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OpenSIPS

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About OpenSIPS


What is OpenSIPS

OpenSIPS (Open SIP Server) (former OpenSER) is a mature Open Source implementation of a SIP server. OpenSIPS is more than a SIP proxy/router as it includes application-level functionalities. OpenSIPS, as a SIP server, is the core component of any SIP-based VoIP solution. With a very flexible and customizable routing engine, OpenSIPS 'unifies voice, video, IM and presence services in a highly efficient way, thanks to its scalable (modular) design.
What OpenSIPS has to offer, comes in a reliable and high-performance flavour - OpenSIPS is one of the fastest SIP servers, with a throughput that confirms it as a solution up to enterprise or carrier-grade class.

OpenSIPS vision

The OpenSIPS vision consists in a "consolidated diversity" - OpenSIPS is an open source project not only as license, but also as policy regarding the contributions, cooperation and community.
The diversity comes from the large number of people involved in the development of the project and from the complexity and feature richness of OpenSIPS (it incorporates and integrates a large variety of functionalities and features, on top of the SIP routing engine).
To reach a high level of reliability and stability, all this diversity needs to be consolidated. The project brings a consolidate d view on the project management, on the project vision, on the design and code, on the work effort, on the project future.

History

OpenSER project was created by Voice System in 2005 by forking the SER project. In the following years, Voice System was the main contributor and supporter behind the project.

In 2008, the OpenSER project was renamed by Voice System as OpenSIPS project due trademark issues over the old name.

More about OpenSIPS

OpenSIPS project is the continuation of the OpenSER project after its renaming. Inheriting the OpenSER spirit of openness to the community and the will of progress (a continuous work to develop, enhance and extend the code), OpenSIPS continues and extends the OpenSER vision through a strong process of consolidation.
Based on the experience provided by the OpenSER project, such a process of consolidation is critical in order to guarantee the best effort and best results inside the project. A consolidated environment is mandatory for large scale project that aims to deliver professional solutions for industry environment.


OpenSIPS Headlines



  • Monday, 15th of February 2010 - unveiling the design for OpenSIPS 2.0 - design description


  • Monday, 21st of December, 2009 OpenSIPS 1.6.1 (minor release) is out



  • VoIP service. Monday, September 21st, 2009. OpenSIPS project provides a completely free VoIP service for the community and not only. This service is based on a residential -like platform provided by Voice System company. You can self-register to get an opensips.org SIP address and to place VoIP call in any SIP network. Registration link : http://www.opensips.org/account/reg/index.php. Login link : http://www.opensips.org/account/index.php.

  • Astricon 2009, Monday, September 21st, 2009. OpenSIPS project will be present at Astricon Conference and Expo Event on October 13-15, 2009, Glandale, Arizona, US http://www.astricon.net/.

  • SVN freeze for 1.6,Thursday,September 17th,2009. http://www.opensips.org/Main/News0044The SVN trunk is freeze for preparing the 1.6 release - no more new code is allowed into SVN, but only fixes to the existing code.

  • New types of scripting routes, Thursday, Sepember 10th, 2009. http://www.opensips.org/Main/News0043The configuration possibilities for OpenSIPS have been enriched with new additions related to the routes. The intention was to extend the action range and also to make the configuration file easier to write and read.
    • Named routes
    • STARTUP route
    • TIMER route
    • Per branch "onreply_route"

  • Stun Server, Monday, September 7th, 2009.http://www.opensips.org/html/docs/modules/devel/stun.htmlWhy an integrated STUN server module?
    • A STUN server uses 2 ips and 2 ports to create 4 sockets on which to listen or respond.
    • This integrated STUN module uses the SIPS sockets created by the OpenSIPS core:
    • In the case of a bad behaving NATs (such as symmetric NAT) only this integrated STUN server will increase the likelihood of STUN to solve the NAT traversal for a wider range of NAT implementations. This translates into less need of media relaying (for NAT traversal purposes).

  • Pseudovariable implementation extended,Thursday,September 3rd, 2009.The pseudovariable implementation has been extended with new operations to give more power to the script writer:
    • Added 2 new operations on pseudovariables values from the script:
      • delete the pseudovariable at a certain index.
      • overwrite the value of a pseudovariable at a certain index; syntax.
    • Added one new function for avp:
    • Added operations for set values of pseudovariables:
    • Added a new transformation: extract a parameter from a To like header
    • Added context meaning to pseudovariables - a new token added in front of the pv type name that specifies the context where the pv should be taken from The implementation is made general - with a mechanism of registering pv context functions, allowing for extensions and addition of custom contexts. For now there are 2 pv contexts: reply and request, registered by the tm module.

  • Sunday, August 30th, 2009, A new complex tutorial is available to describe the design and implementation of a realtime integration of OpenSIPS SIP server and Asterisk media server. The integration make available media services like voicemail, conference and announcements, in a dynamic and transparent way, for all the OpenSIPS subscribers.

