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Common VoIP Faults and How to Fix Them: Troubleshooting Guide

Common VoIP Faults and How to Fix Them: Troubleshooting Guide

Last updated: April 2026

VoIP phone systems are now the standard for UK businesses. They are cheaper, more flexible, and packed with features that traditional lines could never match. But like any technology that depends on your internet connection, things can go wrong.

The good news? Most VoIP faults are predictable, well understood, and fixable. At Compare The Networks, we have supported VoIP systems for UK businesses since 2008. We have diagnosed thousands of VoIP issues across every type of office, from two person startups to 200 seat contact centres. The same problems come up again and again.

This guide covers the most common VoIP faults, what causes them, and exactly how to fix them. We will also help you figure out which problems you can sort yourself and which ones need your provider.


Quick Diagnosis: What Fault Are You Experiencing?

Before we get into the detail, use this table to jump straight to your problem.

SymptomMost Likely CauseDifficulty to Fix
Choppy or robotic audioJitter, packet loss, bandwidthEasy, usually broadband
One-way audio (you hear them, they cannot hear you)NAT/firewall/SIP ALG issueMedium, router config
Dropped calls after 30 secondsSIP session timeout / NAT issueMedium, router config
Echo on callsAcoustic feedback or network delayEasy to medium
No dial tone / phone not registeringRegistration failure, credentials, DNSMedium
Calls going to wrong extensionCall routing misconfigurationEasy, admin portal
Poor quality on WiFi but fine on wiredWiFi congestion or interferenceEasy, use ethernet
Phones dead after power cutNo PoE or UPS in placeEasy, hardware fix
Caller ID showing wrong numberCLI configuration issueEasy, provider setting
Number porting delaysPorting process / losing providerPatience required

Now let us go through each one properly.


Fault 1: Choppy or Robotic Audio

What it sounds like

Words break up mid sentence. Voices sound robotic or metallic. Syllables go missing. It sounds like the other person is talking through a fan.

What causes it

Choppy audio is almost always caused by problems with your internet connection, not with the VoIP system itself. Three things are usually responsible:

Jitter is when data packets arrive at irregular intervals instead of in a steady stream. Your VoIP phone needs packets to arrive in order, at consistent intervals. When they arrive out of order or with variable timing, the audio breaks up.

Packet loss is when some data packets never arrive at all. Your voice is broken into tiny packets, sent across the internet, and reassembled at the other end. If 2% or 3% of those packets go missing, the audio has gaps.

Insufficient bandwidth is when your broadband connection does not have enough upload capacity. Each VoIP call uses about 100Kbps of upload bandwidth. If your upload speed is already maxed out by file uploads, cloud backups, or video calls, there is nothing left for VoIP.

How to fix it

  1. Run a speed test. Go to speedtest.net from a computer on your office network. Pay attention to upload speed (you need at least 5Mbps, ideally 10Mbps+), ping (under 30ms is good), and jitter (under 10ms is good).

  2. Check what else is using your bandwidth. Cloud backups running during business hours are a common culprit. Schedule large uploads and backups for evenings or weekends.

  3. Enable QoS (Quality of Service) on your router. QoS tells your router to prioritise VoIP traffic over everything else. This is the single most effective fix for choppy audio. Your router admin page should have a QoS section. Set VoIP traffic to highest priority.

  4. Use wired connections. Plug your VoIP phones into the router with ethernet cables. WiFi adds latency and jitter that wired connections do not have.

  5. Upgrade your broadband if needed. If your upload speed is under 5Mbps, no amount of QoS tuning will fix choppy audio. We recommend Sky Business Broadband SOGEA 80/20 at GBP35+VAT for VoIP, the 20Mbps upload gives you plenty of headroom.

If you have tried all of this and calls are still choppy, get in touch for a free VoIP assessment. There may be a deeper network issue we can diagnose remotely.


Fault 2: One-Way Audio

What it sounds like

You can hear the other person perfectly, but they cannot hear you. Or the reverse, they can hear you but you hear nothing. The call connects, the line is open, but audio only travels in one direction.

