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

Missing BYEs and CDRs


One of the biggest issues with SIP accounting is the occurrence of missing BYEs. If one leg of a call is abruptly disconnected from the network, the BYE request is not generated. In this case, it is not possible to receive the BYE message and determine the duration of the call correctly. There are some approaches to solve this issue:

  • The SIP session timers

    The SIP session timers, described in RFC 4028, enhance the SIP protocol by adding the capability to refresh the SIP sessions by resending periodic re-INVITEs or UPDATEs. To implement SIP session timers, it is necessary to have support on at least one of the SIP endpoints, the client, or the gateway. The advantage of this method is that it uses only signaling without any control of the media. In the Chapter 8, Managing Dialogs, we have already discussed the setup of the SIP Session Timers (SST).

  • The RTP proxy timeout

    Recently, a timeout socket was included in the RTP proxy. You can use an external program connected...