How Each Part Fails
| Call Element | Typical Bandwidth | Sensitive To | Failure Symptom |
|---|---|---|---|
| Audio | 30–100 Kbps | Jitter, packet loss, CPU spikes | Robotic voice, dropouts, echo |
| Camera video (SD) | 300–600 Kbps upload | Upload stability, Wi-Fi | Pixelated or frozen video |
| Camera video (HD) | 1–2.5 Mbps upload | Upload speed and Wi-Fi stability | Blurry then frozen |
| Screen share | 0.5–5 Mbps upload | Resolution, motion, packet loss | Smeared text, slow refresh |
| Background blur / effects | Minimal extra network | CPU load | Stuttery video, dropped frames |
Why Audio Uses So Little Bandwidth
Modern voice codecs — Opus in Zoom and WebRTC apps, SILK in Teams — compress speech to 30–60 Kbps with remarkably high quality. The challenge is not the size of the data but the delivery timing. Audio must arrive in order and on schedule. A 200ms burst of packet loss turns a clean sentence into a broken one regardless of your plan speed. This is why someone on a 10 Mbps plan with high jitter sounds worse than someone on a 2 Mbps plan with rock-steady delivery.
What "Robotic" Audio Actually Means
Each recognizable audio failure has a specific network cause:
| Sound | Likely Cause |
|---|---|
| Robotic or garbled voice | Packet loss — the codec is interpolating over missing data |
| Choppy, cutting in and out | High jitter causing the jitter buffer to overflow or underflow |
| Echo of your own voice | The far end's speaker audio is being picked up by their microphone |
| Delay or talking over each other | High round-trip latency, often caused by congested routing or VPN |
| One-way audio | Firewall or NAT traversal failure blocking one direction |
Per-Platform Bandwidth Requirements
| Platform | Audio Only | HD Video Call | Group HD Call |
|---|---|---|---|
| Zoom | 60–80 Kbps | 600 Kbps up/down | 1.5–3 Mbps up/down |
| Microsoft Teams | 30 Kbps | 500 Kbps–1.2 Mbps | 1–2 Mbps up/down |
| Google Meet | ~100 Kbps | 500 Kbps–1 Mbps | 2 Mbps up/down |
| Webex | 64 Kbps | 500 Kbps–1 Mbps | 1–4 Mbps group |
Bad Call Triage
- Turn off camera video first — it is the largest single upload consumer.
- Stop screen sharing if audio still breaks — screen share can spike upload unexpectedly during motion.
- Switch from Wi-Fi to Ethernet if possible — eliminates jitter from wireless interference.
- Close cloud sync, uploads, and browser tabs playing media — these compete for upload bandwidth.
- Disable background blur or virtual backgrounds — they add CPU load that causes frame drops.
- Use phone audio (dial-in number) as backup when the internet path is truly unstable — phone calls do not use your home connection.
Jitter Buffers and Why They Matter
Meeting apps use a jitter buffer to smooth out inconsistent packet delivery. The buffer holds a small reserve of audio (typically 20–200ms) so that minor delays do not cause dropouts. When jitter is too high or sustained, the buffer cannot keep up and the audio breaks. Larger jitter buffers introduce delay. This is the tradeoff behind the familiar feeling of a call that sounds "delayed" when the connection is poor — the app expanded its buffer to cope with jitter, adding latency as a side effect.
Frequently Asked Questions
Does audio or video need more bandwidth?
Video needs 10–50 times more bandwidth than audio. Audio only needs 30–100 Kbps, but it is far more sensitive to jitter and packet loss than video. A bad 0.5 Mbps connection with low jitter often produces better audio than a 10 Mbps connection with erratic packet delivery.
What should I turn off first on a bad call?
Turn off camera video first. If audio still breaks, stop screen sharing. Keep audio running at all costs — it carries the meeting. Use the dial-in number as a last resort when the internet path is genuinely unstable.
Why does audio sound robotic?
Robotic audio is almost always packet loss — the codec fills in missing packets with an approximation that sounds artificial. Check your packet loss percentage in a speed test. Even 2–3% sustained loss will cause audible problems in most apps.
Why is my video fine but audio breaks?
Video and audio often use different codecs, buffers, and priority rules. Some routers and ISPs deprioritize small audio packets during congestion. It can also mean audio is running on a separate port that experiences more loss than the video stream. A wired connection almost always fixes this asymmetry.