What causes it

One-way audio is one of the most frustrating VoIP faults because the call technically works, just not properly. It is almost always caused by a network configuration issue, specifically how your router handles NAT (Network Address Translation).

Your VoIP phone sits behind your router on a private IP address (like 192.168.1.x). When it makes a call, the audio stream needs to travel from that private address out to the internet and back again. If your router does not correctly translate between your private network and the public internet, audio can flow one way but not the other.

The most common cause is SIP ALG (Session Initiation Protocol Application Layer Gateway). This is a feature built into most business routers that is supposed to help VoIP traffic pass through NAT. In practice, it almost always makes things worse. It rewrites SIP packets in ways that confuse VoIP systems and break audio paths.

How to fix it

  1. Turn off SIP ALG on your router. Log into your router admin panel. Look for SIP ALG, ALG, or Application Layer Gateway in the firewall or NAT settings. Disable it. This single change fixes one-way audio in roughly 70% of cases.

  2. Enable STUN on your VoIP phone. STUN (Session Traversal Utilities for NAT) helps your phone discover its public IP address and set up audio paths correctly. Your VoIP provider should give you a STUN server address to enter in your phone settings.

  3. Check your firewall. Make sure your firewall is not blocking the RTP port range that VoIP uses for audio (typically UDP ports 10000 to 20000). SIP signalling uses port 5060 (UDP) or 5061 (TLS), but the actual audio stream uses RTP on a different range.

  4. Contact your provider. If disabling SIP ALG and enabling STUN does not fix it, your provider may need to adjust settings on their end, particularly if they use a Session Border Controller (SBC) that needs to be configured for your network.


Fault 3: Dropped Calls

What it sounds like

Calls disconnect after a set time, often exactly 30 seconds or 60 seconds. Or calls drop randomly throughout the day with no pattern.

What causes it

Calls dropping at a fixed interval (30 or 60 seconds) are almost always a NAT timeout issue. Your router assigns a temporary mapping between your phone's private IP and its public IP for each call. If the router times out that mapping before the call ends, the call drops. Many routers have a default UDP timeout of 30 or 60 seconds.

Random drops throughout the day are usually caused by broadband instability. If your connection drops for even half a second, the call drops with it. This is common on older FTTC connections, congested networks, or where the broadband line has a physical fault.

How to fix it

  1. Increase your router's UDP/NAT timeout. In your router settings, look for NAT timeout or UDP session timeout. Increase it to at least 120 seconds, ideally 300 seconds. This gives VoIP sessions enough time to send keep-alive packets.

  2. Enable SIP keep-alives. Most VoIP phones and softphones can send periodic keep-alive packets to prevent NAT timeouts. Set the keep-alive interval to 30 seconds in your phone's SIP settings. This tells the router "this connection is still active, do not drop it."

  3. Check your broadband stability. Log into your router and look at the connection log. If you see frequent resyncs or disconnections, your broadband line may have a fault. Contact your broadband provider. If you are on an unreliable connection, consider switching to a dedicated line. Our broadband guide covers options.

  4. Check for firmware updates. Outdated router firmware is a surprisingly common cause of dropped VoIP calls. Check your router manufacturer's website for the latest firmware.


Fault 4: Echo on Calls

What it sounds like

You hear your own voice coming back to you with a slight delay. It feels like talking in a large empty room. Sometimes the echo is loud, sometimes it is subtle but just enough to be distracting.

What causes it

Echo has two possible sources.

Acoustic echo happens when sound from your phone's speaker feeds back into the microphone. This is common with handsfree/speakerphone mode, cheap headsets with poor noise isolation, and open-plan offices where phones are close together.

Network echo happens when there is excessive latency (delay) on the call. If the round-trip delay exceeds about 150 milliseconds, you start hearing your own voice bounced back. This is more common on international calls or calls that route through multiple network hops.

How to fix it

  1. Reduce speaker volume. If you are using handsfree or speakerphone mode, turn the volume down. The louder the speaker, the more sound leaks into the microphone and creates a feedback loop.

