Quick answer

Nearly every VoIP problem is one of seven: choppy or robotic audio, echo, one-way audio, dropped calls, delay, phones that will not register, and calls that never arrive. And nearly every one traces back to the same few causes: a congested or unstable internet connection, a consumer-grade router without voice prioritisation or with a badly behaved SIP ALG, Wi-Fi, a firewall blocking the right ports, or a misconfigured device. The hosted platform itself is rarely the fault, which is inconvenient because it is the thing people ring about. This guide takes each symptom, gives the likely cause and the fix, and says plainly when to call your provider and when to fix your network.

We diagnose these daily. Most take ten minutes once you know where to look; the rest are a line problem that a test would have found before go-live. If your system was installed without one, get a no-obligation quote and we will start with the test.

First, the ten-minute triage

Before chasing a symptom, three questions locate most faults:

  1. Is it everyone or one person? Everyone points to the connection, router or provider. One person points to their device, cable, headset or, at home, their broadband.
  2. Is it constant or at certain times? Constant points to configuration. Lunchtime and mid-afternoon points to congestion - cloud backups, video calls, someone's download.
  3. Is it wired or Wi-Fi? If Wi-Fi, plug the device in and try again. If that fixes it, the answer is the Wi-Fi.

Then check the provider's status page. If they have an incident, stop diagnosing and wait.

Choppy, robotic or breaking-up audio

Cause: jitter or packet loss - voice packets arriving unevenly or not at all - almost always from a congested connection, a saturated upload, or Wi-Fi. Occasionally a genuinely poor line.

Fix: move desk phones onto wired Ethernet; put a business-grade router in front of the connection with Quality of Service configured to prioritise voice (SIP and RTP) above everything else; find and reschedule the bandwidth hog (backups, CCTV uploads, large syncs) or shape it; check the line's upload speed, which on older copper connections is the bottleneck. If the audio is bad when the line is idle, run a proper line test - see what internet speed VoIP needs - and consider an upgrade to FTTP or a leased line.

Echo

Cause: despite the reputation, echo is rarely the network. It is usually acoustic - a speakerphone or a cheap headset feeding the other party's voice back into the microphone - or a handset volume set so high the earpiece leaks into the mouthpiece. Delay makes existing echo more noticeable, which is why it gets blamed on VoIP.

Fix: turn down the handset volume; try a different headset (a certified one with echo cancellation); switch off the speakerphone; check for a second device on the call picking up audio. If echo appears only on calls to one external number, it is at their end. If every user has echo on every call, then look at the connection's latency.

One-way audio

Cause: the call sets up (SIP) but the audio stream (RTP) cannot get back through your router or firewall in one direction. Classic NAT problem, very often caused by a router's SIP ALG feature "helpfully" rewriting packets and breaking them, or by a firewall blocking the RTP port range.

Fix: turn off SIP ALG on the router (it is enabled by default on many consumer and small-business routers, and providers almost universally recommend disabling it); make sure the provider's SIP and RTP ports are allowed outbound and the firewall is not doing strict inspection on them; if you have two routers in a chain (an ISP router plus your own), remove the double NAT by putting the ISP router in bridge or modem mode. Most one-way audio is fixed by the SIP ALG switch alone.

Dropped calls

Cause: if calls drop at a consistent time - often around 30 seconds, or around 15 or 30 minutes - the cause is a NAT session timing out or, again, SIP ALG interfering with the keep-alive traffic. If they drop randomly, the connection is dropping packets or briefly disconnecting, or a Wi-Fi client is roaming between access points.

Fix: for regular drops, disable SIP ALG and increase the router's UDP session timeout, or enable the provider's keep-alive setting on the handsets. For random drops, check the connection log on the router for disconnections, wire the phones, and look at Wi-Fi roaming settings if the app on a laptop is the victim.

Delay (latency)

Cause: high latency on the connection, a congested path, satellite or heavily loaded 4G links, or a VPN carrying the voice traffic the long way round.

Fix: measure it. Under about 150 ms one way is fine; UK fibre is typically far lower. If it is high, check whether voice is going through a VPN (it should not), whether a firewall is doing deep inspection on RTP, and whether the connection itself is the issue. Delay also amplifies echo, so fixing one often quietens the other.

Phones that will not register

Cause: the handset cannot log in to the platform. Wrong credentials, wrong server address, the SIP port blocked by a firewall, a certificate problem after a firmware change, or a device that lost power and came back before the network did.

