Monday, December 31, 2012

Happy New Year!

This was it for 2012, one of the greatest years in the history of the Kamailio project! One major release was done in the summer (version 3.3.x), another one is few days before testing phase, most probably to be out by mid of February.

The next major release includes several great features and edevelopments:
- support for websockets
- IMS extensions in the main GIT branch
- no more duplicated modules, integration of Kamailio and SER ended

There are big plans for 2013. First is the next major release, together with participation at FOSDEM conference in Berlin. A bunch of other events will open the path to our first dedicated conference for Kamailio project - Kamailio World.

We hope to meet many of you at Kamailio World conference, in Berlin, Germany, during April 16-17. Last organizing bits were sorted out and registration will open in the first days of 2013 -- you can see more details at:


Looking forward to a great 2013 for Kamailio and everyone involved in the project, from developers to community members!

All the best! Happy New Year!

Saturday, December 29, 2012

Kamailio - SER integration fully completed

The development teams of Kamailio and SIP Express Router (SER) decided in November 2008 to join forces and start integration of the Kamailio – formerly OpenSER –  and SER projects. The goal was to get one combined source code base. About one year later, there was a single source code tree, but there were several duplicated modules that kept some confusion around the two projects. We’ve finally sorted out the modules and will soon release a unified release with one set of modules!

Time has shown where to go with the set of modules

As the time passed, we’ve discovered which modules that are in the focus, making the decision easy to sort out the modules that are not frequently used. The past weeks were dedicated to removing duplicated, a process that just ended — at this moment there are no duplicated modules, all of them have been integrated.

Modules are merged or renamed

In summary, the modules from SER that had no conflict were moved as is to the “modules” folder, in some cases renamed. The modules that overlapped with some conflicts were merged, adding missing features to the most developed ones, moving the others to obsolete folder. In a majority of the cases, the Kamailio version of the module was used as the base for merging code, mostly because they had larger set of features (the core in v3.0.0 was based on the SIP Express Router core).

Functionality is still around, but may have changed a bit

Please note that the extra features from the SER module set might not still exist in the same form, but alternatives that should offer same functionality exists in the new or merged modules. Users of the SER flavour of the SIP Router code base  should prepare migration by starting to test the development version in GIT now. We do need your feedback on the feature set and merged functionality.

Use the mailing lists to discuss the merger

Do ask questions on the mailing lists if something that you used to have in your configuration is no longer available, so we can assist in your migration and, hopefully, create a migration guide on the Kamailio wiki.

There might few bits and pieces to tune here and there. Developers will soon get into the code freezing period for testing the next major release, which a good time for such adjustments. It is important that everyone using the former SER modules (modules_s) to report the issues that are found.

Many thanks to all those involved in integration efforts as well as in new developments that kept the project flying!

A Happy New Year and Good Testing!

Monday, December 24, 2012

Merry Christmas!!!

Another year getting to its end! Looking back, looks like one with the greatest achievements in the development of the Kamailio project so far. Year is not done, so that summary is saved for one week later, there is still stuff on its way to our source code repository.

Now we want to thank to everyone promoting and contributing to the project, from developers to community members, and wish Merry Christmas and great winter holidays to all supporters of Kamailio and related projects!

Sunday, December 23, 2012

VoIP Users Conference with Flowroute

The weekly VoIP Users Conference had its last session with Flowroute, a company that contributed to Kamailio project the modules related to JSON.

Bayan Towfiq, the CEO, spoke about their services and how they built the telephony platform based on open standards and many open source projects.

You can see the recorded session at:

Wednesday, December 19, 2012

New developers for Kamailio project

Two new developers have joined Kamailio team in December:
  • Konstantin Mosesov – he has contributed patches to app_python and joins the team to help maintaining and developing this module. His git commit id is: ez
  • Richard Good - he is among those developing the IMS extensions for Kamailio (, to be merged in the next days to the GIT master branch. His git commit id is: richard.good
Warm welcome and looking forward to their contributions!

Tuesday, December 18, 2012

Kamailio v3.3.3 Released

Kamailio SIP Server v3.3.3 stable is out – a minor release including fixes in code and documentation since v3.3.2 – configuration file and database compatibility is preserved.
Kamailio (former OpenSER) v3.3.3 is based on the latest version of GIT branch 3.3, therefore those running previous 3.3.x versions are advised to upgrade. There is no change that has to be done to configuration file or database structure comparing with older v3.3.x.

Resources for Kamailio version 3.3.3

Source tarballs are available at:
Detailed changelog:
Download via GIT:
 # git clone –depth 1 git:// kamailio
 # cd kamailio
 # git checkout -b 3.3 origin/3.3
 # make FLAVOUR=kamailio cfg

Binaries and packages will be uploaded at:
Modules’ documentation:
What is new in 3.3.x release series is summarized in the announcement of v3.3.0:

Monday, December 17, 2012

JavaScript MSRP Library

Peter Dunkley, developer of Kamailio, author of websockets support, has announced that Crocodile-RCS has just open-sourced their MSRP over WebSocket Javascript stack.The project is hosted on Google Code:
The stack is distributed using the MIT License and was developed and tested along side the Kamailio MSRP over WebSocket implementation.

For more about MSRP via websocket, see:
Crocodile-RCS did recently a successful demo of this implementation at WebRTC Expo in San Francisco, video is available at:

Sunday, December 9, 2012

Kamailio World – Conference & Exhibition – April 16-17, 2013

Kamailio project was announcing its first dedicated event – Kamailio World – to take place in Berlin, Germany, during April 16-17, 2013. The event is organized by Asipto in collaboration with Fraunhofer Fokus Institute.

The conference is targeting to present commercial products and services built around Kamailio and related projects, as well as technical workshops of how to use Kamailio for various purposes. Along with the conference will be space for exhibition, that will allow companies to make demos and show cases for participants.

Besides Kamailio, you will have the chance to interact with other open source VoIP projects such Asterisk, FreeSwitch, Open IMS Core, SIP Express Media Server, Homer SIP Capture, Siremis, a.s.o.
For more details, visit the web site of the event:

Thursday, December 6, 2012

Preparing the next major release of Kamailio!

Code freeze for the next major release of Kamailio is now set to January 7th. This release will mark the end of the merge period of Sip Express Router and OpenSER. As we move forward, there will be one set of modules, one set of database schemas and one combined product. We do need your help in this process!

Plenty of time to hack during the holidays! Help the project by starting to test the development version now! Get feedback on all the new stuff – websockets support, shared call appearances and much more! In Open Source, we need more than developers – testers, documentation writers, bug hunters and of course groupies!
  • Take the development version in GIT for a test drive today and give us feedback in the bug tracker or on the sr-dev mailing list!
  • Help us update documentation. Everywhere where it says “OpenSER” or “SER” or “sip-router” we will need to change to Kamailio
  • Point out missing features as we deprecate the modules in modules_s. There will be a lot of work to merge features and move functionality over to the core module set during the coming weeks, but there is of course a risk of not spotting something that is important for you.
Report issues in our bug tracker!

Let’s create a great new release of Kamailio together!

Tuesday, December 4, 2012

Kamailio 3.3.2 release for small embedded systems

For those who like running Kamailio on routers and/or other small embedded systems, the latest Kamailio release is now available for download thanks to recent work by Ovidiu Sas. There has been many commits lately in order to make Kamailio compile and running properly on these systems.

According to Ovidiu, the old OpenSER cross-compilation worked fine, but since the merger with SER the source code base, the new Makefile system was not fully compatible with cross compilation for many embedded systems, something that Ovidiu now has fixed.

NSLU2 Linux

Kamailio 3.3.2 release for small embedded systems

For those who like running Kamailio on routers and/or other small embedded systems, the latest Kamailio release is now available for download thanks to recent work by Ovidiu Sas.

Saturday, December 1, 2012

New Kamailio Module – Shared Call Appearances

A new module named SCA is now available in the development branch of Kamailio, to be part of the next major release. The module was developed by Andrew Mortensen, from University of Pennsylvania, USA.

