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Training -> Bootcamp

Starting with 2011, the Bootcamp program was merged into the eBootcamp program - same content, some trainers, same still, but in remote approach, via a eLearning platform:


  • the eBootcamp is more flexible as time (when you can start) and program (only 4-5 hours per week)
  • Students have more time for labs and to digest a large amount of information
  • You don't need to dedicate a whole week for traveling and training, away from your job/family
  • It lowers the cost, time and logistics related to international traveling.

Where : eBootcamp Web Page

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 closed
  • 15 - 19 November 2010, Edison, New Jersey, USA Successfully Completed

2.  New Bootcamp program for 2010

The new OpenSIPS 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, implement High-Availablity, Presence agent, Load Balancing, Traverse Nat for SIP, generate CDR records and use OpenSIPS as B2BUA. At the end, you will learn how to increase the security of your system and 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, an certification exam, to proof the knowledge consolidation during the training, can be sustained on request after the end of the course.

3.  What's new

  1. Removal of the certification test in the last day of the training. Most people need to study before taking the test and having the test in the last day is counterproductive.
    1. Reduction in redundant themes and presentations. No presentation in this training takes more than 1 hour. The ones with more than 1 hour were split in 45 minute sections.
    2. BYOL format (Bring your own laptop). As of today most students have their own notebook with enough memory and disk space. Each student will make the training in a virtual machine in their own computer and take the VM ready to run back home
    3. Removal of any non-opensips related material such as Asterisk Integration, Wireshark, Sipsak and Sipp. (The old labs will be available as optional).
    4. Inclusion of the following new topics, SIP security, SIP Dialog Awareness and B2BUA.
    5. Removal of the preparation time on Monday. Most students actually arrive in the day before. Rarely, we have seen students arriving in the day of the training. This will open more time for the new topics.

4.  Key Objectives

  • Describe SIP basic call flows, recognize components, addresses and headers
  • Describe OpenSIPS use case scenarios, architecture, advantages and limitations
  • Install, configure and debug OpenSIPS and use its scripting language
  • Integrate with a relational database such as MySQL
  • Install, configure and use the Web Gui, opensips control panel
  • Integrate with PSTN gateways and wholesale providers
  • Implement advanced SIP call flows such as Call Forward, Call Hold and Call Transfer
  • Implement presence using a centralized presence model
  • SIP dialog awareness
  • Understand important aspects of load balancing and high availability
  • Implement SIP NAT traversal using RTPProxy
  • Basic prepaid and postpaid accounting
  • Defend your SIP infrastructure from malicious attacks
  • Learn advanced Topics, with SIPTRACE, STATISTICS, SNMP, B2BUA

5.  Syllabus

  1. SIP Basics
    1. SIP Addresing scheme
    2. SIP Basic call flows
    3. Sessions, Transactions and Dialogs
    4. Initial and Sequential Requests
    5. Media Handling and SDP
  2. OpenSIPS usage scenarios and architecture
    1. Use case scenarios, VoIP Providers, Wholesale routing, Hosted PBX¡­
    2. OpenSIPS architecture, advantages and limitations
    3. Routing Basics and the Standard Configuration
    4. Scripting Basics
    5. Routing Basics
    6. Analyzing the standard configuration files
    7. LAB 3.1 OpenSIPS Installation
    8. LAB 3.2 Connecting two phones to OpenSIPS
    9. LAB 3.3 Running stateful, stateless, with/without record routing
  3. Integration with relational databases
    1. The Auth_DB modules
    2. Register authentication sequence
    3. Invite authentication sequence
    4. Digest authentication
    5. QOP ¨C 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
  4. 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
  5. 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
  6. Advanced SIP Call Flows
    1. Parallel and serial forking
    2. The importance of the messages Subscribe, Notify and Refer
    3. Call Forwarding, unconditional, on busy, on no answer
    4. Call transfer attended and unattended
    5. Call hold
  7. Presence in a centralized model
    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. SIP Dialog Awareness
    1. The dialog module
    2. How Dialog awareness in implemented with OpenSIPS
    3. Dialog variables and dialog profiling
    4. Mi commands used for Dialog control
    5. LAB 9-1 Limiting the number of concurrent calls
  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 10-1 Load balancing & failover for an 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 11-1 Usind RTPProxy for NAT traversal
    11. STUN ¨C Simple Traversal of UDP NAT
  11. Basic Prepaid and Postpaid billing
    1. Authentication, Accounting and Authorization
    2. LAB 12-1 Accounting to a MySQL database
    3. Generating CDRs using the ACC and Dialog Modules for postpaid users.
    4. Session Timeout, integration with RTPProxy
    5. LAB 12-2 Prepaid users, limiting the duration of the call using the Dialog module
  12. SIP Security
    1. Common types of attacks to the SIP environment
    2. Scanning Attacks
    3. Floods
    4. SIP digest leaking
    5. Using PIKE to detect and prevent floods
    6. Using return codes from authentication to identify scanning attacks
    7. Implementing TLS and SRTP
  13. Advanced Topics
    1. Performance analysis using Statistics Monitor
    2. SNMP scalars and traps available for management
    3. Tracing calls for troubleshooting
    4. Topology hiding using B2BUA
    5. LAB 12-1 ¨C Using SIP Trace
    6. LAB 13-1 ¨C Using the B2BUA for topology hiding

6.  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

7.  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)

8.  Course Schedule

  • 08:30-18:00 Mon-Fri

9.  Instructors

  • Bogdan-Andrei Iancu OpenSIPS Solutions / OpenSIPS founder and main developer.
  • Flavio E. Goncalves CEO of V.Office Networks, writer of the book, Building Telephony Systems with OpenSIPS.

10.  Contact

Page last modified on August 29, 2016, at 01:44 PM