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The next OpenSIPS training is scheduled for April, 11th 2015.

1.  Overview

The OpenSIPS Bootcamp is a training program providing in depth coverage of Configuration and Administration. The students will learn step by step how to configure OpenSIPS to authenticate users, forward calls to the PSTN through Dialplan, integrate Media Servers and Voice Mail, Presence agent, Load Balancing, NAT Traversal for SIP and generate CDR records to a database. At the end, you will learn how to use troubleshooting tools to solve end user problems. The content is exactly the same as the OpenSIPS Bootcamp. 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.

2.  OpenSIPS Quick Start

New in this year is the OpenSIPS Quickstart Training. We've moved some of the basic chapters such as SIP in depth, Installation, Web GUI and basic scripting to the Quickstart. The objective is to concentrate in advanced topics in the bootcamp and level the students before the training. If you already know SIP, OpenSIPS installation and have previous experience on routing scripts you can go straight to the bootcamp, otherwise we strongly recommend you to take the Quickstart VoD training. Quickstart will be available starting in March 15th 2016 at http://ebootcamp.opensips.org

3.  How does it work?

The live classes will be taken online by web-conference two sessions per week, 2-3 hours per session. To attend this training you will need to have broadband Internet access. You are going to receive a link to download a virtual machine to run the labs. The virtual machine will be available in the VMWARE format. You can download the free VMWARE player to run the VM. We suggest that you have one separate desktop or server for your VM and at least one IP Phone/ATA in your private labs to complete the training. A LMS (Learning management system) will be available with forums, quizzes and support materials.

4.  Key Objectives

  • Routing basics and the default configuration
  • OpenSIPS authentication using MySQL and Memcache
  • 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 Database
  • How to secure your system against DOS and hacker's attacks
  • Implement complex SIP B2BUA scenarios with OpenSIPS
  • How to use tests, stress-tests and monitoring tools to check your configuration

5.  Syllabus

5.1  Schedule

Session 1 – Live Conference - OpenSIPS Database Integration
  1. Database integration for authentication and location services
  2. Register authentication sequence
  3. Invite authentication sequence
  4. Digest authentication
  5. Quality of protection
  6. Plaintext or hash passwords
  7. The opensipsctl shell utility
  8. Checking From and TO tags
  9. Multidomain support
  10. Inter-domain and intra-domain routing
  11. LAB Installing MySQL Support
  12. LAB Enhancing the script
  13. LAB Installing the opensips-cp
  14. LAB Configuring opensips-cp
Session 2 PSTN connectivity (part 1)
  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. LAB Simple routing to PSTN
Session 3 – In-Class – PSTN connectivity (part 2)
  1. Using the dialplan module
  2. Introduction to Drouting
  3. Drouting tables
  4. LAB Using Dynamic Routing tables
  5. LAB Using the Dialplan module for pre-routing
Session 4 – 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
  6. LAB: Implementing call fwd with avpops
Session 5 – SIP presence ad HA
  1. Presence Agent setup
  2. Publishing Presence from non-SIP devices
  3. Registration-to-Presence conversion (old SIP devices)
  4. Scalability of the presence model
  5. Aggregation of the presence information
  6. OpenSIPS High Availability
  7. Active/Active and Active/Backup setups
  8. SIP and Data Replication
  9. LAB Implementing presence aggregation
  10. LAB Publishing non-SIP Presence
Session 6 – In-Class – SIP Dialog Awareness and Load Balancing
  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. OpenSIPS Load balancing/Dispatching Capabilities
  6. Balancing algorithms
  7. Balancing and failover
  8. Multiple groups of balancing
  9. LAB Limiting the number of concurrent calls
  10. LAB Load balancing & failover for an Asterisk Cluster
Session 7 – In-Class – 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 Usind RTPProxy for NAT traversal
  11. STUN – Simple Traversal of UDP NAT
Session 8 – In-Class - Accounting & Billing, Monitoring Tools
  1. Authentication, Accounting and Authorization
  2. Generating CDRs using the ACC and Dialog Modules for postpaid users.
  3. Session Timeout, integration with RTPProxy
  4. Performance analysis using Statistics Monitor
  5. SNMP scalars and traps available for management
  6. Tracing calls for troubleshooting
  7. Generating traffic with SIPP
  8. LAB Accounting to a MySQL database
  9. LAB Prepaid users, limiting the duration of the call using the Dialog module
  10. LAB Using SIP Trace
Session 9 – In-Class – 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
Session 10 – Advanced Topics
  1. Websockets and WebRTC
  2. Asynchronous Routing
  3. Topology hiding
  4. LAB Topology Hiding
  5. Final work assignment, the final work will be delivered by students after the training

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.  Instructors

  • Bogdan-Andrei Iancu – OpenSIPS Solutions / OpenSIPS founder and main developer.
  • Flávio E. Goncalves – CEO of SipPulse Routing and Billing Solutions for SIP, writer of the book, Building Telephony Systems with OpenSIPS.

9.  Contact & Registration

ebootcamp at opensips dot org


Page last modified on February 27, 2016, at 09:08 PM