SCA (Shared Call Appearances) handles SUBSCRIBE messages for call-info and line-seize events, and sends call-info NOTIFYs to line subscribers to implement line bridging. The module implements SCA as defined in Broadworks SIP Access Side Extensions Interface Specifications, Release 13.0, version 1, sections 2, 3 and 4.

You can read more about the module at:

Monday, November 26, 2012

New kamailio developer: Andrew Mortensen

Kamailio Project is pleased to announce a new developer joining its team: Andrew Mortensen from University of Pennsylvania, USA.

He has contributed a new module: sca (Shared Call Appearances for Broadsoft extensions) – the module is right now in a personal branch, soon to me merged in master branch.

His git commit id is: admorten

An warm welcome and looking forward to his future work within the project!

Tuesday, November 20, 2012

Devel IRC Meeting, Nov 22, 2012

On Thursday, November 22, 2012, 15:00GMT, takes place the developers’ IRC meeting. The main goal is to synchronize everyone regarding the current state of development and set the roadmap to next major release.

A wiki page has been created for this event, collecting details about proposed discussion topics. Feel free to add your notes there.
Everyone is welcome to join the discussions! Just connect to #sip-router IRC channel on server.

Sunday, November 4, 2012

Four Years Kamailio and SER

Fours year ago, was launched as a project to merge the source code of two SIP servers: Kamailio and SER (SIP Express Router) – the announcement is available here. About one year later, the source code trees were merged, opening the testing period for releasing Kamailio v3.0.0, which was done in January 2010. The roots and rules of merging were settled in a face to face meeting in Karlsruhe, Germany, on November 10, 2008 (summaryminutesphotos).

Looking back, those were completely crazy times, but worth it all. There were many developers that spent endless efforts to make the merging happen and keep the best of the two projects. Then, new developers benefited of a more flexible and better scalable core framework that enabled faster innovation and possibility for dozens of improvements.

There were four major releases meanwhile (v3.0.x, v3.1.x, v3.2.x and 3.3.x) – Ohloh statistics reflect better than anything the history and current state of development. From a personal perspective, I wouldn’t have thought four years ago that we will be so far today. Going through some bits of the outcome:
  • asynchronous TCP and TLS
  • asynchronous SIP message processing framework
  • raw UDP sockets and SCTP with multi-homing and multi-streaming
  • onsend and event routes
  • configuration file preprocessor directives
  • extended AVPs
  • configuration message queues
  • step-by-step configuration file debugger and execution trace
  • connectors for memcached, redis and cassandra
  • topology hiding and number portability system
  • embedded Lua, Python and Mono (C#, VisualBasic, etc.) interpreters
  • embedded HTTP and XCAP server
  • broad implementation of SIMPLE presence server (OMA, RCS/RCSe extensions)
  • embedded MSRP relay
  • SDP, XML and JSON operations in configuration file
  • IMS extensions
  • distributed SIP capturing system
  • WebSockets transport layer
  • more you can find in the release notes for each major version and summary about the development branch…
I was deliberately not mentioning any name, because besides the developers, the project’s community had a relevant role in keeping the standards high for stability, performances and innovation, bringing the project at this point. Not to forgot that there is a steady increase of new businesses relying on Kamailio for products and services, ensuring financial power around the project. All these make Kamailio today a truly open source project, combining quality development, with a fantastic community and stronger business opportunities!

There are all premises for an excellent evolution and development of the project from now on! Based on the past, it is going to be hard not to like what is baked by out team members as I write … watch our news closely!

Remember that is open source and you can be part of it – contributing to code, testing, helping on forums or advocating, is first helping you than the others!

Wednesday, October 31, 2012

The 2600Hz project selects Kamailio

The 2600Hz project builds the Kazoo platform, an API-driven telephony platform for carriers. They have announced that they are switching to Kamailo for their SIP proxy needs.
“To our minds, a more active development community means that other users are likely to help solve bottlenecks for us, so we spend less time worrying about our SBC, and more time writing fun new applications!

We’re switching to Kamailio because:
  1. Many of our Major Clients already use it
  2. The Naming Conventions are more sensible
  3. The Pace of Innovation on Kamailio is faster”
Welcome to the Kamailio community 2600Hz! 

Tuesday, October 16, 2012

Kamailio v3.3.2 Released

Kamailio SIP Server v3.3.2 stable is out – a minor release including fixes in code and documentation since v3.3.1 – configuration file and database compatibility is preserved.

Kamailio (former OpenSER) v3.3.2 is based on the latest version of GIT branch 3.3, therefore those running previous 3.3.x versions are advised to upgrade. There is no change that has to be done to configuration file or database structure comparing with older v3.3.x.

Resources for Kamailio version 3.3.2

Source tarballs are available at:
Detailed changelog:
Download via GIT:
    # git clone –depth 1 git:// kamailio
    # cd kamailio
    # git checkout -b 3.3 origin/3.3
    # make FLAVOUR=kamailio cfg

Binaries and packages will be uploaded at:
Modules’ documentation:
What is new in 3.3.x release series is summarized in the announcement of v3.3.0:

Tuesday, October 2, 2012

Upcoming Kamailio Events

Several events related to Kamailio are scheduled in the next months, the organizers will be glad to meet you if it happens to be around or joining the events.
  • Kamailio-VoIP Dinner in Berlin – Monday, October 8, 2012 – come and join many developers and community members for a dinner in Berlin, among them Alex Balashov, Andrei Pelinescu-Onciul, Daniel-Constantin Mierla, Jiri Kuthan and Raphael Coeffic. Exact place and time to be announced soon, send an email to if you want to participate.
  • Astricon 2012 – Kamailio project will be present with it own booth (#20) at the event taking place in Atlanta, GA, USA, October 23-25, 2012.
  • SIP Masterclass – two training sessions organized by and taught by Olle E. Johansson in Stockholm, Sweden and Miami, FL, USA, in October and December 2012
  • Kamailio Advanced Training – two training sessions organized by and taught by Daniel-Constantin Mierla in Berlin, Germany and Miami, FL, USA, both in November 2012

Monday, October 1, 2012

Kamailio Booth at Astricon’12

Kamailio project will be present at Astricon’12 with its own booth in the open source area. The event is taking place in Atlanta, GA, USA, during October 23-25, 2012. The activity at the booth will be coordinated by:
  • Eric Klein – VoIP specialist at Greenfield Technologies , he will be speaking at Astricon on Found in the wild: Telecom Fraud and Security Problems, Wednesday, 11:40-12:15
  • JR Richardson – CTO Ntegrated Solutions – an active community member, using Kamailio as Load balancer and Carrier SIP Trunk Aggregator. He will be speaking at Astricon on Automated Hacker Mitigation, Thursday 11:40-12:15
  • Nir Simionovich – founder of Greenfield Technologies – company focusing on building open source oriented VoIP systems. He will be speaking at Astricon on Asterisk Lock Down – Beyond the Fail2ban, Thursday 10:00-10:35
Kamailio booth number is 20, if you are attending the show, stop by to learn about use cases of Kamailio and what is new in the project. Other companies may join the team till the event takes place.

If you use Kamailio and want to participate at the booth, drop an email to sr-dev [at] .

Thursday, September 20, 2012

Kamailio Advacned Training, Nov5-8, 2012, in Berlin, Germany

The next Kamailio Advanced training in Berlin, Germany is scheduled for November 5-8, 2012 - a class that starts from the ground and gets to advanced topics of routing SIP with Kamailio server.

The class is organized by Asipto and coordinated by me, you can read more details about the content and registration at:

Monday, September 3, 2012

Kamailio at 11 Years

Today the project celebrates 11 years since the first commit was done, back on Sep 3, 2001. Just very few months later I joined the FhG Fokus Institute in Berlin as network communications researcher and since then I am involved in the project, changing meanwhile my job to a VoIP consultant.

I collected some facts related to the development during the past year and made them available at:
Looking forward to the 12th anniversary.

