Training.EBootcamp History

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April 24, 2013, at 05:44 PM by 109.99.235.212 -
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Training -> eBootcamp


(:toc-float Table of Content:)

Annual Calendar

  • May 27th 2013
  • September 2nd 2013
  • November 4th 2013
  • March 3rd 2014

Next scheduled training starts at May 27th, 2013. For more details, please send an email to ebootcamp_at_opensips_dot_org

Overview

The OpenSIPS 1.9 eBootcamp is a seven week e-Training program 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 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.

How does it work?

The live classes will be taken online by web-conference two sessions per week, 2-3 hours per session, for 7 weeks. 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.

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 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

Syllabus

Schedule

Week #1 – Session 1 – Live Conference – Introduction to SIP and OpenSIPS
  1. What SIP is and what SIP is for
  2. SIP Architecture
  3. SIP Addressing scheme
  4. SIP Basic call flows
  5. Sessions, Transactions and Dialogs
  6. Initial and Sequential Requests
  7. Media Handling and SDP
  8. Use case scenarios, VoIP Providers, Wholesale routing, Hosted PBX‘­
  9. OpenSIPS architecture, advantages and limitations
Week #1 – Session 2 – Live conference – OpenSIPS Basics
  1. Routing Basics and the Standard Configuration
  2. Scripting Basics
  3. Routing Basics
  4. Analyzing the standard configuration files
  5. LAB OpenSIPS Installation
  6. LAB Connecting two phones to OpenSIPS
  7. LAB Running stateful, stateless, with/without record routing
Week #2 – Session 3 – Live Conference - OpenSIPS Control Panel and SQL Authentication
  1. Introduction to OpenSIPS Control Panel
  2. Domain administration
  3. User administration
  4. Interface customization
  5. The Auth_DB modules
  6. Register authentication sequence
  7. Invite authentication sequence
  8. Digest authentication
  9. Quality of protection
  10. Plaintext or hash passwords
  11. The opensipsctl shell utility
  12. Checking From and TO tags
  13. Multidomain support
  14. Inter-domain and intra-domain routing
  15. LAB Installing MySQL Support
  16. LAB Enhancing the script
  17. LAB Installing the opensips-cp
  18. LAB Configuring opensips-cp
Week #2 – Session 4 – Live Conference – 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
Week #3 – Session 5 – Live Conference – PSTN connectivity (part 2)
  1. Using the dialplan module
  2. Introduction to Drouting
  3. Drouting tables
  4. LAB Routing calls to the PSTN
  5. LAB Using Dynamic Routing tables
  6. LAB Using the Dialplan module for pre-routing
Week #3 – Session 6 – Live Conference – 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
Week #4 – Session 7 – Live Conference – 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
Week #4 – Session 8 – Live Conference – 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
Week #5 – Session 9 – Live Conference – 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
Week #5 – Session 10 – Live Conference - 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
Week #6 – Session 11 – Live Conference – 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
Week #6 – Session 12 – Back2Back User Agent
  1. B2BUA design
  2. Writing scenarios with B2BUA
  3. Topology hiding using B2BUA
  4. Complex B2BUA scenarios
  5. LAB Using SIP Trace
  6. LAB Using the B2BUA for topology hiding
Week #7 – Session 13 – Live Conference – Final Work
  1. Final work assignment
Week #7 – Session 14 – Live Conference – Final Work
  1. Q & A on Final work

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

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)

Instructors

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

Contact & Registration

ebootcamp at opensips dot org

to:

(:redirect Support.EBootcamp quiet=1:)

April 24, 2013, at 12:24 AM by flaviogoncalves -
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  • May 26th 2013
to:
  • May 27th 2013
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Next scheduled training starts at May 26th, 2013. For more details, please send an email to ebootcamp_at_opensips_dot_org

to:

Next scheduled training starts at May 27th, 2013. For more details, please send an email to ebootcamp_at_opensips_dot_org

March 27, 2013, at 10:49 AM by bogdan -
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Schedule

March 27, 2013, at 10:49 AM by bogdan -
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Schedule

March 27, 2013, at 10:46 AM by bogdan -
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  • May 26th 2013
  • September 2nd 2013
  • November 4th 2013
  • March 3rd 2013

Next scheduled training starts at May 26th, 2013. For more details, please send an email to bootcamp_at_opensips_dot_org

to:
  • May 26th 2013
  • September 2nd 2013
  • November 4th 2013
  • March 3rd 2014

Next scheduled training starts at May 26th, 2013. For more details, please send an email to ebootcamp_at_opensips_dot_org

Deleted lines 199-202:

2013 Schedule

The next eBootcamp will start in February 25th, We accept late registrations until february 18th.

