VOIP Today magazine (www.voiptoday.org) is privileged to conduct exclusive interview with Kamailio (OpenSER) co-founder and core developer of OpenSER and SER projects Daniel-Constantin Mierla.

Kamailio (former OpenSER) is an Open Source SIP Server released under GPL, able to handle thousands of call setups per second. Among features: asynchronous TCP, UDP and SCTP, secure communication via TLS for VoIP (voice, video), SIMPLE instant messaging and presence, ENUM, least cost routing, load balancing, routing fail-over, accounting, authentication and authorization against MySQL, Postgres, Oracle, Radius, LDAP, XMLRPC control interface, SNMP monitoring. It can be used to build large VoIP servicing platforms or to scale up SIP-to-PSTN gateways, PBX systems or media servers like Asterisk™, FreeSWITCH™ or SEMS.

VOIP Today magazine spoke with Daniel-Constantin Mierla about Business , technology and products .

voiptoday.org:   What is different about Kamailio and other open source telephony solutions ?

Daniel-Constantin Mierla: Kamailio is a SIP server, not a pure telephony engine like Asterisk or FreeSWITCH. What means that? Kamailio handles only the SIP signaling traffic, no media (e.g., audio or video) processing, therefore using Kamailio for telephony is just a subset of what can do. Of course, one of the typical use cases is for telephony, but as easy as that you can use Kamailio for instant messaging and presence or machine to machine communication.

Because of its design and purpose, Kamailio targets performance and stability in first places. When we talk about performances we mean more than 5000 call setups per second and more than 300 000 SIP phones registered to same SIP server instance. These figures show also why Kamailio is used very often as load balancer in front of IP PBXes, PSTN gateways or media servers.

voiptoday.org:    What is your average annual growth rate?

Daniel-Constantin Mierla: I could see interesting trends during all these years.

First, there are not many options out there for this kind of needs and all are based on the same initial project SIP Express Router (SER). Since 2008 Kamailio (which started as fork of SER in 2005 under the name OpenSER) and SER teams work again together, so we practically talk about same application, yet different names.

Back to your question, the project started in a research institute, Fraunhofer Fokus, Berlin, in the very early days of SIP and VoIP. After SER was released as open source in 2002, we had a good rate of adoption in academic institutes.

Next wave was realtime communication innovators – they couldn't afford to wait big vendors implement their needs, therefore this was their chance.

Today is clear that SIP is the protocol for telephony services. More and more of the classic telephony companies started using it. I can say also that building SIP trunking services is very popular.

To understand where we stand today as a project, here are some numbers:
•    since its beginning, the project counted over 70 registered developers and hundreds of contributors
•    the users mailing lists have over 63 000 posts
•    the development mailing lists have over 40 000 posts
•    over 2000 commits during 2009
•    over 170 extension modules
•    over 3000 downloads per month only from main web site
•    project management: 11 individuals (from 5 countries)

voiptoday.org:    What were your greatest achievements and implementations since your beginning on 2001?

Daniel-Constantin Mierla: It is hard to count the implementations, I was directly involved in several hundreds so far. In the first stage of the project, the biggest achievement was making it the reference implementation for a SIP server and keep it so.

One of the largest deployments I am aware of is 1&1 Germany – they have about 4 millions phone lines configured, routing more than 1.5 billions minutes per month. These are public data anyone can find on the net, but there are other big deployments and the companies keep them confidentials. Anyone can go to project web site and look at the references page for more examples.

Kamailio received the Best of Open Source Networking Software award at BOSSIE'09 by InfoWorld magazine, a recent project achievement that make us very proud of and confirmed us its spread and popularity.

voiptoday.org:    What are the future plans for Kamailio, both short term and long term?

Daniel-Constantin Mierla: In short term, there is a lot of effort in improving asynchronous processing to face the larger signaling traffic demands brought by instant messaging and presence. Right now the missing part is asynchronous TLS, we have asynchronous SIP message processing and asynchronous TCP since release 3.0.0. TLS should be added in a matter of few weeks and released with 3.1.0. You cannot find these anywhere else.

Another short time goal is to make the maintenance straightforward and adoption easier for new comers. We talk here about a SIP routing server, that means an administrator needs to know pretty much about the protocol itself. People managing telephony systems or IP PBXes don't interact usually with protocol layer, they deal with calls, not much importance for them if beneath is TDM, H323 or SIP. Here we work to make their life easier.

