Bad quality audio can be frustrating, especially if you are in the middle of a business call - words get clipped, voices sound robotic or, in a worse case, the call drops entirely. It can disrupt the negotiations or create a bad first impression for the customer. Therefore, this is the issue which needs to be resolved quickly.
This diagnostic guide breaks down the technical details behind these issues, shows you how to locate what is causing the issue, and outlines step-by-step solutions to restore good call quality.
The Three Causes of Poor VoIP Call Quality
The good news is that poor VoIP performance comes down to three root causes: latency, jitter, and packet loss, and all three are fixable.
Latency
Latency is the delay between speaking and the other person hearing it. For clear, natural-sounding business calls, one-way latency should remain under 150 milliseconds.
The Most Common Causes of Latency Can Be:
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NBN Connection Type: Your physical connection plays a massive role.
- Fibre to the Premises (FTTP) delivers raw light signals straight to your door with minimal delay (typically < 5–15 ms).
- Fibre to the Node (FTTN/B/C) relies on aging copper lines from a neighbourhood cabinet, introducing electrical resistance causing higher base latency (15–45+ ms).
- HFC delivers internet over the pay-TV coaxial cable network, which offers low base latency (10–25 ms) and strong overall bandwidth. However, because HFC relies on a shared node architecture within local neighbourhoods, performance depends heavily on localized traffic.
- Fixed Wireless connection is used, voice data is transmitted through radio frequencies over the air from your roof-mounted antenna to a local NBN transmission tower. Over-the-air transmission inherently introduces higher base ping times (25–80+ ms) and makes data streams susceptible to atmospheric interference.
- Geographic Distance to VoIP Infrastructure: Voice packets travel very fast, but distance still adds milliseconds. If your VoIP provider routes traffic through overseas servers instead of local Australian data centres (e.g., Sydney, Melbourne, or Perth), your latency naturally increases.
- Router Processing Delay: Older or underpowered office routers struggle to process heavy concurrent traffic, causing a bottleneck as data queues up to leave the building.
Jitter
Jitter is inconsistent latency, packets arriving at irregular intervals, causing choppy or robotic audio. In order to have high quality audio during the calls, it needs to be under 30ms. For instance, if Packet A takes 20ms to arrive, Packet B takes 120ms, and Packet C takes 30ms, your phone receives data at a wildly unpredictable rhythm resulting in robotic, unnatural voices, choppy speech, or scrambled sentences.
To combat irregular packet arrivals, VoIP devices and software use a jitter buffer. This buffer temporarily holds incoming packets, reorders them sequentially, and plays them back at a smooth, continuous rate. It’s important to keep it at the right level as jitter buffer being too small causes packets arriving late being discarded, leading to a choppy audio, while if it’s too large, the buffer holds onto packets for too long, introducing noticeable audio delay/latency.
Packet Loss
When data travels across the internet, packets can be dropped along the way due to poor network or hardware. Unlike regular web traffic (like loading a webpage), which can simply request the missing data to be re-sent, VoIP operates in real time using UDP (User Datagram Protocol), therefore it cannot pause a live call to fetch missing audio.
If a packet disappears, that section of your voice is lost forever. Packet loss must remain under 1% for acceptable voice quality. Anything above 2–3% will cause noticeable audio clipping, silent gaps, or dropped calls.
Common Causes of Packet Loss
- Local Network Congestion: Large file downloads, cloud backups, or video streaming can cause the Packet Loss.
- Faulty Hardware: Old switches, degraded Ethernet cables, or failing routers may drop packets during traffic bursts.
- Wi-Fi Interference: Signal can be interrupted because of walls, office equipment, or overlapping wireless channels.
- Misconfigured Routers: Incorrect settings or outdated router firmware may accidentally discard legitimate voice packets.
How to Test Your VoIP Call Quality
If you experience your calls sounding muffled, choppy, or delayed, run a diagnostic test to see what is causing the issue:
A standard internet speed test (such as Speedtest.net) measures raw download and upload bandwidth, but it does not accurately measure VoIP readiness. A connection can have 100 Mbps download speed while simultaneously failing a voice quality test due to high jitter or intermittent packet loss. To verify if your connection is ready for high-quality voice calls, check the following thresholds:
| Over | Target Ideal | Acceptable Limit | Poor Quality Threshold |
| Latency (One-Way) | < 80 ms | < 150 ms | > 200 ms |
| Jitter | < 10 ms | < 30 ms | > 30 ms |
| Packet Loss | 0% | < 1% | > 2% |
Testing Tools
- Dedicated VoIP Network Tests: Run specialized browser-based VoIP quality tests that measure connection stability over several minutes, simulating actual voice streams.
- Third-Party SIP Testing Tools: Tools like Wireshark or dedicated network testing suites can analyse SIP signalling and RTP (Real-Time Transport Protocol) streams across your local area network (LAN).
- Ping & MTR Tests: Use command-line tools like ping or traceroute (or MTR) targeting your VoIP provider's server IP address to detect where latency jumps or packet loss occurs along the hop path.
How to Fix VoIP Call Quality Issues
Most fixes fall into one of four categories: Network Configuration, Connection Type, Hardware, and Provider Infrastructure.
Enable QoS (Quality of Service) on Your Router
This is the single most impactful fix for businesses sharing a single internet connection among multiple employees.
