I reported the similarities and differences between Google Wave and my work TransferHTTP + CAS two years ago. It is fascinating to see another related work called WebRTC. I would not be discussing the similarities and differences between it and my work this time around. Rather, I will present an excerpt of one of my papers that briefly discusses WebRTC. See below.
WebRTC is an open framework that offers web application developers the ability to write rich real-time multimedia applications (e.g. video and gaming applications) on the web without requiring plugins or extensions. Its purpose is to help build a strong Real Time Communication (RTC) platform that works across multiple web browsers and platforms. In an implementation, the WebRTC API will abstract several key components for real-time audio, video, networking and signal [1, 2].
One of the IETF RTCWEB WG  is currently discussing how to integrate WebRTC with deployed SIP equipment and domains. An area of its application is being able to communicate from WebRTC applications to existing deployed SIP/RTP-based Voice/Video-over-IP devices at the signalling and media planes. It may require an interworking middlebox function (e.g. an integrated Web Server module) in the media-plane. However, the deployed devices should communicate using SIP at a signaling layer rather than HTTP. Other protocol implementations, such as XMPP and H.323, can also be achieved.
From the industry perspective, the web browser software industry is also implementing browser-to-browser interaction in various ways. Although WebRTC is currently being standardized, it is however possible that some of the its implementations might require extending an existing terminal (like a web client in our work), a proxy or a server.
- WebRTC, http://www.webrtc.org, Accessed on November 20, 2011.
- IETF WebRTC, http://tools.ietf.org/wg/rtcweb, Accessed on November 20, 2011.
- IETF RTCWeb-SIP WG, http://tools.ietf.org/html/draft-kaplan-rtcweb-sip-interworking-requirements-01, Accessed on November 20, 2011.