VoIP Call Quality Problems: Fixes for AU Business

The Three Factors That Determine VoIP Call Quality

This guide diagnoses and fixes the most common VoIP call quality problems on Australian NBN connections: choppy audio, echo, one-way audio, dropped calls, and crackling. Written by an independent editorial team with direct experience troubleshooting VoIP across every NBN connection type. Unlike generic troubleshooting articles, this covers the specific Australian factors that cause quality issues: FTTN upload limitations, ISP CVC congestion during peak hours, SIP ALG on Telstra smart modems, and QoS misconfiguration on consumer-grade routers.

NBN Call Quality Benchmarks by Connection Type

Not all NBN connections deliver the same VOIP performance. The connection type between your premises and the fibre backbone determines how much latency and packet loss variability you are exposed to. Here is what each NBN technology means for VOIP call quality in Australia:

For VOIP call quality, the acceptable thresholds are: one-way latency under 150ms (under 80ms for good quality), jitter under 30ms, and packet loss under 1%. Most FTTP and FTTC connections stay well within these parameters. FTTN connections on longer copper runs can breach these thresholds, particularly during peak evening hours when CVC contention at the node is highest.

SIP ALG: The Most Common Cause of AU VOIP Problems

SIP ALG (Session Initiation Protocol Application Layer Gateway) is a feature built into many consumer and business routers that was designed to help VOIP calls pass through NAT. In practice, it causes more problems than it solves for modern hosted cloud PBX deployments. SIP ALG inspects and modifies SIP packets as they pass through the router, which frequently corrupts the signalling and causes symptoms including:

• Calls connecting but with one-way audio (you can hear the other person, they cannot hear you, or vice versa)
• Calls that ring but cannot be answered
• Calls that drop after exactly 30 seconds or 60 seconds
• Registration failures where phones cannot register to the SIP server
• Intermittent call quality that varies by time of day or resets after a router reboot

How to disable SIP ALG on common Australian routers:

• TP-Link (Archer series, Deco): Advanced > NAT Forwarding > ALG > disable SIP. On older firmware: Advanced > Security > disable SIP ALG.
• Netgear (Nighthawk, Orbi): Advanced > Advanced Setup > WAN Setup > disable SIP ALG checkbox. Some Netgear firmware hides this -- check the router admin panel at 192.168.1.1.
• ASUS (RT series): Advanced Settings > WAN > disable Enable SIP Passthrough.
• Telstra Smart Modem (Sagemcom, Technicolor): Telstra-branded modems have limited admin access. Your VOIP provider should be able to work around SIP ALG by using SRTP or a VPN tunnel for SIP signalling.
• Ubiquiti (UniFi, EdgeRouter): SIP ALG is not enabled by default on Ubiquiti hardware. If someone has enabled it manually, it can be disabled in the firewall ALG settings.

QoS Configuration for VOIP on Australian Business Routers

QoS (Quality of Service) prioritises VOIP traffic on your network so that a large file download or video stream does not degrade call quality. For most Australian small businesses on NBN 100 or higher, QoS is not strictly necessary -- the connection has enough headroom for simultaneous VOIP and data traffic. It becomes relevant when: your available upload bandwidth is under 10 Mbps (FTTN on a long run), you have multiple staff on calls simultaneously, or you run other bandwidth-intensive applications (video conferencing, large file transfers) alongside VOIP.

QoS settings for common scenarios:

• DSCP marking: Tag VOIP packets with DSCP EF (Expedited Forwarding, value 46) for voice and DSCP AF41 (value 34) for video. Most cloud VOIP providers mark their traffic with DSCP EF by default -- confirm with your provider. Your router needs to honour DSCP markings from upstream for this to work.
• Port-based QoS: Prioritise UDP traffic on ports 5060 (SIP signalling) and 10000-20000 (RTP audio). Specific RTP port ranges vary by provider -- ask your provider what RTP ports they use.
• Bandwidth reservation: Reserve a minimum of 100 kbps upload per concurrent VOIP call. For a 5-seat office with potentially 3 concurrent calls: reserve 300 kbps upload. On an NBN 25 plan (5 Mbps upload), this is 6% of available upload -- negligible. On an FTTN connection with poor sync speed (1-2 Mbps upload), it matters.

