WebRTC is an exciting new technology, perhaps the most exciting thing to happen to voice communication since the invention of Voice over IP. With WebRTC, we are no longer limited to a disjointed communication experience with poor quality audio on antiquated networks. Now we have the ability to put high-definition audio and video where it will have the most impact: right in line with the business processes that benefit the most from it.
This session will present an overview of how WebRTC works, reviewing both the network services that support it and the user-facing software that delivers it. We will look at how Asterisk can be used to give WebRTC additional capabilities that aren’t possible with browsers alone, and how to deploy Asterisk to get the most out of this powerful combination.
As with all new technology, however, there are rough edges. In the final part of this presentation, we will look at the common ways that WebRTC can break down, from technical deployment problems to user interface and design issues. These lessons are drawn from real-world experience deploying WebRTC over the last 3 years and multiple applications that are in production today.
9 дек 2015
