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BubblyPhone

Affordable international calling for everyone. Crystal-clear calls to 100+ countries with transparent per-minute pricing.

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VoIP Speed Test

Check if your internet connection is ready for crystal-clear VoIP calls. Measure latency, jitter, packet loss, and get a MOS score — free, in your browser.

100% Client-Side No Data Sent Results in ~10s

Test Your VoIP Connection

Find out if your network can handle crystal-clear VoIP calls. The test takes about 10 seconds.

How the VoIP Speed Test Works

1

Run the Test

Click Start and the tool measures 5 network metrics in about 10 seconds — all in your browser.

2

Read Your Results

See your latency, jitter, packet loss, download speed, and MOS score with VoIP-specific thresholds.

3

Make Better Calls

Use your results to troubleshoot issues, then call worldwide with BubblyPhone.

Frequently Asked Questions

What internet speed do I need for VoIP calls?

VoIP calls require very little bandwidth — typically 100 kbps per call. Speed is rarely the bottleneck. What matters more is low latency (under 150ms), low jitter (under 30ms), and less than 1% packet loss. Even basic broadband is usually fast enough for VoIP if the connection is stable.

What is a MOS score and what do I need for good calls?

MOS (Mean Opinion Score) is the standard ITU-T measure of voice call quality, rated 1.0 (worst) to 5.0 (best). A score of 4.0+ is excellent — equivalent to landline quality. 3.6-3.9 is good for professional calls. Below 3.5, quality starts to degrade. This tool calculates MOS using the ITU-T E-Model based on your latency, jitter, and packet loss.

How does jitter affect VoIP call quality?

Jitter is the variation in latency between packets. Even if average latency is low, high jitter causes audio to arrive out of order, creating choppy, robotic-sounding audio. VoIP systems use a jitter buffer to smooth this, but the buffer adds latency. Jitter above 30ms starts causing audible issues; above 80ms calls become difficult.

Why is packet loss so damaging to VoIP?

Unlike file downloads where lost packets are retransmitted, VoIP uses UDP for real-time delivery — lost packets can't be resent because they'd arrive too late. Each lost packet creates a brief gap in audio. Even 1-2% loss causes noticeable dropouts. Above 5%, calls become very difficult to hold.

My speed test shows fast speeds but calls are still poor. Why?

Speed (bandwidth) is almost never the limiting factor for VoIP. High latency, jitter, and packet loss are the real culprits. Common causes: congested Wi-Fi (use ethernet), ISP routing issues, VPN interference (disable VPN for calls), or background downloads saturating the connection during calls.

Related Tools

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Carrier Lookup

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Voice Recorder

Record audio in browser

Ready to Make a Call?

Your connection is tested — call any number worldwide with BubblyPhone for crystal-clear quality at affordable rates.