The Dawn of WebRTC

Web Real-Time Communications (WebRTC) was built to provide developers with the ability to create high definition video and audio calls using simple JavaScript APIs. These APIs are embedded directly in the browser and require no plugins, downloads, or installation of any type to get you up and running. Google spent about $200 million to open source the technology giving it to the development community. WebRTC uses several codecs for video and audio giving anyone the ability to create next generation communication apps without the need to pay for licensing or royalties.

What are the Possibilities?

We have only begun to scratch the surface of how WebRTC will change the communications industry. We are seeing all types of applications being created with WebRTC. One of the most iconic examples is Amazon's Mayday Button. It shows the true power of how WebRTC is being harnessed by companies large and small. WebRTC brings many abilities for you to enhance your apps such as:

• Video Communications: Create secure and high definition audio and video streams between browsers
• File Sharing & Messaging: Securely connect and share data between browsers without the need to upload files to the cloud or a network server. Data is sent directly between the connected peers
• Phone to Browser: WebRTC allows for connections between Public Switched Telephone Network (PSTN) and browsers. You can make and receive calls all from one location with the use of the new APIs in HTML5, a SIP Gateway and WebRTC
• Mobile to Mobile: WebRTC is not just for the web, there are native libraries for both iOS and Android that utilize WebRTC's capabilities
• Machine to Machine: WebRTC is embeddable for systems needing to communicate machine to machine such as with the Internet of Things. Google Chromecast is a perfect example of using WebRTC outside the normal use case

Understanding the WebRTC APIs

WebRTC relies on three JavaScript APIs embedded directly into web browsers requiring no client or browser plugin in order to communicate directly with another WebRTC enabled browser. These APIs are:

• MediaStream (aka getUserMedia) allows you to gain access to the camera, microphone, or screen of the device employed by the user. As an added layer of security, the user will have grant access before you will be allowed to stream their media. If the user connects from a secure connection (HTTPS) the user will only need to grant access once for the application but if you connect from a non-secure connection (HTTP) the user will be prompted each time the application needs access
• RTCPeerConnection (aka PeerConnection) allows two users to communicate directly, peer to peer. It encodes and decodes media sent to and from your local machine to a remote peer receiving your media.
• RTCDataChannel (aka DataChannel) represents a bi-directional data channel between two peers. It piggy backs on top of the RTCPeerConnection allowing you to send data directly between the two connected peers securely.

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by Lantre Barr via SitePoint