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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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Anthony Minessale II Anthony Minessale II
Author Profile Icon Anthony Minessale II
Anthony Minessale II
Giovanni Maruzzelli Giovanni Maruzzelli
Author Profile Icon Giovanni Maruzzelli
Giovanni Maruzzelli
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Toc

Table of Contents (17) Chapters Close

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

WebRTC in FreeSWITCH


OK, enough with abstractions, let's see under our belated hood, how WebRTC is implemented by FreeSWITCH.

FreeSWITCH is a complete WebRTC platform, and can act as both WebRTC gateway and WebRTC Application Server. Eg, FreeSWITCH is able to connect WebRTC clients to external "normal" SIP and PSTN services (or PBXes), enabling already existing or legacy services to be accessed by one billion browsers.

At the same time (and this can be literally true: while acting as gateway for some external services) FreeSWITCH is able to directly serve to WebRTC clients (browsers and apps) the exact same audio/video applications is serving to "normal" SIP and PSTN clients, such as PBX, IVR, Call Center, Conferencing System, Calling Card, etc applications.

To FreeSWITCH, WebRTC is just another kind of communication channel, along with SIP, TDM, MGCP, H323, Skype, Jingle, etc. All FreeSWITCH features are available via WebRTC, natively.

WebRTC Signaling in FS: SIP and Verto

As we've already seen...

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