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Uninterrupted, consistent network performance is required for a good experience with the Sharpen platform. Due to the inherent nature of Voice interactions to be real-time, we need consistency in the underlying network. Otherwise, users may experience dropped calls, choppy call quality, latency, or an overall slow experience.

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Assessment

Run this basic speed test to obtain your download, upload, latency, and jitter results.

  • <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 important is making sure you’re not running out.your collection of tools, including Sharpen, have sufficient bandwidth.

    • The best way to determine bandwidth needs is to sample your tool set usage, and extrapolate from there.

    • Sharpen bandwidth utilization can vary widely based on how it is used some base-line examples of usage are as follows

      • Sharpen Q page load for single agent logged into 4 queues

        • ~350 KB transferred

        • ~7 MB page resources

      • 1 minute outbound call from Sharpen Q

        • ~125 KB transferred

        • ~1 MB page resources

      • Reporting/Insights (10 reports) page load

        • ~150 KB transferred

        • ~ 8 MB page resources

  • < 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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