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FreeSWITCH 1.8

You're reading from   FreeSWITCH 1.8 Get to grips with VoIP and WebRTC communication and quickly build robust telephony systems with FreeSWITCH

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Product type Paperback
Published in Jul 2017
Publisher Packt
ISBN-13 9781785889134
Length 434 pages
Edition 1st Edition
Concepts
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Authors (2):
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Giovanni Maruzzelli Giovanni Maruzzelli
Author Profile Icon Giovanni Maruzzelli
Giovanni Maruzzelli
Anthony Minessale II Anthony Minessale II
Author Profile Icon Anthony Minessale II
Anthony Minessale II
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Toc

Table of Contents (23) Chapters Close

Title Page
Credits
About the Authors
About the Reviewer
www.PacktPub.com
Customer Feedback
Preface
1. Architecture of FreeSWITCH FREE CHAPTER 2. Building and Installation 3. Test Driving the Example Configuration 4. User Directory, SIP, and Verto 5. WebRTC, SIP, and Verto 6. XML Dialplan 7. Phrase Macros and XML IVRs 8. Lua FreeSWITCH Scripting 9. Dialplan in Deep 10. Dialplan, Directory, and ALL via XML_CURL and Scripts 11. ESL - FreeSWITCH Controlled by Events 12. HTTAPI - FreeSWITCH Asks Webserver Next Action 13. Conferencing and WebRTC Video-Conferencing 14. Handling NAT 15. VoIP Security 16. Troubleshooting, Asking for Help, and Reporting Bugs

Writing WebRTC Clients


A WebRTC client, in its most popular implementation, is an HTML webpage(s) that loads a JavaScript(s). Together, HTML and JavaScript define the GUI and the behavior of the WebRTC client.

Usually the JavaScript part, loaded by the HTML, leverages one or more JavaScript libraries. Those libraries implement the signaling protocol of choice (in our case SIP or VERTO) and its interaction with WebRTC APIs. We have already seen how WebRTC APIs provide for accessing the local computer multimedia hardware (microphone and camera), manage the peer-to-peer streaming of audio and video with the peer, and a bidirectional data channel.

The session signaling protocol will leverage and complement those WebRTC P2P capabilities, so they become useful for much more than a connection to a pre-known address and port.

SIP and JavaScript

SIP for WebRTC has been notably implemented in theJsSIP JavaScript Open Source library. JsSIP was written by José Luis Millán, Iñaki Baz Castillo, and Sa...

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