WebRTC: the new standard for real-time communication in the Cloud?
The Cloud covers most IT applications ranging from services in SaaS to infrastructure in IaaS. Integrating Communication as a Service (CaaS) into the mix means providing real-time data processing for audio and video communication from browser to browser.
a new standard under development
Developed by the IETF and the W3C, this new standard will make it easier to integrate real-time audio and video communication services (Web Real-Time Communication) into browsers and applications running on the Cloud with no plugins required on the user side.
Major players in cloud computing signaled the need for a new CaaS standard over a year ago. This is now being developed by two organizations: IETF and W3C. Called RTCWeb by IETF and WebRTC by W3C, the new standard will establish the protocol and APIs needed for audio and video communication between users.
chart showing the WebRTC architecture
how it works
A WebRTC client and 2 APIs, called Media Stream and Peer Connection, are built into the browser.
- after login, media is transmitted on the Peer Connection API (RTP/UDP)
- the Media Stream API then manages (In/Out) the various interfaces on the terminal (microphones, speakers, webcams, etc.)
beta versions already available
The WebRTC standard will provide fast access to communication services in an HTML5 browser.
It is seen as a new way to provide communication services in the Cloud for various fixed and mobile devices. It will also make installation and updates much easier.
While the standard has yet to be finalized (coding, NAT traversal, interconnection, etc.), most Web browsers are starting to offer beta versions for use with their technology.
This blog post was originally published in French here.
Photo credit: © Logostylish - Fotolia.com
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