Kamailio Jobs at Sipwise, Vienna, Austria

Sipwise is currently hiring a VoIP System Administrator for an interesting position based in Vienna, Austria.

You have strong skills in Linux system administration (Monitoring and Alerting using SNMP/Nagios/Cacti/MRTG, Scripting in Perl/M4/sh), a deep understanding of highly available system deployments and good knowledge regarding SIP (preferably Kamailio, Sems, Asterisk)?

Sipwise offer you a challenging position to help our team further improving our Kamailio-based Class5 Softswitches, deploying them at customer sites and supporting our customers (large DSL and Cable Providers throughout Europe) and sales teams with technical details.

If you are interested, please send email to Andreas Granig, agranig [at] sipwise.com

SIP Router Development Meeting 2009

Next SIP Router Project Development Meeting coordinates:

Date: Friday, October 2, 2009

Place: Berlin, Germany

The event is co-hosted by FhG Fokus Institute and Technical University Berlin at following address:

FhG Fokus, Room 1008

Kaiserin-Augusta-Allee 31

10589 Berlin

- see venue of the location

Among the goals of the meeting:

• analyze the development and progress so far
  • Kamailio (OpenSER) and SER integration is 99% completed
  • first major release based on SIP router – Kamailio 3.0 – is due in one month – during October
  • OpenIMSCore extensions ready to use with SIP router core
  • what was good and bad?
• find solutions for conflicting modules and namings
  • database table structures
  • database table names
  • module names
• future directions
  • versioning and releasing policies
  • targets for next year
• organizational aspects
  • infrastructure
  • management
• social networking and business environment

The participation is free of charge for anybody upon registration via short email at:

registration [at] lists.sip-router.org

this being required for proper dimensioning of meeting room and needed logistics. Registration must be done before September 28, 2009.

Who you can meet there:

• developers of SIP Express Router (SER), Kamailio (OpenSER), OpenIMSCore, SIP Express Media Server (SEMS) and other open source or commercial VoIP proejcts
• community members of above mentioned projects
• representatives from companies acting in VoIP and SIP business

Who should consider participation:

• people willing to get a close feeling about project development
• people willing to understand how and when SIP Router can be used
• people willing to meet face to face with the others acting within SIP Router project environment

Agenda:

• developer slot: presentations from developers
• community slot: presentations from community and business representatives
• open discussions
• social networking event

Hacking day:

• if there is interest from developers, the days before the public meeting can be organized as hacking session, where you get your hands dirty and code around the project. Do not forget to mention your interest in such event!

You can address general questions about the event via email to:

sr-dev@lists.sip-router.org

More details to follow soon! Stay tuned!

SIP Router: Number Portability Functionality

Courtesy of Henning Westerholt, SIP Router repository includes dedicated module and applications for fast number portability handling - hte feature will be part of upcoming Kamailio (OpenSER) 3.0 release.

pdb server

This server loads serialized routing data from the disk and stores it in memory. It then listens on an UDP port for requests containing a number and returns ID of the carrier which owns the number. This server uses the same data structure as the carrierroute module and provides a really good performance — consumes only a few percent CPU, even you use only one server for your complete call routing cluster.

pdb connector module

This module connects the sip-router server to the pdb server. It supports load-balancing and aggressive timeouts. Normally it does not need more than a few ms to query the remote server and return the reply to the configuration script.

pdb tool (data compiler)

This tool provides the functionality to compile the carrier and number information into the binary data format the pdb server expect. It supports optimizing the generated trie structure, so that for example the complete number to carrier mapping for Germany (app. 150 million numbers) don’t need more than a few hundred megabytes. You can also combine not interesting carriers in order to save even more space and get better performance.

The module can be in the modules/pdb directory, the server and tool is in utils/pdbt. This directory also contains documentation (README for module, utils/pdbt/docs/* for data format and network protocol).

SIP Router: Apply Changes to SIP Message in Config File

One of the most discussed architectural aspects of configuration file was the way changes done to received SIP messages are handled – even new headers are added, old are removed or different parts of message were updated, the changes were not immediately visibile.

