Calls Dropping or Audio Dipping In and Out
If your calls are breaking up, dipping in and out, or dropping unexpectedly, this is typically related to local network performance, even if your internet appears to be working normally.
Why This Can Happen Even When the Internet Seems Fine
Standard internet activities such as browsing websites, sending emails, or using cloud applications may continue to work without noticeable problems.
However, voice calls are real-time communication and are more sensitive to network conditions such as:
Short bursts of network congestion
Packet loss
Latency (delay)
Jitter (variation in delay)
These issues may not affect general internet use but can significantly impact live audio.
Common Causes
Call quality issues are often linked to:
High network usage (downloads, cloud backups, video streaming)
No Quality of Service (QoS) configured for voice traffic
Phones sharing the same network as general office traffic
Network equipment not optimised for real-time communication
ISP line instability or bandwidth saturation
Overloaded or faulty routers, switches, or firewalls
Recommended Actions
We recommend reviewing the following with your IT provider:
Enable Quality of Service (QoS) for voice traffic on your firewall or switches
Use a separate VLAN for hardware phones where possible
Check bandwidth usage during peak business hours
Test for packet loss, latency, and jitter
Inspect network hardware for overload or faults
Confirm ISP line stability
If these measures are already in place, your IT provider can perform a more detailed review of the network environment.
Important Note
Voice services rely on a stable and prioritised network connection.
Even small levels of congestion or packet loss can affect call quality.
Addressing network optimisation typically resolves call dipping or dropping issues.
What the Numbers on a Speed Test Mean
Many customers check their connection with an online speed test and ask whether the result is good enough for calls. The question is almost always about ping — which is usually the least important number on the page.
Ping is a round trip, and calls only travel one way
A speed test measures how long a small piece of data takes to reach a test server and come back again. A voice call only cares about the journey in one direction, which is roughly half that figure. A ping of 144 ms is about 72 ms in each direction.
The international standard for voice quality (ITU-T G.114) treats anything up to 150 ms in one direction as imperceptible — the conversation feels immediate, and nobody notices a delay. Between 150 ms and 400 ms calls remain perfectly usable, with a slight pause on quick exchanges.
A speed test also measures to whichever server it happened to choose, which may be in another country rather than close to you. If your ping looks high, run the test again against the nearest available server before drawing any conclusion from it.
A guide to the numbers
| Measurement | Good | Acceptable | Worth looking into |
|---|---|---|---|
| Ping (round trip) | Under 100 ms | 100–150 ms | Over 150 ms |
| Jitter | Under 20 ms | 20–30 ms | Over 30 ms |
| Packet loss | 0% | Under 1% | 1% or more |
A ping above 150 ms is a prompt to look a little closer — at what type of connection it is, and at the two measurements below it — rather than a sign that calls will sound poor.
Packet loss and jitter matter more than ping
If you only check one thing, do not check ping.
- Packet loss removes small pieces of the audio outright. Above about 1% it becomes audible as gaps or clipped words.
- Jitter is variation in the spacing of arriving audio. Your phone can smooth out a small amount; past that, the late pieces are discarded and it sounds much like packet loss.
- Latency does not damage the audio at all. It only affects how immediate the conversation feels, and only once it becomes high.
A connection with a very low ping but 3% packet loss will sound noticeably worse than one with a high ping and none.
The measurement a speed test does not show you
A speed test measures your connection while it is idle. Calls do not happen on an idle connection.
On many connections, delay rises sharply the moment the line gets busy — someone uploads a large file, a backup runs, a video call starts. A line showing 25 ms at rest can reach several hundred milliseconds under load, and that is what makes calls choppy or delayed. It is known as bufferbloat, and free online tools will measure it by testing your delay while deliberately loading the connection.
If you see a large rise between the idle and loaded figures, prioritising voice traffic on your router resolves it in most cases. See Enhancing NUACOM Phone System Performance: Recommended Network Settings.
How much speed do you actually need?
Less than most people expect. Each call in progress uses about 85 kbps in each direction, so six people on calls at the same time need roughly 510 kbps up and 510 kbps down — around 1 Mbps in total, on top of your normal internet use. Any modern business connection covers this comfortably. See NUACOM Phone System | Essential network settings | Bandwidth for the full figures.
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