The Asterisk SIP Masterclass

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

SIP is the IETF standard for IP realtime communication over private networks and the Internet. It’s a 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 Kamailio – the Open Source SIP server – 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 Asterisk developer and consultant  Olle E. Johansson and the Kamailio 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/Kamailio 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 and Kamailio in enterprise, public sector 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 as a consultant in larger installationsl and have participated in several international SIP interoperability test events with Asterisk.

Daniel-Constantin Mierla co-founded kamailio.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
• 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

If you have questions or suggestions about this class, please do not hesitate to contact us. 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

• This class is now only used for in-house training classes. There will be no more public classes as it is replaced by a new class during the fall of 2012.