  2. Use a headset. A proper headset with a close-talking microphone eliminates acoustic echo almost completely. It does not need to be expensive. A GBP20 to GBP30 USB headset will do.

  3. Check your latency. Run a ping test to your VoIP provider's server. If the round-trip time exceeds 100ms, network echo is likely. This could be a broadband issue, a routing issue, or a problem with your provider's network. Contact us and we can diagnose it.

  4. Update phone firmware. VoIP handsets have built-in echo cancellation algorithms. Older firmware may not handle echo as well as newer versions. Check for updates from your handset manufacturer (Yealink, Snom, Grandstream, etc.).

  5. Check the line at the other end. Sometimes the echo is not on your side at all. If only certain callers have echo, the problem may be with their equipment.


Fault 5: Phone Not Registering (No Dial Tone)

What it sounds like

You pick up the handset and there is no dial tone. The phone's display might show "No Service", "Registration Failed", "401 Unauthorised", or similar. The phone powers on but cannot connect to your VoIP service.

What causes it

Registration failures happen when your phone cannot authenticate with your VoIP provider's server. Common causes include:

  • Wrong credentials. The SIP username, password, or domain entered in the phone is incorrect.
  • DNS failure. The phone cannot resolve the provider's server address.
  • Firewall blocking. Your firewall is blocking SIP traffic on port 5060 (UDP) or 5061 (TLS).
  • Provider outage. Your VoIP provider is having a service issue.
  • IP address change. Your broadband connection got a new public IP and the phone has not re-registered.
  • Expired account or licence. Your VoIP service subscription has lapsed.

How to fix it

  1. Reboot the phone. Unplug it for 10 seconds, plug it back in. This forces a fresh registration attempt. This fixes the problem about 30% of the time.

  2. Check the network cable. Make sure the ethernet cable is firmly connected at both ends, phone and router/switch.

  3. Verify SIP credentials. Log into the phone's web interface (find its IP address on the phone's menu, then type that IP into a browser). Check that the SIP username, password, authentication name, and registrar address are all correct. A single typo will prevent registration.

  4. Check your internet connection. Can other devices access the internet? If your broadband is down, the phone cannot register. Fix the broadband first.

  5. Check your firewall. Ensure ports 5060 (SIP UDP), 5061 (SIP TLS), and 10000 to 20000 (RTP audio) are open for outbound traffic.

  6. Contact your provider. If the phone's credentials are correct and your internet is working, the issue may be on the provider's side. Check their status page first, then call their support line.


Fault 6: Calls Going to the Wrong Extension

What it sounds like

An incoming call rings on the wrong person's phone. Or a caller selects option 2 on your auto-attendant but gets connected to the wrong department.

What causes it

This is a configuration issue in your VoIP system's admin portal, not a technical fault. Common causes:

  • Call routing rules are incorrect. The wrong extension number is assigned to a menu option or ring group.
  • Ring groups have stale members. An employee has left but their extension is still in the ring group, or a new starter has not been added.
  • Time-based routing is wrong. Out-of-hours routing is kicking in at the wrong time, possibly because the system timezone is set incorrectly.
  • DDI (Direct Dial In) mapping is wrong. A direct dial number is pointed at the wrong extension.

How to fix it

  1. Log into your VoIP admin portal. Every VoIP system (3CX, Horizon, 8x8, etc.) has a web-based management interface. Log in and review your call routing.

  2. Check your auto-attendant menu. Verify that each menu option points to the correct extension or ring group.

  3. Review ring groups. Make sure the right people are in the right groups. Remove anyone who has left. Add anyone new.

  4. Check DDI mappings. If you have direct dial numbers for specific staff, make sure each DDI routes to the correct extension.

  5. Check time conditions. If routing changes by time of day, verify the system clock and timezone are correct. British Summer Time changes catch people out every year.

This is something we help businesses with regularly. If your call routing has become a mess over time, get in touch and we can review and rebuild it properly.


