Kamailio v3.1.3 Released

Kamailio SIP Server v3.1.3 stable is out – a minor release including fixes in code and documentation since v3.1.2 – configuration file and database compatibility is preserved.

Kamailio (former OpenSER) 3.1.3 is based on the latest version of GIT branch 3.1, therefore those running 3.1.0, 3.1.1 or 3.1.2 are advised to upgrade. There is no change done to database structure. On 64b operating systems you may need to update the path to loaded modules in configuration file — see the bottom note for more details.

Resources for Kamailio version 3.1.3

Source tarballs are available at:

http://www.kamailio.org/pub/kamailio/3.1.3/src/

Detailed changelog:

http://www.kamailio.org/pub/kamailio/3.1.3/ChangeLog

Download via GIT:

# git clone –depth 1 git://git.sip-router.org/sip-router kamailio
# cd kamailio
# git checkout -b 3.1 origin/3.1
# make FLAVOUR=kamailio cfg

Binaries and packages will be uploaded at:
http://www.kamailio.org/pub/kamailio/3.1.3/

Modules’ documentation:

http://www.kamailio.org/docs/modules/3.1.x/

What is new in 3.1.x release series is summarized in the announcement of v3.1.0:

http://www.kamailio.org/w/kamailio-openser-v3.1.0-release-notes/

Note: on 64b operating systems, the internal libraries and modules are installed under ‘lib64′ folder (e.g., /usr/local/lib64/kamailio/modules/). If you want to install under ‘lib’ directory, then use ‘LIBDIR=lib’ when compiling and installing (e.g., ‘make FLAVOUR=kamailio LIBDIR=lib install’). If you keep the default lib64, then you may need to adjust ‘mpath’ (or ‘loadpath’) parameter in your existing configuration files and change ‘lib’ to ‘lib64′.

If you want to see what is new in development version (to become the future major release v3.2.0), visit the web page:

• http://sip-router.org/wiki/features/new-in-devel

Joint venture on Kamailio to tackle big vendors in telco market

Asipto and Sipwise, two companies involved in the development and management of Kamailio project, announced today the joint venture to create complete and competitive IP Telephony product using Kamailio as core component for SIP routing.

The venture focuses on recently announced SIP:Provider platform targeting IP telephony operators, delivered as out of the box system with features such us authentication, authorization, NAT traversal, call forwarding, call baring, voice mail, web interfaces for monitoring, administration and user portal, post paid billing, interconnection for PSTN with least cost routing, a.s.o.

SIP:Provider CE version is provided free of charge, under open source license, you can install it easily via Debian packages or images for VirtualBox and VMWare. The target is to fill the gab between raw components such as Kamailio and full operational IP telephony system, that one can download, install and then it is ready to go to operate telephony services, without messing to put together various applications and having to understand SIP.

SIP:Provider Pro Edition adds telecom specific SLA along with extra features such as prepaid billing, high availability and redundancy.

You can read the announcement here.

Kamailio 2010 Awards

Here we are, the 4th edition of Kamailio Project Awards, granted for activity during 2010.

The past year was full of events and achieved very important milestones set for our project. First of all was the release of version 3.0, the first as a result of the integration between Kamailio and SIP Express Router (SER), the two being since then one application - see more about 3.0 release here.

More over, another major release was done in 2010, v3.1, worked out by an enlarged development team, brought a big list of new features, including full asynchronous network communication (even TCP and TLS) - see more about 3.1 release here.

All together, 2010 was great, therefore the awards got two new categories - Innovation in Communications for those using Kamailio for services beyond voice and Academic Environment for using Kamailio in research and educational networks.

I was not able to list everyone I wished, trying to stick to the tradition of having each of the category with two winners, listed in alphabetic order. As a rule, I tried to choose people and companies that were not selected in the past editions, but of course I want to thank to everyone contributing to and using Kamailio during 2010.

Let the show begin...

