What exactly is WebRTC and how are the peer-to peer, real-time communication video, audio, and data channels it enables used in different industries such as video conferencing, contact centers, telehealth, insurance, in-context communications, dating and social media, gaming, IoT, and more? Let’s find out.
STUN and TURN are two types of WebRTC signaling servers that can be used to create a real-time, peer-to-peer connection. In this post we will explain why we need them, when we need them, why one is beneficial to the other, and how you can get around the problem altogether using a CPaaS.
Bugs exist. WebRTC is not an exception. Debugging WebRTC applications can be particularly challenging, but not impossible. Arin’s guest for WebRTC Live Episode 49 was our own Senior WebRTC Engineer and Developer Evangelist, Germán Goldenstein. They discuss tools and methodologies for debugging a WebRTC call.
Building in a pre-call test is the most important thing you can do to make your live video application successful. Since we can’t always control call quality, doing a network test, assessing video and audio quality, performing a video and microphone test, and offering a back up plan before the call begins is the best way to ensure customer satisfaction.
