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WebRTC Integrator's Guide电子书

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14人正在读 | 0人评论 9.8

作       者:Altanai

出  版  社:Packt Publishing

出版时间:2014-10-31

字       数:358.7万

所属分类: 进口书 > 外文原版书 > 电脑/网络

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This book is for programmers who want to learn about real-time communication and utilize the full potential of WebRTC. It is assumed that you have working knowledge of setting up a basic telecom infrastructure as well as basic programming and *ing knowledge.
目录展开

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 phonebook

User capabilities and Presence

Unified messaging box

Message history

Rich calls

Call logs

Message history

Multiparty conferencing

WebRTC architecture with RCS modules

Telecom operator's benefit derived from RCS

Voice over LTE

Combination of WebRTC, VOLTE, and RCS

Summary

9. Native SIP Application and Interaction with WebRTC Clients

Support for WebRTC in various operating systems

Windows OS

Native browser support for WebRTC clients

Chrome browser support for WebRTC clients

Mozilla browser support for WebRTC clients

Opera browser support for WebRTC clients

SIP softphones capable of interacting with WebRTC clients

X-Lite

Zoiper

Boghe

WebRTC unsupported browsers interacting with WebRTC clients

Linux OS

Native browser support for WebRTC clients

Chrome browser support for WebRTC clients

Mozilla browser support for WebRTC clients

Opera browser support for WebRTC clients

SIP softphones capable of interacting with WebRTC clients

Kapanga

Linphone

Yate

SFL

Mac OS

Native browser support for WebRTC clients

SIP softphones capable of interacting with WebRTC clients

iDoubs

Jitsi

WebRTC unsupported browsers interacting with WebRTC client

Android OS for mobiles

Native browser support for WebRTC clients

Android phone's/tablet's SIP applications capable of interacting with WebRTC clients

Developing a lightweight Android SIP application

Windows OS for mobiles

Apple iPhone

iPhone/iPad IP applications interacting with WebRTC clients

Developing an iPhone SIP application

Summary

10. Other WebRTC Use Cases

Unified Communicator

Team Communicator

Customized Communicator for specific enterprise segments

Branches and back office communications

The Customer Relationship Management system

Network Operation Center

The human resource management tool

Communicating with candidates for an open post directly from the job portal

Social networking – targeting consumers

Social networking platforms

Dating sites with anonymous call and chat

Retail services

WebRTC online marketing centers

WebRTC contact centers

Users contacting customer care

Health care

Online medical consultation with the doctor

Financial services

Communication with financial services

Insurance claims

Calling from the ATM

Remote management

Surveillance

Managing the connected device

WebRTC games

Two-player games

Multiplayer games

TV experience with WebRTC

Live broadcasting

IPTV integration and streaming

Streaming movies among peers

Interfacing services

WebRTC for e-learning

WebRTC for e-governance

Summary

Index

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