The latest GIT repository includes a new config function – msg_apply_changes() – exported by module textops from modules_k folder. Once this function is used, all changes done up to that point are applied and further processing within config will see the new message content.

The new function can be used only in request route for now. Be carefully when using it, since it changes the expectation you had so far of always working on initially received SIP message. Once the function is used, initial content is lost forever. As it builds a new buffer and re-parses, it is not very recommended to use it extensively.

Example of usage:

append_hf(”My-Header: yes\r\n”);
if(msg_apply_changes())
{
# msg buffer has a new content
if(is_present_hf(”My-Header”))
{
# will get always here
}
}

Readme of the module:

http://sip-router.org/docbook/sip-router/branch/master/modules_k/textops/textops.html

The functionality will be part of upcoming Kamailio (OpenSER) 3.0 release.

Freezing for Kamailio 3.0

Time for first release based on SIP Router Project is approaching. Source code will be frozen Monday, September 14, 2009, to enter the testing phase for releasing Kamailio 3.0.

As usual, work on additional tools and documentation can go on during testing. Developers willing to push brand new features in the next major release have to hurry.

A draft of new features is compiled at:

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

For a complete picture of what Kamailio 3.0 will bring new comparing with 1.5 you have to check inherited features from SIP Express Router as well:

http://sip-router.org/benefits/

Support for ‘include’ in config file

The configuration file language of Kamailio (OpenSER) and SIP Router supports now a new directive that allow including the content of another file during parsing of routing logic.

This allows splitting big configs for easier maintenance, even modularity — building a library of config snippets that are included and combined to build a full configuration file.

The syntax is:

include_file “file_name”

There is no restriction of what the included file should contain, it must be a valid content for the place where the include directive is used.

Here is an example:

route {
   ...
   include_file "/sr/checks.cfg"
   ...
}

--- /sr/checks.cfg ---

  if (!mf_process_maxfwd_header("10")) {
      sl_send_reply("483","Too Many Hops");
      exit;
  }

---

The feature is documented at:

http://sip-router.org/wiki/cookbooks/core-cookbook/devel

Kamailio Jobs at 1&1, Karsruhe, Germany

Interested in working with one of the biggest Voice over IP networks in Europe? 1&1 is hiring and have two open positions in Karlsruhe, Germany.

For more information (requirements, how to apply..) please refer to the openings on comapny's job page (German). Some facts about our VoIP system can be found at:

http://www.kamailio.org/events/2009-linuxtag/presentation.pdf

Shortly:

• over 2 million subscribers
  
• over 1 billion minutes per month
• over 50 servers SIP platform
  

Please feel free to contact Henning Westerholt ( henning.westerholt [at] 1und1.de ) for any question.

Info about positions:

• VoIP Operation, system administration
• VoIP Software development, R&D
  

Kamailio Awarded Best of Open Source Software 2009

InfoWorld has published the Best of Open Source Software Awards 2009. Kamailio (OpenSER) has been awarded within category: Best of Open Source Networking Software.

From InfoWorld site:

"Award winners in network and network management are old favorites Cacti and Nagios, the IPCop firewall, Kamailio SIP proxy server, KeePass password manager, Openfiler SAN/NAS appliance, OpenNMS enterprise monitoring system, PacketFence network access control solution, Puppet configuration management framework, and Untangle network security gateway."

"Kamailio is the open source SIP proxy server formerly known as OpenSER. Used with an Asterisk IP PBX server for phone features, plus a hardware gateway for connection to the outside world, Kamailio brings important call handling and scalability benefits to Asterisk, while also removing the Asterisk server as a single point of failure. Larger organizations get the phone features they need, as well as the added safety of VoIP calls surviving an Asterisk server outage."

Here it is...

On the other hand, Kamailio project is approaching the time for a new major release -- a matter of days to enter the testing phase -- this one will be versioned Kamailio 3.0 to reflect the major enhancements, among them: the http://sip-router.org core, asynchronous TCP, asynchronous SIP message processing API, memcached backend, native topology hiding, nat traversal using kernel space for media relaying ... see more at:

http://sip-router.org/benefits/

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

The news on project's website:
http://www.kamailio.org/mos/view/News/NewsItem/Best-of-Open-Source-Software-Awards-2009/