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bootcamp at opensips dot org

to:

ebootcamp at opensips dot org

March 26, 2013, at 08:00 PM by flaviogoncalves -
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May 26th 2013 September 2nd 2013 November 4th 2013 March 3rd 2013

to:
  • May 26th 2013
  • September 2nd 2013
  • November 4th 2013
  • March 3rd 2013
March 26, 2013, at 07:56 PM by flaviogoncalves -
Changed lines 5-6 from:

Next scheduled training starts at February 25th, 2013. For more details, please send an email to bootcamp_at_opensips_dot_org

to:

Annual Calendar

May 26th 2013 September 2nd 2013 November 4th 2013 March 3rd 2013

Next scheduled training starts at May 26th, 2013. For more details, please send an email to bootcamp_at_opensips_dot_org

Changed line 16 from:

The OpenSIPS 1.8 eBootcamp is a seven week e-Training program 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 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.

to:

The OpenSIPS 1.9 eBootcamp is a seven week e-Training program 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 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.

January 25, 2013, at 04:20 PM by 109.99.235.212 -
Changed line 5 from:

Next scheduled training starts at February 25th, 2012. For more details, please send an email to bootcamp_at_opensips_dot_org

to:

Next scheduled training starts at February 25th, 2013. For more details, please send an email to bootcamp_at_opensips_dot_org

January 08, 2013, at 11:58 AM by flaviogoncalves -
Changed lines 191-195 from:
  • Flαvio E. Goncalves – CEO of V.Office Networks, writer of the book, Building Telephony Systems with OpenSIPS.

2012 Schedule

The next eBootcamp will start in September 10th, We accept late registrations until September 3rd.

to:
  • Flαvio E. Goncalves – CEO of SipPulse Routing and Billing Solutions for SIP, writer of the book, Building Telephony Systems with OpenSIPS.

2013 Schedule

The next eBootcamp will start in February 25th, We accept late registrations until february 18th.

January 08, 2013, at 11:57 AM by flaviogoncalves -
Changed line 5 from:

Next scheduled training starts at September 10th, 2012. For more details, please send an email to bootcamp_at_opensips_dot_org

to:

Next scheduled training starts at February 25th, 2012. For more details, please send an email to bootcamp_at_opensips_dot_org

September 27, 2012, at 03:50 PM by flaviogoncalves -
Changed line 14 from:

The live classes will be taken online by web-conference two sessions per week, 2-3 hours per session, for 7 weeks. To attend this training you will need to have broadband Internet access. You are going to receive a DVD with 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.

to:

The live classes will be taken online by web-conference two sessions per week, 2-3 hours per session, for 7 weeks. 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.

July 04, 2012, at 05:59 PM by bogdan -
Changed line 9 from:

The OpenSIPS 1.6 eBootcamp is a seven week e-Training program 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 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.

to:

The OpenSIPS 1.8 eBootcamp is a seven week e-Training program 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 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.

July 04, 2012, at 05:46 PM by bogdan -
Changed lines 5-6 from:

Next scheduled training starts at February 15th, 2012. For more details, please send an email to bootcamp_at_opensips_dot_org

to:

Next scheduled training starts at September 10th, 2012. For more details, please send an email to bootcamp_at_opensips_dot_org

Changed line 195 from:

The next eBootcamp will start in February 15th, We accept late registrations until February 7th.

to:

The next eBootcamp will start in September 10th, We accept late registrations until September 3rd.

December 12, 2011, at 06:32 PM by 109.99.2.142 -
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Schedule

December 12, 2011, at 06:32 PM by 109.99.2.142 -
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Schedule

