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Domain | Protocol/Port | Purpose |
|---|---|---|
*.s3.amazonaws.com | TCP: 443 | Long-term audio and image file storage |
stun.l.google.com com | UDP: 19302 | WebRTC STUN server |
stun1.l.google.com com | UDP: 19302 | WebRTC STUN server |
stun2.l.google.com com | UDP: 19302 | WebRTC STUN server |
stun3.l.google.com com | UDP: 19302 | WebRTC STUN server |
stun4.l.google.com com | UDP: 19302 | WebRTC STUN server |
*.yealink.com | TCP: 443 | Yealink auto-provisioning |
*.ckeditor.com | TCP: 443 | Visual editor/UI library |
*.loggly.com | TCP: 443 | Logging |
*.ingest.io | TCP: 443 | Client logging |
*.gstatic.com | TCP: 443 | Font library |
*.googleapis.com | TCP: 443 | Font library |
*.fortawesome.com | TCP: 443 | Font library |
*.fontawesome.com | TCP: 443 | Font library |
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Isolation Zone 0 | Your users login to http://app.sharpencx.com
Domain | Protocol/Port | Purpose |
|---|---|---|
*.sharpencx.com | TCP: 443,8089,8090 | App |
*.sharpen.cx | TCP: 443 | Supplemental app domain |
*.cx.shpn.co | TCP: 443 | CX and VCX |
*.sipvbx.com | UDP: 5060 | SIP registration and signaling |
*.fathomvoice.com | TCP: 80,443,9002 | Provisioning, API, webRTC registration |
Isolation Zone 1 | Your users login to http://app.iz1.sharpen.cx
Domain | Protocol/Port | Purpose |
|---|---|---|
*.iz1.sharpen.cx | TCP: 80,443,8089,8090,9002 | Provisioning, app, webRTC registration |
*.cx-iz1.shpn.co | TCP: 443 | CX and VCX |
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<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 important is making sure 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
For a positive experience, voice requires minimal packet loss. If packets are dropped, it will interrupt the audio stream. You may experience disruptive delay or choppy audio. 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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