Acceptable Jitter & Latency for VoIP: UK Business Guide Quick Answer
For good business VoIP call quality, aim for latency below 150ms one way, jitter below 30ms and packet loss below 1%.
Metric Good target What happens when it is poor
Latency
Below 150ms one way
Delay, awkward pauses and people talking over each other
Jitter
Below 30ms
Choppy, robotic or uneven audio
Packet loss
Below 1%, ideally close to zero
Missing words, audio gaps and distortion
Raw broadband speed is only part of the picture, a stable connection with low jitter and packet loss can provide better VoIP call quality than a much faster but congested connection but if your network meets these targets but calls still sound poor, review your VoIP setup, hardware and provider.
See our VoIP call quality guide for a wider troubleshooting checklist, or compare business VoIP systems if the service itself may be the issue.
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Authored by: Ally Cox (Business Technology & B2B Services Specialist 10+ Years’ Experience)
Reviewed by: James Ward (Telecoms Specialist, 12+ Years Experience)
Last Updated: September 2026
Key Takeaways
Latency measures how long voice data takes to travel across the network.
Jitter measures variation in packet arrival times.
Packet loss occurs when some voice packets never reach their destination.
For good VoIP performance, aim for latency below 150ms one way, jitter below 30ms and packet loss below 1%.
Wi-Fi congestion, poor routing and overloaded connections can cause poor calls even when broadband speed is high.
QoS can help prioritise voice traffic over less time-sensitive network activity.
Ethernet is often more predictable than Wi-Fi for fixed VoIP devices.
If your network metrics are healthy but calls remain poor, investigate the VoIP platform or provider.
What Is Latency in VoIP? Latency is the time it takes voice data to travel from one endpoint to another and is measured in milliseconds.
High latency makes conversations feel delayed – so you’ll likely notice:
awkward pauses
people talking over each other
slow responses
a noticeable gap between speaking and hearing a reply
A widely used target for high-quality real-time voice is less than 150ms one-way latency .
One-way latency Practical interpretation
Under 100ms
Very good
100–150ms
Generally acceptable for business VoIP
Above 150ms
Delay may become noticeable
Much higher latency
Conversation can become increasingly difficult
What Is Jitter in VoIP? Jitter is the variation in how long individual voice packets take to arrive.
VoIP works best when packets arrive at regular intervals – when packet timing varies too much, the phone system has to compensate using a jitter buffer and excessive jitter can result in:
robotic voices
broken or uneven speech
audio dropouts
distortion
As a practical business target, keep jitter below 30ms , with lower values preferable.
Jitter Practical interpretation
Under 20ms
Very good
20–30ms
Generally acceptable
Above 30ms
Greater risk of audible quality problems
Jitter is caused by inconsistent packet delay and can be affected by network congestion, queuing and configuration.
What Is Packet Loss? Packet loss occurs when some of the voice data sent across the network fails to arrive at its destination.
Real-time calls cannot simply wait for missing packets to be resent without introducing further delay, so packet loss can directly affect what the listener hears.
Common symptoms include:
missing words or syllables
short audio gaps
distorted speech
one-way or broken audio
For good business VoIP, packet loss should be below 1% and ideally as close to zero as possible.
VoIP Call Quality Thresholds
Metric Good target Higher-risk level
Latency
Below 150ms one way
Above 150ms
Jitter
Below 30ms
Above 30ms
Packet loss
Below 1%
Above 1%
These are practical planning targets rather than hard pass/fail rules. Codec choice, network design, jitter buffers and the VoIP platform itself also affect the final experience.
Why Jitter and Latency Matter More Than Broadband Speed Alone A fast internet connection does not automatically guarantee good VoIP – for example, a connection with hundreds of megabits of download capacity can still produce poor calls if it has:
high latency
high jitter
packet loss
congested Wi-Fi
poor upload stability
network equipment under heavy load
VoIP is a real-time application, so consistency matters as much as capacity, which is why a stable business connection can outperform a faster but unstable one for voice.