Blogging:

• Venture VoIP – very good coverage of Kamailio (OpenSER) & Asterisk related news, like the integration tutorials between the two open source telephony applications. Web link: http://www.venturevoip.com/news.php
• VoIP Today – VoIP news site always keeping an eye on our project activities, publishing news about our releases and major events around the project. Web link: http://www.voiptoday.org

Related Projects:

• SEMS - (aka SIP Express Media Server) programmable and lightweight SIP back to back user agent and media server written in C++, offering features such as signaling B2BUA, Voicemail, audio conferencing, SBC, IVR, a.s.o. The project shares many developers of Kamailio and it has the roots in the same research institute as Kamailio and SER, FhG FOKUS Berlin, Germany. Web link: http://www.ipterl.org/sems/
• SIP:Provider – full featured VoIP servicing platform using Kamailio for SIP routing, offering web management interfaces for administration and users. Among features: postpaid billing, call forwarding, call blocking, speed dial, voice mail, click-to-dial, peering, least cost routing - click here for more. Web link: http://www.sipwise.com/products/spce/

Technical Support:

• Alex Hermann - one of the community members that spotted corner case issues and came with detailed report and patches most of the time. In addition he added enhancements to newly XAVP concept and provided straight answers on our mailing lists. Alex works for SpeakUp, Netherlands
• Timo Reimann - omnipresent at our developer meetings and events as well as on our mailing lists. His development involvements brought many modules, such as dialog, to better structure. Timo works for 1&1, Germany

New Contributions:

• app_lua - the new module allows execution of embedded Lua applications. The latest enhancements to this module allowed writing a pretty complete Kamailio configuration for SIP routing only in Lua, see this page for it. You can see also a presentation done at Fosdem 2011 about Twitter Notifications from Kamailio Configuration - click here for it. Web link: http://kamailio.org/docs/modules/stable/modules/app_lua.html
• presence_conference - this module came as a result of Google Summer of Code, successfully completed by Marius Ovidiu Bucur. Done together with SIP Communicator (btw, the best cross-platform open source SIP softphone I met so far), this extension based on RFC 4353 and 4575 added to our SIMPLE Presence implementations, which along with the newly added embedded XCAP server makes Kamailio the most complete SIMPLE Presence Server implementation that can be found in the open source world. Web link: http://kamailio.org/docs/modules/stable/modules_k/presence_conference.html

Developer Remarks:

• Carsten Bock - member of Kamailio Management team, working for Telefonica O2, Germany, Carsten worked lately a lot with dispatcher, dialog and usrloc modules, plus the newly started efforts to the IMS extensions.
• Marius Ovidiu Bucur - the new developer landed in our project as a result of participation to Google Summer of Code. A student at Polytechnics University of Bucharest, working now part time for 1&1, Marius continued to contribute to Kamailio's SIMPLE Presence server, his latest work to this component focused on increasing the scalability (the code already in our GIT repository).

Advocating:

• Fred Posner - I had the opportunity to meet Fred personally during the last year, a person that carries an amazing bag of experience in VoIP and security. Fred continuously helped in promoting Kamailio, on mailing lists, IRC channels and public events. Besides that, his baker skills are visible at amazing good looking and tasteful cakes by Dream Day Cakes (and yes, I did taste some of them during my last trip in USA, thanks Fred & Yeni - but just trust me, don't look to their site, after that it might be too late and it may cost you a lot by not being able to stop yourself keep ordering).
• Olle E. Johanson - probably it is not really much to add about Olle, the VoIP Olle. However, last year Olle conducted super-human efforts to keep SIP world ahead in communications. Kamailio was always a part of that. I mention here only a few of them: SIPit in Stockholm (organized by Olle himself) where Olle and I setup Kamailio based TLS and IPv6 testbeds to be used by anyone attending there. His VoIP Forum articles kept heads up in regards to IPv6 and security in SIP, then, his involvement made possible the switch to SIP in the entire Portuguese educational network, running now about 300 pairs of Asterisk and Kamailio - deployment presented by Ruben Sousa at Astricon 2010.

