ASIPTO

SIP/SIMPLE and XMPP share a lot of concepts but they are different in many aspects. Being nowadays the leading open protocols for voice, video, instant messaging and presence, the interconnection between them creates an unified communication environment for users in both sides. The workshop aims to bring together people with large expertise in both protocols, interested in development, testing and deployment of SIP/SIMPLE-XMPP solutions. With a permanent focus on innovation, the participants cover open source projects to private enterprises. You can join the event for free.

Date: September 2-5, 2008
Place: INRIA, Paris, France

Main organizers:
- Philippe Sultan, INRIA - contributor to the XMPP/Jingle support in Asterisk, member of XMPP Standards Foundation
- Olle E. Johansson, Edvina - main SIP developer of Asterisk
- Daniel-Constantin Mierla. Asipto - co-founder Openser, developer of XMPP/Jabber gateway

Web page at INRIA:

http://www-c.inria.fr/Internet/events/appointments/sip-simple-xmpp-developer-workshop/

Goal:
- kick up simple-xmpp interoperability

Agenda guidelines:
- identify common issues of simple-xmpp interoperability
- define best-practice solutions and workarounds of delicate issues
- coding sessions in existing applications such as Asterisk, Openser, Jabberd, Freeswitch, eJabberd, libraries, client applications, etc.
- testing sessions
- reports about past experiences and results of the workshop

Target participants:
- developers of simple-xmpp products
- people interested in testing simple-xmpp products
- people interested in building simple-xmpp communication environments

Cost:
- free registration (everybody pays for its traveling and accommodation)

Registration:
- via e-mail at simple-xmpp@asipto.com
- please write the motivation to participate and add bullets into agenda if you like to approach new subjects. The organizers reserve the right to select a group of people that will contribute most as this is mainly a workshop to approach issues, test interoperability and design new solutions.
- updates about the event are posted at: http://www.asipto.com/index.php/simple-xmpp-developer-workshop-2008/

Size:
- up to 20 people

As of Jul 15, 2008, participating companies are:
- INRIA - http://www.inria.fr - SIP-XMPP interoperability in Asterisk (http://www.asterisk.org)
- Edvina - http://www.edvina.net - SIP-XMPP interoperability in Asterisk (http://www.asterisk.org)
- AG Projects - http://www.ag-projects.com - MSRP-XMPP Interoperability (http://www.msrprelay.org)
- Asipto - http://www.asipto.com - SIP-XMPP interoperability in Openser (http://www.kamailio.net)

This service addresses specific needs and issues encountered in SIP-based platforms. For each important topic, ASIPTO has created an eLearning channel. By subscribing to the channel you get one year access to detailed tutorials presenting the issue and solutions for it using KAMAILIO (OPENSER) SIP Server. Outstanding benefits are:

• access to learning materials that improve continuously
• a large set of KAMAILIO (OPENSER) configuration files
• additional tools to ease solving the issue
• updated configuration files and tutorials content once a new stable release is out
• guidelines to upgrade to new versions of KAMAILIO (OPENSER)
• collaborative environment (wiki, forum, chat) to discuss and solve problems related to the topics
• expertise from two co-founders of KAMAILIO (OPENSER) SIP Server project
• subscription fee per year just at the price of several consultancy hours

SIP NAT Traversal channel (ECH-SNT) - learn about types of NATs, their influence in SIP networks, client-side, server-side solutions and combination of them to get the most convenient and optimized results for your scenario.

KAMAILIO (OPENSER) Configuration File (ECH-OCF) - become a master in scripting the configuration file, tips and optimizations for working with pseudo-variables, transformations, flags, routing statements a.s.o. The channel is open now. See more details here.

SIP Authentication, Authorization and Accounting (ECH-AAA) - learn about these three important aspects of a SIP platform, be sure you secure properly the services and your revenue.

ASIPTO will add more eLearning channels, check our web site from time to time. We welcome suggestions on topics you consider important and we do the best to integrate them for mutual benefits.

For more details and registration, please contact us.

ASIPTO offers a series of training classes and certification for OPENSER. Backed up by two co-founders, the certifications attest your level of knowledge about OPENSER.

