Call for speakers

WebRTC Conference 2012

Providing interactive communications between two peers web-browsers is possible already today. Indeed, there are a number of proprietary implementations allowing rich communications using audio, video or any collaboration. But these solutions are not interoperable as they require non-standard extensions or plugins to work.

WebRTC was imagined to solve this interoperability issue. It is an open framework for the web that enables Real Time Communications in the browser. It includes the fundamental building blocks for high quality communications on the web such as network, audio and video components used in voice and video telephone, conferencing and chat applications. These components, when implemented in a browser, can be accessed through a Javascript API, enabling developers to easily implement their own RTC web app.

Achieving Backwards Compatibility and Interoperability

The standards, the W3C is working on, defines a set of APIs that allow local media, including audio and video, to be requested from a platform, media to be sent over the network to another browser or device implementing the appropriate set of real-time protocols, and media received from another browser or device to be processed and displayed locally. This specification is being developed in conjunction with a protocol specification developed by the IETF RTCWEB working group.

Once these APIs are stable, backwards compatibility and interoperability will be achieved. The WebRTC API layer will be the main focus for stability and interoperability.

During the WebRTC conference, to be held in Novotel Wellness & Convention Centre Paris Roissy CDG, from 10 to 12 October, 2012, experts will address all technical and standardization issues that still need to be solved.

The following list of topics is not exhaustive and authors may propose other subjects in keeping within the thematic framework.

• Standardization update
• Reports on Implementations and Interoperability
• Business models for carriers and telcos
• Security issues
• Scalability & management

Web APIs
• API overview
• Developping apps
• HTML5 development toolkits and framework
• Support for teleconferencing

Audio and Video Transport
• Codecs (Opus, VP8 …)
• Packet loss and jitter handling
• Devices for WebRTC (headsets…)
• Wireless WebRTC
• Spatial audio teleconferencing
• Hardware access and control

• Buffers
• Peer to peer connections and NAT traversal
• Efficient and real time transport for audio, video and application
• Security and lawful interception RTP-over-TCP

Abstracts must not exceed one page. They may be submitted by email at: info@upperside.fr or remi.scavenius@wanadoo.fr

This call is now closed. You can find more information here.

  • Closed on: 31st July 2012