📞 The Most Common VoIP Call Problems
| Problem | Most Likely Cause | Quick Fix |
|---|---|---|
| Choppy/robotic audio | Packet loss or jitter | Enable QoS on router |
| Echo on calls | Acoustic feedback or processing delay | Check headset/speaker position |
| One-way audio | NAT traversal failure | Check router STUN/firewall settings |
| Calls dropping after 60s | SIP ALG interference | Disable SIP ALG in router |
| No ringtone for inbound | Firewall blocking UDP ports | Open ports 5060, 10000-20000 |
| Delay / latency | Insufficient bandwidth or QoS | Run jitter test, enable QoS |
🔬 Step 1: Run a VoIP Quality Test
Before changing anything, run a dedicated VoIP quality test (not just a speed test). Go to ping.canopy.tools or voip-test.net from a device on the same network as your VoIP phones. Look for:
- Jitter: Should be under 30ms. Over 50ms causes choppy audio.
- Packet loss: Should be 0%. Even 1% causes noticeably degraded quality.
- Latency (MOS score): MOS above 4.0 = excellent. Below 3.5 = noticeable quality issues.
⚡ Step 2: Enable QoS on Your Router
QoS (Quality of Service) tells your router to prioritise VoIP traffic over everything else. Without QoS, a file download or video stream on someone else's PC can degrade call quality. Most business routers support QoS — look for it in the advanced/traffic settings and set VoIP traffic to highest priority.
🔧 Step 3: Disable SIP ALG
SIP ALG (Application Layer Gateway) is a router feature that's supposed to help VoIP but in practice causes more problems than it solves — including calls dropping after exactly 60 seconds. Disable it in your router's firewall or SIP settings. This single change fixes a large percentage of VoIP drop issues.
📊 Step 4: Check Bandwidth
Each VoIP call uses approximately 80–100kbps (using G.711 codec) or 30–40kbps (using G.729). A 20-user office making 10 simultaneous calls needs around 1Mbps dedicated to VoIP. This is usually not the bottleneck on modern FTTP connections — but on an old FTTC line with 10Mbps upload it can be.
🔒 Step 5: Check Firewall Rules
VoIP requires specific ports to be open. If your firewall is blocking these, calls will fail or have one-way audio. Required ports: UDP 5060 (SIP signalling), UDP 10000–20000 (RTP media). Check your firewall and ensure these are permitted for traffic to your VoIP provider's IP range.
⚠️ Don't assume it's the provider
When call quality issues arise, providers often get blamed prematurely. In our experience, 70% of VoIP quality issues are caused by the router, firewall or local network — not the VoIP platform itself. Always check your infrastructure first.
Still having VoIP issues? Telexico's support team diagnoses and fixes call quality problems for our customers — usually within the same day.
Contact Our Support Team →