For example, in the upcoming release we added interactive configuration file debugger, a nice way to troubleshoot your config as well as understand what is going there. Also, we added include and define directives that allow to separate the SIP routing logic from parameters, so system administrators have to change just few lines to adapt the config to their needs. Asipto released SIREMIS (http://siremis.asipto.com), a complete web management interface that helps to do provisioning easier.

voiptoday.org:    What is the competitive advantage for Kamailio?

Daniel-Constantin Mierla: Reliability was always the strongest point of this project. Configure it properly, start it and forget about it – simple as that. Core team is more or less the same since 2002, one left out of four, but others joined and there are over 50 registered developers, about 20 being very active. SIP v2.0 was released in 2002, this shows the expertise the team carries on.

Processing capacity is also an important characteristics – with an usual server you buy now for less than $2000, one can handle several ten thousands of registered phones and thousands of active calls.

voiptoday.org:    What kinds of platforms like Skype and OCS do Kamailio support?

Daniel-Constantin Mierla: Haven't tried myself, but I heard Skype offers SIP interconnect right now, that makes Kamailio - Skype communication straightforward. Integration with OCS for trunking is possible via TCP or TLS, using SIP as well.

Practically any device that speaks SIP just works with Kamailio. Over the time I combined it with Asterisk, FreeSWITCH and many PSTN gateways or Media servers such as Cisco, Audiocodes, Telcobridge, Teles, etc.

voiptoday.org:    What is kamailio next product and what distinguishes it?

Daniel-Constantin Mierla: Kamailio 3.1.0 is the next major release, scheduled for this autumn. Novelty is not missing at all. Some new features were mentioned already, like config file debugger or asynchronous TLS, the others include embedded Lua and Python on configuration file, interprocess message queues, geo IP API, registration to remote SIP server, tree-based caching container and several more listed at:

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

voiptoday.org:    Does Kamailio ‘s solutions support any open source codecs?

Daniel-Constantin Mierla: Good question since it allows to clarify some things. The answer is short: yes, all that are available and those yet to be released. Why? Because we do not process media. If you build devices that speak SIP for signaling and they are able to transmit 3D video, Kamailio is ready for it. Understanding media stream is an end-to-end affair, we just connect origin with destination.
voiptoday.org:    What is the average cost for an integrated solution provided by Kamailio?

Daniel-Constantin Mierla: Kamailio is the open source SIP server, the application, like Asterisk. Then there are several companies involved in development and building solutions with it, like Digium and other companies for Asterisk. More or less the same, Kamailio has a business environment  around the project.

I can talk about my company, Asipto, solutions span from low range of load balancers to high end complete unified communication platforms, with a save of at least 30% comparing with telco vendors and the main benefit of owning everything and being able to enhance as you wish, the power of code is in your hands.

voiptoday.org:    How did the slow economy affect the business?

Daniel-Constantin Mierla: Global economic crises affected everyone, fortunately we were affected less that others. We play in cost effective solutions market and price became a major criterion to select a product. Furthermore, the reputation of reliability and the number of features keeps us in a good position on the market.

voiptoday.org:    What are your predictions regarding market expansion in the future?

Daniel-Constantin Mierla: We were constantly growing since the project started. We are ahead implementing features that would really get the momentum in very near future. Just want to mention here TLS for secure communication and spit protection, then IPv6 which gains more market every day. Both of them are very well supported right now, so we are ready for that moment.

voiptoday.org:    Do you carry on any special events or exhibitions?

Daniel-Constantin Mierla: We do have dedicated events. At least once a year we have developers' meeting, next is going to be on 8th of June, in Berlin. It is face to face meeting, open for community members as well, were we discuss existing issues and brainstorm future development.

On business side, we used to tie with various events where we had dedicated tracks, running one or more days. Lately we went for smaller events, targeting local or country-level markets. In the business we act, massive call volume routing, you don't have too many potential customers in a region. Trying to get personal is important. In this fashion we run this year The Presence and Future of SIP Routing, first was in London, UK. Soon we will announce dates for other places in Europe.

Besides those, we participate to many related events. SIPit 26 was las week in Sweden, where I participated to test, main interest for me being TLS and IPv6. Then we have presentations at Amoocon 2010 in Rostock and LinuxTag 2010 in Berlin in less than one month – at LinuxTag the project has a booth for the entire 4 days exhibition.

Another group of events are trainings and certifications, next to come are Miami, USA, June 21-23, and Malaga, Spain, July 5-9.