Diagnosing VOIP Call Quality Problems: Is It Your ISP or Your Provider?

When call quality is poor, the problem is in one of four places: your ISP connection, your local network (router, switches, Wi-Fi), your VOIP provider's platform, or the carrier network the other party is on. Blaming the wrong party wastes time. Here is how to narrow it down:

Step 1: Test your internet connection quality. Use a VOIP-specific speed test (not a general speed test -- you need latency, jitter, and packet loss, not just bandwidth). Tools: PingPlotter (free, Windows/Mac), Cloudflare speed test (speed.cloudflare.com) includes latency and jitter. Run the test while call quality is poor, not when things are working fine. If latency is above 150ms or packet loss is above 1%, your ISP connection is the primary suspect.

Step 2: Test via wired vs wireless. Plug a laptop directly into the router via Ethernet and make a test call via softphone. If quality improves, your Wi-Fi is the issue -- not your ISP or provider. Business VOIP phones should always be on wired Ethernet where possible; Wi-Fi VOIP is workable but introduces jitter from wireless interference.

Step 3: Test with a different provider's platform. Most Australian hosted VOIP providers offer a free trial or demo account. If call quality is equally poor on a trial account with a different provider, the problem is your internet connection or local network. If quality is better on the trial, your current provider's platform or routing may be the issue.

Escalate to your ISP when: latency above 150ms on a wired connection, packet loss above 1% on a wired connection, quality degrades at peak hours (5-9pm) consistently, or quality degrades during rain or wet weather (FTTN copper fault indicator).

Escalate to your VOIP provider when: one-way audio persists after SIP ALG is disabled, registration failures on specific handsets or softphones, calls drop at a consistent time interval (30s, 60s), or quality is poor only when calling specific number ranges (interstate, mobile, international).

How Australian NBN Connection Types Affect Call Quality

The biggest variable in Australian VoIP quality is NBN connection type. FTTP (Fibre to the Premises) delivers the most consistent low-latency performance because the fibre runs all the way to your building. FTTN (Fibre to the Node) relies on copper from the node to your premises, and the quality of that copper run has a direct impact on latency and packet loss. If your business phone stopped working recently after an NBN switchover, see Business Phone Stopped Working After NBN: What to Do for immediate steps.During peak evening hours (6-9pm) on congested NBN infrastructure, even FTTP connections can show elevated latency. For business users who need consistent quality during business hours, NBN performance during business hours is generally more reliable than residential peak hours.

How to Measure Your Connection Quality for VoIP

A standard internet speed test does not measure VoIP-relevant metrics. To assess whether your connection can support VoIP well, you need to measure latency, jitter, and packet loss. Tools for this:Ping test to your VoIP provider's SIP server (ask them for the server IP or hostname). Most routers have a built-in ping tool, or use the Windows or macOS command prompt. Run 100 pings and look at average, minimum, maximum, and any lost packets. For a more comprehensive test, use Cloudflare's speed.cloudflare.com which measures latency, download speed, and jitter. Run tests during your business hours, not late at night when the network is quieter.

Codec Selection: G.711 vs G.722 vs G.729

VoIP audio is compressed using a codec (coder-decoder). The codec affects both call quality and bandwidth usage. For Australian NBN connections, the recommended choices are:

QoS: Prioritising Voice Traffic on Your Network

Quality of Service (QoS) configuration on your router ensures that VoIP packets are prioritised over other network traffic. Without QoS, a large file upload or a software update download can introduce enough network congestion to degrade active calls. For a business VoIP setup, QoS configuration is not optional for offices with more than a few staff.Most business-grade routers support QoS. The configuration approach: mark SIP traffic (UDP port 5060) and RTP audio traffic (your provider's RTP port range, typically UDP 10000-20000 or similar) with the highest priority queue. This ensures voice packets are processed ahead of less time-sensitive traffic even during periods of high network utilisation.

