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Mastering FreeSWITCH电子书

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作       者:Anthony Minessale II,Giovanni Maruzzelli

出  版  社:Packt Publishing

出版时间:2016-07-01

字       数:221.3万

所属分类: 进口书 > 外文原版书 > 电脑/网络

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Master the art of advanced VoIP and WebRTC communication with the most dynamic application server, FreeSWITCH About This Book Forget the hassle - make FreeSWITCH work for you Discover how FreeSWITCH integrates with a range of tools and APIs From high availability to IVR development use this book to become more confident with this useful communication software Who This Book Is For SysAdmins, VoIP engineers – whoever you are, whatever you’re trying to do, this book will help you get more from FreeSWITCH. 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 In Detail 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. Style and approach Find out how it works, then put your knowledge into practice - that's how this advanced FreeSWITCH guide has been designed to help you learn. You'll soon master FreeSWITCH and be confident using it in your projects.
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Mastering FreeSWITCH

Table of Contents

Mastering FreeSWITCH

Credits

About the Authors

About the Reviewers

Contributors

www.PacktPub.com

eBooks, discount offers, and more

Why subscribe?

Preface

What this book covers

What you need for this book

Who this book is for

Conventions

Reader feedback

Customer support

Downloading the example code

Errata

Piracy

Questions

1. Typical Voice Uses for FreeSWITCH

Understanding routing calls in FreeSWITCH

Wholesale (provider to providers)

Residential uses of FreeSWITCH

Routing with federated VoIP

Dialers/telemarketing

FreeSWITCH Products and Services

Business PBX services (hosted and on-premises)

Call centers

Value added services and games, prizes, and polls

"Class 4" vs "Class 5" operations (and SBCs)

WebRTC / web services / Internet-only services

Mobile "over-the-top" SIP

Development

Strict on output, broad on input

Very structured, very reusable techniques

Polyglot by vocation and destiny

Extreme scalability, from embedded to big irons

Born internationalist

Telcos internal integration ("FreeSWITCH is the Perl of VoIP")

Rapid new services prototyping

Accounting and billing

Call Detail Records (CDRs)

Mod_nibblebill / CGrateS

Other billing options (open source - commercial)

Summary

2. Deploying FreeSWITCH

Network requirements

Understanding QoS

LANs, WANs, and peering

Testing with SIPp

Running scenarios

Load testing

Logging with FreeSWITCH

Call Detail Records

Monitoring

SNMP

SNMP and FreeSWITCH

Installation and configuration (on Linux)

Getting more information

Monitoring tools

Monitoring with Nagios

Monitoring with Cacti

HA deployment

Storage, network, switches, power supply

Virtualization

Load balancing and integration with Kamailio and OpenSIPS

In the Web world

In the FreeSWITCH world

DNS SRV records for geographical distribution and HA

Summary

3. ITSP and Voice Codecs Optimization

ITSPs – what they do

Routes (to numbers)

DIDs (aka DDIs) – numbers

Quality of routes

White, black, and grey

Codecs and bandwidth

Infrastructure capability

Various important features

Support, redundancy, high availability, and number portability

Summary

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

FreeSWITCH jail

SIP(S) and (S|Z)RTP

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

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

Audio file formats

MP3 and streaming

Music on Hold

Playing and recording audio files and streams

Recording and modifying prompts and audio files

Recording calls

Tapping audio

Summary

6. PSTN and TDM

OpenZap

FreeTDM

I/O modules

Signaling modules

ISDN signaling modules

Analog modules

MFC-R2

SS7

Cellular GSM / CDMA (ftmod_gsm)

FreeTDM installation

Wanpipe drivers

DAHDI drivers

LibPRI

Sangoma ISDN stack

OpenR2

LibWAT

Analog modules

Configuring FreeTDM

Wanpipe

DAHDI

FreeTDM library configuration

FreeSWITCH configuration

Operation

Outbound calls

Inbound calls

Debugging

Checking the physical layer

Enabling ISDN tracing

Audio tracing

Summary

7. WebRTC and Mod_Verto

WebRTC

Browsers are already out there, waitin'

Web Real-Time Communication is coming

Under the hood

Encryption – security

Beyond peer to peer – WebRTC to communication networks and services

WebRTC gateways and application servers

Which architecture? Legacy on the Web, or Web on the Telco?

FreeSWITCH accommodates them ALL

What is Verto (module and jslib)?

Configure mod_verto

Test with Communicator

Build Your Own Verto App

Summary

8. Audio and Video Conferencing

Conference basics

Conference.conf.xml (profiles, DTMF interaction, and so on)

Configuration sections logic

Profile

Caller-Controls group

Conference invocation, dialplan, channel variables

Outbound conference

Moderating and managing conferences – API

Video conference

Video conference configuration

Mux profile settings

Video conference screen layouts

Screen sharing

Screen sharing dialplan extension

Managing video conferences

Conference performances

Summary

9. Faxing and T38

What is Fax on PSTN?

How it works

What is Fax over IP?

Enter T38

T38 terminals and gateways

Fax and FreeSWITCH

The mod_spandsp configuration

mod_spandsp usage

Debugging faxes

How to maximize reliability of fax traffic

PDF to fax and fax to PDF

Fax to mail

HylaFax and FreeSWITCH

ITSPs and Real World Fax Support

Summary

10. Advanced IVR with Lua

Installing IVR

Structure of welcome.lua

Incoming call processing

Before answering

First voice menu

Second and third voice menus

Fourth menu – asynch! Nonblocking! Fun with threads!

After hangup

Utility functions

Summary

11. Write Your FreeSWITCH Module in C

What is a FreeSWITCH module?

Developing a module

Mod_Example outline

Mandatory functions

Load function

Runtime function

Shutdown function

Configuration using XML

Reacting to channel state changes

Receiving and firing events

Dialplan application

API command

Summary

12. Tracing and Debugging VoIP

What can go wrong?

What else can go wrong? (NAT problems)

Other things can go wrong too

SIP, RTP, SDP, RTCP, OH MY!

Tools

Firewall

FreeSWITCH as SIP self tracer

Tcpdum – the mother of all packet captures

ngrep – network grep

tshark – pure packet power

pcapsipdump

sngrep – the holy grail

Sipgrep, Ngrep on steroids for VoIP

Wireshark – "the" packet overlord

Audacity – audio Swiss army knife

SoX – audio format converter

Summary

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

Logs to Homer

FreeSWITCH events to Homer

Summary

Index

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