Thursday, September 24, 2009

What is wrong with VoIP word?

Travels, talks and telephony! There is "a thing" thrown from a corner to the other, enabling fear for some, hope for others. That is VoIP and clear is misused.

Bloggers around the world chopped every news about VoIP providers failure -- let's not give names, searching web will reveal important Telcos or famous venture-capitalist backed up operators closing their VoIP offerings.

Therefore something is wrong, what the heck? Everywhere I get the technology is heavy used. But hated in the same time. In my opinion it started with the way VoIP was brought to the market, since everybody contributed to it and mixed the technology with the service, building FUD, thus confusing environment.

Simply and in first place, VoIP is a telephony technology, on the same layer as TDM, ISDN. Not a service. The operator must not go to enterprise and end user selling VoIP, they must keep selling telephony plus new services. Digital telephony came as added value against analog telephony by allowing dynamic interaction via DTMF and more features to PBX-es. That was one of the attractions that made it worth to deploy and successful.

Today the operators simply fail to present the advantages VoIP brings to customers. In addition, Telco and Mobile operators perform anti-promoting actions to VoIP and they rely on it more than others for backbones and even to end customers -- just that they say it is either IMS based service or what so ever NGN-service.

Therefore I think everyone should review how they show VoIP. If I would be Telco or Mobile operator I would stop saying VoIP provides no QoS, is insecure, a.s.o. Simply that harms them, and I do it with every occasion asking what is what they use to route international calls, what is their new service based on, ...

So, they admit doing VoIP, but on private, secure network. Here we are! What they should better say? Telephony services on VoIP over public network cannot ensure YOU, the end user, QoS, security (well, here lot to debate, but not the scope of this post) and everything else they consider being atu of doing VoIP over their walled-garden infrastructure.

Pure VoIP service providers did the big mistake of advertising VoIP as service, and even worse, free service, thus they cut the main revenue stream, trapping themselves in traffic generation for termination, strongly tied to Telcos, rather than new services business. While a I see it "free as in speech" and not "free as in bear". The freedom I see is I, the customer, have the liberty to choose my terminal and use what ever functionality I like from what VoIP enables.

Everyone should promote what VoIP enables YOU, the customer, to access hell-out of many new services, like presence, instant messaging, video, integration with social networking sites, mobility, multiple identities on the same wire, a.s.o.

First, the average guy does not care what is VoIP. He needs to communicate and businesses should focus on that demand. Look at car manufacture, nobody promotes the technology behind new models, but the better fuel consumption figures, speed and acceleration, a.s.o. -- exactly what the end user cares about.

I, on the other hand, go to operators and sell VoIP solutions. And say, hey, VoIP is the technology to build the future of telephony, showing benefits of new services, maintenance costs, etc. I must promote the technology and scream in all direction VoIP, VoIP, VoIP.

Wednesday, September 23, 2009

Kamailio - amazing autumn

It is very encouraging that the businesses around the project are growing and development speeds up. In the last days there were two requests for Kamailio specialists, so, if you are one of them and willing to work in Germany or Austria, within dynamic teams in challenging markets, check these posts were I summarize the announcements from Kamailio mailing lists.

- jobs at 1&1 Germany

- jobs at Sipwise Austria

Autumn moves forward with couple of events related to the project where you can meet people engaged in the project and learn how you can use Kamailio for various IP communication services:

- VoIP2Day, Madrid, Spain, Sep 24-25, 2009

- SIP and IMS for Next Generation Telecoms Forum 2009

- SIP Router Devel Meeting, Berlin, Germany, Oct 2, 2009

- Astricon 2009, Arizona, USA, Oct 13-15, 2009

- Training - Kamailio SIP Masterclass, Berlin, Germany, Nov 9-13, 2009

All these together with planned release of Kamailio 3.0 in October and further developement of SIP Router core framework announce an amazing autumn ahead.

Monday, September 21, 2009

SIP and IMS for Next Generation Telecoms 2009

On short notice, I will present Understanding SIP/VoIP Architecture Design at SIP and IMS for Next Generation Telecoms 2009, September 23-25, Berlin, Germany.

The presentation is held on Sep 23, 11:30, focusing on:

  • Optimising next generation SIP networks
  • Planning and implementing new SIP functionalities
  • Using SIP for prepaid systems and internet telephony platforms
  • Integrating load balancing and session border control
If you are in Berlin during the event and want to meet, send me an email at miconda [at] .

Wednesday, September 16, 2009

Book: SIP Security

I had it from quite some time now, really enjoyed reading it, so time for blogging it.

First, all authors are former fellows at FhG Fokus Institute, Berlin, Germany and most of them tight involved in SIP Express Router from day one. So this is not something written upon theoretical research and concepts but upon years of hands on experience with SIP networks.

Having technical background, I found interesting the blending of cryptographic mechanisms, security concepts and applicability to SIP networks. Everything needed to fully understand the book is inside.

For me, it is important to mention that lot of scenarios and solutions are exemplified with SIP Express Router, project I was involved pretty much from its beginning, from where I started Kamailio (OpenSER) back in 2005 and I met again last November within SIP Router project.

The foreworld from Philip Zimmermann really synthesize the security concerns about VoIP and SIP. Shortly, the main chapters:
- introduction to cryptographic mechanisms
- introduction to SIP
- introduction to IMS
- secure access and interworking in IMS
- user identity in SIP
- media security
- denial of services attacks on VoIP and IMS service
- spam over IP telephony

The chapter about DoS attacks is comprehensive, covering over 15 type of attacks. I will blog in more details about the chapters I find most interesting for me.

The book is available on Amazon UK:

There you can see complete table of content. A dedicated site for SIP security and this book is put up together by authors at:

Tuesday, September 15, 2009

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]

Monday, September 14, 2009

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]

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:

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


  • 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:

More details to follow soon! Stay tuned!

Friday, September 11, 2009

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).

Thursday, September 10, 2009

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”);
# msg buffer has a new content
# will get always here

Readme of the module:

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

Wednesday, September 9, 2009

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:

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:

Friday, September 4, 2009

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");


The feature is documented at:

Thursday, September 3, 2009

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:

  • 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] ) for any question.

Info about positions:

Tuesday, September 1, 2009

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 core, asynchronous TCP, asynchronous SIP message processing API, memcached backend, native topology hiding, nat traversal using kernel space for media relaying ... see more at:

The news on project's website: