GStreamer WebRTC Wrapper - Introduction - WebRTC Fundaments
| GStreamer WebRTC Wrapper |
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| Introduction |
| Architecture |
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| Evaluating GstWebRTCWrapper |
| Getting the Code |
| GStreamer Pipelines |
| Other WebRTC RidgeRun products |
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What is WebRTC?
WebRTC is a Web Real-Time Communication project started by Google in 2011. It is an open-source and free project that provides real-time communication to mobile applications and web browsers. It is a new standard that extends the web-browsing model to handle the browser in a peer-to-peer manner. Also, WebRTC standards such as the JavaScript API, HTML5, underlying communication protocols, etc. are defined by the World Wide Web Consortium (W3C) and the Internet Engineering Task Force (IETF) jointly in order to make a trustworthy communication channel between future web browsers. Essentially, WebRTC allows defining an API that provides secure access to input peripherals such as microphones and webcams on a device to share or exchange media data and real-time data with a remote device in a peer-to-peer manner.
Why WebRTC?
- It is a plugin-free modern real-time communication technology. It doesn’t require any additional plugins or applications for audio, video streaming, and data sharing.
- It is implemented using JavaScript, application programming interfaces (APIs), and HTML5 to embed the communication technologies within the browser.
- Google Hangouts, WhatsApp, Facebook Messenger, Skype for Web, etc., are all integrated with WebRTC.
- Browsers can directly exchange real-time media with other browsers in a peer-to-peer manner.
- Offers a high level of security, higher than various other streaming systems, without the need for third-party software.
- It is available for free and is operated worldwide, which is the main booster for this technology.
- It has multiple mechanisms that allow two computers to communicate, regardless of the network topologies they may be connected to.
- It extends encryption mechanisms to share information.