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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.

Recommendations

Run this basic speed test to obtain your downoad, 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 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

Microphone permissions

If using the Sharpen Q phone, which is based on WebRTC, it is important to have microphone permissions setup properly on both your operating system and web browser. See the guides below for validating proper setup.

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