Mastering FreeSWITCH

Master the art of advanced VoIP and WebRTC communication with the most dynamic application server, FreeSWITCH

Mastering FreeSWITCH

Mastering
Anthony Minessale II, Giovanni Maruzzelli

5 customer reviews
Master the art of advanced VoIP and WebRTC communication with the most dynamic application server, FreeSWITCH
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Book Details

ISBN 139781784398880
Paperback300 pages

Book Description

FreeSWITCH is one of the best tools around if you’re looking for a modern method of managing communication protocols through a range of different media. From real-time browser communication with the WebRTC API to implementing VoIP (voice over internet protocol), with FreeSWITCH you’re in full control of your projects. This book shows you how to unlock its full potential – more than just a tutorial, it’s packed with plenty of tips and tricks to make it work for you.

Written by members of the team who actually helped build FreeSWITCH, it will guide you through some of the newest features of version 1.6 including video transcoding and conferencing. Find out how FreeSWITCH interacts with other tools and APIs, learn how to tackle common (and not so common) challenges ranging from high availability to IVR development and programming advanced PBXs.

Great communication functionality begins with FreeSWITCH – find out how and get your project up and running today.

Table of Contents

Chapter 1: Typical Voice Uses for FreeSWITCH
Understanding routing calls in FreeSWITCH
FreeSWITCH Products and Services
Development
Accounting and billing
Summary
Chapter 2: Deploying FreeSWITCH
Network requirements
Testing with SIPp
Logging with FreeSWITCH
Call Detail Records
Monitoring
HA deployment
Summary
Chapter 3: ITSP and Voice Codecs Optimization
ITSPs – what they do
Routes (to numbers)
DIDs (aka DDIs) – numbers
Quality of routes
Various important features
Support, redundancy, high availability, and number portability
Summary
Chapter 4: VoIP Security
Latest versions of it all
Default configuration is a demo
Change passwords
Lock all that's not trusted
Dropping root privileges (file permissions)
Fail2ban on all services
Encrypting SIP with TLS (SIPS)
Encrypting (S)RTP via SDES (key exchange in SDP)
Encrypting (S)RTP via ZRTP (key exchange in RTP)
New frontiers of VoIP encryption (WebRTC, WebSockets, DTLS)
Summary
Chapter 5: Audio File and Streaming Formats, Music on Hold, Recording Calls
Traditional telephony codecs constrain audio
HD audio frontiers are pushed by cellphones, right now
FreeSWITCH audio, file, and stream formats
Recording calls
Tapping audio
Summary
Chapter 6: PSTN and TDM
OpenZap
FreeTDM
I/O modules
Signaling modules
FreeTDM installation
Configuring FreeTDM
Debugging
Summary
Chapter 7: WebRTC and Mod_Verto
WebRTC
Summary
Chapter 8: Audio and Video Conferencing
Conference basics
Video conference
Conference performances
Summary
Chapter 9: Faxing and T38
What is Fax on PSTN?
What is Fax over IP?
Fax and FreeSWITCH
ITSPs and Real World Fax Support
Summary
Chapter 10: Advanced IVR with Lua
Installing IVR
Structure of welcome.lua
Incoming call processing
After hangup
Utility functions
Summary
Chapter 11: Write Your FreeSWITCH Module in C
What is a FreeSWITCH module?
Developing a module
Mod_Example outline
Mandatory functions
Configuration using XML
Reacting to channel state changes
Receiving and firing events
Dialplan application
API command
Summary
Chapter 12: Tracing and Debugging VoIP
What can go wrong?
SIP, RTP, SDP, RTCP, OH MY!
Tools
Summary
Chapter 13: Homer, Monitoring and Troubleshooting Your Communication Platform
What is Homer?
Installing Homer and the Capture Server
Feeding SIP signaling from FreeSWITCH to Homer
Searching signaling with Homer
Feeding SIP signaling, QoS, MOS and RTP/RTCP stats from CaptAgent to Homer
Correlating A-leg and B-leg
Feeding logs and events to Homer
Summary

What You Will Learn

  • Get to grips with the core concepts of FreeSWITCH
  • Learn FreeSWITCH high availability
  • Work with SIP profiles, gateways, ITSPs, and Codecs optimization
  • Implement effective security on your projects
  • Master audio manipulation and recording
  • Discover how FreeSWITCH works alongside WebRTC
  • Build your own complex IVR and PBX applications
  • Connect directly to PSTN/TDM
  • Create your own FreeSWITCH module
  • Trace SIP packets with the help of best open source tools
  • Implement Homer Sipcapture to troubleshoot and debug all your platform traffic

Authors

Table of Contents

Chapter 1: Typical Voice Uses for FreeSWITCH
Understanding routing calls in FreeSWITCH
FreeSWITCH Products and Services
Development
Accounting and billing
Summary
Chapter 2: Deploying FreeSWITCH
Network requirements
Testing with SIPp
Logging with FreeSWITCH
Call Detail Records
Monitoring
HA deployment
Summary
Chapter 3: ITSP and Voice Codecs Optimization
ITSPs – what they do
Routes (to numbers)
DIDs (aka DDIs) – numbers
Quality of routes
Various important features
Support, redundancy, high availability, and number portability
Summary
Chapter 4: VoIP Security
Latest versions of it all
Default configuration is a demo
Change passwords
Lock all that's not trusted
Dropping root privileges (file permissions)
Fail2ban on all services
Encrypting SIP with TLS (SIPS)
Encrypting (S)RTP via SDES (key exchange in SDP)
Encrypting (S)RTP via ZRTP (key exchange in RTP)
New frontiers of VoIP encryption (WebRTC, WebSockets, DTLS)
Summary
Chapter 5: Audio File and Streaming Formats, Music on Hold, Recording Calls
Traditional telephony codecs constrain audio
HD audio frontiers are pushed by cellphones, right now
FreeSWITCH audio, file, and stream formats
Recording calls
Tapping audio
Summary
Chapter 6: PSTN and TDM
OpenZap
FreeTDM
I/O modules
Signaling modules
FreeTDM installation
Configuring FreeTDM
Debugging
Summary
Chapter 7: WebRTC and Mod_Verto
WebRTC
Summary
Chapter 8: Audio and Video Conferencing
Conference basics
Video conference
Conference performances
Summary
Chapter 9: Faxing and T38
What is Fax on PSTN?
What is Fax over IP?
Fax and FreeSWITCH
ITSPs and Real World Fax Support
Summary
Chapter 10: Advanced IVR with Lua
Installing IVR
Structure of welcome.lua
Incoming call processing
After hangup
Utility functions
Summary
Chapter 11: Write Your FreeSWITCH Module in C
What is a FreeSWITCH module?
Developing a module
Mod_Example outline
Mandatory functions
Configuration using XML
Reacting to channel state changes
Receiving and firing events
Dialplan application
API command
Summary
Chapter 12: Tracing and Debugging VoIP
What can go wrong?
SIP, RTP, SDP, RTCP, OH MY!
Tools
Summary
Chapter 13: Homer, Monitoring and Troubleshooting Your Communication Platform
What is Homer?
Installing Homer and the Capture Server
Feeding SIP signaling from FreeSWITCH to Homer
Searching signaling with Homer
Feeding SIP signaling, QoS, MOS and RTP/RTCP stats from CaptAgent to Homer
Correlating A-leg and B-leg
Feeding logs and events to Homer
Summary

Book Details

ISBN 139781784398880
Paperback300 pages
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