Book Image

Mastering FreeSWITCH

By : Russell Treleaven, Seven Du, Darren Schreiber, Ken Rice, Mike Jerris, Kalyani Kulkarni, Florent Krieg, Charles Bujold
4 (1)
Book Image

Mastering FreeSWITCH

4 (1)
By: Russell Treleaven, Seven Du, Darren Schreiber, Ken Rice, Mike Jerris, Kalyani Kulkarni, Florent Krieg, Charles Bujold

Overview of this book

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.

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Mastering FreeSWITCH
Russell Treleaven | Seven Du | Darren Schreiber | Ken Rice | Mike Jerris | Kalyani Kulkarni | Florent Krieg | Charles Bujold
Jul, 2016 | 300 pages
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Table of Contents (21 chapters)
Mastering FreeSWITCH
Mastering FreeSWITCH
Credits
Credits
About the Authors
About the Authors
About the Reviewers
About the Reviewers
Contributors
Contributors
www.PacktPub.com
www.PacktPub.com
Preface
Preface
Free Chapter
1
Typical Voice Uses for FreeSWITCH
Typical Voice Uses for FreeSWITCH
Understanding routing calls in FreeSWITCH
FreeSWITCH Products and Services
Development
Accounting and billing
Summary
2
Deploying FreeSWITCH
Deploying FreeSWITCH
Network requirements
Testing with SIPp
Logging with FreeSWITCH
Call Detail Records
Monitoring
HA deployment
Summary
3
ITSP and Voice Codecs Optimization
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
4
VoIP Security
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
5
Audio File and Streaming Formats, Music on Hold, Recording Calls
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
6
PSTN and TDM
PSTN and TDM
OpenZap
FreeTDM
I/O modules
Signaling modules
FreeTDM installation
Configuring FreeTDM
Debugging
Summary
7
WebRTC and Mod_Verto
WebRTC and Mod_Verto
WebRTC
Summary
8
Audio and Video Conferencing
Audio and Video Conferencing
Conference basics
Video conference
Conference performances
Summary
9
Faxing and T38
Faxing and T38
What is Fax on PSTN?
What is Fax over IP?
Fax and FreeSWITCH
ITSPs and Real World Fax Support
Summary
10
Advanced IVR with Lua
Advanced IVR with Lua
Installing IVR
Structure of welcome.lua
Incoming call processing
After hangup
Utility functions
Summary
11
Write Your FreeSWITCH Module in C
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
12
Tracing and Debugging VoIP
Tracing and Debugging VoIP
What can go wrong?
SIP, RTP, SDP, RTCP, OH MY!
Tools
Summary
13
Homer, Monitoring and Troubleshooting Your Communication Platform
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
Index
Index

Searching signaling with Homer

After you configured and started the Capture Agent inside FreeSWITCH, make a call.

Then, log into your Homer, set the Time Range to Today, select calls as transaction kind, and click search.

You'll be greeted by a tabular view of the SIP signaling comprised in your call. If you get no results, check the homer_data database if sip_capture_call_2016* table has been created for today. If not, execute as root /opt/homer_rotate.

If you click on the Method name, a popup springs up showing the whole message, with nice coloring.

If you click on the CallID field, you can peruse the entire end-to-end Call-Flow, in sngrep/wireshark style.

Clicking on a method inside the Call-Flow popup shows a new popup with that whole message.

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