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Fault 7: Poor Call Quality on WiFi

What it sounds like

Calls are fine when the phone is plugged into an ethernet cable, but choppy, delayed, or dropping when connected over WiFi. This applies to softphones on laptops and mobile apps as well as WiFi-connected desk phones.

What causes it

WiFi was not designed for real-time voice communication. It works well for email, web browsing, and file downloads because those applications can tolerate brief delays and retransmissions. VoIP cannot. A 200ms delay loading a web page is invisible. A 200ms delay in a phone call is obvious.

Specific WiFi problems that affect VoIP:

  • Channel congestion. If your WiFi channel is shared with neighbouring networks (common in office buildings and high streets), interference causes packet loss and jitter.
  • Distance from the access point. WiFi signal strength drops with distance. A weak signal means more packet retransmissions and higher latency.
  • Too many devices. Consumer-grade routers struggle when more than 15 to 20 devices are connected simultaneously.
  • Microwave ovens and Bluetooth. Both operate on the 2.4GHz band and cause interference with WiFi.

How to fix it

  1. Use wired connections for desk phones. Always. There is no substitute. Ethernet gives consistent, low-latency connectivity that WiFi cannot match.

  2. Use 5GHz WiFi for softphones and mobiles. The 5GHz band is less congested than 2.4GHz and offers lower latency. Make sure your access point broadcasts a separate 5GHz SSID and connect your VoIP devices to it.

  3. Upgrade your access point. If you are using the WiFi built into your broadband router, consider a dedicated business-grade access point. Ubiquiti, Meraki, and Aruba all make access points with QoS features that prioritise voice traffic.

  4. Reduce WiFi congestion. Move devices that do not need WiFi onto wired connections. This frees up WiFi capacity for the devices that genuinely need it.


Fault 8: Phones Not Working After a Power Cut

What it sounds like

The power goes out. When it comes back, your VoIP phones do not work even though your broadband is back online.

What causes it

VoIP phones need both power and an internet connection. Unlike traditional analogue phones that draw power directly from the phone line, VoIP phones are powered either by a mains adapter or by Power over Ethernet (PoE) from a network switch.

When the power goes out:

  • Your broadband router turns off, so there is no internet
  • Your PoE switch turns off, so phones connected via PoE lose power
  • Your phones lose their registration with the provider

When power returns, your router needs to reconnect to the internet (this can take 2 to 5 minutes), then your phones need to re-register with the VoIP provider (another 1 to 2 minutes). If the router comes back with a different public IP address, phones may need a manual reboot to register on the new IP.

How to fix it

  1. Install a UPS (Uninterruptible Power Supply). A small UPS (around GBP80 to GBP150) can keep your router and PoE switch running for 30 to 60 minutes during a power cut. This keeps your phones operational for short outages.

  2. Set up failover to mobiles. Configure your VoIP system to automatically forward calls to business mobile phones if your desk phones go offline. Most VoIP platforms support this as a standard feature. It means callers never hear a dead line, even during extended outages.

  3. Wait for the full restart sequence. After power is restored, give it 5 minutes. Router first, then switch, then phones. If phones have not registered after 5 minutes, reboot them manually (unplug, wait 10 seconds, plug back in).

  4. Check that your router reconnected. Sometimes routers do not automatically reconnect after a power cut, especially if authentication is required. Log into the router and verify the WAN connection is active.


Fault 9: Caller ID Showing Wrong Number

What it sounds like

When you make outbound calls, the recipient sees the wrong number on their phone. Or no number at all, it shows as "Withheld" or "Unknown" even though you have set a caller ID.

What causes it

Caller ID (CLI, Calling Line Identity) on VoIP calls is configured in your VoIP system settings. Unlike traditional lines where the number was tied to the physical line, VoIP lets you set any number as your outbound CLI, as long as you own that number.

Common causes of wrong caller ID:

  • Default CLI not set. Your VoIP system may be sending its default/system number rather than your main business number.
  • Per-extension CLI not configured. Individual extensions may need their outbound CLI set separately.
  • Number not verified. VoIP providers require you to verify ownership of any number you want to display as CLI. If you have not verified your number, calls may show as withheld.
  • Porting not complete. If you recently ported a number to your VoIP provider, CLI may not update until the port completes fully.

