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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.
Traffic allowed to and from
.sharpencx.com
.sipvbx.com
.fathomvoice.com
.s3.amazonaws.com
.yealink.com
.ckeditor.com
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.
Recommendations
<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.
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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