VoIP network health and call quality

If calls on your Dromlik system sound choppy, delayed, or drop unexpectedly, the cause is rarely the phone system itself. It is usually the local network or internet connection carrying the voice traffic.

VoIP converts voice into real-time data packets. Unlike file downloads or video streams, real-time voice traffic cannot be buffered, paused, or resent — packets must arrive consistently, instantly, and in exact order.

Understanding voice metrics

To diagnose call quality, you need to monitor three core network metrics:

• Latency — the total time it takes a voice packet to travel from speaker to listener.
• Jitter — the variation in arrival time between consecutive voice packets. Inconsistent delivery timing forces systems to drop out-of-order data, creating choppy speech.
• Packet loss — the percentage of voice packets that fail to reach their destination entirely.

Why high bandwidth doesn't guarantee call quality

A common misconception is that upgrading to a faster internet plan will fix bad call quality.

• Bandwidth is capacity, not speed. Bandwidth determines how much data can pass through your connection at once, while latency determines how fast that data actually travels.
• Voice requires minimal data. A single VoIP call consumes only ~80–100 kbps. Even a modest 20 Mbps connection has plenty of raw capacity for dozens of simultaneous calls.
• Timing beats volume. A large pipe does not stop individual voice packets from getting stuck behind a big file upload, hitting wireless interference, or dropping along the route.

Key metric targets

Network health checklist

1. Enable Quality of Service (QoS)

Configure your router or firewall to prioritise real-time voice traffic over non-critical data such as downloads, backups, and streaming. Use DSCP EF (Expedited Forwarding) rules tailored to your network.

2. Implement a dedicated voice VLAN

Isolate IP phones from general workstation, camera, and guest traffic to prevent local congestion and simplify security.

3. Ensure sufficient upload capacity

Upload link saturation from cloud backups, CCTV feeds, or file syncs damages call quality instantly. Reserve dedicated upload capacity for real-time traffic.

4. Check and disable SIP ALG

Application Layer Gateway (ALG) on routers often alters SIP packets incorrectly, leading to one-way audio, failed call setups, or dropped registrations. Disable it if you are seeing signalling issues.

5. Default to wired Ethernet

Wi-Fi introduces latency spikes, signal interference, and packet loss. Use wired connections for fixed phones and workstations wherever possible.

6. Audit bandwidth-heavy background services

Scheduled cloud backups, large updates, or file syncing can temporarily starve voice traffic. Identify recurring lag patterns by tracking network usage against the times calls degrade.

7. Maintain and monitor equipment

Ensure routers, firewalls, and switches have updated firmware and sufficient processor capacity. Re-evaluate network performance whenever you add new users or internet-heavy services.

Quick connection comparison

Troubleshooting matrix

Next steps

Run a baseline diagnostic with the Dromlik network tool at network.dromlik.com during normal business hours — or while a quality issue is actually happening. Share the results with your internal IT team or Managed Service Provider so they can configure your network rules.