Book Image

FreeSWITCH 1.8

By : Anthony Minessale II, Giovanni Maruzzelli
Book Image

FreeSWITCH 1.8

By: Anthony Minessale II, Giovanni Maruzzelli

Overview of this book

FreeSWITCH is an open source telephony platform designed to facilitate the creation of voice and chat-driven products, scaling from a soft-phone to a PBX and even up to an enterprise-class soft-switch. This book introduces FreeSWITCH to IT professionals who want to build their own telephony system. This book starts with a brief introduction to the latest version of FreeSWITCH. We then move on to the fundamentals and the new features added in version 1.6, showing you how to set up a basic system so you can make and receive phone calls, make calls between extensions, and utilize basic PBX functionality. Once you have a basic system in place, we’ll show you how to add more and more functionalities to it. You’ll learn to deploy the features on the system using unique techniques and tips to make it work better. Also, there are changes in the security-related components, which will affect the content in the book, so we will make that intact with the latest version. There are new support libraries introduced, such as SQLite, OpenSS, and more, which will make FreeSWITCH more efficient and add more functions to it. We’ll cover these in the new edition to make it more appealing for you.
Table of Contents (23 chapters)
Title Page
Credits
About the Authors
About the Reviewer
www.PacktPub.com
Customer Feedback
Preface

Making media flow


Now that the SIP messages are flowing properly from FreeSWITCH to the phone, what about the media? A phone call or a videocall is not very eventful if you can't even hear each other, right? We had many problems where the calls would set up properly until the point where NAT would strike the RTP packets that provide the actual media of the call, rendering the call with one-way-audio or even no-way-audio in some cases. In light of this injustice, we created a separate feature that is always enabled and only needs to be manually disabled in a very few set of cases inspired by the fourth pitfall. This feature is called RTP auto-adjust. The reason we need it is because when the phone tries to call us from behind NAT, it will naively advertise its unreachable LAN private address to FreeSWITCH as to where to send the media streams (eg the address written in the text of SDP).

We could guess that since the device is behind NAT, we should really send the audio to the same address...