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

Where does NAT break SIP?


NAT affects SIP because SIP is a session protocol. The translation of the address occurs in the network headers leaving the SIP headers in the session layer untouched. Thus, the server is unable to contact the user using the addresses provided in the SIP headers. The following headers, Contact, Via, Route, and Record-Route are not automatically translated by the NAT device. Session Description Protocol (SDP) is also affected. See the points where some layer-3 addresses appear in a SIP request in the following highlighted code:

U 189.101.207.211:11266 -> 208.109.122.193:5060
INVITE sip:[email protected] SIP/2.0.
Via: SIP/2.0/UDP 192.168.1.160:11266;branch=z9hG4bK-d8754z-1f2cd509;rport.
Max-Fowards: 70.
Contact: <sip:flavio@192.168.1.160:11266>.
To: "8580"<sip:[email protected]>.
From: "flavio"<sip:[email protected]>;tag=99494a4b.
Call-ID: NmYwNjAzMDE3MTE0YWM5MWIxNjNiMWNjZDY3NjI0MWQ..
CSeq: 1 INVITE.
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE,...