VoIP Services:

• Flowroute - early adopter of Kamailio, Flowroute, acting mainly as a SIP interconnect broker and providing quality VoIP routes, keeps pushing the SIP server towards innovation, always looking for better performances and proper security in regard to attacks and fraud detection. Flowroute is also actively involved in promoting Kamailio project, hosting related events at their premises. Web link: http://www.flowroute.com
• XtraTelecom – Spanish telephony provider focused on enterprise market, offering SIP trunking services along with hosted PBX’s. With Inaki Baz Castillo in their team, member of Kamailio's management as well, XtraTelecom relies on a capable group of engineers that can only ensure quality of service. Web link: http://www.xtratelecom.es

Business Initiatives:

• NG Voice - the team coordinated by Carsten Bock working with IMS extensions in Kamailio, also developing other IMS infrastructure applications. It is a new initiative with a lot of potential in business environment in the near future. Web link: http://www.ng-voice.com
• TeamForest - every year, the number of companies offering Kamailio services is growing in USA. Knowing now them personally, TeamForest is another company that you can trust theirs skills in deploying Kamailio and offering professional support services. Web link: http://www.teamforrest.com

Events:

• Cluecon - after missing the 2009 edition, being busy in that year to complete the integration between Kamailio and SER, the 2010 edition was amazing for me. In the first day only, Kamailio was present directly in 5 presentations (one by myself), plus a demo done by Phil Zimmermann using iptel.org sevice which runs SER flavour of our project. Purely amazing for me! I was able to catch up with many members of Kamailio community and FreeSWITCH developers. Web link: http://www.cluecon.com
• LinuxTag - the event taking place in Berlin offered Kamailio the chance to have a booth at the exhibition and a presentation at conference track done by Henning Westerholt. All together we were about 15 Kamailio developers and community chatting with visitors, other open source developers and projects present there. Henning featured also an interview in German for RadioTux - listen the podcast here. Web link: http://www.linuxtag.org

Academic Environment:

• Illinois Institute of Technology - School of Applied Technology - Chicago, IL, USA - they conducted a set of research projects with Kamailio involved (one very interesting is the performance benchmark - click here for results in pdf). Web link: http://voip.itm.iit.edu
• Zilinska University - Slovakia - they published a set of useful tutorials about how they use Kamailio inside the university, see more at: http://nil.uniza.sk/sip/kamailio. Web link: http://nil.uniza.sk

Innovation in communications:

• Ifbyphone - a provider of voice applications for customer interactions - relying on cloud based services to offer call tracking, dynamic inbound call routing with IVR screening, outbound call automation, virtual call center applications and a highly flexible family of API based integration tools. With two presentation at Cluecon by Irv Shapiro and Robin Rodriguez, they showed usage of Kamailio beyond the classic telephony (e.g., video of the talk Web Enabling Voice Applications with Kamailio). Web link: http://www.ifbyphone.com
• NextIX - an innovation company that specializes in universally available information and communication technology solutions. At Astricon 2010, they presented “Asterisk, Kamailio, Openfire and Social Media Integration” - another way of using Kamailio for voice and beyond that. Web link: http://www.nextixsystems.com

As of Personal Facts related to the project, this time I want to underline the release of several complete tutorials, such as: integration with Asterisk or FreeSwitch, scanning attacks protection or SIP SIMPLE Presence server - see all of them at:

• http://kb.asipto.com/kamailio:index.

This is it for 2010. If you want to check the previous turn of awards:

• http://by-miconda.blogspot.com/2010/02/kamailio-openser-2009-awards.html

Asterisk 1.6 and Kamailio 3.1 Realtime Integration Tutorial

A new version of the tutorial about Asterisk and Kamailio realtime integration is out, upgraded to use the latest stable release of Kamailio, v3.1.0. You can find it at:

• http://asipto.com/u/c

Besides making it work for v3.1.x, the Kamailio config file has some new features included:

• IP authentication - can be enabled via define WITH_IPAUTH
• TLS support - can be enabled via define WITH_TLS - TLS to UDP translation and vice-versa is done automatically by Kamailio in case you configure Asterisk on UDP
• detection of DoS attacks - can be enabled via define WITH_ANTIFLOOD - banning automatically traffic from attacker IP addresses for a specific time interval
• restructuring of configuration file for better modularity and highlighting of functionalities such as registrar, location server, within-dialog request routing

SIP Routing Logic in Lua with Kamailio

A tutorial showing a complex SIP routing logic implemented with latest Kamailio development version has been made available at:

• http://asipto.com/u/h

Practically shows how to use Lua scripting language instead of Kamailio’s native configuration language to route SIP requests, taking care of of services such as authentication, registration or user location.

While is not intended as a replacement for Kamailio’s configuration language, Lua, by its nature of small and fast embedded language, is a perfect choice for enhancing SIP routing capabilities. It has dozens of extensions that you can use, including libraries to connect to social networks such as Twitter, allowing you to send notifications from your SIP server configuration.

