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WebRTC Integrator's Guide.
Başlık:
WebRTC Integrator's Guide.
Yazar:
Altanai.
ISBN:
9781783981274
Yazar Ek Girişi:
Fiziksel Tanımlama:
1 online resource (436 pages)
İçerik:
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.
Özet:
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 scripting knowledge.
Notlar:
Electronic reproduction. Ann Arbor, Michigan : ProQuest Ebook Central, 2017. Available via World Wide Web. Access may be limited to ProQuest Ebook Central affiliated libraries.
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