Monday, August 6, 2012

Kamailio 3.3.x and Asterisk 10.7.0 Realtime Integration Tutorial

I released an update to my series of Kamailio and Asterisk Realtime Integration, using the latest stable versions of the two projects, respectively 3.3.1 and 10.7.0. You can find it at:


The tutorial focuses on how to use Asterisk's database structure to perform authentication in Kamailio SIP server, along with user location, nat traversal, instant messaging, presence, a.s.o., offloading processing from Asterisk. Asterisk will still handle all the calls, enabling rich telephony such as MoH, transcoding, ring back, IVR, etc.

Reusing as much as possible the Asterisk database makes the architecture presented in the tutorial easy to be applied to existing installations, without losing management interfaces or other admin tools.

Hope it is useful for many folks out there.

Sunday, August 5, 2012

Kamailio at Cluecon 2012 – Find Me & Chat

Luis Guaman of is presenting Find Me & Chat service at Cluecon 2012 (talk: Kamailio as Geo Location Server), next week in Chicago. The service is using Kamailio as communication engine.

Find Me & Chat enable registered users to publish their location and discover other users nearby that are ready to chat. Upon SIP registration, the location is pinned using GeoIP and a list of contacts within 1km distance is returned. Your buddy list is subscribing to your coordinates and get notifications whenever you are in the neighborhood.

The client side can run the mobile application on iOS, Android or Symbian, visualizing on a map who is near by. Using SIP and Kamailio makes the service fully scalable and secure, by use of TLS, as well as it enables many options to extend or integrate with existing real time communication services for voice, video, presence, a.s.o.. All over, a new service already used by ten thousands of users, showing the flexibility and the potential of quick roll out of social interaction communication system on top of Kamailio and SIP.

In relation to Cluecon 2012and Kamailio, worth to mention the talk about Homer project, the support for HEPv3 being started in Kamailio GIT repository.

Regarding next option to meet Kamailio around the world, in September, two Kamailio-related events are scheduled in Alkmaar, Netherlands (Kamailio Practical Workshop, Sep 10-12, 2012) and Seattle, WA, USA (Kamailio Advanced Training, Sep 24-26, 2012).

Thursday, August 2, 2012

Kamailio v3.3.1 Released

Kamailio SIP Server v3.3.1 stable is out – a minor release including fixes in code and documentation since v3.3.1 – configuration file and database compatibility is preserved.
Kamailio (former OpenSER) 3.3.1 is based on the latest version of GIT branch 3.3, therefore those running previous 3.3.x versions are advised to upgrade. There is no change that has to be done to configuration file or database structure comparing with older v3.3.x.
Resources for Kamailio version 3.3.1
Source tarballs are available at:
Detailed changelog:
Download via GIT:
  # git clone –depth 1 git:// kamailio
  # cd kamailio
  # git checkout -b 3.3 origin/3.3
  # make FLAVOUR=kamailio cfg
Binaries and packages will be uploaded at:
Modules’ documentation:
What is new in 3.3.x release series is summarized in the announcement of v3.3.0:

Thursday, July 19, 2012

Kamailio v3.2.4 Released

Kamailio SIP Server v3.2.4 stable is out – a minor release of the previous stable branch including fixes in code and documentation since v3.2.3 – configuration file and database compatibility is preserved.
Kamailio (former OpenSER) 3.2.4 is based on the latest version of GIT branch 3.2, therefore those running previous 3.2.x versions are advised to upgrade. There is no change that has to be done to configuration file or database structure comparing with older v3.2.x.
Note that the latest stable release series is 3.3.x (at this moment the latest packaged version being v3.3.0) – Kamailio project is officially maintaining the latest two stable branches to allow long term support (approx 2 years) for all deployments. New installation should start with the latest stable version.
Resources for Kamailio version 3.2.4
Source tarballs are available at:
Detailed changelog:
Download via GIT:
  # git clone –depth 1 git:// kamailio
  # cd kamailio
  # git checkout -b 3.2 origin/3.2
  # make FLAVOUR=kamailio cfg
Binaries and packages will be uploaded at:
Modules’ documentation:
What is new in 3.2.x release series is summarized in the announcement of v3.2.0:

Saturday, July 14, 2012

Kamailio v3.1.6 Released

Kamailio SIP Server v3.1.6 stable is out – this is the last minor release packaged from branch 3.1. It includes fixes in code and documentation since v3.1.6 – configuration file and database compatibility is preserved.
Important note: this release has been done to mark the end of official packaging of 3.1.x series. As of July 18, 2012, with the release of v3.3.0, the last two stable branches are 3.2 and 3.3. If you look to install the latest stable version, you have to use v3.3.0 at this moment.
Kamailio (former OpenSER) 3.1.6 is based on the latest version of GIT branch 3.1, therefore those running previous 3.1.x versions are advised to upgrade. There is no change done to configuration file or database structure.
Resources for Kamailio version 3.1.6
Source tarballs are available at:
Detailed changelog:
Download via GIT:
 # git clone –depth 1 git:// kamailio
 # cd kamailio
 # git checkout -b 3.1 origin/3.1
 # make FLAVOUR=kamailio cfg
Binaries and packages will be uploaded at:
Modules’ documentation:
What is new in 3.1.x release series is summarized in the announcement of v3.1.0:

Tuesday, July 10, 2012

Kamailio worldwide in August and September

If you want to meet people around Kamailio project, it might be your chance to be nearby several locations around the world in August and September:
  • Chicago, IL, USA – August 7-9 – during Clucon Conference, Kamailio will be touched in at least two presentations – the talks about mobile geo-location server by Luis Guaman and the one about Homer project by Alex Dubovikov
  • Alkmaar (or Amsterdam Area), The Netherlands – September 10-12 – during Kamailio Practical Workshop, coordinated by Daniel-Constantin Mierla
  • Seattle, WA, USA – September 24-26 – during Kamailio Advanced Training, coordinated by Daniel-Constantin Mierla and Flowroute
Contact us via sr-users community mailing list to get in touch or announce other Kamailio-related events to be listed here.

Monday, July 9, 2012

SIP over Websocket support integrated into Kamailio

Peter Dunkley, an active member of the Kamailio developer team, has integrated support for SIP over the WebSocket protocol into Kamailio. It will be part of the next release and exist only in the developer code today. There are still some issues to sort out to get a production-ready setup. SIP over Websockets, an IETF draft composed by a number of members of the SER/Kamailio/SIP router community – Iñaki Baz Castillo, Victor Pascual and José Luis Millán Villegas – is still a moving target and hopefully on it’s way to become an RFC.
From the IETF draft abstract:
   The WebSocket [RFC6455] protocol enables messages exchange between
   clients and servers on top of a persistent TCP connection (optionally
   secured with TLS [RFC5246]).  The initial protocol handshake makes
   use of HTTP [RFC2616] semantics, allowing the WebSocket protocol to
   reuse existing HTTP infrastructure.

   Modern web browsers include a WebSocket client stack complying with
   The WebSocket API [WS-API] as specified by the W3C. It is expected
   that other client applications (those running in personal computers
   and devices such as smartphones) will also run a WebSocket client
   stack.  The specification in this document enables usage of the SIP
   protocol in those new scenarios.

   This specification defines a new WebSocket sub-protocol (section 1.9
   in [RFC6455]) for transporting SIP messages between a WebSocket
   client and server, a new reliable and message boundary transport for
   the SIP protocol, new DNS NAPTR [RFC3403] service values and
   procedures for SIP entities implementing the WebSocket transport.
   Media transport is out of the scope of this document."

