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WebRTC is an HTML5 specification which can be used to facilitate real-time media communications (video and audio) between browsers and other audio endpoints. The Sharpen Q phone built into the Sharpen Q application leverages WebRTC, allowing for seamless integration to the platform within the browser. The following are necessary for successful webRTC functionality.
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80 HTTP - Default port for web browser traffic
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443 HTTPS - Default secure port for web browser traffic
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8089 TCP - Used for establishing Websocket connections.
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8090 TCP - Used for establishing Websocket connections.
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9002 TCP - Used for establishing Websocket connections.
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*See “Ports and Protocols” above.
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.
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