Clear, reliable calls require a stable internet connection. A good rule of thumb is to allocate approximately 100 kbps per active VoIP call; this estimate covers both the audio stream and the additional data required for transmission:

• Codec usage: A standard call using the G.711 codec requires about 64 kbps for the audio itself.
• Network overhead: Data packets (IP, UDP, etc.) add an additional 20–25 kbps.

Together, this totals roughly 85–90 kbps. Rounding up to 100 kbps per call ensures a safe buffer for minor fluctuations and helps maintain consistent call quality.

Step 1: Run a Speed Test

1. Connect by Ethernet if possible.
2. Open Speedtest.net and run the test.
3. Compare results to your ISP plan. Significant gaps (especially low upload) can explain poor call quality.
   Pro Tip!  Internet plans list “up to” speeds; real-world results are often lower depending on network load and equipment.

Step 2: Check for Packet Loss (Ping Test)

Packet loss and latency spikes cause audio dropouts and delays.

Mac

1. Press Command + Space, type terminal, press Enter.
2. Run:

   ping 8.8.8.8

3. Let it run for ~30–60 seconds, then press Ctrl + C to stop. Look for timeouts or very high time (ms).

Windows

1. Press the Windows key, type cmd, press Enter.
2. Run:

   ping 8.8.8.8

3. After ~30–60 seconds, press Ctrl + C. Frequent “Request timed out” or highly variable latency indicates instability.

Step 3: Optimize Your Router

1. Enable QoS / Traffic Shaping: Prioritize voice traffic so calls stay clear during busy periods.
2. Save changes and reboot the router if prompted.

Step 4: Reboot Your Network

1. Power-cycle your modem, router, and any switches (turn off, wait 20–30 seconds, turn back on).
2. Wait until everything fully reconnects, then test calls again.