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Training -> Bootcamp

1.  2010 Schedule

• 22 - 26 March 2010, Florianopolis, Brazil - successfully completed
• 19 - 23 April 2010, San Francisco, USA - successfully completed
• 20 - 24 September 2010, Frankfurt, Germany Registration Open
• 15 - 19 November 2010, Edison, New Jersey, USA

2.  Overview

The OpenSIPS 1.6 Bootcamp is a full 5 day (40 hours) intensive training providing in depth coverage of OpenSIPS Installation, Configuration and Administration. The students will learn how to download, compile and install OpenSIPS. After the installation, you will start to learn step by step how to configure OpenSIPS to authenticate users, install a GUI to help with daily administration, forward calls to the PSTN through Dialplan, integrate Asterisk and Voice Mail, Presence agent, Load Balancing, Traverse Nat for SIP and generate CDR records to a Radius Server. At the end, you will learn how to use troubleshooting tools to solve end user problems.

All the knowledge that is transferred to you will be strongly backed-up by practice sessions where you will get hands-on experience in handling OpenSIPS SIP Server. The training is structured to be offer 50% - 50% between the theoretical and practical sessions.

Optionally, a certification exam, to proof the knowledge consolidation during the training, can be sustained on request at the end of the course.

The detailed description of the course is available for download.

3.  Key Objectives

• Install OpenSIPS on a Linux Machine
• Routing basics and the default configuration
• OpenSIPS authentication using MySQL and Memcache
• Install OpenSIPS control Panel.
• Connect to the PSTN using Dialplan and Dynamic Routing
• Voicemail integration using Call Forward and AVPs
• Implement a presence agent
• Understand important aspects of load balancing and high availability
• Implement SIP NAT traversal using RTPProxy
• Account Calls to MySQL
• How to use test and monitoring tools to check your configuration

4.  Syllabus

1. Introduction to OpenSIPs
   1. What is OpenSIPS
   2. Main characteristics
   3. Usage scenarios
   4. OpenSIPS architecture
   5. Sessions, Dialogs and Transactions
   6. Message Processing according to the RFC3261
   7. Strict Routing and Loose Routing
   8. SIP and RTP
2. OpenSIPS installation
   1. Hardware requirements
   2. Software requirements
   3. LAB 2.1 – Installing Linux for OpenSIPS (previously installed DVD)
   4. LAB 2.2 – Download, compile and install OpenSIPS
   5. LAB 2.3 – Running OpenSIPS at the Linux Boot
   6. OpenSIPS directory structure and log files
   7. OpenSIPS startup options
   8. Starting OpenSIPS with default configuration script
3. Routing Basics and the Standard Configuration
   1. Scripting Basics
   2. Routing Basics
   3. Analyzing the standard configuration files
   4. LAB 3.1 Connecting two phones to OpenSIPS
   5. LAB 3.2 Running stateful with record routing (packet capture)
   6. LAB 3.3 Running stateless with record routing (packet capture)
   7. LAB 3.4 Running stateless with no record-routing.
4. Adding authentication with MySQL
   1. The Auth_DB modules
   2. Register authentication sequence
   3. Invite authentication sequence
   4. Digest authentication
   5. QOP – Quality of protection
   6. Plaintext or hash passwords
   7. LAB 4-1 Installing MySQL Support
   8. The opensipsctl shell utility
   9. The opensipsCTL resource file
   10. Checking From and TO tags
   11. Multidomain support
   12. Inter-domain and intra-domain routing
   13. LAB4-2 Enhancing the script
5. OpenSIPS Administration using OpenSIPS Control Panel
   1. Introduction to OpenSIPS Control Panel
   2. LAB 5-1 Installing the opensips-cp
   3. LAB 5-2 Configuring opensips-cp
   4. Basic tasks
   5. Domain administration
   6. User administration
   7. Interface customization
6. Connectivity to the PSTN
   1. Introduction to PSTN routing
   2. Accepting calls from the PSTN
   3. The permissions module and the check_source_address() function
   4. Routing a call to the PSTN
   5. DID redirection using Aliases
   6. ACL and Group permissions
   7. Introduction to Drouting
   8. Drouting tables
   9. LAB 6-1 Routing calls to the PSTN
   10. LAB 6-2 Using Dynamic Routing tables
   11. LAB 6-3 Using the Dialplan module for pre-routing
   12. Inter-domain Peering
7. Call Forwarding and Voicemail
   1. Introduction to Call Forwarding
   2. Pseudo-variables and AVPs
   3. AVP functions
   4. Implementing Blind Call Forwarding
   5. Busy or Unanswered forwarding to Voice Mail
   6. LAB 7-1 Testing the Call Forwarding feature
   7. LAB 7-2 Asterisk Integration (optional)
8. Using Presence
   1. SIP presence overview
   2. Presence Agent setup
   3. Publishing Presence from non-SIP devices
   4. Registration-to-Presence conversion (old SIP devices)
   5. Scalability of the presence model
   6. Aggregation of the presence information
   7. LAB 8-1 Implementing presence aggregation
   8. LAB 8-2 Publishing non-SIP Presence
9. Load Balancing and High Availability
   1. OpenSIPS High Availability
   2. Active/Active and Active/Backup setups
   3. SIP and Data Replication
   4. OpenSIPS Load balancing/Dispatching Capabilities
   5. Balancing algorithms
   6. Balancing and failover
   7. Multiple groups of balancing
   8. LAB 9-1 Load balancing & failover foran Asterisk Cluster
10. SIP NAT Traversal
    1. NAT Types
    2. Solving the NAT traversal challenge
    3. Implementing a far end NAT solution
    4. RFC3581 and forc_rport() function
    5. Solving the traversal of RTP packets
    6. Handling Register Requests
    7. Detecting clients Behind NAT
    8. Handling Invite requests behind NAT
    9. RTPProxy installation and configuration
    10. LAB 9-1 Usind RTPProxy for NAT traversal
    11. STUN – Simple Traversal of UDP NAT
11. OpenSIPS accounting and Billing
    1. Authentication, Accounting and Authorization
    2. LAB 10-1 Accounting to a MySQL database
    3. Accounting using a RADIUS server
    4. LAB 10-2 Accounting to a Radius Server
12. Troubleshooting Tools
    1. Built in tools
    2. Packet Capture and Trace Tools
    3. The SIPTRACE module
    4. Predefined and Custom Statistics
    5. Stress Testing Tools
    6. LAB 12-1 – Using SIP Trace
    7. LAB 12-2 Using sipp to stress test OpenSIPS

5.  Audience

VoIP providers seeking “Open Source” platforms to enhance their businesses

• Anyone seeking proficiency in OpenSIPS
• Network Consultants and VARs who need a jump start in the technology
• Developers who want to use OpenSIPS to create new telephony applications and appliances

6.  Prerequisites

• Basic Linux knowledge
• Basic text edition
• Basic SIP protocol (Free online learning)
• Basic OpenSIPS Knowledge (Free online learning)
• Programming logic knowledge (you won’t need to program, but you need to understand logical concepts applied to the dial-plan)

7.  Course Schedule

Please check the PDF - detailed description of the course

8.  Instructors

• Bogdan-Andrei Iancu – OpenSIPS founder and main developer. Also CEO of Voice System, an “know-how” OpenSIPS company.
• Flávio E. Goncalves – CEO of V.Office Networks, writer of the book, Building Telephony Systems with OpenSIPS.

The detailed description of the course is available for download.

9.  Contact