At the end of 2015, we teamed up with AT&T to write a blog series on UX and WebRTC. We’re excited about the future of communication on the web and hope these posts will help teams get started in creating enjoyable user experiences with WebRTC. Check them out!
Post #1: Getting started with UX and WebRTC
An introduction to WebRTC featuring reasons to include it in your application, a handful of questions to consider during the initial stages of planning, and the importance of consistency and seamless implementation.
Post #2: UX considerations for initiating and joining calls
The first major interaction a user will have with WebRTC is initiating or joining a call. This post covers questions to ask and decisions to consider when designing this part of the flow. How do things like buttons, text, and user expectations affect the experience?
Post #3: UX patterns for transparent voice and video calls
Next, here are some useful patterns and techniques for the voice or video call itself. What information or context could help the user during the call? The theme here is clarity and transparency.
Post #4: Ways connectivity and bandwidth affect WebRTC UX
A major part of the WebRTC experience is call quality, affected most by connectivity and bandwidth. In this post, we cover the costs of interactions and ways to design around them and to optimize for them.
Post #5: Ways to enhance the WebRTC user experience
In the last post in the series, discover ways to further enhance the WebRTC experience and make your application more accessible, private, and user-driven.