Best of New in Kamailio 3.1.0 - #11: Asynchronous message queues in config file

One of the main problems while trying to interact with other systems direct from your SIP server was that most of the time such operations are done in blocking mode.

Whether you want to do an http query, send an email, write to a storage system for a specific SIP event, that uses the time and resources of your SIP routing engine and you cannot afford blocking all application processes that handle SIP traffic.

There are a lot of reason you would like to do such operations, for example:

• monitoring activity - notify when the rate of incoming SIP requests exceed a threshold - alert on flood
  
• real time notifications to twitter, facebook or classic email for events such as missed calls or a particular user becomes online
  
• logging purposes - write details about various situations to a storage system

Kamailio v3.1.0 pushed out a new module mqueue, which is message queue system that can be used directly in the configuration file. You can define as many queues as you want, read and write operations are safe even when done from different application processes. You can write a message from a process and read in another one.

For example, a typical usage is to start dedicated processes to consume messages from the queues. You can do that in configu using rtimer module - start separate processes that execute periodically a route block from config, where you process messages from queues.

Next is an example of usage:

• the sip worker process writes in queue "alert" when pike modules triggers alert due to high traffic rate from same IP
• process checks every 5 seconds checks if there are message in queue 'alert' and writes to syslog all the messages in the queue
  

modparam("rtimer", "timer",  "name=ta;interval=5;mode=1;")
modparam("rtimer", "exec",   "timer=ta;route=QMALERT")
modparam("mqueue", "mqueue", "name=alert")

route {
...
   if (!pike_check_req())
   {
       mq_add("alert", "$si:$sp", "pike flood detected [$rm] $fu => $ru");
       exit;
   }
...
}

route[QMALERT] {
   while(mq_fetch("alert"))
   {
       xlog("L_ALERT","ALERT: src [$mqk(alert)] - $mqv(alert)\n");
   }
}

Here you find the online documentation for mqueue module:

• http://kamailio.org/docs/modules/stable/modules/mqueue.html
  

Kamailio Advanced Training, Jan 24-26, 2011, Irvine, CA, USA

Next US and North America edition of Kamailio Advanced Training will take place in Irvine, CA, USA, Jan 24-26, 2011.

Last stable series of Kamailio SIP Server, the 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 in collaboration with Flowroute and will be taught by Daniel-Constantin Mierla, co-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-usa/

Siremis v2.0.0 Released

Siremis v2.0.0 is out – the web management interface for Kamailio (Openser) v3.1.0 and SIP Express Router (SER).

This is a major release, with countless improvements and new features since v1.x series, among them:

• major re-factoring of web interface
• better accessibility
• simplified menu structure
• completely new look
• dashboard with the map of all available tools
• developed on top of Cubi and PHPOpenBiz v2.4 frameworks
• web installation wizard
• added new modules: xcap, dialog, new lcr
• usage of separate database for siremis itself
• management of users that can login to siremis
• management of menu can be done from web interface
• building Apache conf and htaccess file can be done by Makefie
• charts to monitor location transport layers

Step by step installation tutorial, screenshots and demo are available on the web at:

• Installation Tutorial
• Screenshots
• Web Demo

Siremis is used during Kamailio Advanced Training classes for management of SIP server, a good oportunity to learn about Siremis itself, next locations are:

• Kamailio Advanced Training for Carrier Services, November 22-25, 2010, Berlin, Germany
• Kamailio Advanced Training for Carrier Services, January 24-26, 2010, Irvine, CA, USA

Kamailio Advanced Training, Nov 22-25, 2010, Berlin

Next Kamailio Advanced Training will take place in Berlin, Germany, Nov 22-25, 2010.

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, co-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/

Kamailio Business Directory Launched

Today was announced the launch of Kamailio Business Directory.

This is a web page hosted by the Kamailio project site that will list companies and individuals offering products, services or solutions based on Kamailio or SER. It is an open directory that tries to enable more visibility to the business market around the project. If you want to apply to be listed, please follow the instructions at:

• http://www.kamailio.org/w/business-directory-rules/

We have already several companies listed, couple of them to still in review process, therefore check it again soon, the directory is available at:

• http://www.kamailio.org/w/business-directory/

Asterisk, Kamailio, Openfire and Social Media Integration

Courtesy of Kelvin Chua, CTO of NEXTIX, I received the slides presented at Astricon 2010 to the session “Asterisk, Kamailio, Openfire and Social Media Integration”.

