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Network Assessment

VoIP Test: Bandwidth & Call Quality Assessment

Is your network ready for voice and video? This assessment simulates real VoIP traffic across your connection and measures the metrics that determine call quality — jitter, packet loss, latency, and UDP capacity. In 3 minutes, you'll know exactly where you stand and what to fix.

Configure Your Assessment

Set up your test parameters below. The assessment takes approximately 3 minutes.

1
Use a wired connection if possible. Wi-Fi introduces variability that can affect results.
2
Download the BCS testing utility and run it as Administrator for the most accurate results. How do I run as admin?
Download BCS
3
Choose the nearest test server.
4
Choose simultaneous calls to simulate.
Start Assessment
Please enter a value between 1 and 200 for simulated calls.
What This Assessment Measures
Jitter
Packet delay variation that distorts audio and video
Packet Loss
Missing data that causes audio gaps and video corruption
MOS Score
Industry-standard 1-5 call quality rating
Round Trip Time
Latency that creates talk-over and delays
SIP ALG
Firewall interference that silently breaks VoIP
Capacity
Whether your connection can handle the call volume

VoIP bandwidth and call quality, measured end to end

MyConnection Server's VoIP assessment goes beyond a simple bandwidth check. It places real, simulated VoIP calls between your device and your chosen test server, then measures the metrics that actually determine whether voice and video hold up: jitter, packet loss and its distribution, round-trip latency, MOS score, SIP ALG interference, and the UDP capacity your connection can sustain under load.

If you arrived looking for a VoIP speed test, this is what you actually need. Raw download speed tells you very little about call quality — a connection can post a fast speed-test number and still drop calls the moment jitter spikes or capacity runs out. By simulating concurrent calls and measuring quality directly, this assessment shows whether your network is genuinely ready for VoIP, not just how much bandwidth it can burst.

Latency matters as much as loss. This VoIP latency testing measures round-trip time and its consistency, because even a fast connection with unstable latency produces talk-over, echo, and the awkward pauses that make conversations unusable. And if you're qualifying more than one location — a branch rollout, a contact-center migration, or a carrier VoIP assessment spanning multiple interconnects — the same test runs from any site, so you can compare results side by side and prove exactly which network paths are ready for voice.

Frequently Asked Questions
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What does this VoIP assessment test?
This test simulates the number of concurrent VoIP calls you specify, running them from your computer to a server at your chosen location. It measures jitter (packet delay variation), packet loss and its distribution, MOS (Mean Opinion Score), SIP ALG interference, round-trip time, and whether your connection has enough capacity for the call volume.
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What is the capacity test?
The capacity test validates that your connection has enough available bandwidth to support VoIP and media traffic alongside your existing network usage. It sends UDP media packets up to a calculated limit to determine whether adding voice traffic will push your connection past acceptable quality thresholds.
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Why do I need the BCS utility?
VoIP uses UDP packets, which browsers cannot send or receive directly. The BCS utility is a lightweight native client that provides the low-level network access needed for accurate VoIP simulation, capacity testing, and SIP protocol testing. It installs in seconds and requires no admin privileges. Learn more about why this is necessary.
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Is this a VoIP speed test?
Not exactly — and the difference matters. A traditional speed test measures raw download and upload throughput, but a fast speed-test result does not guarantee good calls. VoIP quality depends on jitter, packet loss, latency, and whether your connection has enough UDP capacity to carry concurrent calls. This assessment measures those metrics directly by simulating real VoIP traffic, so it reflects actual call quality rather than peak bandwidth.
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What are good jitter, latency, and packet loss levels for VoIP?
As a general guide, VoIP calls stay clear when jitter is under 30ms (ideally under 5ms), latency (round-trip time) is under 150ms, and packet loss is below 1%. A MOS score of 4.0 or higher indicates good call quality. This assessment measures each of these against those thresholds and flags any that would degrade voice or video.
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How do I test VoIP call quality?
Download the BCS utility, choose the test server nearest your users and the number of concurrent calls to simulate, then start the assessment. It places real VoIP traffic on your connection and reports jitter, packet loss, latency, MOS, SIP ALG interference, and capacity in about three minutes — so you see actual call quality instead of guessing from a speed test.

Deploy this on your own site

This test portal is fully customizable. Brand it, embed it, and use the MCS API to integrate results into your workflows. Give your customers, agents, and remote workers a self-service VoIP readiness check.

Book a Demo
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Initializing
Connecting to test server
You can continue using your computer while the assessment runs.
VoIP Assessment Results
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Overall Result
Pass
Test Metrics
Metric Value Status
Connection Capacity
Metric Value Status

You tested one connection. See what MCS does across your whole network.

MyConnection Server runs this same assessment continuously across every site, branch, and remote worker — automatically, with historical trending and alerting.

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