Choppy audio, dropped calls, and one-way sound are common VoIP complaints - but they're almost always fixable. Here's how to systematically diagnose and resolve business VoIP quality problems.
VoIP (Voice over IP) phone systems are now the standard for Australian businesses. The benefits are well established: lower call costs, flexibility for hybrid teams, integration with Microsoft Teams or other platforms, and far simpler management than legacy PBX systems. But when call quality is poor, the frustration is immediate and the business impact is real - nobody closes a deal over a call where every second sentence is missing.
The good news is that VoIP call quality issues are almost always diagnosable and fixable. The bad news is that the root cause isn’t always obvious, because the problem could sit in any one of several layers: your internet connection, your internal network, your router or firewall, or the VoIP platform itself.
Here’s how to work through it methodically.
Understanding the Three Main Quality Culprits
Jitter
Jitter is the variation in packet arrival time. VoIP audio is transmitted as small packets of data. If those packets arrive at irregular intervals rather than in a steady stream, the receiving end struggles to reassemble them correctly - the result is choppy, robotic-sounding audio.
Jitter is measured in milliseconds. For acceptable voice quality, jitter should be below 30ms. Above 50ms and quality degrades noticeably. Above 100ms and calls become unusable.
Latency
Latency is the delay between when audio is sent and when it’s received. In everyday language, it’s the “lag” on the call - you finish speaking and the other person doesn’t hear it for a beat. One-way latency of more than 150ms is noticeable; above 300ms and conversations become frustrating because both parties end up talking over each other.
Latency is inherent to some degree in any internet-based call, particularly to international destinations. But within Australia, and especially for intra-Melbourne calls, latency should not be a significant issue. If it is, there’s usually a network problem to find.
Packet Loss
Every packet of audio data that gets dropped is a tiny hole in the conversation. Small amounts of packet loss (under 1%) are usually handled by the codec and are imperceptible. Above 3–5% and you’ll hear obvious audio dropouts and broken words. Above 10% and calls become unusable.
Packet loss is often the most serious quality issue because there’s no way to reconstruct lost audio - it’s simply gone.
Step 1: Run a VoIP-Specific Network Test
Before assuming the problem is your internet connection, run a dedicated VoIP quality test from a device inside your network. Tools like PingPlotter, VoIPmonitor, or the web-based MOS test at Meter.net will give you jitter, latency, and packet loss figures from your network to an external server.
Perform this test:
- During business hours (when the problem is occurring)
- From multiple devices (to rule out a single device issue)
- Both over wired Ethernet and over Wi-Fi (separately)
- At different times of day
If Wi-Fi tests show poor quality but wired tests are clean, the problem is your wireless network or the specific device’s wireless card. If both wired and wireless show poor quality, the issue is upstream - your router, firewall, or internet connection.
Step 2: Check Your Internet Connection
VoIP is not bandwidth-hungry - a single concurrent call typically uses 80–100 kbps. Ten concurrent calls uses less than 1 Mbps. The common belief that you need a fast internet connection for VoIP is only partly true. You need a consistent internet connection - one with low jitter and packet loss, not just high throughput.
A 100 Mbps NBN connection with high jitter will produce poor call quality. A 25 Mbps connection with excellent consistency will produce excellent call quality.
Test your connection quality, not just speed. Ask your internet provider about your connection type - NBN FTTN (Fibre to the Node) connections, where the final run is over copper, are more susceptible to quality degradation than FTTP (Fibre to the Premises) or dedicated fibre connections. If your business depends on VoIP quality, an FTTP or dedicated fibre connection is worth the investment.
Step 3: Configure Quality of Service (QoS)
QoS is a router/firewall feature that prioritises certain types of traffic over others. Without QoS, a large file upload or video stream can saturate your connection and steal bandwidth from your VoIP calls, causing immediate quality degradation.
With QoS properly configured, VoIP traffic is tagged as high-priority and gets processed first, regardless of what else is happening on the network.
Most business-grade routers (Cisco Meraki, Fortinet, Ubiquiti) support QoS. Configuration typically involves:
- Identifying VoIP traffic by destination IP range, port (SIP typically uses UDP 5060, RTP uses UDP 10000-20000), or DSCP marking
- Setting it as highest-priority class
- Optionally, capping the bandwidth available to lower-priority traffic (bulk downloads, Windows Update, etc.)
If your router is a consumer-grade device or an ISP-provided modem, it may not support proper QoS. This is a common cause of VoIP quality problems in small businesses that have grown beyond their original setup.
Step 4: Check for Codec Mismatches
A codec is the algorithm used to compress and decompress audio. Common VoIP codecs include G.711 (uncompressed, best quality, most bandwidth), G.729 (compressed, lower bandwidth, slightly lower quality), and Opus (modern, adaptive, used in WebRTC and Teams).
A codec mismatch - where the sending end and receiving end are using different codecs - typically results in no audio rather than poor audio. But misconfigured codec preferences can cause quality issues, particularly if a high-compression codec is being used unnecessarily on a high-bandwidth connection.
Check your PBX or VoIP platform’s codec settings. For calls within Australia, G.711 is usually preferable to G.729 because bandwidth is rarely a constraint and the audio quality is noticeably better.
Step 5: Firewall and NAT Configuration
SIP (Session Initiation Protocol) - the protocol most VoIP systems use for call setup - is notoriously difficult for firewalls to handle correctly. A firewall that isn’t configured to properly handle SIP can cause one-way audio (you can hear the other party but they can’t hear you, or vice versa), calls that drop after exactly 30 seconds, or calls that connect but have no audio at all.
Common fixes:
- Ensure SIP ALG (Application Layer Gateway) is either properly configured or disabled. Many IT professionals disable it entirely on business firewalls because it often causes more problems than it solves.
- Configure proper firewall rules to allow RTP traffic (the actual audio stream) in both directions.
- If you’re using a hosted VoIP provider, check whether they publish a list of IP ranges and ports to whitelist in your firewall.
Step 6: Headsets, Handsets, and Drivers
Don’t overlook the endpoint. A USB headset with a faulty driver, a phone with outdated firmware, or a headset plugged into a noisy USB port can cause quality issues that look like network problems. Test calls using a different headset or handset to rule this out.
For USB headsets, check device manager for driver errors. Update headset firmware if the manufacturer provides updates. Try a different USB port, and if possible, a different computer entirely.
When to Call Your MSP
If you’ve worked through these steps and still have quality issues, it’s time for a professional review. VoIP troubleshooting often requires packet capture and analysis tools to pinpoint exactly where in the call path quality is degrading - this is beyond the scope of most in-house troubleshooting.
CX IT Services manages business phone systems for Melbourne organisations and handles VoIP quality investigations as part of our managed services offering.
Contact us if your business phone quality is letting you down.