The Kamailio SIP over Websocket support

Some customisation of the websocket module is possible through modparams, but for most users the defaults should be OK.  The WebSocket module uses
the xhttp and sl modules for the initial handshake, and (unless you have both a Kamailio installation and WebSocket SIP client supporting GRUU, Outbound[1], and Path[2]) nathelper for request routing and the core force_rport() function for response routing (a new nat_uac_test() has been added to detect whether a message has arrived on a WebSocket).  There is an example kamailio.cfg in the websocket module directory.
  • [1] Kamailio does not currently support Outbound
  • [2] I have not updated the Path module for WebSockets
I believe that, once Kamailio supports Outbound and WebSocket support is added to the Path module (and you have a SIP over WebSocket client that
supports this), it will be possible to use the websocket module without the nathelper module and force_rport() and without needing to change the websocket module or Kamailio core code.
If you want to use secure WebSockets (wss) as well as ordinary WebSockets just configure TLS and listen on an appropriate port.
I have added WebSocket support to some modules, but there are definitely going to be others (modules/lcr, modules/sipcapture, modules_k/nat_traversal, modules_k/path, modules_k/seas, and modules_k/snmpstats, at least) that need updating too.  WebSockets is an unusual transport, so I have put a few notes together for anyone who needs to use it in the code (including adding support to additional modules):
  • A WebSocket server cannot initiate a WebSocket connection.  So a WebSocket connection (over TCP or TLS) is like a TCP/TLS connection coming from behind a NAT.  This is why nathelper aliasing and force_rport() is used for the routing, and “set_…_no_connect()” is always used (it’s set within the websocket module).
  • WebSocket (PROTO_WS) and secure WebSocket (PROTO_WSS) connections are just upgraded TCP and TLS connections, so there are no listening sockets for PROTO_WS and PROTO_WSS.  This means that, when deciding on what transport is being used, you need to look at the proto set in the tcp_connection, receive_info, and/or dest_info structure for the message -looking at the socket_info structure (that the message has arrived on or will be sent on) will not give you the right answer.
  • Although WebSocket (PROTO_WS) and secure WebSocket (PROTO_WSS) are different internal protocols there is only one SIP transport type for both ”;transport=ws” (WS and WSS are explicitly used in Via: headers though).  This means that you can’t tell whether the transport parameter in an R-URI, Route/Record-Route, or Contact-URI is for WebSockets or secure WebSockets.  As long as the message makes it into the WebSocket module everything will be OK as that module sorts it all out, but it has led to slightly more complex checks being required in some of the code relating to record-routing to handle this – and it may have an effect on other modules too.
Please give the new module a go and let Peter know (by writing to the developer mailing list) about any issues you find!

Friday, July 6, 2012

Two new Diameter-IMS modules added to Kamailio repository

Jason Penton of Smile Communications has just  committed CDP and CDP_AVP modules.
CDP (C Diameter Peer) allows Diameter communication from Kamailio and CDP AVP is a helper module for various applications on top of CDP (C Diameter Peer). CDP AVP module adds support for the following applications:
  • Various base AVPs (implemented in base.h) for RFC3588 base AVPs
  • Base AVPs (implemented in nasapp.h) for RFC4005 base AVPs
  • Diameter Credit Control App (implemented in ccap.h) for RFC4006 AVPs
  • EPC (implemented in epcapp.h) for 3GPP Rx, Gx(x) interface AVPs – see TS29l061, TS29.212, TS29.214, TS29.272 and TS29.299
  • IMS (implemented in imsapp.h) for 3GPP IMS AVPs, Cx, Dx, Sh interfaces – see TS29.229 and TS29.329
You can read more about them in the module documentation:

Monday, June 25, 2012

Packages for Kamailio v3.3.0

Debian/Ubuntu packages are ready in the usual APT repository, including nightly builds of the GIT branch 3.3, details at:
RPMs for CentOS/RedHat, Fedora, openSuse (various versions for each OS) are now available on openSuse build factory repository, details at:
There you find the note about Peter Dunkley’s repository, which has also builds for Raspberry Pi.

Tuesday, June 19, 2012

Save the bits! Act now!

It's about the time to announce publicly one of the most important world wide initiatives - "SAVE THE BITS!" Foundation.

If you are not familiar with the concept, the BIT is the smallest entity of computers and Internet data ecosystem, wikipedia is a good place to read about it.

There are many species of bits out there, probably over-exceeding the animal and plant species on the earth. The bits are growing in colonies, by grouping themselves they become live entities, for example being able to move, execute tasks, perform math operations faster than humans - one can clearly hear or even see different kinds of such colonies.

The first focus of our foundation is to take care of real time communication bits, especially those related to rich communication services (RCS - e.g., VoIP or Internet-based telephony, instant messaging, tweeting, a.s.o.). It's not about an endangered category of bit species in terms of high risk of extinction, but about the maltreatment they have to suffer.

IP based real time communication involves migrations of a lot of bits from one device to another one. This process can be very stressful for the tiny, the little cuties bits.

Just think about what kind of abusive, degrading and mercy-less situations the bits have to tolerate, such as:
  • did you know that each bit is electrocuted? And not only once, but by each device it is passing through, like computers, routers, switches, servers, iphones, ipads, etc. During RCS sessions each bit can get thousand of electricity shocks. Imagine that for you and think how you would tolerate that. Stop using RCS, save the bits from paintful treatments!
  • you are in Miami on the beach and start calling your friend in Alaska? Have you thought about thermal shocks? You are sending bits from a very hot temperature to freezing in miliseconds. Did you wear them with proper clothes? Do you like to be parachuted naked in Alaska during Christmas?
  • how many times you've got a call before waking up? Did you brush your teeth before answering? Your mouth flow slaps the bits straight in the face. Eating and drinking crap, smoking & co flood the bits with terrible smell
  • have you heard anyone yelling at the phone? Dirty words, swearing! It's the feeble bits that have to endure all of that
  • do you know that many RCS bits are dying at premature age? Many devices destroy them in short after receiving. Fortunately governments started to act and require long term protection for them, forcing communication companies to keep RCS bits on high quality comfort storage systems (like 5 stars hotels for people), where state agencies (and 'good-will' hackers) can keep an eye on them whenever they wish
It is not honorable for our civilization to allow such maltreatment and humiliations to any of exiting beings on out planet, the colonies of bits, especially the ones in RCS are alive, they move, you can hear them, you can see them -- they are all your voice and video session.

Even it was an initiative that acted pretty discreet so far, we've got lot of support. Recently large mobile operators publicly announced new policies to protect the innocent RCS bits on their infrastructure (e.g., operators from Sweden, Spain, Germany, just to name very few),  by completely blocking usage of VoIP or overcharging for it. In several locations, where this unacceptable situation was truly understood, using bits for VoIP and RCS has been made a crime by legislation, like facing up to 15 years in prison for infringement. Even many vendors take our side by deploying ALG guardians in their home-routers to stop VoIP from working.

Moreover, each human must be aware and join the movement. Don't tolerate abuses again the bits! Use analog telephony and operator's voice plans, by that bits are happy, operators are happy, you are going to be [poor, but] happy! Yeah, everyone happy!

It is not an easy fight, we could call it war, a tough one, but there is visible progress. The foundation urges all mobile and fixed operators to act and start protecting RCS bits, NOW! Block VoIP, block RCS!

To join the foundation or donate for the cause, contact the author of this blog! Remember, each bit matters!

And don't forget to spread the word to the world! Leave a comment and tell us how you protect the bits! Tell us what torturres against bits you witnessed so far!

Facebook fan page, website and other online resources will be launched soon!

Full disclaimer - I do work in RCS business, one of my main activities being the development of an open source SIP server application (Kamailio), used to provide rich communication services. In more than 10 years of activity in this sector, I came to deep understanding of this abusing issue against innocent bits, fighting from inside and offering delicate care to the bits passing through Kamailio, for example:
  • all received bits are stored in clean buffers
  • we have our own friendly manager to care of the bits in memory storage
  • we don't allow unaccredited persons to send us bits, authenticating and carefully checking each packet we receive
  • we don't send the bits to untrusted party, in this way being sure the bits are well cared at destination
  • we add extra protection layer (based on strong TLS security) whenever we have to send the signaling bits through unknown paths

Monday, June 18, 2012

Kamailio v3.3.0 Released

Kamailio SIP Server v3.3.0 is out – a new major release with a very large number of new features and improvements.
On June 18, 2012, Kamailio (OpenSER) 3.3.0 has been released – this release is a result of about 6 months of development and 2 months of testing from the teams of Kamailio (OpenSER) and SIP Express Router (SER) projects.
This version comes with 7 brand new modules in addition to a lot of fresh features in core and old modules. Continue reading full release notes at:
Enjoy SIP routing in a secure, flexible and easier way with Kamailio v3.3.0!

