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Component | Specification |
|---|---|
OS | Windows 7 or greater OSX Yosemite 10.10 or greater |
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CPU | Intel or AMD CPU released after 2010 |
Memory | 6GB RAM or greater |
Network | 10/100 NIC (wired) or greater 802.11n (wireless) or later |
Display | 1680x1050 resolution or greater |
IP Phones | Polycom
Yealink
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Headsets |
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Browser |
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| Note |
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The use of VDI (Virtual Desktop Infrastructure) is not supported. Depending on the solution, VDIs can be configured to work with VoIP solutions such as Sharpen, but Sharpen does not actively test with these solutions to validate a good agent experience. If VDIs are in use, Sharpen will not be responsible for the quality of service and will troubleshoot only up to the edge of our network (webRTC registration servers). |
Ports Protocols and Domains
While most work-from-home users will be all set, it is important to make sure the following ports/protocols are free to communicate with our environment. If, for some reason, your ISP has restricted activity on these ports, you'll need to work with them to allow two-way traffic.
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Domain | Protocol/Port | Purpose |
|---|---|---|
*.s3.amazonaws.com | TCP: 443 | Long-term audio and image file storage |
stun.l.google.com | UDP: 19302 | WebRTC STUN server |
stun1.l.google.com | UDP: 19302 | WebRTC STUN server |
stun2.l.google.com | UDP: 19302 | WebRTC STUN server |
stun3.l.google.com | UDP: 19302 | WebRTC STUN server |
stun4.l.google.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 |
*.pendo.io | TCP: 443 | Analytics and 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 |
Isolation Zone Domains
Access to these domains should be open with respect to which isolation zone your account is built in.
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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 |
Network performance
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.
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 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
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.
Common interruptions
Working from home enables Sharpen users with the flexibility they need while still being connected to the Sharpen environment. With that flexibility comes additional areas of focus to allow for a consistent experience. Most importantly, it is important to remember that Sharpen is a real-time communications platform assuming prompt and consistent network and workstation availability to support the real-time transmission of audio.
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