WebRTC Integrator's Guide
eBook - ePub

WebRTC Integrator's Guide

  1. 382 pages
  2. English
  3. ePUB (mobile friendly)
  4. Available on iOS & Android
eBook - ePub

WebRTC Integrator's Guide

About this book

WebRTC enables real-time communication across the Web and with the whole telecom world behind a single button on a web page. WebRTC promises to bring new reforms and innovation for IP telephony. WebRTC comes with numerous integration features, such as new standards for VoIP services, call control applications, profile and phonebook management, and much more.

This book covers all aspects of building a standalone WebRTC communication platform, making a WebRTC SIP-based Communicator, and shows you how to overcome challenges. It also describes the integration of Rich Services such as voicemail, conference calls, and file transfers, as well as call control mechanisms such as screening and routing. The book then takes you through building a WebRTC project and its integration in the access, network, and service layers of IMS. The book ends with the creation of a commercial-quality web application capable of setting and receiving calls, messages, and conference calls, and other numerous services.

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Information

Year
2014
Edition
1
eBook ISBN
9781783981267

WebRTC Integrator's Guide


Table of Contents

WebRTC Integrator's Guide
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
Downloading the color images of this book
Errata
Piracy
Questions
1. Running WebRTC with and without SIP
JavaScript Session Establishment Protocol (JSEP)
Signal and media flows
Running WebRTC without SIP
Sending media over WebSockets
getUserMedia
RTCPeerConnection
RTCDataChannel
Media traversal in WebRTC clients
WebRTC through WebSocket signaling servers
Node.js
Making a peer-to-peer audio call using Node.js for signaling
Running WebRTC with SIP
Session Initiation Protocol (SIP)
JavaScript-based SIP libraries
Summary
2. Making a Standalone WebRTC Communication Client
Description of the WebRTC client-server model
The sipML5 WebRTC client
Developing a minified webphone application using Tomcat
Developing our customized version of the sipML5 client
The jsSIP WebRTC client
Developing our version of the jsSIP client
SIP servers
SIP-WS to SIP-WS
SIP2SIP
OfficeSIP
SIP WS to SIP and vice-versa
The gateway to convert SIP over WebSocket to native SIP
The WebRTC2SIP gateway
The WebRTC client with Brekeke SIP server
The WebRTC client with the Kamailio SIP server
Limitations of the existing setup
Firewall and NAT issues
Media transcoding
Summary
3. WebRTC with SIP and IMS
The Interaction with core IMS nodes
The Call Session Control Function
Home Subscriber System
The IP Multimedia Subsystem core
The OpenIMS Core
The Telecom server
The Mobicents Telecom Application Server
The Media Server
The FreeSWITCH Media Server
Media Services
WebRTC over firewalls and proxies
The final architecture for the WebRTC-to-IMS integration
Summary
4. WebRTC Integration with Intelligent Network
From mobiles to WebRTC client through GPRS
IMS connectivity to Gateway GPRS Support Node
From mobiles to WebRTC client through GSM
Call processed with the IN service logic
The WebRTC client's communication with the GSM phone through IMS
The WebRTC client's communication with a GSM phone with IN services
The services broker for endpoints and WebRTC in IMS to GSM phone in Intelligence Networks
The WebRTC client's SIP messages to SMS in a GSM phone (SMSC)
The Kannel gateway
Summary
5. WebRTC Integration with PSTN
What is PSTN?
WebRTC connectivity to the PSTN
The PSTN gateway
The PSTN connectivity to IMS via PSTN gateways
The call flow from a WebRTC SIP browser client to a fixed landline phone
The challenges in connecting the WebRTC world to the PSTN landscape
Address mapping
Translation from SIP to ISUP
The call setup
The call termination
The call in progress
The service logic
SIP service logic through application server
IN services via IMSSF
The Service Broker for the orchestration of services
Summary
6. Basic Features of WebRTC over SIP
SIP services
Registering a SIP client
Making audio and video calls using SIP
Text Chat using SIP
Obtaining the online/offline status of users using SIP
Services in the Application Server
Back-to-back user agent
Call screening
Basic call screening
Enhanced call screening
Call hold/resume
Call forwarding
Unconditional call forwarding
Call forwarding when the user is unavailable
Call transfer
Attended call transfer
Unattended call transfer
Generation of call log for tracking
Media Server-based features
Announcement
Media relay
Voicemail
Music on Hold
Interactive Voice Response
Conferencing
Multipart communication
Features of a web application
Geolocation
Authenticating users with OAuth
Import contacts from other accounts
Advertisements in the WebRTC call
Delivering an instant message as a mail
The admin console
Summary
7. WebRTC with Industry Standard Frameworks
The Multitier architecture
The design of a WebRTC client
The Class diagram
The Entity Relationship model
The environment setup
Java Runtime Environment (JRE)
Integrated Development Environment with Java Enterprise Edition (EE)
Databases
The web application server
The web application infrastructure
JSP- / Servlet-based WebRTC web project
Programming the JSP- / Servlet-based web project structure
The development of modules
The User Account module
The Communication module
The Phonebook module
Struts- / Hibernate-based WebRTC web project
Programming the Struts- / Hibernate-based web project structure
The development of modules
The OtherAccount module
Spring 3 MVC-based WebRTC web project
Programming the Spring 3 MVC web project structure
The development of modules
The Geolocation module
Testing
Testing the signal flow
Test cases for WebRTC client validation
Summary
8. WebRTC and Rich Communication Services
Rich Communication Services
Position and adoption of RCS
Business impact of RCS
Technology impact
Rich Communication Services enhanced (RCS-e)
Joyn
The RCS configuration process
RCS specifications
Service discovery by an RCS-enabled device
User capability exchange
Chats with multimedia sharing
The one-to-one text chat over MSRP
File transfer over MSRP
Group chat in a conference session
User availability through XCAP
REST-based notifications
Interoperability and interworking
The RCS ecosystem and WebRTC
RCS services in WebRTC
User profile
Integration with social networks
The enhanced phone...

Table of contents

  1. WebRTC Integrator's Guide

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