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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.
WebRTC (Web Real-Time Communication) is a free and open-source project providing web browsers and mobile applications with real-time communication (RTC) via application programming interfaces (APIs). It allows audio and video communication and streaming to work inside web pages by allowing direct peer-to-peer communication, eliminating the need ...
While no one signalling protocol is mandated by the WebRTC specifications, [3] SIP over WebSockets (RFC 7118) is often used partially due to the applicability of SIP to most of the envisaged communication scenarios as well as the availability of open source software such as JsSIP. In such a case the SBC acts as a gateway between the WebRTC ...
JsSIP uses the SIP over WebSocket transport for sending and receiving SIP requests and responses, and thus, it requires a SIP proxy/server with WebSocket support. Currently the following SIP servers have been tested and are using JsSIP as the basis for their WebRTC Gateway functionality:
FreeSWITCH is a free and open-source telephony software for real-time communication protocols using audio, video, text and other forms of media. The software has applications in WebRTC, voice over Internet Protocol (VoIP), video transcoding, Multipoint Control Unit (MCU) functionality and supports Session Initiation Protocol (SIP) features.
As a result was created RTMFP SIP Gateway, which allowed to make SIP calls from a browser with support for Flash Player. 2013 - The rapid development of WebRTC technology made to implement support of this technology. As a result, version Web Call Server 3 supported both protocols (WebRTC and RTMFP) for SIP calls from a browser.
For example, the Session Initiation Protocol (SIP) communicates the IP address of network clients for registration with a location service, so that telephone calls may be routed to registered clients. ICE provides a framework with which a communicating peer may discover and communicate its public IP address so that it can be reached by other peers.
Simple Gateway Control Protocol; SIP extensions for the IP Multimedia Subsystem; List of SIP response codes; SIP trunking; ... WebRTC Gateway This page was last ...