In order to fix this, log in to your router's administration portal, navigate to the QoS / Traffic Management section, and set rule priorities to elevate SIP and RTP protocols or prioritize the IP addresses assigned to your desk phones. QoS establishes a priority lane for VoIP traffic, ensuring voice packets bypass general web browsing, file downloads, and software updates. Some routers may not have QoS settings, so you may want to check your model or ask your IT provider
Disable SIP ALG
Access your router's advanced firewall or WAN settings and locate SIP ALG (sometimes listed as SIP Passthrough or SIP Module). Most common router brands (Netgear, TP-Link, ASUS) have this in advanced settings. Turn it OFF, save changes, and reboot both your router and desk phones. Application Layer Gateway (SIP ALG) is a default setting on many commercial and home routers originally intended to prevent NAT (Network Address Translation) firewall issues for VoIP calls. In practice, SIP ALG frequently inspects and modifies SIP headers in ways that corrupt voice traffic. It is one of the most common causes of:
- One-way audio (you can hear them, but they cannot hear you).
- Phones unexpectedly unregistering or going offline.
- Dropped calls after a fixed time interval (e.g., exactly 30 seconds).
Switch from Wi-Fi to Ethernet for Desk Phones
Wi-Fi is inherently prone to signal fluctuation. Radio interference from neighbouring office networks, microwaves, Bluetooth devices, and physical obstacles like metal framing or concrete walls introduces random latency bursts and packet loss.
If wiring desk phones to wall plates is not feasible, ensure your wireless endpoints use a dedicated 5GHz Wi-Fi band with minimal local channel congestion, or deploy specialized DECT wireless solutions built explicitly for voice traffic.
Check Your NBN Connection Type
Australia’s National Broadband Network (NBN) utilizes a variety of technologies, each offering vastly different levels of latency stability:
- FTTP (Fibre to the Premises): Best for VoIP. Direct optical fibre delivering low latency (< 10ms) and near-zero jitter.
- HFC (Hybrid Coaxial): Good for VoIP. Capable of high speeds, but susceptible to latency spikes during neighbourhood peak usage hours.
- FTTC / FTTN (Fibre to the Curb / Node): Variable performance. Relies on legacy copper lines. Degraded copper, poor internal building wiring, or wet weather can significantly increase packet loss and noise.
- Fixed Wireless & Satellite: Challenging for VoIP. Highly vulnerable to environmental conditions, atmospheric disruption, and high base latency.
If your office relies on FTTN or Fixed Wireless and experiences recurring call issues, check your address on the NBN Co portal to see if you are eligible for an NBN Technology Choice / FTTP Upgrade. Moving from copper-based FTTN to full-fibre FTTP eliminates a huge proportion of line-noise latency issues.
Check for network congestion
VoIP does not consume massive amounts of data. In fact, a typical high-definition VoIP call uses only 80 to 100 kbps (kilobits per second) of upload and download. Bandwidth is rarely the main issue, but traffic congestion is. Even a 250 Mbps NBN line can experience VoIP stutter if a background application saturates the upload pipe, triggering bufferbloat.
To isolate voice traffic from general business activities you need to:
- Assign desk phones and voice applications to a dedicated Voice VLAN (Virtual Local Area Network).
- Explore a dedicated connection for the voice applications or handsets.
When It’s a Provider Problem, Not a Network Problem
If you have enabled QoS, disabled SIP ALG, wired your phones over Ethernet, and checked that your NBN connection is suitable for VoIP, the issue may be on your VoIP provider’s end.
Signs the Problem is with Upstream
- Localized Failures: Quality issues only occur when calling specific destinations
- Cross-Site Consistency: Remote workers, home offices, and main office sites experience identical voice corruption simultaneously despite using completely different internet connections.
- Good Network Metrics, Poor Audio: Local diagnostic tools confirm 0% packet loss and low jitter, yet call quality remains degraded.
What to Look for in a VoIP Provider
A business-grade VoIP provider builds network architecture specifically engineered to remove latency. You can check the following criteria:
- Australian-Based Infrastructure: Voice servers should be hosted natively in Tier-3 Australian data centres (Sydney, Melbourne, etc.).
- Low-Latency Smart Routing: Voice packets should travel along the shortest physical network route.
- Transparent Service Status Monitoring: Upstream infrastructure issues should be tracked and reported openly in real time. At VoIPline Telecom, live status updates are provided directly at voipline.net.au/status, giving you full visibility into network operations at all times.
Frequently Asked Questions
How much bandwidth does a VoIP call use?
A standard high-definition (HD) VoIP call using the G.722 codec requires approximately 80 to 100 kbps (0.1 Mbps) of symmetrical upload and download bandwidth per active call. 100 Mbps NBN connection can easily support dozens of simultaneous calls, provided the network traffic is properly prioritized.
Why does my VoIP call sound robotic?
Robotic, metallic, or garbled audio is almost always caused by high jitter or packet loss. When packets arrive out of order or disappear, the receiving device attempts to fill in the missing gaps by stretching or compressing the remaining audio data, creating a metallic, artificial distortion.
What is a good MOS score for business VoIP calls?
MOS stands for Mean Opinion Score, a standard numerical rating system used to evaluate voice quality on a scale from 1 to 5:
- 4.3 to 4.5: HD voice quality; crystal-clear audio (the gold standard).
- 4.0 to 4.2: Standard PSTN call quality; clear and acceptable for business.
- Below 3.6: Noticeably poor; frequent audio dropping, distortion, or delay.
Business-grade VoIP connections should consistently achieve an MOS of 4.0 or higher.
Does Wi-Fi affect VoIP call quality?
Yes. Wi-Fi may cause variable packet latency, channel interference, and signal attenuation from walls and metal objects. Connecting desk phones via a physical Ethernet cable provides a much more stable connection.
How do I know if SIP ALG is causing my VoIP problems?
The most telling signs of SIP ALG interference include one-way audio (one caller cannot hear the other), dropped calls at 30 or 60 seconds, or phones randomly unregistering and failing to receive incoming calls. Disabling SIP ALG in your router's advanced settings usually resolves these issues immediately.