This book is a well illustrated, step-by-step guide to building a SIP based network using OpenSER. This book is for readers who want to understand how to build a SIP provider from scratch using OpenSER. Telephony and Linux experience will be helpful but is not essential. Readers need not have prior knowledge of OpenSER.

Build high-speed and highly scalable telephony systems using OpenSIPS About This Book Install and configure OpenSIPS to authenticate, route, bill, and monitor VoIP calls Gain a competitive edge using the most scalable VoIP technology Discover the latest features of OpenSIPS with practical examples and case studies Who This Book Is For If you want to understand how to build a SIP provider from scratch using OpenSIPS, then this book is ideal for you. It is beneficial for VoIP providers, large enterprises, and universities. This book will also help readers who were using OpenSER but are now confused with the new OpenSIPS. Telephony and Linux experience will be helpful to get the most out of this book but is not essential. Prior knowledge of OpenSIPS is not assumed. What You Will Learn Learn to prepare and configure a Linux system for OpenSIPS Familiarise yourself with the installation and configuration of OpenSIPS Understand how to set a domain and create users/extensions Configure SIP endpoints and make calls between them Make calls to and from the PSTN and create access control lists to authorize calls Install a graphical user interface to simplify the task of provisioning user and system information Implement an effective billing system with OpenSIPS Monitor and troubleshoot OpenSIPS to keep it running smoothly In Detail 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. Style and approach This book is a step-by-step guide based on the example of a VoIP provider. You will start with OpenSIPS installation and gradually, your knowledge depth will increase.

Asterisk is the leading Open Source Telephony application and PBX software solution. It represents an effective, easy-to-administer, and accessible platform for running enterprise telephony requirements. The real world, however, offers numerous hurdles when running Asterisk in the commercial environment including call routing, resilience, or integrating Asterisk with other systems. This book will show you some of the ways to overcome these problems. As the follow-up to Packt's highly successful 2005 title Building Telephony Systems with Asterisk, this book presents the collected wisdom of Asterisk Professionals in the commercial environment. Aimed at Administrators and Asterisk Consultants who are comfortable with the basics of Asterisk operation and installation, this book covers numerous hands-on topics such as Call Routing, Network Considerations, Scalability, and Resilience _ all the while providing practical solutions and suggestions. It also covers more business-related areas like Billing Solutions and a Winning Sales Technique. Even if your interest or experience with Asterisk is lower level, this book will provide a deeper understanding of how Asterisk operates in the real world. Asterisk is deployed across countless enterprises globally. Running on Linux, it has constantly demonstrated its resilience, stability, and scalability and is now the advanced communication solution of choice to many organizations and consultants. With a foreword from Mark Spencer, the man behind Asterisk, this book presents the accumulated wisdom of three leading Asterisk Consultants and shows the reader how to get the most out of Asterisk in the commercial environment. Over the course of eleven chapters, this book introduces the reader to topics as diverse as Advanced Dial Plans, Network Considerations, and Call Routing, through to Localization, DAHDI, Speech Technology, and Working with a GUI. The book also covers the more nebulous aspects of being an Asterisk professional such as evaluating customer requirements and pitching for contracts. This book represents the wisdom and thoughts of front line consultants. The knowledge they impart will prove informative, thought provoking and be of lasting interest to Asterisk professionals.

Design a complete Voice over IP (VoIP) or traditional PBX system with Asterisk, even if you have only basic telecommunications knowledge. This bestselling guide makes it easy, with a detailed roadmap that shows you how to install and configure this open source software, whether you’re upgrading your existing phone system or starting from scratch. Ideal for Linux administrators, developers, and power users, this updated edition shows you how to write a basic dialplan step-by-step, and brings you up to speed on the features in Asterisk 11, the latest long-term support release from Digium. You’ll quickly gain working knowledge to build a simple yet inclusive system. Integrate Asterisk with analog, VoIP, and digital telephony systems Build an interactive dialplan, using best practices for more advanced features Delve into voicemail options, such as storing messages in a database Connect to external services including Google Talk, XMPP, and calendars Incorporate Asterisk features and functions into a relational database to facilitate information sharing Learn how to use Asterisk’s security, call routing, and faxing features Monitor and control your system with the Asterisk Manager Interface (AMI) Plan for expansion by learning tools for building distributed systems

Provides information on Asterisk, an open source telephony application.

Widely adopted by service providers to enable IP telephony, instant messaging, and other data services, SIP is the signaling protocol of choice for advanced multimedia communications signaling. Compiled by noted engineering experts Syed Ahson and Mohammad Ilyas, SIP Handbook: Services, Technologies, and Security of Session Initiation Protocol presents a thorough technical review of all aspects of SIP. It captures the current state of IP Multimedia Subsystem technology and provides a unique source of comprehensive reference material on this subject. SIP Applications for Today and Tomorrow The scope of this volume ranges from basic concepts to future perspectives. Divided into three sections, the book begins with a discussion of SIP in peer-to-peer networks and then goes on to examine advanced media integration, migration considerations, mobility management, and group conferencing, while also reviewing home networking and compliance issues. The middle section of the book focuses on the underlying technologies of SIP. Chapters review network architecture, vertical handoffs, NAT traversals, multipoint extensions, and other areas at the forefront of research. Finally, the text examines various security vulnerabilities and provides perspectives on secure intelligent SIP services with a future outlook on a fraud detection framework in VoIP networks. Insights from International Researchers Authored by 65 experts from across the world, this text is sure to advance the field of knowledge in this ever-changing industry and provide further impetus for new areas of exploration. Because of the editors’ pivotal influence and their proximity to both the current market and the latest science, this work is certain to become the definitive text on this emerging technology.

The Institute of Electrical and Electronics Engineers (IEEE) Communications Society designed the IEEE wireless communication engineering technologies (WCET) certification program to address the wireless industry's growing need for communications professionals with practical problem-solving skills in real-world situations. Individuals who achieve th

This newly revised edition of the ground-breaking Artech House bestseller, SIP: Understanding the Session Initiation Protocol gives you a thorough and up-to-date understanding of this revolutionary protocol for call signaling and IP Telephony. The second edition includes brand new discussions on the use of SIP for wireless multimedia communications. It explains how SIP is powerful "rendezvous" protocol that leverages mobility and presence to allow users to communicate using different devices, modes, and services anywhere they are connected to the Internet You learn why SIP has been chosen by the 3GPP (3rd Generation Partnership Program for wireless cell phones) as the core signaling, presence, and instant messaging protocol.

This book is full of practical code examples aimed at a beginner to ease his or her learning curve.This book is written for IT professionals and enthusiasts who are interested in quickly getting a powerful telephony system up and running using the free and open source application, FreeSWITCH.Telephony experience will be helpful, but not required.