You can download the slides from:

• http://www.kamailio.org/events/2010-Astricon/KelvinChuaAstricon2010.pdf

The presentations goes through usage of open source components to build a social media integration system. Here is the abstract:

“The excitement does not end with combining the flexibility of asterisk, the capacity handling of kamailio and the chattiness of openfire, we’ve been seeing this for the past astricons. The real deal is how to use the three powerhouse in a more socially relevant context. This session will introduce you to another way of creating your own facebook or myspace. It aims to discuss what we as asterisk users/developers wanted to see in a social platform and yet nobody did it on facebook. It will discuss a myriad of apps involved with social platforms like iphone/android apps as well as a list of hardware we don’t see integrated with current social websites. it will also touch on rare topics like working on video mixers (MCU) on asterisk using h.264 and how to use this on the platform.”

NextIX is an innovation company that specializes in universally available information and communication technology solutions for the consumer, SME, enterprise and government.

Debian Based Kamailio Distribution

CIITIX has announced CIITIX VoIP 1.0 – a Debian Lenny custom Linux distribution that includes a pre-configured Kamailio v3.1.0. Among enabled features are:

• user authentication against MySQL
• NAT traversal
• accounting to MySQL
• SIP SIMPLE presence
• embedded XCAP server
• DoS detection and protection

You can download the ISO image and read more about at:

• http://ciitix.ciit.net.pk/index.php/home/6-announcements-/14-ciitix-voip-10-released

It is a good and easy way for people to try a Kamailio based VoIP system on a physical server or a virtual machine.

Kamailio v3.0.4 Released

Kamailio v3.0.4 is out – a minor release including fixes in code and documentation since v3.0.3 – configuration file and database compatibility is preserved.

Kamailio (OpenSER) 3.0.4 is based on the latest version of GIT branch 3.0, therefore those running 3.0.0, 3.0.1, 3.0.2 or 3.0.3 are advised to upgrade — there is no change required to be done to configuration file or database.

Note that latest stable version of the project is Kamailio 3.1.0, released on October 06, 2010. It is highly recommended to upgrade directly to latest stable release to benefit of most actual fixes and features. More details about 3.1.0 at:

• http://www.kamailio.org/w/kamailio-openser-v3.1.0-release-notes

Resources for Kamailio version 3.0.4

Source tarballs are available at:

http://www.kamailio.org/pub/kamailio/3.0.4/src/

Detailed changelog:

http://www.kamailio.org/pub/kamailio/3.0.4/ChangeLog

Download via GIT:

# git clone –depth 1 git://git.sip-router.org/sip-router kamailio
# cd kamailio
# git checkout -b kamailio_3.0 origin/kamailio_3.0

Binaries and packages will be uploaded at:

http://www.kamailio.org/pub/kamailio/3.0.4/

Modules’ documentation:

http://www.kamailio.org/docs/modules/3.0.x/

What is new in 3.0.x release series is summarized in the announcement of v3.0.0:

http://www.kamailio.org/w/kamailio-openser-v3.0.0-release-notes/

Kamailio v1.5.5 Released

Kamailio v1.5.5, a new release in 1.5 series, is out. Kamailio (OpenSER) 1.5.5 is based on the latest version of branch 1.5, including many fixes in code and documentation, therefore those running 1.5.0, 1.5.1, 1.5.2, 1.5.3 or 1.5.4 are advised to upgrade.

Note that latest stable version of the project is Kamailio 3.1.0, released on October 06, 2010. It is highly recommended to upgrade directly to latest stable release to benefit of most actual fixes and features. More details about 3.1.0 at:

http://www.kamailio.org/w/kamailio-openser-v3.1.0-release-notes

This release marks the end of official maintenance for branch 1.5 by development team. That means no new packaging will be done for 1.5.x, but fixes can be added in SVN branch 1.5 repository. The development team officially maintain last two stable branches, these are now 3.0.x and 3.1.x.

