Book Image

Building Telephony Systems with OpenSIPS Second Edition - Second Edition

By : Flavio E. Goncalves, Bogdan-Andrei Iancu
Book Image

Building Telephony Systems with OpenSIPS Second Edition - Second Edition

By: Flavio E. Goncalves, Bogdan-Andrei Iancu

Overview of this book

OpenSIPS is a multifunctional, multipurpose signalling SIP server. SIP (Session Initiation Protocol) is nowadays the most important VoIP protocol and OpenSIPS is the open source leader in VoIP platforms based on SIP. OpenSIPS is used to set up SIP Proxy servers. The purpose of these servers is to receive, examine, and classify SIP requests. The whole telecommunication industry is changing to an IP environment, and telephony as we know it today will completely change 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. Starting with an introduction to SIP and OpenSIPS, you will begin by installing and configuring OpenSIPS. You will be introduced to OpenSIPS Scripting language and OpenSIPS Routing concepts, followed by comprehensive coverage of Subscriber Management. Next, you will learn to install, configure, and customize the OpenSIPS control panel and explore dialplans and routing. You will discover how to manage the dialog module, accounting, NATTraversal, and other new SIP services. The final chapters of the book are dedicated to troubleshooting tools, SIP security, and advanced scenarios including TCP/TLS support, load balancing, asynchronous processing, and more. A fictional VoIP provider is used to explain OpenSIPS and by the end of the book, you will have a simple but complete system to run a VoIP provider.
Table of Contents (21 chapters)
Building Telephony Systems with OpenSIPS Second Edition
Credits
About the Authors
About the Reviewers
www.PacktPub.com
Preface
Index

SIP session timers


One of the main issues with SIP is to make sure that the call and dialog finishes properly. However, in the SIP protocol, there is no guarantee that either the UAC or UAS will send the BYE request to terminate the call. Network and device issues can lead to unfinished calls. These calls, also called hanged calls, will stay in the dialog table until its default timeout, by default, 43,200 seconds or, in other words, 12 hours. This time is changeable in the default_timeout parameter.

The Internet Engineering Task Force (IETF) defined a mechanism to detect hanged calls and lack of communication with a specific device. It is now a standard described in RFC 4028. The SIP session timers send periodic messages that continuously update the timeout at the proxy. The SIP Session Timer (SST) module in OpenSIPS implements this feature.

How the SIP session timer works

The module receives notifications from the dialog module of any new or updated dialogs. It will look for the value in...