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Training

Training -> Bootcamp

1.  2017 Schedule

2.  New Bootcamp program for 2017

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, Load Balancing, Traverse Nat for SIP, generate CDR records. 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. We have increased the depth of OpenSIPS knowledge on scripting and routing
  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.
  4. 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
  • Learn advanced Topics, with SIPTRACE, STATISTICS

5.  Syllabus

  1. SCRIPTING LANGUAGE AND ARCHITECTURE
    1. CONFIG FILE STRUCTURE
    2. ROUTES AND SUB-ROUTES
    3. REPLY AND FAILURE ROUTES
    4. OTHER TYPES OF ROUTES
    5. SCRIPT STATEMENTS & OPERATIONS
    6. SCRIPT VARIABLES
    7. TRANSFORMATIONS
    8. LAB 1: INSTALLING OPENSIPS 2
  2. SIP ROUTING AND TRANSACTIONS
    1. THE TRANSACTION MODULE
    2. SIP TIMERS
    3. SIP ROUTING WITH OPENSIPS
    4. SIP MESSAGE OPERATIONS (ADD AND REMOVE HEADERS)
    5. MANAGEMENT INTERFACE (FIFO, XML_RPC)
    6. ROUTING VARIABLES
    7. LAB 2: RUNNING THE DEFAULT SCRIPT WITHOUT NAT
  3. DATABASE INTEGRATION
    1. 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. MULTIDOMAIN SUPPORT
    9. INTER-DOMAIN AND INTRA-DOMAIN ROUTING
    10. LAB CONFIGURING MYSQL SUPPORT FOR AUTHENTICATION AND LOCATION
  4. 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
  5. PSTN CONNECTIVITY (PART 2)
    1. INTRODUCTION TO DYNAMIC ROUTING
    2. RULES, CARRIERS AND GATEWAYS
    3. PARTITIONING
    4. SEQUENTIAL AND WEIGHT BASED GATEWAY SELECTION
    5. PROBING MODE
    6. LAB USING DYNAMIC ROUTING TABLES
    7. USING REGULAR EXPRESSIONS TO SELECT SERVICES
    8. CALLER AND CALLEE NUMBER TRANSLATIONS
    9. LAB USING THE DIALPLAN MODULE FOR PRE-ROUTING
  6. ADVANCED SIP CALL FLOWS
    1. PARALLEL AND 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. HUNT GROUPS, PARALLEL AND SERIAL
    7. LAB: IMPLEMENTING CALL FWD
    8. LAB – IMPLEMENTING A HUNT GROUP
  7. HANDLING SIP DIALOGs
    1. THE DIALOG MODULE
    2. DIALOG VARIABLES
    3. DIALOG PROFILING
    4. MI COMMANDS USED FOR DIALOG CONTROL
    5. TERMINATING CALLS IN THE SERVER SIDE
    6. LAB LIMITING THE NUMBER OF CONCURRENT CALLS
  8. OPENSIPS AS A FRONT-END FOR ASTERISK AND FREESWITCH
    1. OPENSIPS AS A MID-REGISTRAR
    2. LOAD-BALANCING MEDIA CLUSTERS
    3. HANDLING REFER’S
    4. HANDLING CONFERENCE ROOMS
    5. BALANCING FOR CALL TRANSFER AND CONFERENCE SERVICES
    6. LAB LOAD BALANCING & FAILOVER FOR A MEDIACLUSTER
  9. NAT TRAVERSAL
    1. NAT TYPES
    2. SOLVING THE NAT TRAVERSAL CHALLENGE
    3. IMPLEMENTING A FAR END NAT SOLUTION
    4. THE OPENSIPS UNIVERSAL SOLUTION STUN+TURN
    5. RTPPROXY INSTALLATION AND CONFIGURATION
    6. LAB USING RTPPROXY FOR NAT TRAVERSAL
    7. STUN – SIMPLE TRAVERSAL OF UDP NAT
    8. RUNNING OPENSIPS BEHIND NAT
  10. ACCOUNTING AND MONITORING TOOLS
    1. AUTHENTICATION, ACCOUNTING AND AUTHORIZATION
    2. MANUAL AND AUTOMATIC ACCOUTING
    3. EXTRA ACCOUTING
    4. MULTILEG ACCOUTING
    5. GENERATING CDRS USING THE ACC AND DIALOG MODULES FOR POSTPAID USERS.
    6. SESSION TIMEOUT, INTEGRATION WITH RTP PROXY
    7. PERFORMANCE ANALYSIS USING STATISTICS MONITOR
    8. LAB ACCOUNTING TO A MYSQL DATABASE
    9. LAB PREPAID USERS, LIMITING THE DURATION OF THE CALL USING THE DIALOG MODULE

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 - OpenSIPS Quickstart elearning.opensips.org)
  • Basic OpenSIPS Knowledge (Free online learning - OpenSIPS Quickstart elearning.opensips.org)
  • 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 May 15, 2017, at 10:49 AM