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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.
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:
WebRTC allows browsers to stream files directly to one another, reducing or entirely removing the need for server-side file hosting. WebTorrent uses a WebRTC transport to enable peer-to-peer file sharing using the BitTorrent protocol in the browser. [29]
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.
Jitsi Videobridge is a video conferencing solution supporting WebRTC that allows multiuser video communication. It is a Selective Forwarding Unit (SFU) and only forwards the selected streams to other participating users in the video conference call, therefore, CPU horsepower is not that critical for performance.
STUN is a tool used by other protocols, such as Interactive Connectivity Establishment (ICE), the Session Initiation Protocol (SIP), and WebRTC. It provides a tool for hosts to discover the presence of a network address translator, and to discover the mapped, usually public, Internet Protocol (IP) address and port number that the NAT has ...
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.