SIP is the most important VoIP protocol and OpenSIPS is clearly the open source leader in VoIP platforms based on pure SIP. The whole telecommunication industry is changing to an IP environment, and telephony in the way we know today will disappear in less than ten years. SIP is the protocol leading this disruptive revolution and it is one of the main protocols on next-generation networks. While a VoIP provider is not the only kind of SIP infrastructure created using OpenSIPS, it is certainly one of the most difficult to implement.
This book will give you a competitive edge by helping you to create a SIP infrastructure capable of handling tens of thousands of subscribers. You can extend the examples given in this book easily to other applications such as a SIP router, load balancing, IP PBX, and Hosted PBX as well. This book is an update of the title Building Telephony Systems with OpenSER.
The book starts with the simplest configuration and evolves chapter by chapter teaching you how to add new features and modules. It will first teach you the basic concepts of SIP and SIP routing. Then, you will start applying the theory by installing OpenSIPS and creating the configuration file. You will learn about features such as authentication, PSTN connectivity, user portals, media server integration, billing, NAT traversal, and monitoring. The book uses a fictional VoIP provider to explain OpenSIPS. The idea is to have a simple but complete running VoIP provider by the end of the book.
A practical guide to building an efficient SIP telephony system
What you will learn from this book :
* Identify how SIP transactions are routed including initial and sequential requests
* Accelerate the processing of SIP sequential requests with the help of Loose Routing
* Install OpenSIPS in a Linux platform and integrate a media server such as Asterisk
* Acquire authentication and persistency by enabling a MySQL back-end for OpenSIPS
* Administer the server with the help of graphical web interfaces such as OpenSIPS control panel and serMyAdmin
* Connect to a PSTN gateway to send and receive calls
* Enable dynamic dial plans and routing by using the DIALPLAN module DROUTING module
* Traverse NAT using STUN and TURN
* Bill your costumers or simply check your expenses by generating CDRs (Call Detail Records)
* Monitor your SIP infrastructure to keep it running smoothly
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