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 defines several APIs together with a collection of communication processes and protocols. From a development perspective, core functions are encapsulated into three main JavaScript APIs: getUserMedia, RTCPeerConnection and RTCDataChannel. These APIs are incorporated into browsers that support WebRTC, hence a web developer who has JavaScript programming experience can bring an interactive video collaboration experience to the web.
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.
Within the financial services sector, real-time video and audio will be extremely useful for discussing transactions and agreements efficiently and securely. For example, when securing a loan the customer visits the website to check the lender's rates. They decide they want more information, but instead of having to arrange a meeting with a lender representative at a later date, they can launch an instant video or audio call in their browser to agree the deal. WebRTC signifies the next stage in B2C management.
WebRTC lends itself particularly well to telemedicine, specifically in terms of aftercare and therapy (not emergency treatment) as it removes the need for patients or medical professionals to travel excessively. This improves the overall care for, and recovery of, the patient, as well as reducing the costs associated with the treatment.
Polycom can already see where some of its customers could make use of WebRTC. For example, NHS Lothian is using Polycom RealPresence CloudAXIS for patient follow up in home as part of a breastfeeding clinic. But the average new mother probably won't have a brand new PC or mobile device at home, so their older technology probably won't be capable of running WebRTC, as it puts a greater load on the host device.
Within the social and public sector there are clear opportunities for applications, as part of the drive towards digital access and inclusion. Often governmental regulations can be complex for citizens to understand, and the opportunity to ask questions, face-to-face and immediately, would improve the level of service provided to citizens at a local and national level, as well as increase levels of compliance.
Many of us will have used the chat function on retailers' websites. For sales advisors and customer service departments, the move to a more human interaction over video that works at the click of a button, would be great in terms of improving sales and customer satisfaction.
Issues and considerations
Worldwide there are over three million devices which function through SIP, ISDN or H323, and all of these will need to communicate with the WebRTC standard. Given the average time for IT refresh cycles (four to six years before legacy hardware is supplanted by soft clients), these devices will not disappear before WebRTC enters the market.
Businesses would benefit from using a standards-based system to support WebRTC, as they would be able to mix existing IT deployments with the new standard in an interoperable environment.
In terms of actually getting WebRTC to the market, the industry will probably look at creating specialised clients based on the core building block of WebRTC. But until all browsers can support WebRTC and can do so with the requisite quality, it makes sense to maintain plug-in alternatives to fill the gaps in WebRTC functionality.
When it comes to hardware and networking requirements, a WebRTC video application will demand a reasonable host. The bandwidth will not be high, and may even run at a better bandwidth consumption level than today's video clients because the algorithms used are highly efficient. These algorithms are efficient but complex, so they need greater processing power built into the actual device.
This means they may not work on outdated devices, including current tablets and smartphones. It may well be the case that the devices released in the last two years will be capable of running WebRTC based on the in-development H.265 video compression code, which is 50% more efficient than H.264 but requires four to five times more processing power. In this sense, from a performance point of view, WebRTC doesn't come for free.
- Andrew Hug is Vice President, Sales Engineers EMEA, at Polycom
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