Asipto has recently announced two new training and consultancy public events where you can learn how to use Kamailio to build your rich communication services, read more about them at:

Friday, June 15, 2012

New Kamailio Developer – Vicente Hernando

Kamailio SIP Server project announced Vicente Hernando as a new registered developer – he submitted lately very useful patches to ndb_redis module (e.g., array support in replies, redis free function for config), new ones being on the pipe — watch our mailing list where he is going present better his plans or see his contributed commits.

Mainly he will be in charge with ndb_redis module, being a heavy user of it, but there have been submissions from him to other parts of code, we welcome improvements and new good features anywhere!

Welcome and everyone is looking forward to his contributions!

Thursday, June 14, 2012

Updates of Kamailio Debian Packages

Courtesy of Jon Bonilla, Debian packaging got a bit of updates in preparation to release Kamailio v3.3.0.

Next is his announcement on the Kamailio's users mailing list.

With the upcoming release of version 3.3.0, we have updated our Kamailio Debian repositories.
  • Debian 5.0 “lenny” will no longer be supported. This means that no more nightly builds for this distribution will be triggered. We’ll provide builds for stable releases of 3.0, 3.1 and 3.2 branches though.
  • Ubuntu 12.04 “Precise” is now supported for nightly builds and 3.2+ versions. Next time a 3.2.x version is tagged we’ll provide the build. Meanwhile you can download the nightly build of 3.2 branch for this distro.
  • There will be no more nightly builds of 3.1 branch. We’ll provide stable builds in case another version is released though.
  • We now support nightly builds of branch 3.3 (same as 3.2 and master). You can test latest git versions of branch 3.3 until a stable release is built using these repositories:
deb squeeze main
deb wheezy main
deb lucid main
deb precise main

All the information and repositories can be found in Kamailio wiki as usual:

Wednesday, June 6, 2012

Kamailio Packages for Raspberry Pi

Peter Dunkley, a registered Kamailio developer, has built some RPMs of upcoming Kamailio 3.3.0, including for the Raspberry Pi (running Fedora 17).

If anyone wants to try them, they are available via Yum repository:
The post at Peter Dunkley’s blog:

Wednesday, May 30, 2012

Remarks about Kamailio at LinuxTag

The participation of Kamailio at LinuxTag this year was filled with several related events and, as usually, a great opportunity for community meeting and social networking.

Again, five of the management team members were at the booth in various occasions, answering questions and making demos: Andreas, Carsten, Daniel, Henning and Ramona. Marius and Mario from 1 & 1 Romanian branch and Germany completed our staff. We shared the space with related SEMS project, Stefan, Rafael, Vladimir and Alena being there for it.

On Wednesday, Carsten had his presentation in the LinuxTag Conference track, Daniel did the Project Fast Forward presentation at Open Source Arena. On Thursday, Daniel gave one hour workshop about how to install Kamailio and configure it for secure unified communication services – a well received event, the seminar room being fully attended. On Saturday, the Project Fast Forward presentation was repeated by Daniel. Links to presentations:
Most of visitors were from Germany, but we had several people coming from neighborhood countries, like Poland and Denmark.

We learned about interesting deployments using Kamailio, notable being a relatively large Instant Messaging and Presence communication platform, using the SIP SIMPLE presence modules (including RLS) – the average online user base being 40 000, with peaks at 60 000. Out there are much more larger Kamailio deployments, but mainly targeting telephony services. This deployment proves that SIP SIMPLE presence extensions are becoming more and more an attraction, not being a dead direction. Our upcoming Kamailio v3.3.0 has a lot of improvements in scalability of presence related modules, which along embedded XCAP server and MSRP relay modules, makes Kamailio the most complete SIP SIMPLE implementation out there.

Now LinuxTag 2012 is gone, time to focus on the release of Kamailio v3.3.0!

Tuesday, May 29, 2012

Planning for Kamailio v3.3.0 Release

The volume of reported issues on mailing lists and tracker has been close to zero lately, suggesting that upcoming Kamailio v3.3.0 is going well. Therefore it’s time to set the timelines for the new major release – by now there is more than one month since the development was frozen and testing phase started.

In my side, I (Daniel-Constantin Mierla) have 3.3.0 running in a production-like environment for about 4 weeks without problems and another one in a pre-production phase.

Having in consideration all these, the proposed date to branch 3.3 is set to June 11, 2012. If everything looks very good at that time (or actually few days before), we may do the release also by that date or next day. If not, it can be done a week later.

Branching will allow new features development on git master, there are already several contributed patches with new features flowing around on mailing lists and tracker.

Therefore, if you are aware of any issue, please report asap on mailing list (sr-dev [at] or tracker (preferable –

Other suggestions about planning the release 3.3.0 are welcome, as well.

If you want to check the draft of new features in v3.3.0, see:
Testing and feedback is very appreciated!

Friday, May 25, 2012

Kamailio and SEMS in a Risk-of-Life Service

An interesting story sent by Jeremy A. on SEMS mailing list about use of Kamailio and SEMS in emergency services.

This is a belated report on the use of SEMS in a risk of life service.

The system uses Kamailio in a distributed architecture of dozens of Fire & Rescue stations. This is heavily based on distributed and replicated DNS.

A single ’911′ style headquarters has duplicate hot swap-over control rooms at other locations.
The headquarter and alternate posts have servers that service HQ operator positions with SIP phones. These provides sidecar indication of F&R Station state for up to 64 F&R stations – using BLF. These phones are hooked into an integrated analogue audio management system.

Each Fire and Rescue station has an embedded SIP based controller that runs Kamailio and proprietary software to control the F&R station electrical and safety systems as well as provide public address functions to alert the F&R staff of a new emergency. These PA announcements are SIP based using a DSL network and are live from the HQ positions, plus computer synthesized voice, as well as alerting tones.

Each station also has multiple SIP phones for in-station and station to station calling.
The network is decentralized, so failure of the central control system still allows point to point communications between Fire and Rescue stations.

The headquarter systems uses SEMS as the primary operator manager to perform multiple simultaneous deployment calls to remote Fire and Rescue stations. SEMS is used to create a dynamic conference between an operator and multiple Fire & Rescue stations. These are automatically initiated by SEMS and answered by the F&R embedded systems. This means an operator can broadcast a deployment message and initiate station control activities at up to five stations (fifth alarm) This is only constrained by the bandwidth available at the headquarters. Our SEMS packages have been designed to handle non-answered calls to the conference and provide operator indication by ‘SMS’ messages to the handsets and audio feedback.

The system provides full forensic recording by using rtpproxy at all locations. These recordings are archived by an out-of-band process.

Control of the system is purely SIP based – so every item in the system is a SIP based entity. This includes servers, embedded systems, and phones.

The phones are physically integrated into operator positions that also handle PSTN, PBX, and radio traffic. The interface is purely keyboard on the operator phones.
Options for integration of the SIP system into CAD (Computer Aided Dispatch) are obvious. The only drawback is the rusty and ancient systems and the unbelievable process required to get approval to integrate.

The system as provided provides at least 5 nines reliability. Probably a lot better. The only downside is the DSL network (provided by others at amazing expense) that provides a system with a lousy 2 nines reliability. We are in the process of developing an offering using redundant DLS/3G routing to improve this.

The field stations are a hybrid Centos 5/Slax system running out of flash. The HQ systems are straight Centos 5 systems running off disk or off flash. Future versions will be pure Centos out of flash with no fancy memory overlay – flash is well good enough.

The system has been live for over a year with no major issues. I can’t say how many lives have been saved, but certainly quite a few. At least we haven’t been sued yet!

