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Recommendations
Run this basic speed test to obtain your downoad, upload, latency, and jitter results.
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| title | Speed Test |
|---|
| Iframe | ||
|---|---|---|
|
<150ms latency to *.sipvbx.com (example: us1-webrtc-11.sipvbx.com for east coast US, or us2-webrtc-02.sipvbx.com for west coast US)
Average latency variation < 30ms
High variation represents interruption to your connection. This may be a result of competing network traffic, or general hardware/network instability
While high latency on its own simply means delay, latency variation typically comes coupled with packet loss, which will mean dropped calls and/or choppy audio
>10Mb/s internet connection
While voice itself is a fairly light operation, we recommend having enough bandwidth to handle all your operations. This value is more of a guideline, rather than a requirement. Most importantly is making sure you’re not running out.
< 1% packet loss
Voice requires basically no packet loss. If any packets are dropped, it will interrupt the audio stream. For this reason, packet loss directly influences the quality of a call. Enough packet loss will cause dropped calls
< 30ms jitter
Jitter is the variation in delay of packets. Having high jitter will also cause poor call quality
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