January 6th, 2015 by The Modulis Team
Using webRTC for collaboration
“WebRTC is the ability to communicate live with somebody or something as if you were right there next to them. WebRTC fills a critical gap in the web platform as you can communicate in real-time just by loading a web page,” said Justin Uberti, Tech Lead on WebRTC, Google.
In simple words, WebRTC is about real-time peer-to-peer communication via a web browser! This communication may be in the form of voice, data or video. Robust, plugin, and latency-free communication over complex web have been a dream for as long as we can remember. WebRTC is the answer to this long-awaited quest. The vision of WebRTC is to realize real-time communication capabilities (voice, data, and video) in all web browsers.
Previously, in order to place audio or video calls from a computer, users needed to download proprietary software for which they had to create accounts in order to make calls. WebRTC leverages a recent trend in which a web browser IS the “application,” facilitating browser-to-browser communication with no software downloads or registration needed. Web browsers themselves include the capabilities needed to support standards. WebRTC standardizes communications between browsers, enabling audio and video along with data bridges to support text chat or file-sharing.
Bringing SIP to a web browser is one of the major aspects of WebRTC. By enabling SIP applications in a web browser, we can use the potential of WebRTC APIs to create many telephony solutions, from online calling systems to contact center management systems. The WebRTC offer/answer model fits very naturally into the idea of a SIP signaling mechanism. There are, however, some technical issues that make SIP somewhat of a challenge to implement with WebRTC: connecting to SIP proxies via WebSocket, and sending media streams between browsers and phones.
The Word Wide Web Consortium (W3C) and Internet Engineering Task Force (IETF) are jointly working on the standardization of WebRTC and its API development. The goal behind standardization of the API is to create secure mechanisms for devices to communicate with onboard hardware peripherals (like cameras, audio, etc.) and to establish a reliable and secure peer-to-peer real-time media and data session with different remote devices.
Main Applications of WebRTC:
Some of the major applications of WebRTC include browser based audio and video communication–an easy platform for file sharing. These are also very important applications for call centers. Users do not need to enable third-party software to communicate since a simple browser application will enable audio and video communication between local and remote peers. For an organization’s PBX platform, WebRTC will enable real-time communication with different local or remote PBX users via standard Chrome, Firefox or Internet Explorer browsers. Some other applications of WebRTC include:
- Click-to-Call banners
- CallMe-for-free buttons for yellow pages
- Webphone on site
- Desktop Adobe AIR softphone
- Social network phone apps
- Voice and video mail
- E-learning–seminars and elections by phone
- Internet-TV–running commentary by phone
- Internet-TV/Radio Games–players calls from mobile
Take a look at an interesting WebRTC video chat application worth exploring: apprtc.appspot.com
Key Players in the WebRTC Industry:
W3C is standardizing APIs, and IETF is standardizing the protocols for WebRTC. Google, Mozilla, and Opera are also joining the band. Some of the major softswitch and media gateway vendors like Genband, Sonus, Sansay are also embracing WebRTC and developing next generation communication platforms that support WebRTC related functionality. Asterisk 11 also supports WebRTC.
Modulis has already incorporated WebRTC into its Cloud PBX platform. At Modulis, our team can help you build robust WebRTC based applications to enhance your communication needs. WebRTC is a new frontier in real-time web communications, offering remarkable business opportunities.