likely causes | Where to look first |
|---|---|---|
| Choppy, robotic or breaking-up audio | Jitter or packet loss, often from a congested upload, weak Wi-Fi or a VPN | Bandwidth at busy times, QoS on the router, Wi-Fi coverage |
| One-way audio or no audio | SIP ALG rewriting messages, audio ports blocked, NAT changing mid-call | Router and firewall settings |
| Echo | Sound leaking from a speaker into the microphone, a poor headset, or delay that makes normal echo noticeable | Headsets and speakerphones at the far end, then delay |
| Delay and talking over each other | High latency from distance, congestion, VPN tunnels or slow routing | Round-trip time tests, VPN use, routing |
| Calls dropping | Firewall or proxy blocking, UDP timeouts, NAT changes, Wi-Fi roaming, a failing connection | Firewall session timers, Wi-Fi, connection logs |
| Phones not ringing or showing offline | Registrations expiring behind the firewall, SIP ALG | Session timers, SIP ALG, the platform's admin portal |
Microsoft's own troubleshooting notes point the same way: calls with static that cut out or sound robotic suggest jitter or packet loss, and calls that keep dropping suggest firewall or proxy blockers. Cisco notes that echo cancellers are needed once one-way delay passes 25 ms, which is why delay and echo often travel together.
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VoIP call quality test: how to measure your calls
A good VoIP call quality test measures the right things, at the right time, from the right place.
- Note the pattern: which users, which devices, inbound or outbound, internal or external, and at what time of day. A problem on every call from one site points to the network; one user points to their device or Wi-Fi
- Test the connection at a busy time: upload and download speed, latency, jitter and packet loss while normal work is going on, not first thing in the morning
- Use the platform's own tools: each platform records quality data for real calls, which beats any synthetic test
- Test wired and on Wi-Fi: if a wired phone or laptop sounds fine and Wi-Fi does not, the answer is in the wireless network
- Check from outside the office: try the same call on a mobile app over mobile data to separate the office connection from the platform
The platform tools we use:
| Platform | Tool | What it shows |
|---|---|---|
| Microsoft Teams Phone | Call Quality Dashboard (CQD), plus per-user call analytics in the Teams admin centre | Organisation-wide quality trends in CQD, and the network and device details of an individual user's calls |
| Webex Calling | Monitoring > Troubleshooting in Control Hub | Calls graded Good or Poor once they end, hop-by-hop details and whether the problem sits with the caller or the callee |
| 3CX | The 3CX Firewall Checker | Whether firewall and port settings for an on-premise 3CX system are right; 3CX recommends running it after configuring the firewall |
For Teams, Microsoft's Network Planner also works out the bandwidth each site needs before you change anything.
How to improve VoIP call quality
Once the cause is clear, the fixes are usually straightforward:
- Turn SIP ALG off on the router or firewall, a common cause of one-way audio and failed registrations
- Prioritise voice: mark audio DSCP EF (46), give it a priority queue and shape upload. See VoIP QoS settings
- Fix the firewall: allow the platform's published ports and addresses, keep NAT stable for UDP and set session timers longer than phone re-registration. See VoIP router settings
- Take calls off the VPN: Microsoft recommends split-tunnelling so Teams traffic goes straight to Microsoft
- Improve Wi-Fi: WMM, the 5 GHz band, band steering and non-overlapping channels where people take calls
- Upgrade devices: a good wired or DECT headset, or a desk phone instead of a laptop's built-in microphone. See VoIP headsets
- Add capacity: more upload, a second connection or a leased line when the line is simply full at peak times
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When the connection is the problem
If tests show the line itself runs out of upload at busy times, or latency and jitter climb whenever the office is busy, no setting will fully fix it. The options are more bandwidth, a dedicated connection or a second line for voice.
Broadband is shared at busy times, while a leased line is uncontended and symmetric. Our sister sites compare leased line vs broadband and explain broadband for VoIP. For home workers on their own broadband, a wired connection to the home router and a good headset often make the biggest difference.
How we help
Forever Systems can review call quality on the platforms we supply, read the platform's call data alongside your network settings, and recommend the fixes, whether that is a router setting, a QoS policy, better Wi-Fi or a different connection. Much of that work can be done remotely. If you are planning a new system, getting the VoIP requirements right at the start helps you avoid problems later.
Contact us with the symptoms you hear, when they happen and the platform you use, and we can help you find the cause.
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
How do I run a VoIP call quality test?
Test at a busy time of day, measuring upload speed, latency, jitter and packet loss. Then use your platform's call data: Call Quality Dashboard and call analytics for Teams, Control Hub troubleshooting for Webex Calling, and the Firewall Checker for an on-premise 3CX system. Compare wired, Wi-Fi and mobile data to narrow down the cause.
Why is my VoIP call quality bad?
Most poor VoIP call quality comes from jitter or packet loss on a congested upload, weak Wi-Fi or a VPN, or from router settings such as SIP ALG. Echo usually comes from a speakerphone or headset at the other end. Matching the symptom to its cause narrows it down quickly.
What causes one-way audio on VoIP calls?
One-way audio is usually a router or firewall problem: SIP ALG rewriting call set-up messages, audio ports being blocked, or NAT changing during the call. Turning SIP ALG off and allowing the platform's published ports normally fixes it.
Is VoIP call quality as good as a landline?
It can be. On a connection with enough upload bandwidth and voice prioritised on the network, VoIP calls are clear, and wideband codecs carry more of the voice than a narrowband line. Problems come from congestion, Wi-Fi or settings, all of which can be found and fixed.
Does Wi-Fi affect VoIP call quality?
Yes. Weak coverage, interference and busy channels cause jitter and packet loss. Microsoft recommends WMM or QoS on the wireless network, the 5 GHz band, band steering and non-overlapping channels for real-time calls.
How can I improve VoIP call quality for home workers?
Use a wired connection to the home router where possible, a good headset, and keep calls off the corporate VPN with split-tunnelling. If the home broadband upload is small, avoid large uploads during calls.
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