The data and comms industry is well known for its acronyms and abbreviations, and now we have another one – webRTC. WebRTC is an open source project aiming to enable the web with Real Time Communication (RTC) capabilities. Not only is it new, but it also has its origins in the non-Telco world of the world wide web, creating the intriguing possibility of a whole range of innovative web based comms applications and opportunities. But what does this mean for the voice and data channel, and is webRTC an opportunity or a threat?
The web has been immensely successful in storing and distributing information to millions of users, but if you want to use real time communications over the web, you need to download multiple clients and apps to make a voice call or send a message. It can be a messy and sometimes frustrating experience - I am sure we have all experienced those delayed webinars as your PC attempts to download the required plug in.
WebRTC is an open source technology based on HTML5, pioneered by Google, which launched the technology following their acquisition of Global IP Solutions and their resulting ownership of a number of VoIP patents. It is not, on its own, a service or a product, but rather a technology enabler for almost any type of communication delivered from a browser. It is still in development, with standards and codec choice still being finalised by the W3C and IETF, and it presently does not have the full support of Microsoft and Apple.
Google and other leading vendors developed webRTC to provide support for real time communications - voice, video and data comms - delivered from a browser without the need for any downloads or plugins. As an open source standard, it is designed to have the widest possible appeal by enabling, with a few lines of JavaScript code, developers to build peer to peer communications between browsers. Developers can also select the most suitable signaling protocol to control the media stream (this is not defined within the webRTC standard). For instance for voice apps, it is possible to use webRTC in conjunction with SIP signaling, thereby exposing traditional voice features over the internet and embedding them in web based services. This can potentially provide a great opportunity for hosted SIP services to be integrated into a wide variety of cloud based business applications.
The browser bit is highly significant for a number of reasons. There are presently reckoned to be about 1 billion devices with webRTC enabled browsers, and this is forecast to increase to 4 billion by the end of 2016, resulting in an increasing number of voice and video calls originating or terminating via a webRTC enabled fixed or mobile device.
The standard is also supported not only by Google and Mozilla in their browsers, but also familiar names such as Genband, Cisco, Oracle, Plantronics, Avaya, Dialogic etc., thereby building a substantial ecosystem of supported equipment. This includes components such as webRTC/SIP gateways and SBCs that service providers will deploy in their networks to enable services to be offered across IP, web and PSTN networks for a variety of different end user devices.
As it’s a web technology, it also means that millions of web developers worldwide can start to play around and quickly create new and innovative services by experimenting with real time comms within websites using JavaScript APIs. These APIs will be provided as a service by wholesale SPs, offering a simple toolkit and connectivity for developers to rapidly launch new services and connect to multiple networks, without them knowing anything about SIP or telephony. This is likely to result in lots of new websites offering comms features such as voice or video chat to communities of users, providing bespoke services at a much lower cost than previously possible.
With this type of momentum and numbers, and a multitude of different communication applications, it is expected that the role of voice will change, both in the way an enterprise communicates with its market, and the way individuals communicate with one another. WebRTC will allow the easy and scalable use of multiple media (not just voice) streams simultaneously in one encounter, expanding the comms experience and providing more ‘value add’ over and above a simple phone call. And as it’s browser based, the user experience can be consistent across smart phone, tablet and desktop. It will also make video conferencing and collaboration easier to build and use, as well as substantially reducing its cost.
However it’s not just the ability for webRTC to provide a simple way to offer more effective unified communications. Because it’s browser based it can provide ‘contextual’ communications, e.g. providing browsing history to a sales agent. Voice comms will no longer be a standalone silo, but increasingly will be embedded with other media streams in the apps that we use when we work, contributing to a more engaging ‘user experience’. This will result in, for instance, better contact centre agent communications, more effective on line sales support and more sales closed which, in turn, will increase revenues for businesses willing to adopt these new ecommerce features.
There are likely to be many different models that will emerge over the coming months and years, and it is impossible to predict who and what will be the winners and losers. What is certain is that webRTC has the potential to be a major disrupter in the comms world, with new entrants launching new services resulting in a more fragmented market. With an abundance of enabled devices, ubiquitous bandwidth and an acceptance of cloud services, it is inevitable that we will see strong growth in use of webRTC over the next few years. End users will need guidance, and as with all technology changes, the channel has the opportunity to help customers make the migration and use this exciting technology to create real business benefit.