December 12, 2011, at 06:32 PM by 109.99.2.142 -
Changed lines 39-48 from:
  1. .What SIP is and what SIP is for
  2. .SIP Architecture
  3. .SIP Addressing scheme
  4. .SIP Basic call flows
  5. .Sessions, Transactions and Dialogs
  6. .Initial and Sequential Requests
  7. .Media Handling and SDP
  8. .Use case scenarios, VoIP Providers, Wholesale routing, Hosted PBX‘­
  9. .OpenSIPS architecture, advantages and limitations
to:
  1. What SIP is and what SIP is for
  2. SIP Architecture
  3. SIP Addressing scheme
  4. SIP Basic call flows
  5. Sessions, Transactions and Dialogs
  6. Initial and Sequential Requests
  7. Media Handling and SDP
  8. Use case scenarios, VoIP Providers, Wholesale routing, Hosted PBX‘­
  9. OpenSIPS architecture, advantages and limitations
Changed lines 50-57 from:
  1. .Routing Basics and the Standard Configuration
  2. .Scripting Basics
  3. .Routing Basics
  4. .Analyzing the standard configuration files
  5. .LAB OpenSIPS Installation
  6. .LAB Connecting two phones to OpenSIPS
  7. .LAB Running stateful, stateless, with/without record routing
to:
  1. Routing Basics and the Standard Configuration
  2. Scripting Basics
  3. Routing Basics
  4. Analyzing the standard configuration files
  5. LAB OpenSIPS Installation
  6. LAB Connecting two phones to OpenSIPS
  7. LAB Running stateful, stateless, with/without record routing
Changed lines 59-78 from:
  1. .Introduction to OpenSIPS Control Panel
  2. .Domain administration
  3. .User administration
  4. .Interface customization
  5. .The Auth_DB modules
  6. .Register authentication sequence
  7. .Invite authentication sequence
  8. .Digest authentication
  9. .Quality of protection
  10. .Plaintext or hash passwords
  11. .The opensipsctl shell utility
  12. .The opensipsCTL resource file
  13. .Checking From and TO tags
  14. .Multidomain support
  15. .Inter-domain and intra-domain routing
  16. .LAB Installing MySQL Support
  17. .LAB Enhancing the script
  18. .LAB Installing the opensips-cp
  19. .LAB Configuring opensips-cp
to:
  1. Introduction to OpenSIPS Control Panel
  2. Domain administration
  3. User administration
  4. Interface customization
  5. The Auth_DB modules
  6. Register authentication sequence
  7. Invite authentication sequence
  8. Digest authentication
  9. Quality of protection
  10. Plaintext or hash passwords
  11. The opensipsctl shell utility
  12. Checking From and TO tags
  13. Multidomain support
  14. Inter-domain and intra-domain routing
  15. LAB Installing MySQL Support
  16. LAB Enhancing the script
  17. LAB Installing the opensips-cp
  18. LAB Configuring opensips-cp
Changed lines 79-85 from:
  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
to:
  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
Changed lines 87-93 from:
  1. .Using the dialplan module
  2. .Introduction to Drouting
  3. .Drouting tables
  4. .LAB Routing calls to the PSTN
  5. .LAB Using Dynamic Routing tables
  6. .LAB Using the Dialplan module for pre-routing
to:
  1. Using the dialplan module
  2. Introduction to Drouting
  3. Drouting tables
  4. LAB Routing calls to the PSTN
  5. LAB Using Dynamic Routing tables
  6. LAB Using the Dialplan module for pre-routing
Changed lines 95-101 from:
  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
to:
  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
Changed lines 103-113 from:
  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
to:
  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
Changed lines 115-125 from:
  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
to:
  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
Changed lines 127-138 from:
  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
to:
  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
Changed lines 140-150 from:
  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
to:
  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
Changed lines 152-159 from:
  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
to:
  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
Changed lines 161-167 from:
  1. .B2BUA design
  2. .Writing scenarios with B2BUA
  3. .Topology hiding using B2BUA
  4. .Complex B2BUA scenarios
  5. .LAB Using SIP Trace
  6. .LAB Using the B2BUA for topology hiding
to:
  1. B2BUA design
  2. Writing scenarios with B2BUA
  3. Topology hiding using B2BUA
  4. Complex B2BUA scenarios
  5. LAB Using SIP Trace
  6. LAB Using the B2BUA for topology hiding
Changed lines 169-170 from:
  1. .Final work assignment
to:
  1. Final work assignment
Changed line 172 from:
  1. .Q & A on Final work
to:
  1. Q & A on Final work
December 12, 2011, at 06:29 PM by 109.99.2.142 -
Changed lines 9-11 from:

The OpenSIPS 1.6 eBootcamp is a seven week e-Training program 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. 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.

to:

The OpenSIPS 1.6 eBootcamp is a seven week e-Training program 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 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.

Changed line 14 from:

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 DVD with 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.

to:

The live classes will be taken online by web-conference two sessions per week, 2-3 hours per session, for 7 weeks. To attend this training you will need to have broadband Internet access. You are going to receive a DVD with 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.

December 12, 2011, at 06:27 PM by 109.99.2.142 -
Changed lines 5-6 from:

Next scheduled training starts at February 15th, 2012. For more details, please send an email to bootcamp_at_opensips_dot_org

to:

Next scheduled training starts at February 15th, 2012. For more details, please send an email to bootcamp_at_opensips_dot_org

