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WebRTC Integrator's Guide

By : Altanai Bisht
Book Image

WebRTC Integrator's Guide

By: Altanai Bisht

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WebRTC Integrator's Guide
Altanai Bisht
Oct, 2014 | 382 pages
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Table of Contents (17 chapters)
WebRTC Integrator's Guide
WebRTC Integrator's Guide
Credits
Credits
About the Author
About the Author
About the Reviewers
About the Reviewers
www.PacktPub.com
www.PacktPub.com
Preface
Preface
Free Chapter
1
Running WebRTC with and without SIP
Running WebRTC with and without SIP
JavaScript Session Establishment Protocol (JSEP)
Running WebRTC without SIP
WebRTC through WebSocket signaling servers
Running WebRTC with SIP
Summary
2
Making a Standalone WebRTC Communication Client
Making a Standalone WebRTC Communication Client
Description of the WebRTC client-server model
SIP servers
Limitations of the existing setup
Summary
3
WebRTC with SIP and IMS
WebRTC with SIP and IMS
The Interaction with core IMS nodes
The IP Multimedia Subsystem core
The Telecom server
The Media Server
WebRTC over firewalls and proxies
The final architecture for the WebRTC-to-IMS integration
Summary
4
WebRTC Integration with Intelligent Network
WebRTC Integration with Intelligent Network
From mobiles to WebRTC client through GPRS
From mobiles to WebRTC client through GSM
Call processed with the IN service logic
The WebRTC client's SIP messages to SMS in a GSM phone (SMSC)
Summary
5
WebRTC Integration with PSTN
WebRTC Integration with PSTN
What is PSTN?
WebRTC connectivity to the PSTN
The challenges in connecting the WebRTC world to the PSTN landscape
The service logic
Summary
6
Basic Features of WebRTC over SIP
Basic Features of WebRTC over SIP
SIP services
Services in the Application Server
Media Server-based features
Features of a web application
The admin console
Summary
7
WebRTC with Industry Standard Frameworks
WebRTC with Industry Standard Frameworks
The Multitier architecture
The design of a WebRTC client
The environment setup
Testing
Summary
8
WebRTC and Rich Communication Services
WebRTC and Rich Communication Services
Rich Communication Services
The RCS ecosystem and WebRTC
Telecom operator's benefit derived from RCS
Voice over LTE
Summary
9
Native SIP Application and Interaction with WebRTC Clients
Native SIP Application and Interaction with WebRTC Clients
Support for WebRTC in various operating systems
Summary
10
Other WebRTC Use Cases
Other WebRTC Use Cases
Unified Communicator
Customized Communicator for specific enterprise segments
The Customer Relationship Management system
Network Operation Center
The human resource management tool
Social networking – targeting consumers
Retail services
Health care
Financial services
Remote management
WebRTC games
TV experience with WebRTC
Interfacing services
Summary
Index
Index

Summary

In this chapter, we learned that a WebRTC communication process is divided into two parts: signaling, where the session setup and teardown is agreed to, and media transactions, which deals with the actual RTP streams that contain voice/video/data that the user has sent. We saw how to program the three basic APIs of WebRTC media stack, namely, getUserMedia, RTCPeerConnection, and DataChannel. The Running WebRTC without SIP section described signaling done over JSON via XMLHttpRequest using Node.js as the intermediately signaling server to connect the peers and prepare for the media flow. The next section, Running WebRTC with SIP, listed the libraries or WebRTC clients that use SIP over WebSocket to take care of the signaling between WebRTC peers. In the following chapters, we will see how to use WebRTC media APIs over the SIP WebSocket protocol in detail.

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