Source tarballs are available at:

http://www.kamailio.org/pub/kamailio/1.5.5/src/

Detailed changelog:

http://www.kamailio.org/pub/kamailio/1.5.5/ChangeLog

Download via SVN:

svn co https://openser.svn.sourceforge.net/svnroot/openser/branches/1.5 kamailio

Tag for this release can be browsed at:

http://openser.svn.sourceforge.net/viewvc/openser/tags/1.5.5/

Project site at SourceForge.net (still using old name):

http://sourceforge.net/projects/openser/

Binaries and packages will be uploaded at:

http://www.kamailio.org/pub/kamailio/1.5.5/

Modules’ documentation:

http://www.kamailio.org/docs/modules/1.5.x/

What is new in 1.5.x release series is summarized in the announcement of v1.5.0:

http://www.kamailio.org/w/kamailio-openser-v1.5.0-release-notes

Note: Kamailio is the new name of OpenSER project. First version under Kamailio name was 1.4.0. Older versions will continue to use OpenSER name.

Best of New in Kamailio 3.1.0 - #10: Registration to Remote SIP Servers

Many DID providers require to register to their servers in order to route the calls to your server. While the recommended way is to get a peering agreement where all calls to your block of DIDs are sent to the IP address of your SIP server, in some cases, like SOHO or SMB, you usually register to some VoIP services that offer free DIDs. Then is hard to get a peering agreement just for one or two DIDs.

Normally you used such providers only for incoming calls, for outgoing you have a least cost routing system selecting the best route from many termination providers.

Kamailio 3.1.0 introduces a handy way to configure your SIP proxy to register to other SIP servers for incoming calls. You have to load uac module and add records to uacreg table. The table stores following attributes:

• l_uuid - local unique user id, e.g.,: 12345678
• l_username - local user name, e.g.,: daniel
• l_domain - local domain, e.g.,: mysipserver.com
• r_username - remote username, e.g.,: daniel123
• r_domain - remote domain, e.g.,: sipprovider.com
  
• realm - remote relam, e.g.,: sipprovider.com
• auth_username - authentication username, e.g.,: daniel123
• auth_password - authentication password, e.g.,: xxxxxx
• auth_proxy - SIP address of authentication proxy, e.g.,: sip:sipprovider.com
  

The module takes care of sending REGISTER and refresh registrations before they expire.

When calls come in, you have to run uac_reg_lookup() that will detect if the call is coming from a remote SIP provider and can change the R-URI to local username@domain. Afterwards you can run location lookup.

if(uac_reg_lookup("$rU", "$ru")) {
  xlog("request from a remote SIP provider [$ou => $ru]\n");
}
lookup("location");

The documentation for uac module is available at:

• http://kamailio.org/docs/modules/stable/modules_k/uac.html
  

Apt Deb Repositories for v3.1.0

Courtesy of Jon Bonilla, Kamailio's Debian APT repository is offering now the packages for version 3.1.0 as well as nightly builds from stable branch 3.1.

So, if you want to get the latest version of branch 3.1, set your apt sources to:

deb http://deb.kamailio.org/kamailio31-nightly lenny main 

Supported OSes are Debian Lenny, Squeeze and Ubuntu Lucid.

Check for more details the wiki page:

• http://www.kamailio.org/dokuwiki/doku.php/packages:debs

Best of New in Kamailio 3.1.0 - #9: Load Balancer

SIP load balancer extension - dispatcher module - comes in Kamailio v3.1.0 with a bunch of new features:

• weight-based load balancing - you can assign weights to addresses in a destination group and Kamailio will take care to distribute calls accordingly. For example, if you want to distribute 80% of calls to one server and the rest to another one, just set weight=80 to first address and weight=20 to second one

2 sip:192.168.178.20:5080 0 0 weight=80
2 sip:192.168.178.28:5082 0 0 weight=20

• call-load-based dispatching - the module can track active calls and select least loaded destination to distribute the traffic. Note that the module includes a light-weight call tracing system which works even in transaction stateless mode. There is no dependency on heavy modules such as dialog, making it suitable for embedded systems as well.
  

• configurable list of valid codes for SIP ping replies - the module has the ability to send OPTIONS requests to addresses in destination sets in order to discover whether they are active or not. In some particular cases, the reply code could be different than 200 (for example the system is asking for authentication), but still the system should be able to handle new calls. With module parameter ds_ping_reply_codes you can define a list of reply codes or reply code classes that are valid to consider that destination is active. For example - all 2XX and 3XX replies, together with 403 and 488:
  

modparam("dispatcher", "ds_ping_reply_codes", "class=2;code=403;code=488;class=3")

The value of this parameter can be changed at runtime without restartion Kamailio.

