WebRTC Integrator's Guide

Name: WebRTC Integrator's Guide
Brand: Packt Publishing
Availability: OnlineOnly

Altanai Altanai(Autor*in)

Packt Publishing

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Erschienen am 31. Oktober 2014

382 Seiten

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978-1-78398-127-4 (ISBN)

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Inhalt

Intro
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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