How to fix it

  1. Check your system CLI settings. In your VoIP admin portal, look for "Outbound CLI", "Caller ID", or "Presentation Number". Set this to your main business number.

  2. Check per-extension settings. Some systems allow each extension to have its own CLI. Make sure each person's extension shows the correct number.

  3. Verify the number with your provider. If you are trying to display a number that you ported in, confirm with your provider that the port is fully complete and the number is verified for CLI use.

  4. Check OFCOM compliance. Under UK regulations, you must present a valid, callable number as your CLI. Displaying a fake or non-working number is against the rules.


Fault 10: Number Porting Delays

What it sounds like

You have requested to move your existing phone number to your new VoIP provider. It has been weeks and the port has not happened. Or the port date keeps being pushed back.

What causes it

Number porting in the UK should take 10 to 15 working days for a standard geographic number. But delays are common. Typical causes:

  • The losing provider is slow to release. Some providers drag their feet on porting, especially if you are leaving a contract early.
  • Incorrect information on the port request. If the account name, address, or number details on the port request do not exactly match what the losing provider has on file, the port gets rejected.
  • Multi-line ports. If you are porting multiple numbers or an entire number range, the complexity increases and so does the time.
  • ISDN to VoIP ports. Porting from ISDN/PBX systems to VoIP can take longer than standard line ports.

How to fix it

  1. Double check the information on your port request. The account holder name and address must match exactly what your current provider has on file. Even a difference between "Ltd" and "Limited" can cause a rejection.

  2. Contact the losing provider. Ask them for the status of the port. If they are the bottleneck, a polite but firm call can speed things up.

  3. Use a temporary number. While waiting for the port, your VoIP provider can give you a temporary number. Set up a divert from your old number to the temporary one so you do not miss calls.

  4. Escalate through your new provider. Your new VoIP provider can chase the port on your behalf. At Compare The Networks, we handle the porting process for you and chase any delays directly. Get in touch if your port is stuck.


When to Fix It Yourself vs. Call Your Provider

Not every VoIP fault needs a support call. Here is a guide to what you can handle and what needs professional help.

Fix It YourselfCall Your Provider
Reboot phones and routerRegistration failures after checking credentials
Enable QoS on your routerPersistent one-way audio after disabling SIP ALG
Disable SIP ALGNumber porting issues
Switch from WiFi to wiredService-wide outages
Adjust speaker volume for echoPersistent call quality issues after broadband checks
Update phone firmwareCLI/Caller ID not working after configuration
Check call routing in admin portalFirewall or network issues beyond basic settings
Install a UPS for power resilienceSBC or server-side configuration

If you have worked through the self-help steps and the problem persists, do not keep guessing. A five minute call to your provider (or to us) can often identify an issue that you would spend hours chasing on your own.


Broadband Requirements for Reliable VoIP

Many of these faults come back to broadband. Here is what you need for trouble-free VoIP.

MetricMinimum for VoIPRecommendedWhat Happens If Too Low
Upload speed5Mbps10Mbps+Choppy audio, dropped calls
Download speed5Mbps20Mbps+Poor audio on incoming calls
Latency (ping)Under 50msUnder 20msEcho, audio delay
JitterUnder 30msUnder 10msRobotic/choppy audio
Packet lossUnder 1%Under 0.1%Missing words, gaps in audio

Each simultaneous VoIP call uses approximately 100Kbps of upload bandwidth. For a 10 person office where 5 people might be on calls at once, you need at least 500Kbps just for VoIP, plus whatever your other applications need.

We recommend Sky Business Broadband SOGEA 80/20 at GBP35+VAT for VoIP-dependent businesses. The 80Mbps download handles everything else. The 20Mbps upload gives you plenty of capacity for VoIP with headroom to spare. Check our broadband guide for more details.


