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SIP Masterclass training

Madrid, date to be confirmed.

For the first time in Madrid, Avanzada7 proudly presents, in collaboration with its training partner Edvina, SIP Masterclass training.

This course is targeted  for VoIP engineers who already have extensive experience with Asteriskand wish to expand their knowledge, combining Asterisk with Kamailio and their ability to getthe perfect solution for VoIP operators and large corporations with more than 1000 extensions

This 1 week intensive training is taught in English by 2 gurus and major projectdevelopers of Asterisk and Kamailio (OpenSER) and requires prior knowledge of Linux, networking and Asterisk (min dCAP level) for an optimal learning.

The teacher are:


Olle E. Johansson and Daniel-Constantin Mierla.

The ultimate objectives of the course are to learn from the 1st hand of the programmers,how  Asterisk source code is structured, specially regarding the SIP protocol. Identify communication frames, analysis tools, integration with other supported platforms, etc.

Training price is 3200Eur + VAT and includes breakfast and lunch during all days of the week. It´s possible to subsidize part of the cost with Fundación Tripartita.


Course schedule is as follows:

Day Monday Tuesday Wednesday Thursday Friday
Block 1 Introduction to Asterisk The Asterisk SIP channel SIP transfers Asterisk, SIP and Video LAB:Building a failover SIP network
Block 2 Asterisk overview SIP debugging Presence, IM
Block 3 Asterisk overview
Lab: Setting up Asterisk
Kamailio – SIp express router Asterisk SIP channel advanced features Asterisk SIP realtime
Block 4 Asterisk NAT support Lab: Setting up Kamailio Lab: Asterisk and Kamailio LAB: Asterisk SIP realtime

What you will learn in this class:

  • Asterisk basics – a recap
    • A quick update on Asterisk on a technical level, the core design, channel architecture, codecs, formats and various modules
  • SIP – an introduction to the protocol
    • An introduction to the SIP protocol. Design ideas, basics, methods, transactions and call features. SDP – the Session Initiation Protocol and RTP, the Real Time Protocol is also covered.
  • SIP proxys and network infrastructure
    • What’s the role of a SIP proxy, SIP location server, SIP registrar? What’s the relationship between a user agent (phone) and the server infrastructure?
  • The Asterisk SIP channel – introduction
    • An introduction to the Asterisk SIP implementation – what is supported? Adding phones, implementing voicemail, subscriptions, connecting to service providers, working with outbound proxies and NATs.
  • Traversing firewalls and NAT devices
  • Kamailio – SIP express router
    • A quick introduction to the SIP proxy from kamailio.org. Design ideas, modules, concepts, configuration.
  • SIP phones and ATAs for audio and video
    • An overview of various devices and their functionality
  • SIP presence and Instant messaging
    • How to integrate new SIP features in an Asterisk/SER network
  • Building a SIP network with Asterisk and SIP proxys
    • An extensive lab session where we build a SIP network with clients behind NAT and on the same network, with Asterisk and Kamailio servers communicating with each other and delivering services to the network.
  • SIP test tools and debugging
    • A brief introduction to various SIP test tools and their usage