Fix: reboot the phone after the network is up; check the provisioning credentials; make sure the firewall allows outbound SIP (commonly 5060/5061 or the provider's alternative) and the RTP range; check whether the provider has changed a server address or certificate. If several phones fail at once after a change, the change is the cause.

Calls that never reach you

Cause: the phone is registered but inbound calls go straight to voicemail or ring out. Usually a call-flow rule - do not disturb, a forward left on, an out-of-hours schedule still active, a hunt group with nobody in it - rather than a fault. Occasionally a NAT timeout means the platform cannot reach the phone, which also shows as calls arriving only when the phone has recently made a call.

Fix: check the user's presence and forwarding, the time schedule, and the group membership in the portal. If inbound only works right after an outbound call, it is a NAT timeout: keep-alives or a shorter registration interval fix it.

Wi-Fi

Not a symptom but the cause of so many of the others that it deserves its own heading. Wi-Fi adds jitter, contention and roaming gaps that voice cannot tolerate and web browsing never notices. Wire every desk phone. If phones or laptops must be wireless, give voice its own network on business-grade access points with enough capacity, and expect the mobile app on 4G to outperform bad office Wi-Fi.

When it is the provider

Sometimes it is. Signs: a status page incident; every customer of theirs affected at once; problems that persist after your network has been ruled out with wired tests during quiet hours; audio problems on calls between two of your own sites that both test clean. A good provider will look at their side quickly and tell you what they see - call detail records show packet loss and jitter per call from the platform's point of view, which is the fastest way to separate network from platform. If your provider cannot show you that, it is a data point about the provider; see how to choose a hosted telephony provider.

Preventing all of it

  • A line test before go-live, repeated at busy hours.
  • A business-grade router with QoS, SIP ALG off, no double NAT.
  • Wired desk phones, PoE switch, voice VLAN where the network allows.
  • A 4G or 5G failover for the day the line fails.
  • Firmware kept current on router and handsets.
  • A provider whose support answers the phone.

The bottom line

VoIP problems are almost always network problems wearing a VoIP disguise: congestion, Wi-Fi, a consumer router with SIP ALG on, a firewall in the way. Triage by who, when and wired-or-not; fix the router and the cabling; and expect the platform to be the last thing wrong rather than the first. Installed properly, hosted telephony has fewer faults than the lines it replaced.

If your phones were installed without the groundwork, we will do the test and fix the network before touching the platform: get a no-obligation quote or call 0333 006 9399. Our hosted telephony service page explains how the assessment works.

Not sure who is calling one of your numbers? Our free UK phone number checker shows which network or provider Ofcom allocated any number to, and whether the range is in service — no sign-up.

Frequently asked questions

Why does my VoIP call sound choppy or robotic?

Jitter or packet loss - voice packets arriving unevenly or not at all - usually from a congested connection, a saturated upload or Wi-Fi. Wire the phones, prioritise voice with QoS on a business router and move heavy uploads off peak. If it persists on an idle line, test the line itself.

How do I fix one-way audio on VoIP?

Turn off SIP ALG on your router, check the provider's RTP port range is allowed through the firewall, and remove any double NAT by putting the ISP router into modem mode. One-way audio is almost always a NAT or ALG problem.

Why do my VoIP calls drop after 30 seconds?

A NAT session timing out or SIP ALG interfering with keep-alive traffic. Disable SIP ALG, increase the router's UDP timeout or enable keep-alives on the handsets. Random drops, rather than regular ones, point to the connection or Wi-Fi.

What causes echo on VoIP calls?

Usually acoustic feedback from a speakerphone, a cheap headset or an over-loud handset earpiece, made more noticeable by any delay. Lower the volume, try a certified headset, avoid speakerphone. Echo on every user's every call is the only case that points at the network.

Is Wi-Fi OK for VoIP phones?

Not for desk phones - wire them. Wi-Fi adds jitter, contention and roaming gaps that voice cannot tolerate. If devices must be wireless, use a dedicated voice network on business-grade access points, and expect the mobile app on 4G to beat poor office Wi-Fi.

What is SIP ALG and should I turn it off?

A router feature that rewrites VoIP packets to help them through NAT and, in practice, usually breaks them - causing one-way audio, registration problems and dropped calls. Almost every hosted telephony provider recommends disabling it. It is on by default on many routers.

How do I know if the problem is my provider?

Check their status page, ask whether other customers are affected, and rule out your network with wired tests at quiet times. Ask the provider for the call detail records showing jitter and packet loss from their side; a good provider will share them within the hour.