Wednesday, May 16, 2012

LinuxTag Workshop: Building secure UC service with Kamailio

Daniel-Constantin Mierla (n.r. the blog owner), co-founder and member of management board for Kamailio project, will provide one hour workshop during LinuxTag Conference & Exhibition, Berlin, Germany. Scheduled on Thursday, May 24, 2012, the workshop will focus on how to build yourself a secure unified communication platform, using Kamailio in server side and Jitsi as application in client side.

You can read more about the workshop at:
As you get at LinuxTag show, just drop by at Kamailio project booth to have a chat, you can meet five of the management team members, along with other developers, community members as well as the friends from SIP Express Media Server (SEMS) project. The booth is located in Halle 7.2b, Stand 278.

Monday, May 14, 2012

LinuxTag Talk: Rich Communications using SIP and Kamailio

Carsten Bock, developer and member of management team for Kamailio project, will give a presentation at LinuxTag Conference & Exhibition in Berlin. The date of the talk is Wednesday, May 23, 2012.
Scheduled about noon, you can learn how to use Kamailio build platforms for providing RCS/RCSe and IMS-based services (communication beyond voice using SIP – instant messaging, presence, IPTV, a.s.o.).
You can read more about the presentation at:
As you get at LinuxTag show, just drop by at Kamailio project booth to have a chat, you can meet five of the management team members, along with other developers, community members as well as the friends from SIP Express Media Server (SEMS) project. The booth is located in Halle 7.2b, Stand 278.

Tuesday, May 8, 2012

SEMS v1.4.3 Released

On the 4th of May, 2012, the SIP Express Media Server project announced the availability of the SEMS 1.4.3 release.

This is a SEMS 1.4 series bugfix release and should be a drop-in replacement to 1.4 installations. For details of the changes see:
Sources are available at:
More about the project:
SEMS is a project tight related to Kamailio, started at the same institute in Berlin, Germany, having many shared developers. It can be used together with Kamailio to provide services such as voicemail, audio conferencing, announcements, IVR menus or back-to-back user agent functionality.

Monday, May 7, 2012

Kamailio Project at LinuxTag 2012

Kamailio project will be present with an exhibition booth at LinuxTag show in Berlin, Germany, May 23-26, 2012.

If you visit the event, drop by stand 278 in hall 7.2b and have a chat with members of management team, developers or other community members. You can see demos of services and solutions from companies such as Asipto, NG Voice or Sipwise.

SIP Express Media Server (SEMS), a sister project, will be collocated with us, giving you the opportunity to learn how to use it together with Kamailio in order to provide media services such as voicemail, audio conferencing, IVR menu or back-to-back user agent functionality.

Thursday, April 26, 2012

Development frozen for Kamailio v3.3.0

Next major version of Kamailio SIP Server, to be numbered 3.3.0, entered in the last phase before release – development of new features has ended and testing phase began.

After about 1-1.5 months of testing, the new release should be out, just in time for the summer.
Here are the guidelines to install this version, which as this time is still the GIT master branch:
You can see the draft of the list with the new features at:
Feedback will be appreciated, email us to sr-dev [at]

Wednesday, April 25, 2012

New developer joins Kamailio team

Kamailio project got a new developer on board: Richard Fuchs.

Richard has been working at Sipwise for over 4 years now and has already contributed many new features and bug fixes for our Kamailio packages.

Among his contributions are the period matching in tmrec module, r-uri matching in lcr module and some IPv6 patches.

Thursday, April 19, 2012

Kamailio v3.2.3 Released

Kamailio SIP Server v3.2.3 stable is out – a minor release including fixes in code and documentation since v3.2.2 – configuration file and database compatibility is preserved.

Kamailio (former OpenSER) 3.2.3 is based on the latest version of GIT branch 3.2, therefore those running previous 3.2.x versions are advised to upgrade. There is no change that has to be done to configuration file or database structure comparing with older v3.2.x.

Resources for Kamailio version 3.2.3

Source tarballs are available at:
Detailed changelog:
Download via GIT:
  # git clone –depth 1 git:// kamailio
  # cd kamailio
  # git checkout -b 3.2 origin/3.2
  # make FLAVOUR=kamailio cfg

Binaries and packages will be uploaded at:
Modules’ documentation:
What is new in 3.2.x release series is summarized in the announcement of v3.2.0:
FYI, next major release, v3.3.0, is planned to be out before mid of summer – you can check the draft of the list with new features at:
Btw, since Kamailio is a Hawaian word (kama’ilio = converse), 3.2.3 version translates to ‘Ekolu-’Elua-’Ekolu.

Monday, March 26, 2012

Next Events in Berlin

After the Easter holidays, few events related to our project will take place in Berlin. If it happens to be around, drop by and meet developers or other community members.
  • Social Networking Event, during the evening on the 25th of April – planned during Kamailio Advanced Training, it is a good chance to catch up with latest news about the project and its plans to next major release 3.3.0
  • Booth at LinuxTag – the project will be participating again at the biggest Linux show in Germany, May 23-26. We will have our stand there, every day several people will be available to make demos and show our latest cool features
You can contact us via email on mailing list or via contact form.

Friday, March 23, 2012

ITSPA Awards 2012 – Open Source VoIP Projects

ITSPA UK has unveiled the winners of its 4th annual Awards, an event designed to celebrate innovation and best practice in the VoIP industry.
The event took place at the House of Commons Members Dining Room, Palace of Westminster, London, on 21st of March 2012, hosted by Dr Julian Huppert MP, Vice Chair of the Parliamentary Internet Communications and Technology Forum.

With main focus on awarding IP Telephony businesses in UK, this year they introduced a new category, “Members’ Pick Award”, to endorse something or someone that has provided real value to VoIP Industry. Open Source VoIP Projects as a group was introduced in this category, made it do the final and ultimately won the category.

I (Daniel-Constantin Mierla, as co-founder and member of management board of Kamailio SIP Server project) attended the event and was selected to pick up the award.

The members of ITSPA acknowledged the major role of open source VoIP projects for their businesses, many of them would have not existed without these projects. Awarding Open Source VoIP Projects as a group was decided because most of the deployments combine several projects to build a complete IP telephony platform. It is a common practice to mix applications such as Kamailio/SER, Asterisk or FreeSwitch to build large VoIP systems and provide a broader range of services.
It is yet another confirmation of reliability and quality solutions provided by Open Source environment for real time communications.

This is also an opportunity to send best wishes and regards to all the people behind Open Source VoIP Projects, developers or community members, that dedicate work and time to develop and improve the quality of the applications and act in the true spirit of Open Source: freedom and fairness!

Tuesday, February 28, 2012

Kamailio 2011 Awards

End of winter, therefore it's now time for the 5th edition of Kamailio Project Awards, granted for the activity during 2011, like in the past, each category has two winners.

First, I want to thank to everyone contributing to and using Kamailio/SER during 2011, their effort made possible to release a new major version with tons of new features - see release notes for v3.2.0.

The unconventional winners of 2011 are:
  • the project itself - it succeeded to survive over 10 years since it started. There were many critical moments in the past (forks, renames and such), but it went through, becoming even stronger now, with an exceptional support from an amazing community.
  • the development team - it succeeded to get on board lot of new contributors. This made it very hard to select the winners for Developer Remarks awards. According to Ohloh statistics, there were over 30 registered developers contributing code during the past 12 months.

The new category this year is Friends of Kamailio, for persons that helped for many years in various occasions, in background or foreground, and were always promoting and saying good stories about the project.

Next are presented the categories and the winners.