Changed line 38 from:
Week #1 – Tuesday – Live Conference – Introduction to SIP
to:
Week #1 – Session 1 – Live Conference – Introduction to SIP and OpenSIPS
Deleted lines 45-46:
Week #1 – Thursday – Live conference – Introduction to OpenSIPS
Added lines 48-49:
Week #1 – Session 2 – Live conference – OpenSIPS Basics
Changed lines 54-58 from:
  1. .LAB 3.1 OpenSIPS Installation
  2. .LAB 3.2 Connecting two phones to OpenSIPS
  3. .LAB 3.3 Running stateful, stateless, with/without record routing
Week #2 – Tuesday – Live Conference - SQL Databases
to:
  1. .LAB OpenSIPS Installation
  2. .LAB Connecting two phones to OpenSIPS
  3. .LAB Running stateful, stateless, with/without record routing
Week #2 – Session 3 – Live Conference - OpenSIPS Control Panel and SQL Authentication
  1. .Introduction to OpenSIPS Control Panel
  2. .Domain administration
  3. .User administration
  4. .Interface customization
Changed line 67 from:
  1. .QOP ¨C Quality of protection
to:
  1. .Quality of protection
Deleted line 68:
  1. .LAB 4-1 Installing MySQL Support
Changed lines 74-87 from:
  1. .LAB4-2 Enhancing the script
Week #2 – Thursday – Live Conference – OpenSIPS Control Panel
  1. .Lab 4 - Review
  2. .Introduction to OpenSIPS Control Panel
  3. .LAB 5-1 Installing the opensips-cp
  4. .LAB 5-2 Configuring opensips-cp
  5. .Basic tasks
  6. .Domain administration
  7. .User administration
  8. .Interface customization
Week #3 – Tuesday – Live Conference - Connectivity to the PSTN
  1. .Lab 5 - Review
to:
  1. .LAB Installing MySQL Support
  2. .LAB Enhancing the script
  3. .LAB Installing the opensips-cp
  4. .LAB Configuring opensips-cp
Week #2 – Session 4 – Live Conference – PSTN connectivity (part 1)
Added lines 86-88:
Week #3 – Session 5 – Live Conference – PSTN connectivity (part 2)
  1. .Using the dialplan module
Changed lines 91-96 from:
  1. .LAB 6-1 Routing calls to the PSTN
  2. .LAB 6-2 Using Dynamic Routing tables
  3. .LAB 6-3 Using the Dialplan module for pre-routing
Week #3 – Thursday – Live Conference – Advanced SIP Call Flows
to:
  1. .LAB Routing calls to the PSTN
  2. .LAB Using Dynamic Routing tables
  3. .LAB Using the Dialplan module for pre-routing
Week #3 – Session 6 – Live Conference – Advanced SIP Call Flows
Changed lines 101-102 from:
Week # 4 – Tuesday – Live Conference – SIP presence
to:
  1. .LAB: Implementing call fwd with avpops
Week #4 – Session 7 – Live Conference – SIP presence ad HA
Changed lines 109-112 from:
  1. .LAB 8-1 Implementing presence aggregation
  2. .LAB 8-2 Publishing non-SIP Presence
Week # 4 – Thursday – Live Conference – SIP Dialog Awareness
to:
  1. .OpenSIPS High Availability
  2. .Active/Active and Active/Backup setups
  3. .SIP and Data Replication
  4. .LAB Implementing presence aggregation
  5. .LAB Publishing non-SIP Presence
Week #4 – Session 8 – Live Conference – SIP Dialog Awareness and Load Balancing
Deleted lines 119-125:
  1. .LAB 9-1 Limiting the number of concurrent calls
Week #5 – Tuesday – Live Conference – Load Balancing and High Availability
  1. .Lab 8 - Review
  2. .OpenSIPS High Availability
  3. .Active/Active and Active/Backup setups
  4. .SIP and Data Replication
Changed lines 124-127 from:
  1. .LAB 9-1 Load balancing & failover for an Asterisk Cluster
Week #5 – Thursday – Live Conference – SIP NAT Traversal
  1. .Lab 9 - Review
to:
  1. .LAB Limiting the number of concurrent calls
  2. .LAB Load balancing & failover for an Asterisk Cluster
Week #5 – Session 9 – Live Conference – SIP NAT Traversal
Changed line 137 from:
  1. .LAB 9-1 Usind RTPProxy for NAT traversal
to:
  1. .LAB Usind RTPProxy for NAT traversal
Changed line 140 from:
Week #6 Tuesday – Live Conference - OpenSIPS Accounting and Billing
to:
Week #5 – Session 10 – Live Conference - Accounting & Billing, Monitoring Tools
Deleted line 141:
  1. .LAB 12-1 Accounting to a MySQL database
Deleted lines 143-154:
  1. .LAB 12-2 Prepaid users, limiting the duration of the call using the Dialog module
Week #6 Thursday – Live Conference – 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
Week #7 – Tuesday – Advanced Topics
Added lines 147-163:
  1. .Generating traffic with SIPP
  2. .LAB Accounting to a MySQL database
  3. .LAB Prepaid users, limiting the duration of the call using the Dialog module
  4. .LAB Using SIP Trace
Week #6 – Session 11 – Live Conference – 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
Week #6 – Session 12 – Back2Back User Agent
  1. .B2BUA design
  2. .Writing scenarios with B2BUA
Changed lines 165-173 from:
  1. .LAB 12-1 ¨C Using SIP Trace
  2. .LAB 13-1 ¨C Using the B2BUA for topology hiding
Week #7 – Thursday – Final Work – Live Conference
  1. .Lab 12 - Review
  2. .Q&A on Final work assignment
to:
  1. .Complex B2BUA scenarios
  2. .LAB Using SIP Trace
  3. .LAB Using the B2BUA for topology hiding
Week #7 – Session 13 – Live Conference – Final Work
  1. .Final work assignment
Week #7 – Session 14 – Live Conference – Final Work
  1. .Q & A on Final work
Changed lines 195-197 from:

2011 Schedule

The next eBootcamp will start in September 19th, We accept late registrations until September 12th.

to:

2012 Schedule

The next eBootcamp will start in February 15th, We accept late registrations until February 7th.

