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

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.

Detalles Bibliográficos
Clasificación:Libro Electrónico
Autor principal: Altanai
Formato: Electrónico eBook
Idioma:Inglés
Publicado: Birmingham, UK : Packt Pub., 2014.
Edición:Successfully build your very own scalable WebRTC infrastructure quickly and efficiently.
Colección:Community experience distilled.
Temas:
Acceso en línea:Texto completo
Texto completo
Tabla de Contenidos:
  • 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.