What Causes High VoIP Latency? Often, causes include:
long network routes
internet congestion
VPN routing
overloaded routers
slow or unstable broadband
poor provider routing
geographically distant servers or call destinations
Latency cannot be eliminated entirely because every network has some transmission delay, but excessive delay should be investigated.
What Causes High Jitter? High jitter is often associated with inconsistent network conditions which can be:
Wi-Fi interference
network congestion
poor traffic prioritisation
overloaded routers
large uploads and downloads
misconfigured network equipment
unstable broadband
What Causes Packet Loss? Packet loss can result from:
network congestion
weak Wi-Fi
faulty cables
overloaded equipment
ISP issues
routing problems
network errors
Even relatively small amounts of packet loss can be noticeable on voice because missing packets may translate directly into missing speech.
How Do I Test Jitter, Latency and Packet Loss? Do not rely on a standard download-speed result alone! A useful VoIP network test should measure:
latency
jitter
packet loss
upload capacity
download capacity
Test under normal working conditions, including at busy times when staff are:
joining video calls
uploading files
using cloud applications
running backups
making multiple simultaneous VoIP calls
A quiet-network test can look excellent while peak-time performance is poor.
Should I Test Round-Trip or One-Way Latency? Many network tools report round-trip time (RTT) , while VoIP guidance often refers to one-way delay.
As a rough approximation, one-way latency is often estimated as around half the round-trip result, although real network paths are not always perfectly symmetrical.
How to Reduce VoIP Latency and Jitter 1. Test With Ethernet
2. Reduce Network Congestion
3. Configure Quality of Service
4. Review Your Router and Network Equipment
5. Improve Wi-Fi
6. Check Upload Performance
7. Review VPN Routing
8. Escalate Persistent Problems
Does QoS Help VoIP? Yes, where congestion is the problem.
Quality of Service can classify and prioritise voice packets ahead of traffic that can tolerate delay.
For example, a large software download can take a few extra seconds without affecting the user significantly, while a delayed voice packet can immediately harm a live conversation.
Microsoft’s Teams guidance explains that QoS is specifically intended to prioritise real-time voice and video over less delay-sensitive traffic.
QoS does not fix an unreliable internet connection, but it can improve how a healthy network handles competing traffic.
Does Wi-Fi Increase Jitter? It can! Wi-Fi is affected by factors that wired Ethernet does not face to the same degree, including:
radio interference
signal strength
distance
walls and building materials
channel congestion
other wireless devices
This does not mean VoIP cannot work over Wi-Fi. Well-designed business Wi-Fi can support high-quality calling, but if jitter is high, testing over Ethernet is one of the quickest diagnostic steps.
Microsoft Teams Jitter and Latency Microsoft Teams uses latency, jitter and packet-loss measurements when assessing call and meeting network quality which they describe as:
jitter as variation in network packet delaypacket loss as packets that fail to arrivelatency as the network time delay
Microsoft also provides call-health and Call Quality Dashboard tools to help administrators diagnose individual or organisation-wide quality problems.
For a broader Microsoft calling comparison, see our Microsoft Teams Phone guide .
Is High Jitter a VoIP Provider Problem? Not necessarily!! Jitter can originate from:
your local network
Wi-Fi
your ISP
network congestion
routing
the VoIP provider
the wider internet path
First determine whether the problem affects:
one device
one location
all employees
all call destinations
only particular external numbers
When Do You Need Better Business Connectivity for VoIP? You do not automatically need a leased line for VoIP.
Standard business broadband or FTTP can be suitable where:
call volumes are modest
the connection is stable
upload performance is reliable
network congestion is controlled
Stronger connectivity becomes more relevant where:
you operate many simultaneous calls
voice is business-critical
you run a call centre
your connection regularly becomes congested
you need stronger service guarantees
you need backup connectivity
In those cases, compare business leased lines or see our guide to leased line costs .
Do Leased Lines Improve VoIP Quality? A leased line can provide more predictable connectivity because it offers dedicated business capacity and symmetrical speeds.
However, a leased line does not automatically solve every VoIP problem.
