Daniel-Constantin Mierla of Asipto, co-founder of Kamailio SIP Server project, will provide Kamailio Developer Seminar for one day and a half, Oct 24-25, 2011,  in San Mateo, south of San Francisco, California.

Among topics to be approached:

• internal architecture of Kamailio/SER SIP server
• SIP parser
• memory manager
• locking manager
• database API
• configuration file language structure and interpreter
• RPC/MI control interface
• pseudo-variables and transformations framework
• module interface
• how to write your own extensions in C as module
• inter-module APIs – transaction management, SIP SIMPLE Presence, asynchronous processing, a.s.o.
• working with embedded interpreters for high level programming languages: Lua, Perl, Python

Who should attend:

• VoIP/Telecom developers intending to write own extensions to Kamailio SIP Server
• VoIP/Telecom administrators interested to learn about the internals of Kamailio SIP Server in order to know how to optimize the SIP routing and build proper configuration file
• VoIP/Telecom professionals interested to learn about Kamailio SIP Server, how and where can it be used, its current features and future development

The event is a good place for networking with other professionals from VoIP/Telecom and Real Time Communications fields, past similar events having dozens of attendees.

The price per attendee is 250 USD.

Number of seats is limited and access will be granted in first come first served fashion.

Registration or requests for more details can be done via:

• registration web form – click here for it
• or email to: registration [at] kamailio.org .

Next Kamailio Advanced Training will take place in Berlin, Germany, Oct 10-13, 2011.

Last stable series is 3.1.x (Oct 06, 2010, see release notes), continues the work done within SIP-Router.org project. Among brand new features in v3.1.0, starting with the previous major version, 3.0.0, you can run mixed Kamailio (OpenSER) and SIP Express Router (SER) modules in the same SIP server instance, giving you the most powerful tools to build stable, very performant and features rich VoIP and Unified Communication platforms.

The class is organized by Asipto and will be taught by Daniel-Constantin Mierla, founder and core developer of Kamailio SIP Server project.

Read more details about the class and registration at:

• http://www.asipto.com/index.php/kamailio-advanced-training/

Daniel-Constantin Mierla of Asipto, founder and core developer of Kamailio (OpenSER), will present at Cluecon conference in Chicago, USA, Aug 9-11, 2011.

His speech focuses on how to build strong security for large VoIP operators, a topic where Asipto has broad expertise accumulated over the past 10 years. Some of our experiences were already shared via knowledge base site.

If you attend the conference or you are in Chicago area during these days and want to meet with Daniel, do not hesitate to contact us.

In the past few weeks, there were couple of articles showing alternatives for Skype (like this one) – applications or/and services.

Why not running your own Skype-like service? Even better, why not do it using open source and free applications and spend 8.5 billions for something else?

The target is to build your own communication service that offers:

• peer-to-peer and secure communication (encryption of the content sent between users)
• voice calls between two users
• video calls between two users
• group voice calls (audio conferencing)
• screen sharing
• chatting – instant messaging between users
• contacts list and presence status notifications

The service is going to use the open standard protocol – SIP (Session Initiation Protocol). Two open source application are playing the major roles in the network:

• Kamailio SIP Server – for user authentication server and ‘super-node’ functionality of relaying the voice/video packets when necessary
• Jitsi – the client application, a cross platform implementation running on Linux, Mac OS X and Windows

Installation of the SIP server is exemplified on a Debian Squeeze system, but any flavour of Linux can be used.

The step-by-step tutorial is available at:

• http://asipto.com/u/2h

It took me about 15 minutes to get the service up and running following the guidelines presented (being very familiar with the two applications), however it should not take more than 1 hour to get everything in action.

Enjoy! Communicate secure and freely!

March 1, 2011 – Berlin, Germany and Vienna, Austria

In the current era of a rapidly changing telecommunication world, Asipto and Sipwise are pleased to announce the merge of their IP telephony system offerings in order to strengthen the position in the market and to consolidate the development of their unified communication solutions.

With Asipto’s know-how and role in the SIP routing development process, and the experience of Sipwise in integrating and operating highly available communication systems based on Sipwise carrier grade communications platforms portfolio, the newly formed joint-venture positions itself as an extremely competitive vendor in the IP telephony market. This is a solid base for further growth and adoption of new technologies to become the reference in Open Source IP telecommunication industry.

Daniel-Constantin Mierla of Asipto, co-founder Kamailio Project, says: “Asipto and Sipwise were successful partners in the past years, building strong confidence between us. Our successful collaboration was not limited to the business side only, as Sipwise is a relevant contributor to the Open Source Kamailio project. Having a strong background in research, Asipto’s lines of products focused a lot on novelty in communications. The evolution of the market demands it more than ever and we will continue to do that. In order to accommodate the SLA requirements for operators and be able to professionally handle the increased demand for Kamailio based solutions, the new joint venture with Sipwise is the perfect option.

