Getting Started with WebRTC -- Table of Contents -- Getting Started with WebRTC -- Credits -- About the Author -- About the Reviewers -- www.PacktPub.com -- Support files, eBooks, discount offers and more -- Why Subscribe? -- Free Access for Packt account holders -- Preface -- What this book covers -- What you need for this book -- Who this book is for -- Conventions -- Reader feedback -- Customer support -- Downloading the example code -- Errata -- Piracy -- Questions -- 1. An Introduction to Web-based Real-Time Communication -- Introducing WebRTC -- Uses for WebRTC -- Try WebRTC yourself right now! -- Browser compatibility -- Chrome and Firefox on the PC -- Chrome and Firefox on Android -- Opera -- Microsoft -- Apple -- Staying up-to-date -- Summary -- 2. A More Technical Introduction to Web-based Real-Time Communication -- Setting up communication -- The general flow -- Connect users -- Start signals -- Find candidates -- Negotiate media sessions -- Start RTCPeerConnection streams -- Using WebSockets -- Other signaling options -- MediaStream API -- RTCPeerConnection API -- The caller's flow -- Register the onicecandidate handler -- Register the onaddstream handler -- Register the message handler -- Use getUserMedia to access the local camera -- The JSEP offer/answer process -- The callee's flow -- Register the onicecandidate handler -- Register the onaddstream handler -- Register the message handler -- Use getUserMedia to access the local camera -- The JSEP offer/answer process -- Where does RTCPeerConnection sit? -- RTCDataChannel API -- Summary -- 3. Creating a Real-time Video Call -- Setting up a simple WebRTC video call -- Using a web server to connect two users -- Setting up a signaling server -- Creating an offer in the caller's browser -- Creating an answer in the callee's browser -- Previewing the local video streams.

Establishing peer-to-peer streams -- Stream processing options -- Extending this example into a Chatroulette app -- Summary -- 4. Creating an Audio Only Call -- Setting up a simple WebRTC audio only call -- The HTML user interface for audio only calls -- Adding an audio only flow to the signaling server -- Audio stream processing options -- Summary -- 5. Adding Text-based Chat -- Adding text-based chat to our video chat app -- The HTML user interface for text-based chat -- Adding JavaScript functions to enable chatting -- Handling text-based chat signals on the server -- Other text message processing options -- Summary -- 6. Adding File Sharing -- Adding file sharing to our video chat app -- The HTML user interface for file sharing -- Adding JavaScript for enabling file sharing -- Adding files using the element -- Adding support for drag-and-drop -- Adding JavaScript for transferring files via WebSockets -- Handling the file-sharing signals on the server -- Sending a thumbnail preview before the entire file -- Providing progress updates -- Establishing an RTCDataChannel connection -- Transfering files via an RTCDataChannel connection -- Other file-sharing options -- Summary -- 7. Example Application 1 - Education and E-learning -- Applying WebRTC for education and e-learning -- Overall application architecture -- Educators -- Students -- WebRTC capable browser -- Existing or new web application -- Signaling server -- TURN server -- Archive server -- Potential issues that may be faced -- Privacy -- Copyright and intellectual property -- Restrictive networks -- Restrictive SOEs -- Outdated student browsers -- Interoperability -- Benefits that can be delivered -- The opportunity for educators -- Summary -- 8. Example Application 2 - Team Communication -- Applying WebRTC for team communication -- Overall application architecture -- Managers.

Team members -- WebRTC capable browser -- New and existing web applications -- Signaling server -- TURN server -- Messaging server -- Potential issues that may be faced -- Privacy -- Data security -- Restrictive networks -- Restrictive SOEs -- Interoperability -- Timezones -- Benefits that can be delivered -- The opportunity for managers -- Summary -- Index.