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.

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Building Telephony Systems with OpenSIPS Second Edition - Second Edition
Flavio E. Goncalves | Bogdan-Andrei Iancu
Jan, 2016 | 384 pages
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Table of Contents (21 chapters)
Building Telephony Systems with OpenSIPS Second Edition
Building Telephony Systems with OpenSIPS Second Edition
Credits
Credits
About the Authors
About the Authors
About the Reviewers
About the Reviewers
www.PacktPub.com
www.PacktPub.com
Preface
Preface
Free Chapter
1
Introduction to SIP
Introduction to SIP
Understanding the SIP architecture
The SIP registration process
Types of SIP servers
SIP request messages
The SIP dialog flow
SIP transactions and dialogs
Locating the SIP servers
SIP services
The SIP identity
The RTP protocol
Session Description Protocol
The SIP protocol and OSI model
The VoIP provider's big picture
Additional references
Summary
2
Introducing OpenSIPS
Introducing OpenSIPS
Understanding OpenSIPS
OpenSIPS capabilities
Usage scenarios for OpenSIPS
Who's using OpenSIPS?
The OpenSIPS design
Summary
3
Installing OpenSIPS
Installing OpenSIPS
Hardware and software requirements
Generating OpenSIPS scripts
Working with the log files
Startup options
Summary
4
OpenSIPS Language and Routing Concepts
OpenSIPS Language and Routing Concepts
An overview of OpenSIPS scripting
The OpenSIPS configuration file
Scripting capabilities
SIP routing in OpenSIPS
Summary
5
Subscriber Management
Subscriber Management
Modules
The AUTH_DB module
The REGISTER authentication sequence
The INVITE authentication sequence
Digest authentication
Plaintext or prehashed passwords
Installing MySQL support
Analysis of the opensips.cfg file
The opensipsctl shell script
Using aliases
Handling the CANCEL requests and retransmissions
Lab – multidomain support
Lab – using aliases
IP authentication
Summary
6
OpenSIPS Control Panel
OpenSIPS Control Panel
The OpenSIPS control panel
Installation of OpenSIPS-CP
Configuring administrators
Adding and removing domains
Manage the access control lists or groups
Managing aliases
Managing subscribers
Managing permissions and IP authentication
Sending commands to the management interface
A generic table viewer
Summary
7
Dialplan and Routing
Dialplan and Routing
The dialplan module
PSTN routing
Summary
8
Managing Dialogs
Managing Dialogs
Enabling the dialog module
Creating a dialog
Dialog matching
Dialog states
Dialog timeout and call disconnection
Dialog variables and flags
Profiling a dialog
Disconnecting calls
Topology hiding
Validating a dialog and fixing broken dialogs
Displaying the dialog statistics
SIP session timers
Summary
9
Accounting
Accounting
Progress check
Selecting a backend
Accounting using RADIUS
Lab – accounting using a FreeRADIUS server
Missing BYEs and CDRs
Summary
10
SIP NAT Traversal
SIP NAT Traversal
Port address translation
Where does NAT break SIP?
Types of NAT
Solving the SIP NAT traversal challenge
Building the solution
Understanding the solution flow
Summary
11
Implementing SIP Services
Implementing SIP Services
Where to implement SIP services
Explaining RFC 5359 with SIP service examples
Playing announcements
Call forwarding
Implementing call forward on busy or unanswered
Debugging the routing script
Lab – testing the call forwarding feature
Call transfer
Summary
12
Monitoring Tools
Monitoring Tools
Built-in tools
Trace tools
Troubleshooting routing scripts
Summary
13
OpenSIPS Security
OpenSIPS Security
Configuring a firewall for OpenSIPS
Preventing DNS and registration poisoning
Enabling Transport Layer Security
Enabling Secure Real-time Protocol
Enabling the anti-fraud module
Summary
14
Advanced Topics with OpenSIPS 2.1
Advanced Topics with OpenSIPS 2.1
Asynchronous operations
Asynchronous support in the OpenSIPS script
Binary replication
TCP handling
Summary
Index
Index

Call forwarding

Now, we will work with call forwarding on OpenSIPS. There are usually three kinds of call forwards:

  • Call forward unconditional: The system redirects all the INVITE requests sent to this phone number to the phone stored in the usr_preferences table. The SIP server will simply replace the request URI before sending the request ahead.

  • Forward on busy: In this case, we will use the failure_route feature to intercept the 486 Busy Here message and create a new leg sending the INVITE request to the final destination.

  • Forward on no answer: If a phone replies to an INVITE request with a 480 Temporarily Unavailable or the SIP proxy generates a timeout with a 408 request Timeout, then OpenSIPS will again use failure_route to intercept the message and create a new leg sending the INVITE to the final destination.

To implement any of these services, we need to store data related to the called number. During the authentication process, it is possible to load data from the caller but not...

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