Summarizing, with the new weight and call load dispatching, the module offers now a large range of algorithms to select destination, from old ones I would mention: round-robin, priority level, random, hashing of SIP message attributes. you can hardly find a competitor to beat the performances and features of dispatcher module for SIP load balancing.

You can browse module's documentation online at:

• http://kamailio.org/docs/modules/stable/modules_k/dispatcher.html
  

Best of New in Kamailio 3.1.0 - #8: Configuration File

Kamailio v3.1.0 is shipped with a refurbished configuration file. It has a new structure to reflect better SIP server functionalities such as:

• SIP location server
• SIP registrar server
• SIP presence server
• NAT traversal management
• PSTN routing
• SIP message format sanity checks
  
• routing within-dialog SIP requests

Several values were replaced by defines making easier to maintain and understand the logic of configuration file:

• value of db_url parameters is set by DBURL define - if you change the access to database server you have to update a single value
• multi-domain parameter value
• used flags (for accounting, missed calls or NAT traversal) are also defined values, using more meaningful ID name
  

Couple of new features are included in the configuration file and can be easily enabled or disabled by defining or un-defining specific IDs:

• TLS support - controlled by #!define WITH_TLS
• multi-domain support - controlled by #!define WITH_MULTIDOMAIN
• flood detection and protection - controlled by #!define WITH_ANTIFLOOD
• database aliases lookup - controlled by #!define WITH_ALIASDB
• source IP authentication - controlled by #!define WITH_IPAUTH
• XMLRPC control interface -controlled by #!define WITH_XMLRPC

Have a look at configuration file (located in /etc/kamailio/kamailio.cfg or /usr/local/etc/kamailio/kamailio.cfg) and you will be surprised to discover how easy is now to manage it and turn on/off features. Also, you can browse it online on our GIT repository.

Kamailio v3.1.0 Released

Kamailio (OpenSER) v3.1.0 is out – major release with impressing number of new features and improvements.

On October 06, 2010, Kamailio (OpenSER) 3.1.0 has been released – this release is a result of more than 8 months of development from the teams of Kamailio (OpenSER) and SIP Express Router (SER) projects. Backed up by a solid development group, we are proud to announce that this release brings a large set of features, many for first time on the SIP server market, such as asynchronous TLS, UDP raw sockets, embedded HTTP and XCAP servers, embedded Lua, configuration file debugger. All together, there are over 15 new modules and countless improvements to old components.

Since last major release (version 3.0.0, which was out in January 10, 2010), the two SIP servers, Kamailio and SER, are practically the same application, the name making the difference regarding the database structure and the extensions used for certain features, such as user database based authentication or location service. Therefore another development direction was towards smooth integration of Kamailio and SER extensions, previously duplicated modules such as auth, sl, ratelimit or sms were merged during this development cycle.

Continue reading the release notes at:

• http://www.kamailio.org/w/kamailio-openser-v3.1.0-release-notes/

Best of New in Kamailio 3.1.0 - ToC

Kamailio 3.1.0 - a new major release of the open source SIP server - is scheduled for October 6, 2010. The amount of new feature is astonishing, it is the outcome of 7 months development made by Kamailio and SER teams.

With blog series named "Best of New in Kamailio 3.1.0 ..." I try to detail some of most relevant new additions this release brings to light.

Here is the table of content:

• 1 - Interactive configuration file debugger
• 2 - Embedded Lua
• 3 - Embedded HTTP server
• 4 - Embedded XCAP server
• 5 - GeoIP API
• 6 - Asynchronous TLS
• 7 - Extended configuration file pre-processor directives
• 8 - Default configuration file
  
• 9 - Load balancing enhancements
• 10 - Registration to remote SIP servers
• 11 - Asynchronous message queues in config file
• 12 - Generic tree-based caching system
• 13 - RFC3326 - Reason header for CANCEL
  
• 14 - UDP raw sockets
• 15 - Additional tools
  

Chapters will be added over the time and this page will be updated with the links to specific posts. Check it from time to time to see its new content...