Workstation Requirements
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.
Hardware/Software Requirements
Component | Specification |
---|---|
OS | Windows 7 or greater OSX Yosemite 10.10 or greater |
|
|
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
|
Headsets |
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Browser |
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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.
Access to these domains need to be open regardless of the isolation zone (IZ0,IZ1) in which your account is built. These include some supporting services and libraries which allow Sharpen to run as designed.
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 |
*.chameleon.io | TCP: 443 | Application enablement |
*.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.
Isolation Zone 0 | You login to 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 | You login to 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 |
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.
SIP ALG is enabled
Perhaps contrary to its name, SIP ALG is not compatible with most enterprise VoIP solutions, such as Sharpen. Depending on the manufacturer, network device configurations will show up as “SIP ALG”, “SIP”, “VoIP”, or something similar. Intermittent disconnection, dropped calls, one-way audio, and the inability to register are common symptoms when SIP ALG is enabled.
Depending on your device manufacturer or ISP, it may be difficult to get a straight answer on confirming this setting is off. Commonly, ISPs will have this setting enabled, because it supports their own options for VoIP solutions. It is not uncommon to have to work through a couple layers of support or technical team members to validate the proper setting is turned off.
UDP timeout is set less than 240 seconds
Sharpen’s expected SIP registration interval is 4 minutes (240 seconds). If your network is set to “timeout” UDP connections at less than that, it will disconnect an active registration. Depending on when this happens, you’ll see the following symptoms.
Dropped calls
Inability to be reached on the phone which has lost its connection.
You’ll be able to dial outbound without issue, since registration is established on an attempted outbound call, if it does not already exist.
You’re probably seeing a sawtooth pattern in your latency graphs.
ISP provided “combination” network equipment
Especially if you’re working from home on your residential internet connection, be weary of the Internet Service Provider (ISP) controlled settings which may exist on these managed devices. “Combination” devices are typically those which integrate modem, router, and wifi into one device. While in principle, the integration of these functions problematic, they can sometimes come with configuration hurdles which are difficult to overcome since you, as the borrower of the device, do not have administration access to the devices. It is not uncommon for settings like SIP ALG to be enabled but invisible to you as the user. These situations require that you work with your ISP’s support team to change a setting.
As a result, Sharpen discourages the use of combination network equipment. Instead we recommend purchasing a stand alone modem which is compatible with your ISP, and connecting a router of choice to it. This allows you full control of any potentially conflicting setting. Most self-managed devices have these problematic settings disabled by default.
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.
VPN
VPNs have many valid use cases, especially for work from home users. However, VPNs are not always configured to be optimal for voice traffic. Adding the additional virtual layer to the network, in most situations, will cause a recognizable degradation in network performance. Voice requires low latency, with minimal packet loss. As a result, if the VPN introduces too much interruption, your quality of service will be impacted. If VPN is necessary, it is recommended to configure it so Sharpen traffic can be omitted through the use of split tunneling. Sharpen will not support voice quality issues involving the use of VPNs.