As a long time Open Source advocate, I am glad that we can offer a free version of our newly branded product: the sip:provider Community Edition. Small and medium sized operators can start their voice services without any commercial constraints. With my new role as Director of Innovations at Sipwise, I am eager to launch the next release of sip:provider CE and Pro in the near future, to include features such as secure communication over TLS, IPv6, rich presence services and IMS extensions.”

Atilla Ceylan, co-founder and CMO of Sipwise: “Decisions involving the core of a carrier’s network are not made every day, and service providers are placing their confidence in Sipwise’s vision and capabilities to provide them with industry-leading solutions today and over the years to come. Sipwise is currently engaged in projects with Europe’s leading cable operators supporting the conversion of their MGCP based voice infrastructure to a more flexible, more reliable and more economic SIP based Class 5 softswitch platform.

The Joint Venture with Asipto improves our business model and the business case for shifting investment from legacy equipment to now even more compelling next generation solutions from Sipwise. As operators select the partners who will usher them into the future very carefully, I believe that the joint approach of Asipto and Sipwise will provide the best choice by leading the change in voice infrastructure solutions based on Open Source for the new public networks.

The launch of our first jointly developed products – the sip:provider CE and PRO – have been architected with the global market in mind. With operators around the world embracing the transition to reliable, scalable and affordable solutions, our platforms dramatically lower the cost structure and enable the delivery of enhanced services that provide a competitive advantage.”

Based in Berlin, Asipto is led by co-founders and core developers of Kamailio SIP Server project (former OpenSER). The team built around Daniel-Constantin Mierla and Elena-Ramona Modroiu worked with SIP-based telephony since the early times of this protocol. They designed and deployed unified communication solutions for large VoIP providers around the world, with the core of the systems being the Open Source SIP server Kamailio.

Sipwise, a system development and integration company located in Vienna, builds IP telephony appliances based on Open Source technologies, using Kamailio as the core SIP routing engine. Targeting residential, mobile and carrier services, Sipwise products are deployed at operators world-wide, accompanied with long term support and professional SLAs.

Existing customers of Asipto and Sipwise, as well as related projects, will not be directly affected by the merge, existing contracts and collaboration will continue unchanged. The new telephony products will be developed and commercialized under the Sipwise brand. The Asipto brand will continue with a focus on services like trainings, development and consultancy for Kamailio and cutting-edge SIP services.

The new venture will operate in both locations Berlin and Vienna – whenever you come around, let us know and we will be happy to meet and show you our latest products and services.

Contact:
Asipto
Elena-Ramona Modroiu
Phone: +49 30 21480730
Web: http://www.asipto.com
Email: office@asipto.com

Sipwise
Atilla Ceylan
Phone: +43 1 2521522
Web: http://www.sipwise.com
Email: office@sipwise.com

UC Expo 2011 takes place in London, UK, between March 08-09, 2011. Asipto representatives will be present this year as well at the event, meeting many of our UK customers base that will exhibit at the show.

UC Expo describes itself as the show mirroring the diversity of Unified Communications by bringing together all the key technologies and key people of this rapidly evolving world.

If you want to meet with Daniel-Constantin Mierla of Asipto, co-founder and core developer of Kamailio SIP Server project, feel free to contact us:

• Go to Contact Page

CeBIT 2011, the biggest digital show, takes place in Hanover, Germany, March 01 – 05, 2011. Daniel-Constantin Mierla of Asipto, co-founder and core developer of Kamailio SIP Server project, is visiting the event.

The exhibition has dedicated pavilions for Telecommunication Industry, from equipment providers to software integrators. Asipto is glad to see several customers exhibiting there and we will be delighted to meet and discuss with the other participants at the event, as well.

If you want to schedule a meeting during the CeBIT 2011, don’t hesitate to contact us:

• Go to Contact Form

Call Center World 2011 is taking place in Berlin, Germany, Feb 21-24, 2011. Localized in the same city with Asipto, the event is a great opportunity to meet and discuss our latest offerings and developments.

Strictly related to the topic of the congress, our load balancing and security solutions are ideal for scaling and guarding IVR systems.

We will be around as usual during these days, feel free to contact us if you want to meet:

• Go to Contact Form

Daniel-Constantin Mierla gave a presentation and the European Open Source Developer event FOSDEM 2011, in Brussels, Belgium.

The presentation focused on integration SIP and Web2.0 worlds for real time communication using Kamailio SIP Server and Lua programming language.

The links to the presentation and slides:

• Unifying SIP and Web Worlds with Lua – slides available here

Besides this presentation, Kamailio project organized a dinner and had a second presentation in the Open Source Telephony dev room, more details at:

• http://asipto.com/u/18