Diagnosing Specific Call Quality Problems

See our NBN VoIP setup guide for step-by-step instructions on checking your NBN line quality and configuring your router.

When to Call Your ISP Versus Your VoIP Provider

One of the most frustrating aspects of VoIP troubleshooting is determining who is responsible for a call quality issue. Both your ISP and your VoIP provider will naturally lean toward attributing the problem to the other party. Having a clear framework for isolating the issue saves time and prevents the blame-shifting loop that can leave a problem unresolved for weeks.

If you are unsure how your NBN connection type affects VoIP, our NBN compatibility guide breaks it down by technology type.

The ISP is responsible for everything from the NBN connection point to your router WAN port. This includes: the quality and stability of the NBN line itself (line sync rate, error rates on FTTN copper, HFC coaxial noise), the latency from your premises to the ISP's point of interconnect, packet loss at the NBN or ISP network level, and congestion on the ISP's internal network during peak hours. To isolate ISP responsibility, run a ping test from a device on your network to your ISP's gateway IP (usually findable in your router admin interface or by running "traceroute" to an external IP and noting the first hop). If this test shows elevated latency or packet loss, the issue is with your ISP or the NBN line, not your VoIP provider.

Your VoIP provider is responsible for everything from their SIP platform to your phone registration. This includes: SIP server availability and response time, codec configuration on your account, media server routing quality, and the accuracy of your account configuration. To isolate provider responsibility, run a ping test directly to your provider's SIP server address (ask them for it). Compare the results to your ISP gateway ping. If the ISP gateway ping is clean but the SIP server ping shows packet loss or high latency, the issue is between the ISP and the VoIP provider, or within the provider's own infrastructure. Your router and local network (including SIP ALG, QoS, and firewall rules) are your own responsibility to configure correctly before escalating to either party.

Call Quality During Peak Hours on Australian NBN

A specific pattern affects Australian VoIP users on the NBN: call quality that is consistently good during business hours but degrades noticeably between 6pm and 9pm. This is peak hour congestion on the NBN, and it is a known characteristic of the network, particularly for residential-grade NBN plans.

The cause is CVC (Connectivity Virtual Circuit) contention. CVC is the capacity that ISPs purchase from NBN Co to carry their customers' traffic. During peak hours, many customers are simultaneously streaming video, gaming, and making video calls. ISPs that have not purchased sufficient CVC capacity experience congestion at their network layer, which shows up as increased latency and packet loss across all traffic, including VoIP calls. The problem is worse on residential NBN plans because the CVC commitments on residential services are lower than on business services.

For businesses whose staff work office hours and whose customers call during business hours (9am to 6pm), peak-hour congestion may never be a material issue. For businesses with staff who work from home in the evenings, or whose customers call after hours, it is worth testing call quality during the 6pm to 9pm window specifically. If you consistently observe degraded quality during this period, the most effective solutions are: upgrading to a business NBN plan (which typically carries stronger CVC commitments and SLA guarantees), switching to an ISP with better CVC investment (quality varies significantly between providers), or routing after-hours calls to a mobile rather than relying on the home internet connection during peak congestion.

Long-Term Call Quality Monitoring

Reactive troubleshooting, where you only investigate call quality when a problem is already affecting customers, is less effective than proactive monitoring. Most enterprise-grade hosted PBX platforms include call quality reporting that captures MOS (Mean Opinion Score) data for every call on your account. Understanding how to use this reporting turns call quality management from a reactive exercise into a proactive one.

MOS (Mean Opinion Score) is the standard measure of voice call quality. It is calculated from underlying metrics including latency, jitter, and packet loss, and expressed as a number from 1 to 5. A score above 4.0 is considered toll-quality (equivalent to PSTN). A score between 3.5 and 4.0 is acceptable for business voice. Below 3.5, quality is noticeably degraded. Below 3.0, calls are difficult to conduct. Your provider's call quality reports will show MOS scores per call, often with the underlying metrics (latency, jitter, packet loss) available for deeper investigation.

