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Mastering FreeSWITCH

You're reading from   Mastering FreeSWITCH Advanced tips and tricks for advanced multimedia communication

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Product type Paperback
Published in Jul 2016
Publisher Packt
ISBN-13 9781784398880
Length 300 pages
Edition 1st Edition
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Concepts
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Authors (8):
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Darren Schreiber Darren Schreiber
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Darren Schreiber
Russell Treleaven Russell Treleaven
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Russell Treleaven
Kalyani Kulkarni Kalyani Kulkarni
Author Profile Icon Kalyani Kulkarni
Kalyani Kulkarni
Seven Du Seven Du
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Seven Du
Charles Bujold Charles Bujold
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Charles Bujold
Ken Rice Ken Rice
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Ken Rice
Florent Krieg Florent Krieg
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Florent Krieg
Mike Jerris Mike Jerris
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Mike Jerris
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Table of Contents (15) Chapters Close

Preface 1. Typical Voice Uses for FreeSWITCH 2. Deploying FreeSWITCH FREE CHAPTER 3. ITSP and Voice Codecs Optimization 4. VoIP Security 5. Audio File and Streaming Formats, Music on Hold, Recording Calls 6. PSTN and TDM 7. WebRTC and Mod_Verto 8. Audio and Video Conferencing 9. Faxing and T38 10. Advanced IVR with Lua 11. Write Your FreeSWITCH Module in C 12. Tracing and Debugging VoIP 13. Homer, Monitoring and Troubleshooting Your Communication Platform Index

Encrypting SIP with TLS (SIPS)


TLS, as SSL, depends on certificates issued by a Certification Authority that guarantee the identity of the certificate bearer. You can buy a TLS certificate from the same CAs that sell Web HTTPS certificates. You can then use that same certificate with WebSockets, WebRTC and mod_verto too (and for the HTTPS website with the same name as your SIP registrar, for example, https://pbx.freeswitch.org).

Also, you can use free and valid certificates from https://letsencrypt.org/, (see the automatic script in FreeSWITCH Confluence about verto_communicator demo installation on Debian 8).

The tool you use to generate the various certificates involved is (aptly named) gentls_cert:

/usr/local/freeswitch/bin/gentls_cert command -cn pbx.freeswitch.org -alt DNS:pbx.freeswitch.org -org freeswitch.org

(Instead of pbx.freeswitch.org and freeswitch.org, use the FQDN your clients will use as SIP registrar and SIP domain).You will use the same utility with the same arguments, but...

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