Preventing VoIP Faults Before They Happen

The best approach to VoIP troubleshooting is to prevent faults in the first place. Here is what we set up for every client:

  1. Proper broadband. We check your upload speed, latency, and jitter before recommending a VoIP system. If your broadband is not up to the job, we sort that first.

  2. Router configuration. SIP ALG off. QoS enabled with VoIP prioritised. NAT timeouts set correctly. Firewall rules configured for SIP and RTP.

  3. Wired connections. Every desk phone on ethernet. No exceptions. WiFi is for laptops and mobiles, not desk phones.

  4. Failover routing. Calls automatically redirect to business mobiles if desk phones go offline. No dead lines, ever.

  5. Regular firmware updates. VoIP handsets and routers kept up to date. Manufacturers like Yealink, Grandstream, and Snom release updates that fix bugs and improve call quality.

  6. Monitoring. We keep an eye on call quality metrics so we can spot problems before you notice them.


Frequently Asked Questions

Why does my VoIP sound fine in the morning but terrible in the afternoon?

This is almost certainly a bandwidth issue. In the morning, fewer people in your office and in your neighbourhood are using the internet. By afternoon, bandwidth is more congested. Cloud applications, video calls, file syncing, and other offices sharing the same exchange all compete for capacity. Enable QoS on your router to prioritise VoIP traffic, and consider upgrading your broadband if the problem persists.

Can I use my old analogue phones with VoIP?

Not directly. Analogue phones use a different signalling protocol. However, you can buy an Analogue Telephone Adapter (ATA) that converts analogue signals to VoIP. ATAs cost around GBP30 to GBP50 and let you plug in a standard phone. That said, you will miss out on many VoIP features (visual voicemail, call transfer screens, directory) that only work with proper IP phones. A Yealink entry-level VoIP phone starts from around GBP50, which is barely more than an ATA.

My VoIP provider says the problem is my broadband. My broadband provider says the problem is VoIP. Who is right?

This is incredibly common, and frustrating. In our experience, the VoIP provider is right about 80% of the time. The majority of VoIP faults are caused by broadband issues (insufficient upload speed, high jitter, SIP ALG interference). Start by running speed tests at different times of day. If upload speed is under 5Mbps or jitter is over 30ms, the broadband needs attention. If broadband metrics are fine, then the VoIP configuration or provider may be the issue. If you are stuck between the two, call us. We work with both sides and can pinpoint where the fault actually sits.

Do I need a separate broadband line just for VoIP?

For most small businesses (under 20 users), no. A single good quality broadband connection with QoS enabled can handle both VoIP and data traffic. For larger offices or businesses where call quality is absolutely critical (e.g. sales teams, contact centres), a dedicated voice line can be worth the extra cost. It guarantees bandwidth for VoIP without competing with other traffic.

How long should it take for my VoIP phones to come back online after a broadband outage?

Typically 5 to 10 minutes. Your router needs to reconnect to the internet first (2 to 5 minutes), then each phone needs to re-register with the VoIP provider (1 to 2 minutes). If phones have not come back after 10 minutes, try rebooting them manually. If they still will not register, contact your provider as there may be a registration issue related to your IP address changing.

Will the PSTN switch-off in January 2027 affect my existing VoIP system?

If you are already on a hosted VoIP system, the PSTN switch-off should not affect you directly. Your calls already travel over the internet, not copper lines. However, if your broadband is delivered over a line that requires the PSTN (ADSL, for example), you may need to switch to a SOGEA or FTTP broadband product. Check with your broadband provider. Our PSTN switch-off guide covers this in detail.


Still Stuck? We Can Help.

We have been fixing VoIP problems for UK businesses since 2008. Whether it is a simple router setting or a complex network issue, we will find it and fix it. We work with all the major VoIP platforms including 3CX, and we can recommend the right broadband to underpin it all.

Get a free VoIP assessment. We will review your current setup, diagnose any faults, and recommend the right solution, whether that is fixing what you have or switching to something better.

Compare The Networks. OFCOM regulated. 4.3/5 on Trustpilot from 1,000+ reviews. Helping UK businesses get their phones right since 2008.

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