WebRTC integrator's guide : successfully build your very own scalable WebRTC infrastructure quickly and efficiently / Altanai.

Format:

Book

Author/Creator:

Altanai, author.

Series:

Community experience distilled.

Community experience distilled

Language:

English

Subjects (All):

Telematics.

Communication--Technological innovations.

Communication.

Peer-to-peer architecture (Computer networks).

Physical Description:

1 online resource (382 p.)

Edition:

Successfully build your very own scalable WebRTC infrastructure quickly and efficiently

Place of Publication:

Birmingham : Packt Publishing, 2014.

Language Note:

English

System Details:

text file

Biography/History:

Bisht Altanai: Altanai, born into an Indian army family, is a bubbly, vivacious, intelligent computer geek. She is an avid blogger and writes on Research and Development of evolving technologies in Telecom (http: //altanaitelecom. wordpress. com). She holds a Bachelor's degree in Information Technology from Anna University, Chennai. She has worked on many Telecom projects worldwide, specifically in the development and deployment of IMS services. She firmly believes in contributing to the Open Source community and is currently working on building a WebRTC-based JS library with books for more applications. Her hobbies include photography, martial arts, oil canvas painting, river rafting, horse riding, and trekking, to name a few. This is her first book, and it contains useful insight into WebRTC for beginners and integrator in this field. The book has definitions and explanations that will cover many interesting concepts in a clear manner. Altanai can be contacted at tara181989@gmail. com.

Summary:

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.

Contents:

Cover; Copyright; Credits; About the Author; About the Reviewers; www.PacktPub.com; Table of Contents; Preface; Chapter 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

Chapter 2: Making a Standalone WebRTC Communication ClientDescription 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 issuesMedia transcoding; Summary; Chapter 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; Chapter 4: WebRTC Integration with Intelligent Network; From mobiles to WebRTC client through GPRS

IMS connectivity to Gateway GPRS Support NodeFrom 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, 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; Chapter 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 phoneThe 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; Chapter 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

Notes:

Includes index.

Description based on online resource; title from PDF title page (ebrary, viewed November 13, 2014).

ISBN:

9781783981274

178398127X

OCLC:

896723285