  • Thursday, August 27th, 2009, OpenSIPS 1.5.3 (minor release) is out

  • Tuesday, August the 18th, 2009, OpenSIPS 1.6 gets a new AAA API and an improved RADIUS support - the target is to gain more flexibility and simplicity in configuration and to open the path for a new up to date DIAMETER support. The Radius support was enhanced to allow custom queries and access directly from script - find out more from the opensips web site

  • Tuesday, August the 4th, 2009, the next OpenSIPS webinar on the topic "Types of Routs in OpenSIPS" is scheduled for Wednesday, August 12, 2009 . The registration is free under http://www.opensips.org/Training/Webinar. This webinar will help you to understand how routing is done by OpenSIPS. How the SIP messages (requests/replies) are processed inside OpenSIPS and what types of scripting routes are available to help us with this.

  • Monday, August the 3rd, 2009, OpenSIPS 1.6.0 becomes more than a SIP proxy - it extends with a signalling B2BUA support. This implementation gives OpenSIPS power over the calls and makes possible building some PBX specific services. It is not a full B2BUA as it does not handle media, but only creates and controls media sessions. To learn more on this and to read the OpenSIPS B2BUA tutorial, see http://www.opensips.org/Main/News0036

  • Thursday, July 30th, 2009, OpenSIPS 1.6.0 comes with a valuable enhancement in the DB area - virtual DB connection, a DB connection mixer for failover, parallel and LB support. Find more on http://www.opensips.org/Main/News0035

  • Thursday, July 23th, 2009, OpenSIPS 1.6.0 supports codec manipulation. A new set of functions for searching codecs, for removing codecs or changing the codec priorities are available. Find more on http://www.opensips.org/Main/News0034.

  • Thursday, July 16th, 2009, OpenSIPS 1.6.0 has support for memcached daemon interfacing. A separated, decoupled memory caching application offers several advantages over the local caching - find here a detailed description.

  • Wednesday, July 15th, 2009, OpenSIPS 1.5.2 (minor release) is out



  • Monday, June 29th, 2009, A large set of changes pushed to the REGISTRAR module to allow more flexibility in configuring the behaviour of the module. More or less, these changes aimed to move most of the global options (per module options) to per AOR or function options. So, you can configure the behaviour per AOR (like how many contacts are allowed, if branches should be appended, if PATH support should be used and how, etc).

OpenSIPS news archive





OpenSIPS Webinars


OpenSIPS project started a series of free Webinars to help people to better understand and work with OpenSIPS / OpenSER.

In the webinars section, you can find the topics of the webinars, the recordings of past webinars, the schedule for the next webinars and registration forms.


OpenSIPS Events


Events around the world where you can meet OpenSIPS folks or OpenSIPS has dedicated sessions or talks.

2010

  • 15 - 19 November 2010, Edison, OpenSIPS Bootcamp (5 days training and certification), New Jersey, USA,
  • 20 - 24 September 2010, OpenSIPS Bootcamp (5 days training and certification), Frankfurt, Germany
  • 19 - 23 April 2010, OpenSIPS Bootcamp (5 days training and certification), San Francisco, USA
  • 1 - 4 March 2010 OpenSIPS devel Course (4 days training), Bucharest, Romania
  • 22 - 26 February 2010, OpenSIPS Bootcamp (5 days training and certification), Florianopolis, Brazil,

2009



2008

  • SIPit 23th - Lannion, France, 13th to 17th Ocrober. See the event report
  • OpenSIPS Advanced Programming Course, 11th to 14th of August, Bucharest, Romania. Free registration.
  • ClueCon, 05th of August, Chicago, USA - OpenSIPS fronting and load-balancing media servers and PBXs



OpenSIPS 2.0 Design


Why a new design


The current (pre 2.0) OpenSIPS internal design is based on some concepts older than 7 years. At the time, the requirements were few (simple stateless UDP proxy) and the decisions were made accordingly. But with all the additions, both in SIP and functionality (like TCP/TLS, script manipulation, dialog support, external integration, etc), the existing design is not able to cope with the requirements and use-cases any more.

List of issues that the new design wants to solve].

Preliminary discussions


For almost an year discussions were carried and feedback was received about what issues need to be addressed by the new design.

In other words, the whole community was invited to contribute in generating the set of requirements for drafting the new design for OpenSIPS.

Requirements for the new design

New design description


The first step in making the new design a reality was to compile all the information to generate a proposal / draft for the design OpenSIPS.

Full and detailed description of a the new design.




Resources

Web site

http://www.opensips.org

Documentation

Cookbook - modules, script variables, routing
Documentation - Tutorials, Troubleshooting, presentations

Mailing lists

devel at lists dot opensips dot org
users at lists dot opensips dot org
See more...

SandBox

SandBox for developing and converging ideas

Project hosting

http://sourceforge.net/projects/opensips

Trackers

Multiple trackers are available here

Development

Development Process and status


Practical Examples




Platforms


OpenSIPS is know to work on
  • most of Linux flavours (OS used for development is mainly Debian)
  • 64b Linux flavours
  • SUN/Solaris
  • FreeBSD
  • OpenBSD
  • NetBSD


Turnkey solutions based on OpenSIPS


  • Multimedia Service Platform - a turnkey solution for delivery of SIP services like voice and video over IP (VoIP), instant messaging (IM) and Presence to SIP end-points


Monitoring OpenSIPS



OpenSIPS and Web Interfaces



OpenSER Configuration Auto-generator





Created by: admin,Last modification on Mon 22 of Feb, 2010 [10:31 UTC] by bogdan_iancu


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