The short answer
Nearly every VoIP quality problem traces back to one of three things: inconsistent packet delivery (jitter), congested or slow internet (bandwidth), or a hardware capture problem (your microphone). Whatever the symptom, the fastest path to better calls is a wired Ethernet connection and a pair of headphones.
Start here: run a speed test
Before anything else, test your connection at fast.com or speedtest.net. Here's what a VoIP call needs.
- Downstream
- At least 1 to 3 Mbps for a video call, or 100 kbps for voice only.
- Upstream
- The same as downstream, and the one people most often overlook.
- Latency (ping)
- Under 150ms for good quality. Under 100ms is ideal.
- Jitter
- Under 30ms.
If your numbers look fine, the issue is something else, so keep reading. If they don't, your connection is the problem, and most of the fixes below apply directly.
Choppy or robotic audio
What it sounds like
Voices cut out in short bursts, sound mechanical or garbled, and syllables get swallowed.
What's causing it
This is almost always jitter, meaning inconsistency in when packets arrive. VoIP audio travels as a continuous stream of data packets. When they show up at irregular intervals, the audio engine can't play them back smoothly, and you get that choppy sound.
How to fix it
Switch to a wired connection. Wi-Fi is convenient but naturally variable. An Ethernet cable to your router wipes out most of the jitter caused by wireless interference, and this one change fixes choppy audio more often than anything else.
Move closer to your router. If a cable isn't an option, cutting the distance and the obstacles between you and the router helps a lot.
Close other bandwidth-heavy apps. A big download or a video stream in the background can cause the packet bursts that create jitter. Pause them during your call.
Restart your router. Routers get bogged down over time. A restart clears their internal state and often steadies the connection.
Audio delay (latency)
What it sounds like
A clear gap between when you speak and when the other person hears you, or the other way around. Conversations start to feel like you're on walkie-talkies.
What's causing it
Network latency, the time it takes packets to travel from your device to theirs. For calls routed through a relay server (TURN), that includes the round trip to the server. High latency is common on long-distance international calls and congested networks.
How to fix it
Use a wired connection. Again, it's the single best thing you can do for call quality in general.
Check your router's Quality of Service (QoS) settings. Many routers can prioritize real-time traffic like VoIP over file downloads, which pushes background data out of the way of your call.
Try a different network. On a mobile hotspot or a shared office network, the congestion may be upstream of you. Moving to your home broadband often cuts latency a lot.
For browser calls specifically, Chrome tends to give the best WebRTC performance, since its VoIP engine is the most mature.
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Echo
What it sounds like
You hear yourself come back through the other person's speaker, slightly delayed. Or they hear themselves echoing through your end.
What's causing it
Echo happens when one person's speakers play the other's voice loudly enough for their own microphone to pick it back up and send it returning. This is acoustic echo, and it's the most common type.
How to fix it
Both people should use headphones. This is the most reliable fix. Headphones keep your speaker audio from reaching your microphone in the first place.
Turn down speaker volume. Lower output means less sound bleeding into the mic.
Use an external microphone with a tight pickup pattern, like a headset or a desktop cardioid mic, instead of the built-in laptop microphone that hears the whole room.
Some browser and hardware setups cancel echo automatically (WebRTC includes acoustic echo cancellation by default), but it only works well when the echo isn't too loud. Headphones are still the most reliable fix.
One-way audio
What it sounds like
You can hear the other person, but they can't hear you, or the reverse.
What's causing it
Usually a microphone permission or a mute issue. Once in a while it's a firewall configuration problem.
How to fix it
Check browser microphone permissions. Click the lock icon in your address bar and confirm the site is allowed to use your microphone.
Make sure you're not muted. It sounds obvious, but the mute at the OS level (your system sound settings) is separate from the mute in the call interface.
Try a different browser or reload the page. Browser microphone access can get stuck in an odd state.
Check your operating system's microphone settings. On Mac it's under System Settings, Privacy & Security, Microphone. On Windows it's Settings, Privacy, Microphone.
If nothing else works, a corporate or school firewall may be blocking the media ports WebRTC uses. Try the call from a home network or a mobile hotspot to confirm.
Pixelated or frozen video
What it sounds like
The video freezes or turns into a blocky, pixelated mess, even when the audio is fine.
What's causing it
Video needs far more bandwidth than audio. When bandwidth is tight, WebRTC's adaptive bitrate drops video quality to protect the audio. If it drops further, the video freezes entirely.
How to fix it
Check your upload speed specifically. Video sends more upstream than down during a call, and many broadband plans have asymmetric upload speeds that are much slower.
Reduce background bandwidth use. Cloud sync from Dropbox, iCloud, or Google Drive can quietly eat your upload bandwidth.
Switch to voice-only. If your connection genuinely can't carry video, turning off the camera frees up headroom for stable audio.
The common thread
Nearly every VoIP quality issue traces back to one of three things: inconsistent packet delivery (jitter), congested or slow internet (bandwidth), or a hardware capture problem (the microphone). Whatever the symptom, the fastest path to better calls is a wired Ethernet connection and headphones. Those two changes alone address most of the root causes above.
If you want to understand what's happening under the hood when these problems show up, our explainers on how VoIP works and the WebRTC technology behind browser calls walk through the packets, codecs, and connection paths in plain language.
Frequently asked questions
- How much internet speed do I need for a VoIP call?
- A voice-only call needs about 100 kbps in each direction. A video call wants roughly 1 to 3 Mbps. Just as important are latency under 150ms and jitter under 30ms. Upload speed matters as much as download, and it's the number people most often forget to check.
- Why is my call worse on Wi-Fi than on a cable?
- Wi-Fi signal strength varies moment to moment as it competes with other devices and works around walls and interference. That variability shows up as jitter, which makes audio choppy. A wired Ethernet connection removes almost all of it, which is why it's the single most effective fix.
- How do I stop hearing an echo on calls?
- Echo usually means one side's speakers are loud enough for their microphone to pick up the other person's voice and send it back. The most reliable fix is for both people to wear headphones. Lowering speaker volume and using a microphone with a tight pickup pattern also help.
- The other person can't hear me. What's wrong?
- One-way audio is almost always a microphone permission or mute issue. Check that your browser is allowed to use the mic (the lock icon in the address bar), confirm you're not muted at the OS level, and check your operating system's microphone privacy settings. If it still fails, a firewall may be blocking WebRTC's media ports, so try a different network.
- Does the browser I use affect call quality?
- It can. For browser calls, Chrome generally offers the most mature WebRTC engine and the most consistent performance. If a call is misbehaving in one browser, switching to Chrome or reloading the page is a quick thing to rule out.
- Why does my video freeze while the audio keeps working?
- Video uses far more bandwidth than audio, so when your connection tightens, WebRTC deliberately sacrifices video first to keep the audio intelligible. Check your upload speed, pause background cloud sync, and if the connection genuinely can't carry video, switch to voice-only.
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