  • CSDN - the "Chinese Software Developer Network" - for spreading out news and tutorials about Kamailio, our Asian user base grew substantially lately.
  • Jason Penton and Richard Good (the two being colleagues, were tied together here) - for many useful posts of how to deal with Kamailio on Solaris. The two are contributing code to the IMS extensions in our project.
Related Projects:
  • Gemeinschaft - a project that aims to provide a secure, scalable and easy to use open source PBX platform built with Kamailio and FreeSWITCH, developed by Amooma and sponsored by the German Federal Office for Information Security (BSI).
  • Homer - SIP Capturing Server started by Alexandr Dubovikov, developed first time for Kamailio SIP Server project and adopted by other VoIP applications. In one side, Kamailio can be configured to mirror the SIP traffic, on the other side Kamailio can receive the mirrored traffic and save it to database from where it can be searched for various keywords via a web interface. Then, it can build diagrams from results or the results can exported in pcap format.
Technical Support:
  • Jon Bonilla - for setting up the Debian/Ubuntu APT repositories, updating the .deb specs and maintaining them. It made the spreading through packages and installation lot more easier for these distributions.
  • Laura Testi - for substantial feedback, testing and many patches on our SIP SIMPLE Presence extensions. I I would award development topics, SIMPLE Presence extensions would have been among the top ones in 2011 (user presence, dialog states (aka blinking lamps), resource list services, embedded XCAP server, SIP-XMPP presence gateway). Laura helped a lot to improve and develop further these components.
 New Contributions:
  • dmq - Distributed Message Queue on top of SIP - the module presents huge potential and building a native distributed platform with Kamailio, personally I expect several modules to use it soon as underneath layer for data exchange between SIP server nodes. The developer is Marius-Ovidiu Bucur.
  • json - JSON Parser and JSONRPC Client - json format has become very popular, being able to interact directly from the config file, opens the way to integrate easier with backend structures used for Web 2.0 services. As a plus, JSONRPC client uses the asynchronous processing framework added starting with v3.0.0. The developer is Matthew Williams.
Developer Remarks:
  • Peter Dunkley - he and Crocodile RCS team contributed major enhancements to SIP SIMPLE Presence extensions towards RCS/RCSe (e.g., user presence services, resource lists services, embedded xcap server), as well as scalability. For a fair acknowledgement of contributed work, I am mentioning also his colleagues Paul Pankhurst, Hugh Waite and Andrew Miler.
  • Timo Terras - by contributing SQLite database connector, running Kamailio on embedded systems entered in a new era. The primary choice in the past as DB backed for embedded devices was dbtext module, but it has limitations dealing with large records. With SQLite, any module can be fully functional on very small devices, from user authentication and authorization, sqlops, to location and presence services.
  • Carsten Bock - for participating at, organizing and sponsoring several events related to Kamailio SIP server as well as publishing blogs and news about the project.
  • Henning Westerholt - another year with Henning on traveling to most of our public events, speaking about the over 3 000 000 VoIP users platform he is involved in, with special credits for taking care of our LinuxTag 2011 presence, booth and slides.
VoIP Services:
  • Portugal Academic Network - for running a large grid of Kamailio and Asterisk servers (about 500) to provide communication services. Credits to Ruben Sousa (at Astricon 2010) and Olle E. Johansson (at 10 Years SER event - the slides from previous link) for sharing their experiences in deploying this VoIP network with us.
  • Telio - one of the oldest companies with development contributions to SER/Kamailio project, Telio continued to have in 2011 good revenue growth. I am adding also a special remark for its movement towards mobile networks, with launch of Goji application.
Business Initiatives:
  • Crocodile RCS - one of the few players in the vendors market with a clear target for Rich Communication Services, their team was very active in the development process of Kamailio during last year. Mobile operators are in a rush to roll out value added services, especially in the social networking area - real time instant messaging and presence are going to play a crucial role for operators in their battle with the big Internet companies.
  • Frafos - with a business oriented mainly to SIP Express Media Server (SEMS), the project started in the same place as SIP Express Route (SER), having many common developers, the company can help deploying addition components in Kamailio-based VoIP platform, to achieve functionalities such as media server (voice mail, conferencing, IVR), back to back user agents or session border controllers.
  • 10 Years SER - the project crossed first time the age of 2 digits, a good opportunity to meet and look at the past, present and future. About 50 people spent a great day at FhG Fokus in Berlin, from first developers to our latest users. There were 15 presentations during the day and a relaxing grill party in the evening.
  • LinuxTag - we were several time in a raw at LinuxTag with a booth for the project and presentations about it, but the last year we had the opportunity to meet in the same place 5 out of 11 members of the management team, pretty rare situation as we are distributed over many countries.
Academic Environment:
  • Columbia University - for using our SIP server to conduct various research projects since 2002, results from some of them being useful and relevant for development and scalability of SER/Kamailio platforms. The Green VoIP paper showing interest results, like ability to cope with about 43 000 active TLS connections on a server with 2GB memory allocated for the SIP server. Worth to mention that Jan Janak, one of the major contributors to the SIP server, has been involved in some of the research projects.
  • FhG Fokus Research Institute - for being a host of 10 Years SER event, but overall for starting in 2001 and funding the development of our SIP server project for many years. Also, during the recent past years, Fokus hosted several developer meetings. Another relevant contribution to the VoIP environment coming from Fokus is the OpenIMSCore project, developed on top of SER/Kamailio, several components from it being integrated in our project.
Friends of Kamailio:
  • Olivier Taylor - for helping to organizes our meetings in Brussels, at Fosdem events. By now, for the last several years in a raw, Olivier did his best to select nice places to meet and enjoy great time with folks around Kamailio project and VoIP arena.
  • Suzanne Bowen - an exceptional friend of open source telephony applications, Suzanne was ready to help always with written interviews and podcasts, free invitations to conferences and exhibitions, setting up connections, recommendations and discussions to various business entities, to reveal in the best way the benefits and strong points of Kamailio.
As of Personal Facts related to the project, I continued to release complete tutorials of using Kamailio, very useful and actual would be Bridging IPv4-IPv6 VoIP Networks, Secure Communications with Kamailio (building own Skype service alternative) - see all of them at:
This is it for 2011. If you want to check the previous turn of awards, visit:

Monday, February 27, 2012

Social Networking Event, London, March 6, 2012

Several folks involved in Kamailio SIP Server and VoIP in general (including me, Daniel-Constantin Mierla, co-founder Kamailio) are organizing a social networking event in London, during the evening of 6th of March.

If you want to participate, send us a short note via registration form:
Be sure your email is valid in order to send you the details of the location.

The event will be dinner/pub-drinks style, with the goal of discussing the latest developments of Kamailio project as well as what is new in VoIP and Unified Communications world.

Wednesday, February 15, 2012

Unified Communications Expo 2012

UC Expo 2012 takes place in London, UK, between March 06-07, 2012. Asipto representatives will be present this year as well at the event, meeting many of our UK customers base that will exhibit at the show.

UC Expo describes itself as the show mirroring the diversity of Unified Communications by bringing together all the key technologies and key people of this rapidly evolving world.

If you want to meet with me (Daniel-Constantin Mierla of Asipto, co-founder and core developer of Kamailio SIP Server project), feel free to contact us:

Tuesday, February 14, 2012

Call Center World 2012

I will be present at Call Center World congress in Berlin, February 27 – March 01, 2012. If happens for you to be around and want to meet, feel free to contact us.

The event gathers over 250 exhibitors from around the globe, mainly focusing on help desk and support solutions, integrating SIP/IP and TMD for call centers. Along with the exhibition, there are workshops and the conference.

Asipto’s offerings include reliable solutions to scale and enhance security as well as add new features to call center oriented systems. For example, the load balancing solutions can be used to scale call center capacity in a transparent and flexible manner.

Thursday, February 9, 2012

CeBIT 2012

CeBIT 2012, the biggest digital show, takes place in Hanover, Germany, March 06 – 10, 2012. Daniel-Constantin Mierla (me in other words) of Asipto, co-founder and core developer of Kamailio SIP Server project, is visiting the event.

The exhibition has dedicated pavilions for Telecommunication Industry, from equipment providers to software integrators. Asipto is glad to see several customers exhibiting there and we will be delighted to meet and discuss with the other participants at the event, as well.

If you want to schedule a meeting during the CeBIT 2012, don’t hesitate to contact us:

Wednesday, February 8, 2012

Presentation at Fosdem 2012

Fosdem 2012 included a DevRoom for Open Source Telephony, Daniel-Constantin Mierla of Asipto participated and presented “Secure SIP Communication with Kamailio”.

