OpenSIPS At Amoocon (Former AsteriskTag)

http://snapvoip.blogspot.com/
OpenSIPS project will be presenting papers on May 4th - 5th, 2009 at AMOOCON (Formerly AsteriskTag)in Rostock (Germany). The bilingual conference on Open-Source VoIP and Open-Source communication. AMOOCON was formally Asterisk-Tag and thay changed the name to fit the changing field of Open Source VoIP. AMOOCON now includes Yate and FreeSwitch in addition to Asterisk.

Among many hot topics related to Open Source VoIP will be presented by various organizations and OpenSIPS will focus its efforts on very interesting subjects.

The papers already submitted include subjects covering,

Load-balancing (using the new load-balancer module) for media servers, Workshop

Deploying and delivering Presence based Services with OpenSIPS, Workshop

SIP clustering (OpenSIPS + PBX integration), Workshop

All are invited.
For more information visit OpenSIPS and AMOOCON

Tags: OpenSIPS, Open Source VoIP, AMOOCON, AsteriskTag, Asterisk-Tag, Yate, FreeSwitch, SIP clustering, Presence based Services with OpenSIPS, Load-balancing Media Servers

OpenSIPS 1.4.4 And A OpenSIPS Bootcamp, Awaits you.

http://snapvoip.blogspot.com/
In the footsteps od OpenSIPS 1.4.3 (production) release, OpenSIPS is offering us a new minor release, OpenSIPS 1.4.4. This release fixes a set of major and minor fixes on the OpenSIPS 1.4 branch.
The OpenSIPS team recommends to upgrade to this release, as it brings stability improvements to the already a fantastic system.
For those who are interested, complete change log could be found here.
OpenSIPS 1.4.4 is now available for download at OpenSIPS and Sourceforge.

For those interested in training on the OpenSIPS system, there is a OpenSIPS Bootcamp in sunny Brazil, 16-20 February, 2009, Florianopolis.

Tags: OpenSIPS Bootcamp, OpenSIPS 1.4.3, OpenSIPS 1.4.4, Sourceforge, Open Source VoIP, OpenSIPS, SIP

VoIP Conferencing Server From FreeSwitch And SIPFoundry

SIPFoundry and FreeSwitch has joined hands to create a fully featured conferencing server. The conferencing server is based on FreeSWITCH and sipXecs adds ease-of-use Web based management, cluster management, and all the other services provided by sipXecs already. More info is available on the sipXecs Wiki .

Some of the key features of the new conferencing solution include:

• Every user of sipXecs can have a personal conference bridge. When a new user is created the system can automatically create a bridge extension for that user.
  
• Users are able to fully control the conference bridge from their user portal (dynamic conference control)
• FreeSWITCH is integrated as a service of sipXecs. It auto-installs and all configuration is done automatically by sipXconfig
• Several FreeSWITCH based conferencing servers can be configured in the system, each on separate hardware. This allows the creation of a massively scalable conferencing solution all with centralized Web management for configuration and process control
• The conferencing server supports a variety of codecs and does proper mixing

The new conferencing application is available in the current development branch of sipXecs (release 3.11) and will get released with the upcoming 4.0 release of sipXecs.

Tags: FreeSWITCH, VoIP Conferencing, Open Source VoIP, sipXecs, VoIP Conferencing Server, SIPFoundry

CafeSIP, Where Java And SIP Come Together, Test Your SIP Here.

The CafeSip.org hosts a number of open-source SIP projects. These cover the areas of SIP server framework, SIP applications and SIP test tools. The idea is to provide free and open-source products for users and organizations developing or offering SIP applications and products.

Following three projects are in the works at CafeSIP and each provide unique but inter connected products and services and according to the site;

1. Jiplet is a framework for developing SIP server-side applications using the Java programming language. It provides a container for hosting "SIP servlets". In concept, it is very similar to a Java Servlet (web application). Instead of using the HTTP protocol, a jiplet supports the SIP protocol. The Jiplet container can be run as a standalone application or it can be run as a service inside a J2EE container. The Jiplet Container has been tested extensively with the JBOSS.
   
2. SipUnit is a unit testing tool for SIP server-side applications. It extends the JUNIT framework (http://www.junit.org) which is a popular tool for testing Java applications. Using SipUnit, one can develop test programs that simulate a SIP User Agent (a SIP phone). It provides a low-level as well as high-level API for communicating with a SIP server.
   
3. SipExchange is a soft switch application. It supports many features offered by a small switch including subscriber registration, call handling, call routing, etc. It is built at the top of the Jiplet framework and JBOSS J2EE server. It is extensible and in future, new features will be added to provide capabilities such as multi-media contact center, unified messaging, etc.
4. Common Projects: Consists of a set of common components used by multiple CafeSip projects. They are designed to be used by outside projects as well.
   

CafeSIP has announced the public release of SipUnit r0.0.7b. This is a Beta release to give users a working system to evaluate and test. Feedback from the development community is encouraged for further enhancing the system.

So far I have tested / used the SIPUnit and if that is any indication, others also must rock. I certainly will be looking into creating Jiplets to host some of our SIP ideas.

So I invite all those who are ready to get hands dirty with coding and compiling pay visit to the project site.

CafeSIP Project home page.

Tags: VoIP, SIP Testing, VoIP Testing, SIP, Open Source SIP, CafeSIP, SipUnit, Jiplet, SIPExchange, SIP server framework, SIP applications, SIP test tools