Network diagnostic

Will your internet handle VoIP?

Pick your codec and how many people will be on calls at once. We’ll measure your connection’s speed, latency, jitter, and packet loss, then tell you whether it’s ready for business phone service.

Voice codec

Not sure? G.711 is the most common codec for hosted PBX and the safest number to plan around.

The number of people likely to be on the phone at the same time, not total employees. A common rule of thumb is one concurrent call per 3 employees.

Bandwidth your calls will need (each direction) 0.87 Mbps
Live readout
Download
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Mbps
Upload
—
Mbps
Latency
—
ms round-trip
Jitter
—
ms
Loss (est.)
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%

Latency samples, most recent on the right

MetricYour resultTarget for VoIPRating

Techmode runs the network assessment for you

Every TechmodeGO deployment starts with Premier Launch onboarding: a dedicated project manager and install team who assess your network, configure QoS, and test call quality before go-live. Your phone system runs on private, dedicated AWS infrastructure, backed by U.S.-based Concierge support and a lifetime configuration guarantee.

99.999% uptime Private AWS per client NPS 85.7 (industry avg ~31) A+ BBB rating 24/7 U.S.-based support
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How this test works.

Speed is measured by timing real data transfers. Latency is the median round-trip time of repeated small requests, and jitter is the average variation between consecutive samples. Because browsers can’t send the UDP packets that live calls actually use, packet loss shown here is an estimate based on request timeouts; a hard-wired network assessment measures it precisely. The call quality score (MOS) is calculated from latency, jitter, and loss using a simplified ITU E-model. Results reflect this device on this network right now: Wi-Fi, VPNs, and other traffic all affect the numbers.

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