Changed line 199 from:

bootcamp@opensips.org

to:

bootcamp at opensips dot org

December 12, 2011, at 06:03 PM by 109.99.2.142 -
Changed lines 5-6 from:

Next scheduled training starts at September 19th, 2011. For more details, please send an email to bootcamp_at_opensips_dot_org

to:

Next scheduled training starts at February 15th, 2012. For more details, please send an email to bootcamp_at_opensips_dot_org

Changed lines 14-15 from:

The live classes will be taken online by web-conference every Tuesday and Thursday 03:00PM GMT, 11:00AM EDT 08:00AM, PDT. To attend this training you will need to have broadband Internet access. You are going to receive a DVD with 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.

to:

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 DVD with 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.

Changed lines 29-30 from:
  • Account Calls to MySQL
  • How to use test and monitoring tools to check your configuration
to:
  • 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
July 28, 2011, at 12:27 PM by 109.99.2.142 -
Changed lines 5-6 from:

Next scheduled training starts at May 2nd. For more details, please send an email to bootcamp_at_opensips.org

to:

Next scheduled training starts at September 19th, 2011. For more details, please send an email to bootcamp_at_opensips_dot_org

Changed line 194 from:

The next eBootcamp will start in May 2nd, We accept late registrations until April 23

to:

The next eBootcamp will start in September 19th, We accept late registrations until September 12th.

April 10, 2011, at 07:08 PM by flaviogoncalves -
Changed line 5 from:

Next scheduled trainind starts at May 2nd. For more details, please send an email to bootcamp_at_opensips.org

to:

Next scheduled training starts at May 2nd. For more details, please send an email to bootcamp_at_opensips.org

April 10, 2011, at 07:08 PM by flaviogoncalves -
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Overview4

to:

Overview

April 10, 2011, at 07:08 PM by flaviogoncalves -
Changed lines 5-6 from:

Overview

to:

Next scheduled trainind starts at May 2nd. For more details, please send an email to bootcamp_at_opensips.org

Overview4

Changed line 194 from:

The eBootcamp will start at February – 28, We accept late registration until February 21

to:

The next eBootcamp will start in May 2nd, We accept late registrations until April 23