Poor LAN configuration, faulty hardware, bad Wi-Fi or VoIP-platform issues can still affect calls.
It is therefore better to diagnose the problem first and upgrade connectivity only where the underlying connection is actually the constraint.
Remote Workers and VoIP Jitter Remote workers can experience inconsistent VoIP quality because each employee may use a different:
broadband provider
router
Wi-Fi setup
VPN
home network
For call-heavy remote employees:
test their actual connection
use Ethernet where practical
avoid weak Wi-Fi locations
review VPN routing
consider mobile or secondary connectivity as backup
See our VoIP for remote teams guide for wider planning advice.
Latency, Jitter or Packet Loss: Which Is Causing the Problem?
Symptom Likely metric to investigate
Long conversational delay
Latency
People repeatedly talk over each other
Latency
Robotic or metallic voice
Jitter or packet loss
Words disappear
Packet loss
Audio quality changes constantly
Jitter
Calls worsen when network traffic increases
Congestion, jitter and packet loss
Only Wi-Fi calls are poor
Wireless instability/jitter
Our Verdict: What Is Acceptable Jitter and Latency for VoIP? For a reliable business VoIP deployment, use these as practical targets:
Latency: below 150ms one wayJitter: below 30msPacket loss: below 1%, ideally close to zero
If calls are poor but those measurements look healthy, the issue may lie elsewhere in your network, hardware, configuration or VoIP platform.
Our VoIP call quality guide provides a full troubleshooting process.
If your existing phone system itself is no longer reliable or suitable, compare options through our business VoIP phone systems guide .
Frequently Asked Questions What is acceptable latency for VoIP?
A widely used target for high-quality VoIP is below 150ms one-way latency. Lower latency generally produces more natural conversations.
What is acceptable jitter for VoIP?
A practical target is below 30ms jitter. Lower values are preferable because excessive variation in packet arrival times can make speech sound choppy or robotic.
How much packet loss is acceptable for VoIP?
Packet loss should be below 1% and ideally as close to zero as possible. Higher packet loss can cause missing words and audio distortion.
Is 100ms latency good for VoIP?
Yes. Around 100ms one-way latency is comfortably within commonly used business VoIP targets.
Is 200ms latency bad for VoIP?
At around 200ms one-way latency, conversational delay is more likely to become noticeable. The call may still work, but the experience can feel less natural.
Is 30ms jitter bad?
Around 30ms is commonly treated as the upper end of a useful target range. Lower jitter is preferable, particularly for call-heavy businesses.
Why does high jitter make VoIP sound robotic?
High jitter means voice packets arrive at inconsistent intervals. If the phone system cannot smooth that variation effectively, speech can become broken, metallic or robotic.
Does faster broadband reduce latency?
Not necessarily. Broadband speed and latency are different measurements. A fast connection can still have high latency, jitter or packet loss.
Can Wi-Fi cause high jitter?
Yes. Interference, weak signal and congestion can make wireless packet delivery less consistent. Testing the same VoIP call over Ethernet can help identify whether Wi-Fi is contributing.
Can QoS reduce jitter?
QoS can help when network congestion is delaying voice packets by prioritising real-time traffic. It cannot fix faults outside the network or an unstable broadband service.
What latency is acceptable for Microsoft Teams?
Teams performance depends on several network measurements including latency, jitter and packet loss. Microsoft provides call-health and Call Quality Dashboard tools to help identify poor network conditions.
Do I need a leased line for low-latency VoIP?
No. Many businesses can run VoIP successfully over stable business broadband or FTTP. A leased line becomes more relevant for high call volumes, critical voice services or businesses requiring stronger connectivity guarantees.
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Hi, I’m Ally Cox , a senior copywriter and blogger at CompareYourBusinessCosts.co.uk, the UK’s trusted platform for comparing business services.over a decade of experience in the B2B sector, I specialise in simplifying complex topics like leased lines, VoIP, business energy, HR and payroll solutions, accounting software, and EPOS systems .HR, copywriting, and business operations- from clocking-in systems to card machines and office technology .informed, confident decisions when choosing products and services that improve efficiency and save money.