Certified OPENSER Administrator - this certification verifies your skills as an OPENSER SIP Sever administrator. It consists of 150 questions related to the last stable release of OPENSER and a practical exercise where you have to build the OPENSER configuration file according to given specifications.

To pass the certification program, you need to have advanced knowledge of OPENSER and SIP, sustained by a long experience with these technologies. It is recommended that you read the documentation available at OPENSER SIP Server web site, have familiarity with the configuration file and attend a course about basics of SIP and OPENSER SIP Server.

Enrolling for certification program:

• contact us via this form, give your contact coordinate and we will send back further details
• you will get a list with documents to read and the fields you should insist on for better understanding and learning
• you will get the set of questions and have to return back the answers in a predefined time
• you will get the specifications for the practical exercise and have to return back the configuration file in a predefined time
• you can attempt two more times to pass the certification in 3 months time since enrolling in the program
• you will receive a paper certificate and the correct answers to all questions via mail
• you choose the privacy level about advertising your certification via ASIPTO
  • public - the name and date of certification will be listed on a dedicated ASIPTO web page
  • private - you get an unique key and a web page link where to introduce the key to show your name and date of certification

For more details and registration, please contact us.

“Since beginning of 2002, as core developer of SIP Express Router or KAMAILIO (OPENSER) and IP Telephony consultant, I have designed, developed and deployed close to hundred of VoIP platforms, witnessing the evolution of VoIP services world wide.” — Daniel-Constantin Mierla, CEO ASIPTO.

A list with relevant deployments running KAMAILIO (OPENSER) - based solutions:

• AG Projects’ Multimedia Service Platform
• CEM Solutions
• Localphone
• SIPit
• Truphone
• VoIPuser
• Voztelecom

For more details, please contact us.

• edvina.net - main partner for Asterisk-OPENSER training and consultancy
• AG Projects - partnership for SIP SIMPLE Interactive Messaging System (IM)
• goes.com - partner for VoIP products and consultancy for USA and North America
• itsyscom.com - partner for VoIP products and consultancy for Europe
• sipwise.com - partner to develop large and distributed VoIP platforms
• voipembedded.com - partner for VoIP products and consultancy for Canada and North America

For more details, please contact us.

ASIPTO technical leaders and our partners represent an experienced team trained over the year to offer you the best available courses that cover KAMAILIO (OPENSER) SIP Server and integration with other applications, such as Asterisk.

Main teachers:

• Daniel-Constantin Mierla, co-founder and main developer KAMAILIO (OPENSER) SIP Server
• Olle E. Johansson, main Asterisk SIP developer
• Elena-Ramona Modroiu , co-founder and main developer KAMAILIO (OPENSER) SIP Server

The courses are organized on demand, at customer premises, in a private environment, or periodically on sites selected by ASIPTO, with open registration. For the second, the place and date is announced for each course, via email and web. If you want to get notifications, please contact us.

Each course come with digital and printed materials and are applied to latest stable version of KAMAILIO (OPENSER) SIP server.

Testimonials

“Just finished the class here in Orlando and we met people from Peru, Columbia, USA and Canada. It was a nice gathering and everybody left with strong feeling of satisfaction.

I never had installed OpenSER before and we had the guts of a carrier-class architecture worked-out and prototyped in one day.

Seldom does a class gives you a career boost of such magnitude. Merci beaucoup to Daniel and Olle for caring to share their incredible knowledge.“

Francois D. Menard, Project Manager, Xit Telecom Inc.

Course: OPENSER 3W

One day introduction of KAMAILIO (OPENSER) SIP Server - what is, where to use and when is the time for it. The course focuses to outline the capabilities, use cases and production deployments.

Course: OPENSER Administration

Standard course spans over 3 days, covering the basic principles of KAMAILIO (OPENSER) SIP Server configuration file and common functionalities required within a VoIP platform. It is the course suitable for Level 1 support or for those willing to reduce the time to learn by themselves.

Advanced course spans over 5 days, continuing the standard version with complex aspects of the VoIP services, such as high availability, redundancy, media services, instant messaging and presence extensions.

This course include lectures and labs.

Course: OPENSER Development

Three days covering the internal structure of KAMAILIO (OPENSER) SIP Server, the APIs for connecting to database, parsing SIP messages, synchronization, shared memory and pseudo-variables, writing new modules and MI commands.

