GStreamer WebRTC Wrapper - Introduction - WebRTC Fundaments
| GStreamer WebRTC Wrapper |
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| Introduction |
| Architecture |
| Evaluating GstWebRTCWrapper |
| Getting the Code |
| GStreamer Pipelines |
| Other WebRTC RidgeRun products |
| Contact Us |
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 JavaScript API, HTML5, underlying communication protocols, etc. are defined by the World Wide Web Consortium (W3C) and Internet Engineering Task Force (IETF) jointly in order to make a trustworthy communication channel between the future web browsers. Essentially, WebRTC allows defining an API that provides secure access to the input peripherals such as microphones and webcams on a device to share or exchange the media data, 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 to use Javascript, application programming interfaces (APIs), and HTML5 to embed the communication technologies within the browser.
- Google Hangouts, Whatsapp, Facebook Messenger, Skype for Web, etc. all are integrated with WebRTC.
- Browsers are able to 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.
