The short version
- Voice needs very little bandwidth but hates delay, jitter and packet loss.
- Upload congestion is the number-one cause of bad calls on the NBN. Prioritise voice traffic with QoS or "smart queue" settings.
- Turn off SIP ALG on your router, use wired connections for desk phones, and keep phones on their own VLAN in an office.
- Measure before and after. Your provider's portal or a simple test can show jitter and loss.
What good looks like
A VoIP call needs only around 30–100 kbps in each direction, so raw speed is rarely the problem. What matters is how consistently packets arrive. Common targets are:
| Metric | Good | Noticeable | Bad | What you hear |
|---|---|---|---|---|
| Latency (one way) | < 150 ms | 150–300 ms | > 300 ms | People talking over each other, awkward pauses |
| Jitter | < 20 ms | 20–40 ms | > 40 ms | Robotic or warbling audio |
| Packet loss | < 0.5% | 0.5–2% | > 2% | Words dropping out, clipped syllables |
| MOS (quality score) | 4.0+ | 3.6–4.0 | < 3.6 | Overall perceived quality on a 1–5 scale |
The 150 ms latency guideline comes from the ITU's G.114 recommendation and has been the industry rule of thumb for decades.
Step 1: Fix upload congestion with QoS
Most NBN plans have far less upload than download. When someone uploads a large file, syncs a cloud backup or joins a video call, the upload link fills up and voice packets queue behind everything else. The result is jitter and loss on your calls even though your "speed test" looks fine.
The fix is to make the router send voice packets first:
- Enable QoS or Smart Queue Management (SQM) on the router. Set the upload limit to around 85–90% of your real measured upload so the queue forms in your router, where it can be managed, rather than in the NBN network.
- Prioritise voice. Either mark the IP addresses or VLAN of your phones as highest priority, or trust DSCP markings: most VoIP phones mark voice as
EF (46)and signalling asCS3 (24). - Choose the right plan. If you have more than a handful of concurrent calls plus video meetings, an NBN plan with higher upload (or symmetrical business fibre) is worth it.
Run a speed test while you're on a call and watch whether the audio breaks up. If it does, you've confirmed an upload congestion problem, and QoS will usually fix it.
Step 2: Turn off SIP ALG
Many consumer and small business routers ship with a feature called SIP ALG (Application Layer Gateway) that tries to "help" VoIP traffic through NAT by rewriting SIP messages. In practice it often breaks things: one-way audio, calls that drop after 30 seconds or 15 minutes, phones that unregister at random. Unless your provider specifically tells you to keep it on, turn it off. The setting is usually under NAT, firewall, or "advanced" in the router admin page.
Step 3: Get phones off Wi-Fi where you can
Wi-Fi is shared, half-duplex radio. Microwaves, neighbours' networks and people walking between you and the access point all add jitter. For desk phones, use Ethernet. For softphones on laptops, a wired connection is best. If you must use Wi-Fi, use 5 GHz or 6 GHz, stay close to a modern access point, and enable Wi-Fi Multimedia (WMM), which most routers do by default.
Step 4: In an office, separate voice traffic
- Put phones on a dedicated voice VLAN, usually assigned automatically using LLDP-MED from a managed switch.
- Use PoE switches so phones don't need power packs. Put that switch on a UPS.
- Make sure switches honour QoS markings end to end (CoS/DSCP trust on the voice VLAN).
- Avoid sending voice through a VPN or web proxy unless you have to. Every hop adds delay.
Step 5: Tune the phone and the system
- Registration and NAT keep-alives. If phones drop calls or stop receiving calls after a while, the router may be timing out the NAT session. Shorten the registration interval (e.g. 300 seconds or less) or enable keep-alives.
- Use the right codec. See our codec guide. Opus for softphones on variable networks; G.722 for desk phones.
- Jitter buffer. Adaptive jitter buffers trade a little extra latency for smoother audio. Defaults are usually right; don't push them too high.
- Firmware. Keep phone and router firmware current. Fixes for audio and security issues are common.
Diagnosing specific problems
| Symptom | Likely cause | Try |
|---|---|---|
| One-way audio or no audio | NAT/SIP ALG, firewall blocking RTP ports | Disable SIP ALG; check RTP port range is allowed; ask provider about STUN/TURN |
| Calls drop at a fixed time (e.g. 30 s, 15 min) | Session timers, NAT timeout, SIP ALG | Disable SIP ALG; adjust registration/keep-alive; check session timer settings |
| Robotic, choppy audio | Jitter and loss from congestion or Wi-Fi | Enable QoS/SQM; move to wired; check upload usage |
| Echo | Handset/headset acoustic issue or high latency | Try another headset; check latency; avoid speakerphone in hard rooms |
| Phone won't register | Wrong credentials, DNS, or blocked SIP port | Check settings and that outbound SIP (5060/5061) isn't blocked |
Measure, don't guess
Most business VoIP platforms record per-call jitter, loss and MOS. Look at the worst calls of the week and you'll usually see a pattern: same site, same time of day, same users on Wi-Fi. Fix that pattern and quality improves for everyone.