OpenSIPS advanced training

@ Cluecon 2017

Using all the tools in the OpenSIPS ecosystem to improve, expand and monitor your FreeSWITCH installations and Telephony platforms


The concept of this training course is to spend the day building from the ground up a fully operational VoIP system consisting of an OpenSIPS front-ending a cluster of FreeSWITCH: PBXes:

  • Registration throttling and aggregation with OpenSIPS acting as a mid-registrar
  • OpenSIPS front-end taking over the traffic authentication and authorization
  • Performing Dynamic and Realtime Balancing over the traffic to the FreeSWITCH servers (with failover)
  • Routing outbound traffic via OpenSIPS for LCR and carrier interfacing
  • Realtime traffic capturing and monitoring with Homer
  • Implementing centralized Presence/BLF support in OpenSIPS front-end

Training Attendees will be provided with virtual machines/containers to follow along and get a hands on experience building the final functional product, plus access to a video library of “OpenSIPS basics” prior the event study fundamentals or brush up on their knowledge

An official OpenSIPS "Certificate of Attendance" will be provided for succesfully completing the OpenSIPS and FreeSWITCH integration training.

For the Virtual Machines/containers, an in-class cloud will be available - each attendend will have his own hosted enviroment in our local cloud. A laptop is required only for SSH and WEB access to the training environment, nothing more and nothing simpler. At the end of the class, the student may copy their Virtual Machine image for further self study.

From start to finish, each phase will be lead by a different instructor who specializes in that particular component. The idea being that the same system used in the first phase will be built upon during the ones that follow.

Bogdan is the OpenSIPS project founder with an experience of 15+ year in the SIP world. Practicing the symbioses between managing the Open Source project and building commercial products around OpenSIPS, gives the best results in producing a viable SIP Server software for the read-life needs.
OpenSIPS developer for 4 years, have both developed a series of modules (Sangoma transcoding, REST client, math operations) and done extensive troubleshooting / improvements in both the project´s critical areas of code (i.e. SIP transactions, SIP dialogs, TCP and UDP processing and scaling) and its scripting language. Also experienced with troubleshooting SIP / VoIP and glue scripting in bash 4.0+ and Python 2.
OpenSIPS maintainer and developer, involved in both design and development of new modules, as well as core functions
He is employed as Senior Voice Engineer for QSC AG, one of the major German voice and data providers. Alexandr holds a diploma in physics of Odessa State University. He has 20 years of experience in telecommunication techniques and has contributed to many OpenSource projects like FeeeSwitch, SER, Kamailio, SEMS, Asterisk, SIPP, Wireshark. Alexandr is the main developer of Homer SIP Capture project. He is also founder of IRC RusNet Network, one of the biggest national IRC networks in the world.
He is founder of Amsterdam based QXIP BV, Co-Founder and Developer of HOMER / SIPCAPTURE Project and voice specialist of the NTOP Team. Formerly a Sound Engineer, Lorenzo has been deeply involved with telecommunications and VoIP for well over a decade and has contributed ideas, design concepts and code to many voice-related Open-Source and commercial projects specializing in active and passive monitoring solutions with his team. Currently he is Sr. Voice Engineer and Designer for the largest international cable operator worldwide.


Registration to the training is just $299 (per person)

or use your discount code if you have already purchased a Cluecon Ticket.

Event Location

OpenSIPS training @ Cluecon 2017

Swissotel Chicago
323 E Upper Wacker Dr
Chicago, IL 60601
Register Online