Monday, January 22, 2007

OpenSER v1.1.1.Released

Version 1.1.1 of OpenSER has been released. It is an update of version 1.1.0 and includes the fixes for the issues discovered in version 1.1.0, therefore all people using 1.1.0 should upgrade to this new version.

For people using sms, permissions, postgres, unixodbc, osp, it is highly recommended to move to this release, in the case no CVS updates were done after v1.1.0 (if you installed and maintain OpenSER from CVS, then you have to use branch rel_1_1_0, as it was for v1.1.0). The compatibility of the configuration file and database has been preserved (you do not need to do database reinstallation).

Some details about v1.1.1 can be found here

Detailed changelog:

Sources can be downloaded from:

The documentation for modules is posted at:

If someone wants to contribute with packages for different distros, please send the link from where can be downloaded to

Sunday, January 21, 2007

Kamailio (OpenSER) - Summary of 2007

2007 was a year full of achievements and events for Kamailio (OpenSER). We had a major release in summer (1.1.0), and a continuous increase in features set and robustness. What was new in 1.1.0 was summarized in release news:

Since then we went though many milestones. We had the first Kamailio (OpenSER) summit where we were able to meet many folks from the lists, have nice chats and present interesting VoIP solutions based on OpenSER.

A very important statistic is the number of new modules and features introduced in current development version after 1.1.0:
- domainpolicy - policies to connect federations
- imc - instant messaging conferencing
- mi_fifo and mi_xmlrpc - FIFO and XMLRPC transports for the new management interface (MI)
- perl - embed perl programming in configuration file
- presence - SIMPLE Presence Server implementation
- pua, pua_mi, pua_usrloc - presence user agent client implementations for user location records and management interface
- seas - connector to SIP Application Server - WeSIP - Java SIP Servlet Application Server (
- snmpstats - SNMP (Simple Network Management Interface) interface to OpenSER statistics
- sst - SIP session timer support
- xmpp - transparent SIP-XMPP gateway

Documentation and news about all these:

A lot of other significant improvements and addons were included in existing code (management interface, fetch database support, postgres, usrloc, dialog, enum, acc modules polishing), openser web administrator was developed. There is not very long time to the next release, along with few pending new features, the next period will be allocated to core components like timers, dns and configuration file flexibility. By end of January we should approach a new testing phase.

This year the OpenSER community collaboration exploded, dokuwiki has lot of content, many tutorials were submitted, new developers and many modules sponsored by companies, mailing lists and web forum are places to get reliable information about Kamailio (OpenSER)...

We want to thank to all of you (developers, contributors with patches, tools and documentation, testers and Kamailio (OpenSER) users) for supporting the project. 2006 was a very good year for Kamailio (OpenSER) and 2007 looks very challenging.

Kamailio (OpenSER) vs SER performances

Following lot of questions and comments related to the results published by Iptelorg guys related to openser vs ser performances on their site, I will give some clarifications.

First, the tests were done only by Iptelorg guys. They didn't ask anybody of openser development team if could help them to make accurate tests and make sure that openser and ser have same internal parameters. For people familiar with these SIP servers, it is well known that the applications uses internal hash tables, different number of processes, memory sizes, and so on -- this parameters can influence a lot if they have different values.

Because they continued saying tests are fair and reflect reality, we had to investigate and debug ser which ended in something pretty much expected -- they did performance improvement, but the processing was totally wrong - we reported the bug, and more or less it was admitted that it was the source of the performance improvement in ser stable version (although some from Iptelorg tried to say the results of the bug were not critical -- but was a bug). Follow the thread on mailing list for more details:

Removing that bug, ser gets back more or less to same performances.

Kamailio (OpenSER) will do performance measurements in the testing period for version 1.2.0. But it will be only for openser, as we won't try to claim we know ser that well to be sure that any eventual comparison is fair.

In open source it is expected to have friendly relationship with similar projects, not unsustainable accuses and unfair approaches and comments about the others.