After completing this course, you should be able to write new extensions to KAMAILIO (OPENSER) SIP Server by yourself, understand better and be able to optimize accordingly the configuration file.

Course: ASTERISK - OPENSER SIP Masterclass

Detailed information is presented in SIP Masterclass page.

For more details, please contact us.

World’s top Asterisk and KAMAILIO (OPENSER) training focused on SIP, a partnership program with Edvina, brings to you five days full of intensive course, practical examples and open discussions.

Learn from Olle E. Johansson, the main Asterisk SIP developer, and Daniel-Constantin Mierla, co-founder and main developer of KAMAILIO (OPENSER), the mechanisms to build professional VoIP platforms.

It is more than a learning opportunity, the classes are well known for networking and business opportunities.

You will meet people with professional expertise in different area of IP telephony, discuss about cutting edge technologies, discover brand new solutions and alternatives to solve your business ideas.

Overview of course content and more details at http://edvina.net/training/sipmasterclass/index.shtm

Next places and dates:

- Orlando, Fl, USA, April 21-25, 2008

- Barcelona, Spain, May 5-9, 2008

Price and registration at edvina.net

ASIPTO-NTS comes to solve first and one of the biggest issue in VoIP networks: NAT traversal. This is not a truly Session Border Controller, with all required features, it is a box that solves the NAT traversal for you. You can place it in front of your SIP server, close to it or on a far site.

Using ASIPTO-NTS allow tailoring complex VoIP services inside your VoIP platform, forgetting about NAT. The box benefits of full flexibility KAMAILIO (OPENSER) SIP Server provides, becoming easily a security check point, traffic optimizer or routing engine.

ASIPTO-NTS brings:

• seamless integration with any SIP server
• fully functional VoIP calling network
• media stream path optimization (QoS)
• media traffic distribution
• core VoIP platform with no NAT traversal concept

For more details, please contact us.

You would need certain VoIP platform features not available in the KAMAILIO (OPENSER) SIP server, but you want to avoid delays introduced by self training of your people for coding them.

There is now the possibility to ask for a core-developer of the open source project to do this task for you in the shortest time possible, having also the guarantee of optimal solutions.

Our management team, made up of two KAMAILIO (OPENSER) SIP Server project co-founders, committed to offer best development services on open source SIP server.

ASIPTO is offering training courses for KAMAILIO (OPENSER) SIP Server extension development. It is the shortest option to write new functionalities by yourself and be sure you do it right, using at maximum the internal API.

For more details, please contact us.

Sooner or later, your VoIP traffic will reach the capabilities of your Application server, so you need to add another one or more. No matter your Application server is Asterisk, a PSTN gateway, prepaid calling card, a.s.o., ASIPTO-LBD helps scaling to meet VoIP traffic demands.

With load balancing and traffic dispatching you can modularize and distribute VoIP platform components for a better QoS and service availability.

• hot plug/unplug of new components in the VoIP platform
• customizable load balancing and dispatching rules
• redirect traffic based on DID or service type
• secure your server network by using a single point of entry, protecting the applications servers against the wild world
• high availability and failover of load balancing and dispatching server

For more details, please contact us.

With a deep knowledge of SIP, extensive experience with KAMAILIO (OPENSER) SIP Server programming and deploying it in production VoIP platforms, our specialists are happy to help you with discovering and get recovered from failures of your VoIP platform. Time is very important in such situations, we can give solutions in the shortest time possible.

We commit to Level 2 or higher of support agreements. ASIPTO offers various KAMAILIO (OPENSER) SIP Server training courses to prepare in-house support team, ready to take Level 1 support queries and maintain your VoIP platform in a day-by-day basis.

For more details, please contact us.

With years of hands-on SIP/VoIP experience starting in the early time of SIP, our team is able to provide the right answers to any of the VoIP issues you may have. We significantly contributed to build KAMAILIO (OPENSER) SIP Server project, we can help you get the best of it.

Turn-key solutions to real-life deployments, robust platforms that easily integrate into your environment will leave you more time and resources to spend for developing the business.

