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 services


Beyond making and receiving calls, you can implement a series of SIP services. These services include, but are not limited to, Call Transfers, Call Pickup, Call Hold, Call Forward, and many others. Fortunately, RFC 5359 (SIP services) defines a standard way to accomplish these tasks. Most SIP phones comply with the way SIP services are implemented; however, to make them work, you need to make sure that all the components in the network support some specific RFCs. As an example, the call transfer requires the support of the REFER method defined in RFC 3515 and the Referred-By and Replaces headers defined in RFCs 3891 and 3892, respectively. If you intend to provide PBX-like services using a SIP proxy, you have to make sure that all the components, including phones and gateways, support it. SIP services are implemented in phones, gateways, media servers, and proxies. All the components must collaborate in order to implement each specific service. The following are some of the services defined in RFC 5359:

  • Call Hold

  • Consultation Hold

  • Music on Hold

  • Transfer—Unattended

  • Transfer—Attended

  • Transfer—Instant Messaging

  • Call Forwarding—Unconditional

  • Call Forwarding—Busy

  • Call Forwarding—No Answer

  • 3-Way Conference—Third Party Is Added

  • 3-Way Conference—Third Party Joins

  • Find-Me

  • Incoming Call Screening

  • Outgoing Call Screening

  • Call Park

  • Call Pickup

  • Automatic Redial

  • Click to Dial

It would be counterproductive to describe in detail each service here. Refer to the specified RFC for details.