As the web platform continues to evolve, tools have emerged for connecting people and computers in new and interesting ways. Web Real-Time Communication (WebRTC) stands out as one of the most significant and disruptive of these emerging technologies, allowing developers to embed peer-to-peer real-time communication in the browser without proprietary plugins, while breaking away from the traditional client-server paradigm.
Since Google released and open-sourced the WebRTC project in early 2011, the Internet Engineering Task Force (IETF) and the World Wide Web Consortium (W3C) have been working together to formalize the WebRTC standard and 1.0 stable release. Companies like Twilio and Vidyo have adopted WebRTC as a protocol for video chat in the browser, and established telco and VoIP players such as Cisco, Ericsson, and Telefónica have also lent support to the project.
At the most recent meeting of the IETF, Simon Pietro Romano, author of Real-Time Communication with WebRTC, hosted a panel to discuss developments in the WebRTC community and the road ahead. The panel, who are driving forces behind ongoing standardization and implementation, included:
- Justin Uberti, tech lead for the WebRTC team at Google
- Eric Rescorla of Mozilla
- Dan Burnett, editor of the PeerConnection and getUserMedia W3C WEBRTC specification
- Cullen Jennings, Cisco Fellow and Co-Chair of the IETF RTCWeb
I’d encourage you to listen to the whole conversation, but to get started, you might explore these highlights.
What is WebRTC?
Simon starts off the discussion [2:54] with a brief introduction to the basics of WebRTC from Justin and Cullen. As Justin explains, at its core, WebRTC is a set of standards and APIs that allow supported browsers (Google Chrome and Mozilla Firefox) to share peer-to-peer audio, video and data connections. Cullen points out [3:47] we’ve been using voice and audio over the web for a while now, but one of the reasons WebRTC is taking off is the ease and accessibility it offers developers when it comes to building applications.
More than just a set of APIs
Justin reminds us [4:44] that WebRTC is also more than just a set of APIs, but also a set of protocols for how audio, video and data communication should be done on the Web. This is a far reaching effort and goes beyond the browser to all different kinds of modern devices communicating using these protocols.
No more plugins
User privacy and security is important
While users want the ability to share their screen with WebRTC this poses a complicated issue in terms of security [10:11]. The panel discusses steps to creating a modern and secure infrastructure where sensitive user data cannot be scraped and the challenges that may not be obvious to end users.
An evolving mobile experience
Switching gears, Cullen discusses how mobile development with WebRTC [13:20] has allowed for cross-pollination between developer and designer environments and accelerated the ability for rapid change.
Controversial or designed to work together?
Simon asks [14:45] the group to discuss the ORTC (Object Real-time Communications) initiative which was originally seen as a disruptive solution for web real-time that aimed to offer a low-level API and fine grain control for developers. However, the group points out that there is room to have it both ways in the community and the existing high-level WebRTC API and low-level API are ultimately designed to work together. Finally, the group closes out with a discussion of the role of telcos [20:52] and how WebRTC brings together two previously separated camps of web development and telecommunications. A number of telcos and VoIP companies have brought in support and expertise to the party, and while friction still exists across groups, different perspectives will continue to bring innovation to the field.