Professional carrier grade voip platforms, designed and tailored to specific needs, can be deployed in a very short time taking advantage of our know-how.

Moreover, our specialists are prepared to teach and transfer the knowledge to your staff, helping to build in-house competent support and maintenance team in no time.

As integrators, we work with top VoIP applications to build effective solutions for your demands. ASIPTO competence covers:

• KAMAILIO (OPENSER) SIP server
• SIP Express Router (SER)
• Asterisk
• FreeSWITCH

For more details, please contact us.

If you look for quick revenue or to secure your investment, you got in the right place. ASIPTO-PPS extends KAMAILIO (OPENSER) SIP Server to become a reliable call stateful proxy that can control the duration of each call as you wish, just by provisioning control rules in database.

ASIPTO-PPS scales to carrier-grade traffic. It was designed to handle SIP signaling only, not interfering at all with the media stream — in this way the bandwidth exhausting and media streams management bottlenecks are avoided.

Capability highlights:

• real-time charging
• call disconnect
• per-second precision billing
• shared credit account among many users
• multiple charging plans
• simultaneous calls from same user
• keepalive mechanism to detect the failure of ongoing calls
• cost and time unit per destination

For more details, please contact us.

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Built on top of KAMAILIO (OPENSER), integrating Asterisk and several other popular Open Source applications and technologies, ASIPTO-ITP offers the shortest time to roll out your production VoIP/SIP system, leaving open the way to extend to new functionalities as you go. You have to focus on the business aspects, we will take care of technical side.

Outline of out-of-the-box features

• voice and video calling with NAT traversal support
• presence and instant messaging
• customizable web front-ends (soap/xml)
• dns based addressing, multiple domain enabled
• rich telephony features (shortdial, forwarding, barring, accept/reject, voicemail, 911/112)
• postpaid with anti-fraud or prepaid accounting
• rating engine with tariffs per SIP account, domain or IP address
• least cost routing and enum peering engine
• real-time web based SIP tracing facility
• CDR mediation

For more details, please contact us.

Although it is frankly an young venture, ASIPTO has a strong background in KAMAILIO (OPENSER) and SIP/VoIP. Employing two co-founders of KAMAILIO (OPENSER) SIP Server Project, the knowledge of ASIPTO team is built on the experiences in VoIP since beginning of 2002, with tens of production deployments and active development of KAMAILIO (OPENSER).

Executive team

Daniel-Constantin Mierla - CEO - he co-founded in June 2005 openser, 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.

He is in charge with technical aspects, conducting the executive and products management.

Elena-Ramona Modroiu - COO - she joined SER (SIP Express Router) in the spring of 2003, only a few months after the project was publicly released, becoming in short time one of the most active contributors. Mrs Modroiu holds a master degree in Computer Science and Engineering from the Polytechnic University of Bucharest, Romania, completing the studies at Polytechnic University of Valencia, Spain, and Fraunhofer FOKUS Institute, Berlin, Germany. In summer 2005, she co-founded the OPENSER SIP Server project.

She is in charge with business management, customer care and company strategy.

ASIPTO has an extended partnership program, making available for you a world-wide network of IP telephony experts.

For more details, please contact us.

Applications and Services for IP Telephony

ASIPTO provides professional expertise to help businesses take full advantage of KAMAILIO (OPENSER) SIP Server and SIP technology. Ready to go VoIP solutions, consultancy, development and training stand on top of our portfolio.

KAMAILIO (OPENSER) Configuration File e-Learning channel is open. You can enroll now. More about here…

ASIPTO can help you build customized IP PBX systems, VoIP Service Platforms, VoIP Load Balancing boxes.

SIMPLE-XMPP Developer Workshop - INRIA, Paris, France, September 2-5, 2008. Click for more details.

Get to KAMAILIO (OPENSER) Devel Guide…

** ASTERISK is a trademark of DIGIUM.

ASIPTO is continuously looking for enthusiastic professional people in the VoIP/SIP area, willing to contribute and promote open source software. We appreciate individual skills and ideas, giving you the possibility to become one of our young and dynamic team members contributing to the success of our company.

VoIP Professionals Openings

• C Network Communications Programmer, 3 years experience

Student Flexible Time Openings

• Junior C Programmer

Apply or ask for more details