Kamailio (OpenSER) Summit 2006 - Closing notes

Last posts on this blog are short reviews of the presentations held at Kamailio (OpenSER) Summit 2006. It was an opening session on the afternoon of 7th November, continued by a full day on the 8th. Totally were 17 speeches, sustained by breaks for face-to-face talks with developers, executives and community members.

We look forward to next Kamailio (OpenSER) event, the place was not decided yet, is very likely to happen in Europe or North America. When settled, news will be published on this blog, Kamailio (OpenSER) mailing lists and website.

You can get all slides and view pictures with attendants of Kamailio (OpenSER) Summit 2006 from:

Kamailio (OpenSER) Summit 2006 - Asterisk

We had the chance of a good talk held by Olle E. Johansson, Edvina, the father of SIP channel in Asterisk. He provided a good overview of how OpenSER and Asterisk can be used together to provide feature rich VoIP service. Furthermore, he talked about Asterisk 1.4 and future plans for SIP.

Aside this a good remark was lack of good collaboration between major players in Open Source telephony. We all have to work on this, to collaborate to build an Open Telephony Platform. There will be no application that will achieve that alone. We should gather our knowledge and forces to make it possible.

I end with same topic as he did. Lot of people try to promote security as a big hole in VoIP versus PSTN. Next image from the slides is very much relevant (:-) in this respect: pstn-security.jpg.

Download slides of Asterisk Presentation

Sunday, January 14, 2007

Kamailio (OpenSER) Summit 2006 - The Challenge of Service Diversity

Bogdan-Andrei Iancu, one of Kamailio (OpenSER) Project developers, talked about the implications of VoIP diversity. Features base and deployment characteristics have big impact in maintenance and management of the VoIP platform.

The discussion covered four use cases and solutions in VoIP: carrier, hosted services, residential services and billing systems. Each case has its set of special issues that has to be taken care of: scalability, load balancing, dynamic routing, peering, system resources... The conclusions summarized several points that should be considered when designed a complex VoIP system.

Download slides of The Challenge of SIP Diversity

Saturday, January 13, 2007

Kamailio (OpenSER) Summit 2006 - INRIA - Deployment Notes Working Group aims to interconnect academic institutes all over the world via VoIP/SIP. We had as special guest Phillipe Sultan from INRIA, the National Research Institute of France. He gave a comprehensive overview of their goals and achievements so far.

Most of the components used in deployments are open source. The first step was to make available existing user directory via VoIP. The email address is used as principal VoIP ID. In addition, local PBX extensions are made available on VoIP. So far, major universities and research institutes of US and Europe joined in project: MIT, Yale, Harvard, Columbia University, INRIA, more here ...

Download slides of INRIA - Deployment Notes

Kamailio (OpenSER) Summit 2006 - Development of convergent J2EE applications for OpenSER

A more technical approach of writing VoIP Java applications for Kamailio (OpenSER) and WeSIP was given by Elias Baixas of Voztelecom. He went through SIP servlet concept and how its API eases the development of VoIP application without any need of technical details about SIP/VoIP internals.

With about two slides of Java code you can embed in your web application a VoIP click to dial feature. Also, as showed in the slides, WeSIP provides seamless interoperation of SIP and HTTP Servlets. web developers will find lot of similarities that will make VoIP SIP Servlets something easy to do if you have HTTP Servlet knowledge.

Download slides of Development of convergent J2EE applications for OpenSER

Kamailio (OpenSER) Summit 2006 - Managing a Highly Available VoIP System

Andreas Granig of UPC Austria introduced causes for VoIP services downtime and options to solve for them. It is clear that a reliable service cannot resist without high availability. With perfect hardware you will get software failures, with perfect software you will get a hardware crash at a point in time.

Two concept should guide your service as much as possible: simplicity and modularity. Try to keep your deployments as simple as possible for the set of services you want to offer. The maintenance overhead is lowered. Second, try to make the platform as modular as possible -- is very unlikely that all components will fail at the same time -- you will have features downtime, but not service downtime.

Download slides of Managing a Highly Available VoIP System.