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WebRTC Gateway : ウィキペディア英語版 | WebRTC Gateway WebRTC Gateway connects between WebRTC and an established VoIP technology such as SIP . WebRTC (Web Real-Time Communication) is an API definition drafted by the World Wide Web Consortium (W3C) that supports browser-to-browser applications for voice calling, video chat, and messaging without the need of either internal or external plugins. ==Usage Scenario==
To enable browsers using different application providers to communicate with each other (e.g. a user logged in to application providers X wants to call someone that is logged in to application provider Y) a so-called WebRTC trapezoid can be used. In this case the two providers use a widely used VoIP signalling protocol such as SIP to federate between them. However, each of their respective browser-based clients signals to its server using proprietary application protocols built on top of HTTP and Websockets. This component that mediates between WebRTC and SIP is referred to as a WebRTC Gateway. Beside connecting different WebRTC applications, a WebRTC gateway also enables the communication between a WebRTC phone and a VoIP or even a PSTN phone. Thereby, a WebRTC gateway extends the scope of a WebRTC applications and enables much wider reach and usage scenarios.〔(【引用サイトリンク】 title= WebRTC:How and Why? )〕
抄文引用元・出典: フリー百科事典『 ウィキペディア(Wikipedia)』 ■ウィキペディアで「WebRTC Gateway」の詳細全文を読む
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