First part focused on an overview of the project, history and latest new features, then continued to present the tools offered by Kamailio to achieve strong security in your VoIP deployments.

You can find the slides (pdf) at:
If you have questions related about SIP security or look for consultancy in this area, our experienced team can help you, feel free to contact us.

Kamailio at VoIPUsersConference, Feb 17, 2012

Daniel-Constantin Mierla (co-founder) and Alex Balashov (member of management board) will coordinate a new session of VoIP Users Conference (VUC) about Kamailio SIP Server, on Friday, the 17th of February, 2012, starting at 17:00GMT. There will be other developers and users around, ready to answer whatever questions you may have.

VUC is the well know weekly online conference that allow everyone to connect via SIP (voice) and/or IRC (text) to interact with the guests. The host and moderator will be Randy Resnick. You can read more details about this weekly event at:
To participate, choose your preferred way to connect:
  • SIP:
  • Skype: or
  • IRC: #vuc channel on
  • PSTN: +15672522286
The session targets first to give an update on latest developments: asynchronous TLS layer, asynchronous SIP processing, SIMPLE presence extensions, embedded HTTP-XCAP server and MSRP relay, no-SQL storage systems (Redis, Cassandra), embedded interpreters (Lua, Python, C#), a.s.o.

Besides hearing what is new, it is a good chance for everyone to ask and learn about how Kamailio can be used by service providers to meet today’s communication demands: integrated voice, video, instant messaging and presence services, load balancing and least cost routing, security and confidentiality, scalability and redundancy, SIP in IPv6, interaction with web 2.0 and social networking services, …

The previous VUC session about Kamailio and SIP Express Router was done almost two years ago, you can read more and listen the recorded podcast at:

Tuesday, January 31, 2012

Kamailio v3.2.2 Released

Kamailio SIP Server v3.2.2 stable is out – a minor release including fixes in code and documentation since v3.2.2 – configuration file and database compatibility is preserved.
Kamailio (former OpenSER) 3.2.2 is based on the latest version of GIT branch 3.2, therefore those running previous 3.2.x versions are advised to upgrade. There is no change that has to be done to configuration file or database structure comparing with older v3.2.x.
Resources for Kamailio version 3.2.2
Source tarballs are available at:
Detailed changelog:
Download via GIT:
# git clone –depth 1 git:// kamailio
 # cd kamailio
 # git checkout -b 3.2 origin/3.2
 # make FLAVOUR=kamailio cfg
Binaries and packages will be uploaded at:
Modules’ documentation:
What is new in 3.2.x release series is summarized in the announcement of v3.2.0:

Sunday, January 29, 2012

Presentations at Fosdem 2012

Fosdem 2012 includes again a dev room for Open Source Telephony, Kamailio SIP Server Project having a dedicated presentation “Secure SIP Communication with Kamailio”, by me (Daniel-Constantin Mierla).

Andreas Granig, also a developer of the project, will present another talk about SIP:Provider solution, which has Kamailio as the core components for SIP routing.

Related to our eco-system, Stefan Sayer will talk about SIP Express Media Server and its SBC functionality.

The schedule for telephony dev room is available at:

Friday, January 27, 2012

Kamailio & friends dinner at Fosdem 2012

Fosdem 2012 is approaching and we are going to have our traditional dinner at the event on the evening of Saturday, February 4.

At this moment should be over 15 participants, many Kamailio developers and community members, among them:
  • Henning Westerholt
  • Andreas Granig
  • Daniel-Constantin Mierla
  • Stefan Sayer
  • Olivier Taylor
  • Peter Dunkley
  • Raphael Coeffic
If you want to join us, send an email to Register as early as possible, since we need to make a reservation, also the place cannot accommodate too many people. All friends of Kamailio, SER, SEMS, Asterisk, FreeSWITCH and SIP/VoIP are welcome!

Tuesday, January 17, 2012

New Module – Cassandra DB Connector

Courtesy of Anca Vamanu, of 1 & 1 Internet AG, Germany, a new module is available in the development version of Kamailio SIP Server (to become the 3.3.0 release). Here is an adapted excerpt done for this web news from the announcement sent to mailing list.

“The module is named db_cassandra and offers a DB interface that can be used by other modules to perform DB operations instead of other DB modules (like db_mysql for example).
Because Cassandra is a NoSQL storage system, it is not possible to run all kind of SQL-like queries on it and this is the reason why the module has some limitations.

It is especially suited for applications that store large data or that require data distribution, redundancy or replication. One usage example is a distributed location system in a platform that has a cluster of SIP servers, with more proxies and registration servers accessing the same location database.

This was actually the main usage we had in mind when implementing the module. It has been tested with usrloc and auth_db modules, but it can also be used with other modules that have similar queries.
You can read more about this module in the README file:
Inside the module directory you can find an example that modifies the default configuration file to enable a location service with a Cassandra backend. If you have a Cassandra installation, it should be very easy to test it.

We hope you find this module useful and are glad to receive any feedback, comments about it.”

Monday, January 16, 2012

New module – embedded MSRP relay

A new module in development branch of Kamailio SIP Server, named msrp, provides a MSRP routing engine, a.k.a. MSRP relay. The core specification of MSRP (Message Session Relay Protocol) is defined by RFC4975, the extensions for a MSRP Relay being covered in RFC4976. One of typical use case for MSRP is to do Instant Messaging sessions negotiated with SIP via INVITE-200OK-ACK.

The msrp is controlled from configuration file via actions in event_route[msrp:frame-in]. The module is a full, embedded MSRP relay, it does not require any external application nor library. It uses the core transport layer components, thus it benefits of the scalable and asynchronous TCP/TLS support implementation already existing in the project for many years now.

Kamailio, with msrp module loaded, can handle SIP and MSRP traffic received on the same port. But you can configure Kamailio as a stand alone instance to deal only with MSRP traffic, leaving the SIP traffic to another Kamailio instance. Also, another option is to configure Kamailio to listen on different TCP/TLS sockets (e.g., different ports or IP/network interfaces) and direct SIP and MSRP to different ports — then in the config file you can take care of filtering (dropping) inappropriate content on specific ports. With all this flexibility, you can choose a configuration that will not affect at all the routing of SIP messages with Kamailio.

The embedded MSRP relay, built on top of the SIP server, offers many benefits such as:
  • reuse mature code tested over the past 10 years, msrp module itself being really small piece of code in regards to MSRP protocol
  • MSRP is done over TCP/TLS, thus implicitly the forwarding is done asynchronously, offering great performances
  • IPv4 and IPv6 support
  • MSRP is for transmission of a SIP session content, going to be used by the SIP users in your UC platform — there is no need to manage a different user profile
  • the configuration and MSRP routing is done via the same flexible language and format as for SIP traffic, you being in control of what is passing through your server
  • access to all existing extensions that are related to SIP request routing, for example: IP address checking, flood detection, many database connectors, accounting, a.s.o.
You can read more about the msrp module in the documentation file:
At this moment, Kamailio offers a set of extensions that allows building a complete Unified Communication platform, within a single SIP server instance for small deployments as well as a grid of servers, each one doing particular functions:
  • voice, video, screen sharing, etc. sessions with content communication via RTP
  • end to end presence – this is purely SIP routing
  • SIMPLE-based presence (aka, presence server or presence agent model) via presence* and pua* modules — user presence, dialog states notification (aka, blinking lamps), resource lists service (including OMA/RCS extensions), user location states notification and replication, audio/video conference mixer notifications, a.s.o.
  • embedded XCAP server – management of user contact lists, presence policies, user agent configuration files, a.s.o. There is also an XCAP client extension
  • embedded HTTP server – for admin and user interaction with the service via pure HTTP or XMLRPC requests
  • embedded MSRP relay – for relaying and fine controlling of the message-based content of SIP sessions
  • IRC-style instant messaging conference via imc module
  • storage of instant messages for offline users and relay to them when they become again online via msilo module
All above components are built on the same solid foundation, practically is Kamailio core plus a selected set of modules, no extra dependencies, just configuration options.