If you're not familiar with WebRTC, it's the technology which will cause the next drive towards mass adoption of voice, video and file collaboration. In essence, it will enable high quality video and audio in your web browser, which is one of the most deployed applications in the world.
It has been drafted by W3C (World Wide Web Consortium) with protocol work done by IETF (Internet Engineering Task Force). WebRTC ultimately simplifies the incorporation of real-time communications into a web browser.
If you want to have videoconferencing on a PC, tablet or mobile, you currently download a client, which means a transfer of a software application. This causes some challenges, particularly in terms of file size (some clients are very large in terms of software) and download time. Also, in many organisations your PC will be locked down so you can't download software for security reasons.
The original idea behind the development of WebRTC was specifically targeted at real-time communications applications, including video, audio and content sharing, where any delays to delivery make the information meaningless.
The objective of WebRTC is to trigger a real-time session in your browser, without having to download anything, simply functioning as part of the normal operation of browser. Everything required to deliver a high quality experience at the endpoint is supported natively within a WebRTC-capable web browser.
How does the tech work?
WebRTC allows a mesh-based technology to enable users to send and receive streams to and from each other. This is not a new concept, but each stream operates independently, which reduces the strains of conferencing applications (as bandwidth doesn't aggregate to a single choke point) unless, of course, bandwidth inefficiencies come into play.
In theory the mesh approach to a multipoint session can accommodate an infinite number of participants on a call. In practice, however, the more parties that join a call, the more bandwidth that call consumes. Bandwidth inefficiencies can mount quickly, as each device connected to the call receives and transmits multiple transmissions. If the available bandwidth expires the quality suffers and the call can ultimately fail.
On these more complex calls, signalling factors in as well. In the past, Session Initiation Protocol (SIP) has provided a way to register users and identify them uniquely, as well as to manage call notifications and modifications. WebRTC in its infancy does not include a concrete means of signalling, leaving some basic call functionality up in the air. Without protocols for connecting, disconnecting and identification, disorder can ensue.
In terms of security, WebRTC has robust measures built in as a basic standard. All media channels are encrypted using SRTP (Secure Real-time Transport Protocol) and encryption keys are exchanged via DTLS (Datagram Transport Layer Security). Even within the browser, the end user must presently give explicit permission for the browser to access local media resources such as microphones, cameras, etc, and they must renew this permission every session.
There are still some security issues to be considered around other aspects, such as content sharing, but the likelihood is that these additional functionalities would be provided by a third-party, such as Polycom. In that way, the user would benefit from the exceptional security standards of an enterprise level solution.
What is the potential for businesses?
Although WebRTC isn't confined solely to web applications, embedding real-time communications directly into web browsers has been the focus for most of the industry. With WebRTC it becomes possible to embed real-time video into a range of vertical applications, including for business, medical and education purposes.