Below are the summary and the presentation slides.
Web browsers will now use WebRTC (Web Real-Time Communication) to communicate with one another. WebRTC can also be used to communicate with existing telecommunication networks. As a result, voice services in existing telecommunication networks will likely drop as customers will pay more for data services in order to use WebRTC. In addition, the number of VOIP & IM applications on a PC would likely reduce. WebRTC is a game changer, and it is bound to take some of the market share of voice and data services from the telco operators. This talk will take a look at WebRTC and its potentials. It will also include a short demonstration.
CPUT FOSHS ’12 has come and gone. Many thanks to our presenters, supporters and colleagues (Dr. Ernest Pineteh and Wilhelm Rothman) that help organized it. On day 1, Wilhelm Rothman gave an interesting talk on VPN (Virtual Private Network)) and did some demonstrations using OpenVPN. In addition, I gave a talk on ARM devices. I also did some demonstrations using Pandaboard. Day 2 was for the industry players to come present. All the presenters gave mind-blowing talks, such as Using Chef for System Automation and MongoDB Geospatial. Below are the videos of some of the talks.
Schalk Heunis (Ph.D) and I were invited to the Google IO (Extended) last Thursday to talk on Arduino and arm devices, respectively. The event was an interesting one with participants from the academic environment and the industry. The event took place a day after Google announced the release of Nexus 7, Nexus Q, the Google Event app, and Android 4.1 (Jelly Bean). It was all fun, and I thought of recording the talks for anyone that might be interested. I was only able to cover the first part of Schalk’s talk; after it, the recording application started acting up and would not record the second part and my own presentation.
It’s nice to see various implementations of SIP on the Web; .I also wrote about the interworking of SIP and WebRTC sometime ago. Below is an excerpt from the concluding part in my doctoral thesis.
In summary, while session handoff has been widely explored, content sharing and the proxy services are relatively new services in the Web-browsing context. These services could encourage collaboration and community interaction between the Internet users. In addition, having shown that the integration of a SIP stack into a Web browser makes no significant change on the memory footprint or quality of experience, the inclusion of SIP in commercial Web browsers is not only feasible, but also will offer new services to end users. SIP is an extensible protocol that is not only used in multimedia services provisioning, but also in control and automation, such as smart homes. Should Free Open Source Software (FOSS) and Open Standards be widely adopted, many more innovative solutions, like this project, would be introduced into the Web browsing experience and found in this Web 2.0 era as services are rapidly converging.
SIP is good at what it does, and the future is the web .
Mayowa Mulero had her graduation this morning. She has now earned an M.Tech degree in Information Technology. Mayowa used to be one of my diligent masters students; she completed her masters degree (by dissertation) in less than 18months. In addition, she published a couple of papers in the course of studies (IST-Africa ’11, IASTED CIIT ’12, e.t.c.), and her thesis sailed through external examination with no request for corrections. I wish you, Mayowa, the very busy in all your future endeavours.
The ITU Kaliedoscope ’11 conference came to an end yesterday. It is one of those conferences I have always wanted to attend, and I am glad this year’s conference took place at my backyard. It was at the University of Cape Town. It was nice meeting folks from Italy, most notably University of Bologna, where I have got some collaborators. The paper presentations were great, and I am really impressed with the diverse papers on policies, services, rural development and so on.
Many thanks to those that made it happen. I remember Paul Inglesby told me at a function (I think AfricaCom ’10) at CTICC that he wanted the next ITU Kaliedoscope conference to take place here.
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.