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To use the Sharpen platform as it is intended, we recommend the following specifications to provide a positive user experience while running with your other tools. Though the application may function while operating below these standards, we will focus our support on systems compliant with our recommendations.

Component

Specification

OS

Windows 7 or greater

OSX Yosemite 10.10 or greater

 

  • 64-bit when using desktop app

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
  • VVX Series

  • IP Series

  • Soundpoint Series

  • Txx Series

  • Wxx Series

Headsets

  • Any USB or 3.5mm headset properly recognized by the client workstation and web browser

  • Wireless headsets are discouraged due to increased probability of signal interference which can resulted in degraded call quality

  • Sharpen does not officially certify or support any particular headset due to the wide variety of suitable devices

Browser

  • Google Chrome (Past 3 revisions)

  • Microsoft Edge Chromium (Past 3 revisions)

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.

  • SIP ALG - Present in many network device configurations as SIP ALG, SIP, VoIP, settings which enable SIP ALG are contrary to a successful network setup with multiple VoIP devices. Intermittent disconnection, audio drops, and the inability to register are common symptoms when SIP ALG is enabled. This must be shut off for optimal performance.

  • Sharpen strongly discourages the use of “combination” network equipment such as all-in-one modem/router devices. ISPs typically provide or rent these out to customers. They are known for having issues with VOIP traffic as well as having limited access to critical settings. If you are using a combination device and experiencing issues, the first step is to acquire a stand-alone modem and router. 

    • If you can not acquire a stand-alone modem and router, we recommend reaching out to your ISP to see if they can place your combination device in “bridge” mode, then purchasing and connecting a 3rd party router to handle the local networking. 

  • Ensure UDP timeouts are greater than 240 seconds (phones register every 240 sec). If you’re seeing a sawtooth pattern in your latency graphs, or your phones are sometimes unable to be reached, the UDP timeout is probably incorrect.

  • Disable Stateful Packet Inspection (SPI) as it often flags VOIP traffic incorrectly. 

  • Disable any VOIP specific functions that come pre-setup on your network equipment.

  • Sharpen traffic over a VPN is discouraged due to likely latency and quality of service concerns.

  • Sharpen traffic over an MPLS is discouraged due to potential inefficient route paths to voice resources.

Ports and Protocols

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.

  • 80 HTTP - Default port for web browser traffic

  • 443 HTTPS - Default secure port for web browser traffic

  • 8089 TCP - Used for establishing Websocket connections.

  • 8090 TCP - Used for establishing Websocket connections.

  • 9002 TCP - Used for establishing WebRTC connections.

  • 10000-20000 UDP - Port range used for media transmission through WebRTC.

Due to the dynamic nature of Sharpen infrastructure, whitelisting is not recommended. The following items may be changed without advance notice.

If whitelisting traffic, please allow the following.

  • *.sharpencx.com - main app domain

  • *.sharpen.cx - supplemental main app domain

  • *.cx.shpn.co - dotCX domain

  • *.sipvbx.com - voice infrastructure domain

  • *.fathomvoice.com - legacy api domain

  • *.s3.amazonaws.com - long-term storage domain for audio and image files

  • *.yealink.com - Yealink registration domain

  • *.ckeditor.com - visual editor/UI library domain

  • *.loggly.com - logging tool domain

  • *.pendo.io - logging tool domain

  • *.ingest.io - logging tool domain

  • *.gstatic.com - font library domain

  • *.googleapis.com - font library domain

  • *.fontawesome.com - font library domain

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.

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