Review your call quality reports weekly during the first month after go-live. Look for patterns: are low-MOS calls concentrated at specific times of day (suggesting congestion), on specific extensions (suggesting a local network issue with that phone or switch port), or on calls to specific destinations (suggesting a routing issue on the provider's network)? Patterns in the data point you toward the right fix. A provider that does not offer call quality reporting in their platform is a provider that cannot help you proactively manage quality. This should be a requirement, not a nice-to-have, when evaluating providers.

What Most People Get Wrong About VoIP Quality

1. Blaming the VoIP provider when the problem is your network. In the vast majority of cases, call quality issues are caused by your local network or internet connection, not your VoIP provider. Before calling them, run the diagnostic steps in this guide.2. Running a speed test and assuming everything is fine. Speed tests measure throughput, but VoIP quality depends on jitter and packet loss, which a standard speed test does not measure. Use a VoIP-specific test that reports jitter (under 30ms needed) and packet loss (under 1% needed).3. Upgrading the NBN plan without fixing the real problem. Going from NBN 50 to NBN 100 will not fix call quality if the problem is SIP ALG, QoS, or CVC congestion during peak hours. Diagnose before spending more.

Your Next Steps

1. Disable SIP ALG on your router. This fixes the problem in roughly half of cases.
2. Enable QoS for voice traffic. Prioritise SIP (port 5060) and RTP (ports 10000-20000).
3. Test during peak hours (7-9pm). This is when CVC congestion hits hardest on NBN.
4. Check your upload speed. You need at least 1 Mbps upload per 5 concurrent calls. FTTN connections may struggle.
5. Consider a dedicated VLAN for voice. Separating voice and data traffic prevents interference. Our setup guide walks you through this.
6. If problems persist, contact your ISP about CVC congestion. If quality is consistently poor during peak hours, your ISP may be undersizing their CVC allocation.Still having issues? Get a free recommendation and we will help you assess whether your connection can support VoIP or if you need to consider alternatives.

Power outages are a call quality issue that most businesses do not plan for until the first blackout takes the phones offline. For the full picture on keeping calls up during outages, see our guide to NBN battery backup for VOIP, including which UPS devices work with NBN equipment and what the NBN Co battery subsidy program actually covers.

Power Outages: When Call Quality Drops to Zero

The most overlooked VoIP call quality issue is not jitter or packet loss. It is a complete outage. On NBN, your VoIP system depends entirely on mains power. When the power goes out, call quality does not degrade gracefully. It stops completely.

Unlike the old copper PSTN (which carried its own power from the exchange), NBN connections require powered equipment at your premises: the NTD (Network Termination Device), your router, and your VoIP phones or ATA.

Mitigation options:

• UPS (uninterruptible power supply): A basic UPS (~$150-250 AUD) on your NTD and router gives 1-2 hours of phone service during outages. Check current pricing before purchasing.
• Mobile failover: Some hosted PBX providers (including Maxotel) can automatically redirect calls to mobile numbers during outages, so callers still reach your team.
• 4G/5G backup: A mobile broadband failover router keeps your internet (and VoIP) running when NBN drops, though call quality depends on mobile signal strength at your location.

Call recording is a standard feature on most cloud phone systems and is frequently used for quality monitoring and dispute resolution. Before enabling it, review your obligations under Australian law. See Call Recording Laws in Australia for a full guide.

One of the most common causes of poor VOIP call quality in Australian businesses is an undersized SIP trunk count relative to concurrent call demand. Our guide to how many SIP trunks you need walks through the concurrent call calculation, seasonal peak adjustments, and the correct sizing formula for Australian SMBs using hosted PBX or on-premise systems.

Voicemail delivery problems are sometimes mistaken for call quality issues — missed voicemail notifications look like missed calls in the call log. If your team is experiencing unreliable voicemail delivery, our guide to voicemail to email setup for Australian businesses covers the configuration steps and common causes of email delivery failure, including SMTP relay, SPF/DKIM, and provider-specific settings.

