Building Telephony Systems with OpenSIPS 1.6
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- Build a VoIP Provider based on the SIP Protocol
- Cater to scores of subscribers efficiently with a robust telephony system based in pure SIP
- Gain a competitive edge using the most scalable VoIP technology
- Learn how to avoid pitfalls using precise billing
- Packed with rich practical examples and case studies on the latest OpenSIPS version 1.6
Book Details
Language : EnglishPaperback : 284 pages [ 235mm x 191mm ]
Release Date : January 2010
ISBN : 1849510741
ISBN 13 : 9781849510745
Author(s) : Flavio E. Goncalves
Topics and Technologies : All Books, Networking & Telephony, Open Source
Table of Contents
Preface
Chapter 1: Introduction to SIP
Chapter 2: Introduction to OpenSIPS
Chapter 3: OpenSIPS Installation
Chapter 4: Script and Routing Basics
Chapter 5: Adding Authentication with MySQL
Chapter 6: Graphical User Interfaces for OpenSIPS
Chapter 7: Connectivity to the PSTN
Chapter 8: Media Services Integration
Chapter 9: SIP NAT Traversal
Chapter 10: OpenSIPS Accounting and Billing
Chapter 11: Monitoring Tools
Index
Flavio E. Goncalves
Code Downloads
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Submit Errata
Please let us know if you have found any errors not listed on this list by completing our errata submission form. Our editors will check them and add them to this list. Thank you.
Errata
- 5 submitted: last submission 09 Jan 2012type: Technical | Page number: 18
Last sentence: ...(that is, "487 busy here")... Should be: ...(that is, "486 busy here")...
type: Code | Page number: 168
On the last line of the page: countrycode)=contrycode;$avp(s:areacode)=areacode") Should be countrycode)=countrycode;$avp(s:areacode)=areacode")
type: Others | Page number: 165
"We will have two gateways " should be "We will have three gateways"
Errata type: Code | Page number: 150
Missing closing bracket in the following line:
if (!check_source_address("0"). It should be if (!check_source_address("0"))
Errata type: Technical | Page number: 207 | Errata date: 22 July 11
"It seems that force_rtp_proxy was replaced by rtpproxy_offer() and rtpproxy_answer()"
Sample chapters
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- 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
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
This is a practical, hands-on book based around a fictitious case study VoIP Provider that you will build on a development server using OpenSIPS 1.6. The case study grows chapter by chapter, from installing your local development server, right up to the finished VoIP provider.
This book is for readers who want to understand how to build a SIP provider from scratch using OpenSIPS. It is suitable for VoIP providers, large enterprises, and universities.
Telephony and Linux experience will be helpful but is not essential. Readers need not have prior knowledge of OpenSIPS. This book will also help readers who were using OpenSER but are now confused with the new OpenSIPS.