January 17, 2011, at 03:41 PM by bogdan -
Changed line 187 from:
  • Bogdan-Andrei Iancu – OpenSIPS founder and main developer. Also CEO of Voice System, an “know-how” OpenSIPS company.
to:
  • Bogdan-Andrei Iancu – OpenSIPS founder and main developer, an “know-how” OpenSIPS person.
January 17, 2011, at 03:39 PM by bogdan -
Changed lines 37-41 from:
  1. .SIP Components
  2. .SIP Requests and SIP Responses
  3. .Main SIP headers
  4. .Transactions and Dialogs
to:
  1. .SIP Addressing scheme
  2. .SIP Basic call flows
  3. .Sessions, Transactions and Dialogs
  4. .Initial and Sequential Requests
  5. .Media Handling and SDP
Changed lines 44-64 from:
  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
Week #2 – Tuesday – Live Conference - 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
Week #2 – Thursday – Live Conference – Routing Basics
  1. .Lab 2 Review
to:
  1. .Use case scenarios, VoIP Providers, Wholesale routing, Hosted PBX‘­
  2. .OpenSIPS architecture, advantages and limitations
  3. .Routing Basics and the Standard Configuration
Changed lines 49-56 from:
  1. .Analysing the standard configuration files
  2. .LAB 3.1 Connecting two phones to OpenSIPS
  3. .LAB 3.2 Running stateful with record routing (packet capture)
  4. .LAB 3.3 Running stateless with record routing (packet capture)
  5. .LAB 3.4 Running stateless with no record-routing
Week #3 – Tuesday – Live Conference - Authentication
  1. .LAB 3 Review
to:
  1. .Analyzing the standard configuration files
  2. .LAB 3.1 OpenSIPS Installation
  3. .LAB 3.2 Connecting two phones to OpenSIPS
  4. .LAB 3.3 Running stateful, stateless, with/without record routing
Week #2 – Tuesday – Live Conference - SQL Databases
Changed line 59 from:
  1. .QOP – Quality of protection
to:
  1. .QOP ¨C Quality of protection
Changed line 63 from:
  1. .The opensipsctl resource file
to:
  1. .The opensipsCTL resource file
Changed line 65 from:
  1. .Multi-domain support
to:
  1. .Multidomain support
Changed lines 68-70 from:
  1. .Optional LAB – Authentication using TLS
Week #3 – Thursday – Live Conference – OpenSIPS Control Panel
to:
Week #2 – Thursday – Live Conference – OpenSIPS Control Panel
Changed line 79 from:
Week # 4 – Tuesday – Live Conference – Connectivity to the PSTN
to:
Week #3 – Tuesday – Live Conference - Connectivity to the PSTN
Changed lines 92-105 from:
  1. .Inter-domain Peering
Week # 4 – Thursday – Live Conference – Media Server Integration and Presence
  1. .Lab 6 - Review
  2. .Introduction to Call Forwarding
  3. .Pseudo-variables and AVPs
  4. .AVP functions
  5. .Implementing Blind Call Forwarding
  6. .Busy or Unanswered forwarding to Voice Mail
  7. .LAB 7-1 Testing the Call Forwarding feature
  8. .LAB 7-2 Asterisk Integration (optional)
  9. .Lab 7 - Review
  10. .SIP presence overview
to:
Week #3 – Thursday – Live Conference – 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
Week # 4 – Tuesday – Live Conference – SIP presence
Added lines 110-116:
Week # 4 – Thursday – Live Conference – 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
Changed lines 126-127 from:
  1. .LAB 9-1 Load balancing & failover foran Asterisk Cluster
to:
  1. .LAB 9-1 Load balancing & failover for an Asterisk Cluster
Deleted line 142:
  1. .Lab 10 - Review
Changed lines 144-159 from:
  1. .LAB 10-1 Accounting to a MySQL database
  2. .Accounting using a RADIUS server
  3. .LAB 10-2 Accounting to a Radius Server
Week #6 Thursday – Live Conference – Troubleshooting Tools
  1. .Lab 11 - Review
  2. .Built in tools
  3. .Packet Capture and Trace Tools
  4. .The SIPTRACE module
  5. .Predefined and Custom Statistics
  6. .Stress Testing Tools
  7. .LAB 12-1 – Using SIP Trace
  8. .LAB 12-2 Using sipp to stress test OpenSIPS
  9. .Final Work Assignment – Build a telephony system based on specifications
Week #7 – Tuesday – Final Work – Live Conference
to:
  1. .LAB 12-1 Accounting to a MySQL database
  2. .Generating CDRs using the ACC and Dialog Modules for postpaid users.
  3. .Session Timeout, integration with RTPProxy
  4. .LAB 12-2 Prepaid users, limiting the duration of the call using the Dialog module
Week #6 Thursday – Live Conference – 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
Week #7 – Tuesday – 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
Week #7 – Thursday – Final Work – Live Conference
Deleted lines 169-173:
Week #7 – Thursday – Certification Testing - (Optional)
  1. .Last date to send the scripts for the final work
  2. .Q&A on Final work assignment
  3. .Q&A on the Certification Testing
  4. .Certification Test
January 17, 2011, at 02:54 PM by bogdan -
Changed lines 207-208 from:

2010 Schedule

The eBootcamp will start at August – 31, We accept late registration until August 16

to:

2011 Schedule

The eBootcamp will start at February – 28, We accept late registration until February 21

July 13, 2010, at 08:06 PM by flaviogoncalves -
Changed line 189 from:

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

to:
  • VoIP providers seeking “Open Source” platforms to enhance their businesses
July 13, 2010, at 07:52 PM by bogdan -
Added lines 32-33:

Schedule

Deleted lines 187-190:
Deleted lines 200-201:

Course Schedule

Please check the PDF - detailed description of the course

July 13, 2010, at 07:49 PM by bogdan -
Changed line 32 from:

Week #1 – Tuesday – Live Conference – Introduction to SIP

to:
Week #1 – Tuesday – Live Conference – Introduction to SIP
Changed line 40 from:

Week #1 – Thursday – Live conference – Introduction to OpenSIPS

to:
Week #1 – Thursday – Live conference – Introduction to OpenSIPS
Changed line 50 from:

Week #2 – Tuesday – Live Conference - OpenSIPS Installation

to:
Week #2 – Tuesday – Live Conference - OpenSIPS Installation
Changed line 60 from:

Week #2 – Thursday – Live Conference – Routing Basics

to:
Week #2 – Thursday – Live Conference – Routing Basics
Changed line 70 from:

Week #3 – Tuesday – Live Conference - Authentication

to:
Week #3 – Tuesday – Live Conference - Authentication
Changed line 87 from:

Week #3 – Thursday – Live Conference – OpenSIPS Control Panel

to:
Week #3 – Thursday – Live Conference – OpenSIPS Control Panel
Changed line 97 from:

Week # 4 – Tuesday – Live Conference – Connectivity to the PSTN

to:
Week # 4 – Tuesday – Live Conference – Connectivity to the PSTN
Changed line 112 from:

Week # 4 – Thursday – Live Conference – Media Server Integration and Presence

to:
Week # 4 – Thursday – Live Conference – Media Server Integration and Presence
Changed line 132 from:

Week #5 – Tuesday – Live Conference – Load Balancing and High Availability

to:
Week #5 – Tuesday – Live Conference – Load Balancing and High Availability
Changed line 143 from:

Week #5 – Thursday – Live Conference – SIP NAT Traversal

to:
Week #5 – Thursday – Live Conference – SIP NAT Traversal
Changed line 157 from:

Week #6 Tuesday – Live Conference - OpenSIPS Accounting and Billing

to:
Week #6 Tuesday – Live Conference - OpenSIPS Accounting and Billing
Changed line 164 from:

Week #6 Thursday – Live Conference – Troubleshooting Tools

to:
Week #6 Thursday – Live Conference – Troubleshooting Tools
Changed line 175 from:

Week #7 – Tuesday – Final Work – Live Conference

to:
Week #7 – Tuesday – Final Work – Live Conference
Changed line 179 from:

Week #7 – Thursday – Certification Testing - (Optional)

to:
Week #7 – Thursday – Certification Testing - (Optional)
July 13, 2010, at 07:48 PM by bogdan -
Changed lines 32-143 from:
  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
to:

Week #1 – Tuesday – Live Conference – Introduction to SIP

  1. .What SIP is and what SIP is for
  2. .SIP Architecture
  3. .SIP Components
  4. .SIP Requests and SIP Responses
  5. .Main SIP headers
  6. .Transactions and Dialogs

Week #1 – Thursday – Live conference – 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

Week #2 – Tuesday – Live Conference - 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

Week #2 – Thursday – Live Conference – Routing Basics

  1. .Lab 2 Review
  2. .Scripting Basics
  3. .Routing Basics
  4. .Analysing the standard configuration files
  5. .LAB 3.1 Connecting two phones to OpenSIPS
  6. .LAB 3.2 Running stateful with record routing (packet capture)
  7. .LAB 3.3 Running stateless with record routing (packet capture)
  8. .LAB 3.4 Running stateless with no record-routing

Week #3 – Tuesday – Live Conference - Authentication

  1. .LAB 3 Review
  2. .The Auth_DB modules
  3. .Register authentication sequence
  4. .Invite authentication sequence
  5. .Digest authentication
  6. .QOP – Quality of protection
  7. .Plaintext or hash passwords
  8. .LAB 4-1 Installing MySQL Support
  9. .The opensipsctl shell utility
  10. .The opensipsctl resource file
  11. .Checking From and TO tags
  12. .Multi-domain support
  13. .Inter-domain and intra-domain routing
  14. .LAB4-2 Enhancing the script
  15. .Optional LAB – Authentication using TLS

Week #3 – Thursday – Live Conference – OpenSIPS Control Panel

  1. .Lab 4 - Review
  2. .Introduction to OpenSIPS Control Panel
  3. .LAB 5-1 Installing the opensips-cp
  4. .LAB 5-2 Configuring opensips-cp
  5. .Basic tasks
  6. .Domain administration
  7. .User administration
  8. .Interface customization

Week # 4 – Tuesday – Live Conference – Connectivity to the PSTN

  1. .Lab 5 - Review
  2. .Introduction to PSTN routing
  3. .Accepting calls from the PSTN
  4. .The permissions module and the check_source_address() function
  5. .Routing a call to the PSTN
  6. .DID redirection using Aliases
  7. .ACL and Group permissions
  8. .Introduction to Drouting
  9. .Drouting tables
  10. .LAB 6-1 Routing calls to the PSTN
  11. .LAB 6-2 Using Dynamic Routing tables
  12. .LAB 6-3 Using the Dialplan module for pre-routing
  13. .Inter-domain Peering

Week # 4 – Thursday – Live Conference – Media Server Integration and Presence

  1. .Lab 6 - Review
  2. .Introduction to Call Forwarding
  3. .Pseudo-variables and AVPs
  4. .AVP functions
  5. .Implementing Blind Call Forwarding
  6. .Busy or Unanswered forwarding to Voice Mail
  7. .LAB 7-1 Testing the Call Forwarding feature
  8. .LAB 7-2 Asterisk Integration (optional)
  9. .Lab 7 - Review
  10. .SIP presence overview
  11. .Presence Agent setup
  12. .Publishing Presence from non-SIP devices
  13. .Registration-to-Presence conversion (old SIP devices)
  14. .Scalability of the presence model
  15. .Aggregation of the presence information
  16. .LAB 8-1 Implementing presence aggregation
  17. .LAB 8-2 Publishing non-SIP Presence

