Still in its relative infancy, the cutting edge technology of WebRTC is already demonstrating its potential to transform the nature of business communications.
Setting A New Standard
WebRTC is an HTML5 specification that you can use to add real-time media communications directly between web browsers and devices. The technology enables web applications and sites to capture and optionally stream audio and / or video media. It also enables the exchange of arbitrary data between browsers, without requiring an intermediary.
WebRTC consists of a set of standards which make it possible to share data and perform teleconferencing on a peer-to-peer basis, without requiring users to install plug-ins or any other third-party software.
The Potential To Transform
WebRTC consists of several interrelated application programming interfaces (APIs) and protocols, which work with the well-established Media Capture and Streams API to bring powerful multimedia capabilities to the web. These include support for audio and video conferencing, file exchange, identity management, and integration with legacy telephone systems. Media streams can consist of any number of tracks of information.
Using WebRTC, connections between communicating nodes or peers can be made without the need for special drivers or plug-ins – and they can often be made without any intermediary servers. What this does in effect is open up multimedia access across a range of devices. So voice and video communication can be made to work inside web pages using the same mechanism that allows a user to gain access to the microphone on their desktop system, or the camera on their phone.
Besides voice and video, WebRTC may be used for data delivery, group calling services, or to host recordings. What’s more, the technology isn’t limited to browsers and web pages – it can be used in mobile apps, as well.
WebRTC In Action
Media tracks transmitted via WebRTC may contain one of a variety of data types including audio, video, and text (such as subtitles or even chapter names). WebRTC media streams consist of various tracks, and typically include at least one audio track and a video track. This enables them to be used for sending and receiving both live media and stored media information (such as a streamed movie).
The WebRTC data exchange capabilities allow for the transmission of back-channel information between peers, the exchange of metadata, game status packets, and file transfers. The mechanism may even be used as a primary channel for data transfer.
Though the technology is still relatively new, a number of WebRTC applications are already under way. WebRTC is currently available in most modern browsers, including Google Chrome, Mozilla Firefox, and Microsoft Edge. Apple has announced its intention to build WebRTC support into its Safari browser. This functionality allows users to make and receive phone calls directly from their computers, through their desktop browser.
The WebRTC open source code has already been used in a number of mobile apps, and there are software development kits (SDKs) for both mobile and embedded web environments, giving developers the opportunity to create new business models and use cases.
The new net2phone business communications platform is fully integrated with WebRTC. With net2phone, email support, Live Chat, voicemail, text, and instant messages may be brought together over multiple devices and multiple channels – all of which can be easily administered from a single dashboard.
If you’d like to know more about how WebRTC can transform your business communications, get in touch with the experts at net2phone.