2015: The Deciding Year for WebRTC?
WebRTC (Web Real-Time Communication) is an API definition drafted by the World Wide Web Consortium (W3C) that supports browser-to-browser applications for voice calling, video chat, and P2P file sharing without the need of either internal or external plugins.
WebRTC is not exactly new, having been in existence since about 2011. Until now, however, adoption has been limited to a few situations such as conferencing solutions, and at proof-of-concept stage, for interview or story-telling applications.
At GS Lab, our work in the video communications product space led us to observe the adoption of WebRTC to implement plugin-free browser as an endpoint for video conferencing systems. However, since screen sharing is not officially supported on released browsers due to security concerns, these end points are also not completely plugin free.
I attended the WebRTCWorld conference in San Jose in November Nov 2014, where I saw some exciting applications and concepts, apart from the standard gaming and conferencing ones. Offbeat use cases, such as one for context sensitive voice and video communication for support were demonstrated. As an example, if a user browsing an application (e.g. banking) needs assistance in real time, she has to leave the application and get on the phone to talk to the customer service executive. Now with WebRTC communication embedded into the application, rich context (based on user navigation of web pages and data exchange) is available for the support executive when an audio or video call is directly placed from within the application. This leads to a faster, better and focussed customer service experience. Another interesting application is using the data-channel for secure communication among devices.
A very insightful overview of the WebRTC ecosystem (check details here) which explains the ‘webification of communication’ and forecasts major adoption from 2015 was interesting. Although 2015 started with positive notes such as AT&T announce commercial support of webrtc and some presence at CES, there are still hurdles which preclude widespread adoption.
- Both codec H264 and VP8/9 need to be supported by major technology companies – it’s the same old browser story.
- There is a need for more native clients for devices such as mobiles or tablets. (Android L will support WebRTC through WebView but again it’s limited and not native.)
- For webification of communication to happen, application developers require an easy to build platform, but WebRTC does not provide all the necessary components. For example:
- Complicated standards for restricted network traversal such as STUN, NAT and Firewall are difficult for web developers to understand.
- As presence needs to be managed outside the application, SIP or some proprietary implementation becomes necessary.
- Peer to Peer (P2P) communication is available out of the box, but if there is a need for auditing or some conferencing feature, then the solution needs to run the WebRTC client on the server as an end point (B2B user-agent) and a lot of custom coding is required.
To overcome the above problems, application developers have to work with proprietary platforms which provide them easy to use APIs. We see that most of the communication platform providers are bridging these gaps and providing ‘peace of mind’ to the application developers.
We also see another disruptive innovation for streaming media directly over HTTP, which hides networking complexity, happening with technologies such as HLS/MPEG-DASH/HSS. However, these are fragmented and do not address real time communication use cases. Let’s wait and watch 2015 closely and check how the adoption of WebRTC moves along.