Week #5 – Tuesday – Live Conference – Load Balancing and High Availability

  1. .Lab 8 - Review
  2. .OpenSIPS High Availability
  3. .Active/Active and Active/Backup setups
  4. .SIP and Data Replication
  5. .OpenSIPS Load balancing/Dispatching Capabilities
  6. .Balancing algorithms
  7. .Balancing and failover
  8. .Multiple groups of balancing
  9. .LAB 9-1 Load balancing & failover foran Asterisk Cluster

Week #5 – Thursday – Live Conference – SIP NAT Traversal

  1. .Lab 9 - Review
  2. .NAT Types
  3. .Solving the NAT traversal challenge
  4. .Implementing a far end NAT solution
  5. .RFC3581 and forc_rport() function
  6. .Solving the traversal of RTP packets
  7. .Handling Register Requests
  8. .Detecting clients Behind NAT
  9. .Handling Invite requests behind NAT
  10. .RTPProxy installation and configuration
  11. .LAB 9-1 Usind RTPProxy for NAT traversal
  12. .STUN – Simple Traversal of UDP NAT

Week #6 Tuesday – Live Conference - OpenSIPS Accounting and Billing

  1. .Lab 10 - Review
  2. .Authentication, Accounting and Authorization
  3. .LAB 10-1 Accounting to a MySQL database
  4. .Accounting using a RADIUS server
  5. .LAB 10-2 Accounting to a Radius Server

Week #6 Thursday – Live Conference – Troubleshooting Tools

  1. .Lab 11 - Review
  2. .Built in tools
  3. .Packet Capture and Trace Tools
  4. .The SIPTRACE module
  5. .Predefined and Custom Statistics
  6. .Stress Testing Tools
  7. .LAB 12-1 – Using SIP Trace
  8. .LAB 12-2 Using sipp to stress test OpenSIPS
  9. .Final Work Assignment – Build a telephony system based on specifications

Week #7 – Tuesday – Final Work – Live Conference

  1. .Lab 12 - Review
  2. .Q&A on Final work assignment

Week #7 – Thursday – Certification Testing - (Optional)

  1. .Last date to send the scripts for the final work
  2. .Q&A on Final work assignment
  3. .Q&A on the Certification Testing
  4. .Certification Test
July 13, 2010, at 07:41 PM by bogdan -
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The OpenSIPS 1.6 eBootcamp is a seven week e-Training program 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. 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.

to:

The OpenSIPS 1.6 eBootcamp is a seven week e-Training program 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. 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.

Changed lines 12-13 from:

The live classes will be taken online by web-conference every Tuesday and Thursday 03:00PM GMT, 11:00AM EDT 08:00AM, PDT.To attend this training you will need to have broadband Internet access. The labs will be available in a Computing Grid,with public access to the Internet. You will have a hundred hours of access to the grid to build your own labs. Time extensions will be available as needed. We suggest that you have at least one IP Phone in your private labs to complete the training. A LMS (Learning management system) will be available with forums, quizzes and support materials.

to:

The live classes will be taken online by web-conference every Tuesday and Thursday 03:00PM GMT, 11:00AM EDT 08:00AM, PDT. To attend this training you will need to have broadband Internet access. You are going to receive a DVD with 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.

July 13, 2010, at 07:38 PM by bogdan -
July 13, 2010, at 06:51 PM by bogdan -
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The detailed description of the course is available for download.

to:
July 13, 2010, at 06:50 PM by bogdan -
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The detailed description of the course is available for download.

July 13, 2010, at 06:49 PM by bogdan -
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Overview

to:

Overview

The OpenSIPS 1.6 eBootcamp is a seven week e-Training program 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. 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.

How does it work?

The live classes will be taken online by web-conference every Tuesday and Thursday 03:00PM GMT, 11:00AM EDT 08:00AM, PDT.To attend this training you will need to have broadband Internet access. The labs will be available in a Computing Grid,with public access to the Internet. You will have a hundred hours of access to the grid to build your own labs. Time extensions will be available as needed. We suggest that you have at least one IP Phone in your private labs to complete the training. A LMS (Learning management system) will be available with forums, quizzes and support materials.

The detailed description of the course is available for download.

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

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

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

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)

Course Schedule

Please check the PDF - detailed description of the course

Instructors

  • Bogdan-Andrei Iancu – OpenSIPS Solutions / OpenSIPS founder and main developer.
  • Flαvio 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.

2010 Schedule

The eBootcamp will start at August – 31, We accept late registration until August 16

Contact & Registration

bootcamp@opensips.org

July 13, 2010, at 06:29 PM by bogdan -
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Training -> eBootcamp


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Overview


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