Why do my VoIP calls sound good in the morning but bad in the afternoon?

This pattern typically indicates network congestion on shared infrastructure, most commonly FTTN or HFC NBN connections. As more users in the area are active in the afternoon, the shared link from the node or pillar to the NBN exchange becomes more congested, increasing latency and packet loss. Solutions include: requesting an NBN upgrade to FTTP (where available), using QoS to maximise priority of voice traffic on your internal network, or contacting your NBN provider to report congestion.

How much bandwidth does VoIP use?

Each active VoIP call uses approximately 80-100 kbps of bandwidth (with G.711 or G.722 codecs). Ten simultaneous calls use less than 1 Mbps. Bandwidth is almost never the limiting factor for VoIP on Australian NBN connections. Even an NBN25 connection can support multiple simultaneous calls. Latency and jitter are far more important than raw bandwidth speed.

Will a better NBN plan improve my VoIP quality?

Upgrading your NBN speed tier (e.g. from NBN25 to NBN100) rarely improves VoIP quality because bandwidth is not usually the constraint. However, upgrading your NBN technology type (e.g. from FTTN to FTTP, where FTTP upgrade is available in your area) can significantly improve quality because it replaces the variable copper run with consistent fibre. Contact your NBN provider or check NBN Co's website for upgrade availability at your address.

Is my VoIP provider responsible for call quality problems?

Your VoIP provider is responsible for quality on their network from their data centre outward. They are not responsible for the quality of your NBN connection or your internal network. In practice, most call quality problems originate in the last mile (your NBN connection) or internal network configuration. Start troubleshooting at your router and NBN connection before escalating to your VoIP provider.

If you have worked through this guide and still have call quality issues, the problem is diagnosable. A good VOIP provider with Australian support staff should be able to work through this with you. Troubleshooting your call quality is part of what you are paying them for, not an extra. If your provider cannot or will not help you diagnose the issue, that tells you something useful too.

My calls have an echo. What causes it and how do I fix it?

Echo on VoIP calls is almost always caused by acoustic feedback at the far end, not at your end. The person you are speaking with has either a handset with a poorly sealed earpiece (allowing sound to leak from the speaker into the microphone) or speaker phone is active and the room acoustics are causing their microphone to pick up your voice from the speaker and send it back to you. Ask the other person to check their handset seal or switch from speakerphone to handset. If echo occurs on all calls regardless of who you are speaking with, the issue is more likely a sidetone configuration problem in the phone firmware, which your provider can adjust through remote provisioning.

What does "one-way audio" mean and how is it fixed?

One-way audio means you can hear the other person but they cannot hear you, or vice versa. It is caused by a routing asymmetry in the audio path, most commonly from a firewall or NAT device blocking the return path of RTP (audio) packets. The most common cause is SIP ALG on your router modifying SIP packets in a way that confuses the media routing. Disable SIP ALG on your router, check that your firewall is not blocking the RTP port range your provider uses (typically UDP 10000-20000, confirm with your provider), and test again. One-way audio that persists after these steps usually indicates a misconfiguration in the phone's SIP settings that your provider can correct remotely.

How much internet bandwidth do I need for VoIP?

A single simultaneous VoIP call using the G.711 codec requires approximately 100kbps of upload and download bandwidth. G.729 (a compressed codec) reduces this to around 32kbps per call at the cost of slightly lower audio quality. In practice, the bandwidth requirement for VoIP is very low. A four-person office where all four people are on calls simultaneously needs 400kbps of upload capacity for voice, which virtually every NBN plan exceeds. The more important measure is latency and packet loss on your upload path, not raw speed. A 25Mbps connection with low latency and no packet loss delivers excellent VoIP quality. A 100Mbps connection with high jitter and 2 percent packet loss delivers poor quality.

Related reading:

• NBN VoIP setup guide
• NBN compatibility for VoIP