Maintaining quality and performance as complexity increases and requirements change is very difficult to do in any application, particularly in WebRTC applications. Testing is a must, and automated testing can make it much easier. Episode 52 of WebRTC Live covers the basics and types of automated testing, continuous integration and deployment, verifying tests, testing call quality, and more. Watch it here!
Whether to have recording capability in your WebRTC video or audio application is incredibly important to decide on BEFORE you build. You must also consider how much recording, recording layouts, where and how long you keep it, and how secure it needs to be. Let’s look at how recording affects your app’s architectural choices, as well as the question of recording as composite or individual streams.
Timely and adequate notifications in your WebRTC application are essential in delivering a quality user experience. Opportunities to shine or to fail abound, from simply letting someone know another wants to connect to technology issues such as cameras, microphones, and internet strength.
There are a variety of ways you can use traditional telephony in your WebRTC video app, as well as different architectures you can choose to support it from commercial to open source. Let’s explore the reasons you might want to integrate a dial-in or dial-out capability into your WebRTC video or audio application and look at a sample architecture.
