The Asterisk SIP Master Class!
5 days of SIP labs that will turn you into a SIP master!

Asterisk™ is becoming more popular each day, and is rapidly becoming a very important part in a SIP infrastructure. SIP is the IETF standard for IP based telephony over private networks and the Internet. It's the IP realtime communications platform with support for many applications beside audio and video telephony - Instant messaging, whiteboarding, games...

Asterisk is not a SIP proxy, but works alongside with the proxy and adds a lot of features to the SIP network - voicemail, PSTN gateways, follow-me scripts and application integration to name a few. If you are building a larger SIP network, then make sure you build it in a long term way, supporting future SIP applications.

This class is focusing on Asterisk in a SIP network. With a combination of theory and practical labs, you will learn how to setup and configure Asterisk and OpenSER - the Open Source SIP proxy - in a scalable enterprise or service provider network. We will go through various kinds of setups and give you insight in the design of real SIP networks with Asterisk running in enterprise and service provider networks. By meeting the maintainer of the Asterisk SIP channel, Olle Johansson and the OpenSER developer Daniel-Constantin Mierla, you will also get a lot of insight into current and future features, bugs and implementation details in a way that's hard to get otherwise.

Target group: Technical people with experience of Asterisk, working in large enterprises or with Internet Telephony Service providers.

You need to be comfortable with Linux and Asterisk before registering for this class.

Teachers - a professional SIP/VoIP consultant with many years of Asterisk and SER/OpenSER knowledge

The teachers are Olle E. Johansson .and Daniel-Constantin Mierla.

Olle E. Johansson is one of the Asterisk developers with many years of experience in teaching networking. Olle is an active Asterisk developer with many years of experience of running Asterisk in enterprise and service provider networks. Olle has been teaching Asterisk since early 2005 and have previously taught many classes in networking, IP, IP security, LDAP, XML and other topics.

Olle currently works on enhancing the Asterisk SIP channel and have participated in two international SIP interoperability test events with Asterisk. He's also a member of the Digium Asterisk Advisory Board.

Daniel-Constantin Mierla co-founded OpenSER.org, a scalable and flexible open source SIP server, being also core developer of SIP Express Router (SER) from its early beginning in 2002. He has a Master degree in Computer Science and Engineering from the Polytechnics University of Bucharest. His experience was accumulated working as consultant for Orange Romania, branch of French Orange mobile operator, and researcher in network communications at FOKUS Fraunhofer Institute, Berlin, Germany.

The Asterisk SIP Masterclass - Overview

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
• OpenSER - SIP express router
  A quick introduction to the SIP proxy from openser.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 OpenSER servers communicating with each other and delivering services to the network.
• SIP test tools
  A brief introduction to various SIP test tools and their usage

If you have questions or suggestions about this class, please do not hesitate to contact us. As this class is still being in construction, we reserve the right to modify the agenda, even though we won't change the generic content and goal for this class.

Details about this training

• Price: 3,200 EURO, payment in advance by bank wire (You will get an invoice with instructions)
  The price includes training documentation, lunch and refreshments.
  For students within the European Union, Swedish VAT is added unless we get your companies VAT registration ID at registration.
  We do not give refunds on late cancellations or no-shows. Cancellation has to be sent to us at least one week in advance.
• Times: The training starts at 10 AM on Monday, ends 3 PM on Friday
  Tuesday-thursday 9 AM to 5 PM.
• Location: Address of our training centers and hotel recommendations is found here
• Each student will have access to a desktop computer. Feel free to bring your own laptop and SIP equipment to the training. Make sure that all equipment is insured by you or your company.