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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

NAT and WebRTC (SOLVED)

OK, WebRTC has been defined barely yesterday, actually is being defined right now. It's completely unencumbered by legacy shortcomings, particularly when dealing with NAT. So let's say using WebRTC as transport solves all NAT problems.

You define your ext-rtp-ip and ext-sip-ip in SIP profiles, then your ext-rtp-ip in verto.conf.xml, and you're all set.

The magic of ICE, used by default by all WebRTC clients to determine how to connect, and implemented on FreeSWITCH side too, is almost sure to guarantee a perfect flow of signaling and media.

If you have a very strange situation, for example you want to use WebRTC inside LANs without using public addresses or DNS names, then you will need to implement your own ICE/STUN server and configure your WebRTC clients to use that server(s). But if you have such a project, you probably already know you...

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