By MiConDa

Kamailio at VoIPUsersConference, Feb 17, 2012

2012-02-08T13:29:00.000+01:00

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:

http://vuc.me

To participate, choose your preferred way to connect:

SIP: 200901@login.zipdx.com
Skype: vuc.me or ld.vuc.me
IRC: #vuc channel on irc.freenode.net
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:

http://www.kamailio.org/w/2010/03/vuc-listen-the-sip-router-project-podcast/

Kamailio v3.2.2 Released

2012-01-31T13:27:00.000+01:00

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:

http://www.kamailio.org/pub/kamailio/3.2.2/src/

Detailed changelog:

http://www.kamailio.org/pub/kamailio/3.2.2/ChangeLog

Download via GIT:

# git clone –depth 1 git://git.sip-router.org/sip-router kamailio
 # cd kamailio
 # git checkout -b 3.2 origin/3.2
 # make FLAVOUR=kamailio cfg

Binaries and packages will be uploaded at:

http://www.kamailio.org/pub/kamailio/3.2.2/

Modules’ documentation:

http://www.kamailio.org/docs/modules/3.2.x/

What is new in 3.2.x release series is summarized in the announcement of v3.2.0:

http://www.kamailio.org/w/kamailio-openser-v3-2-0-release-notes/

Presentations at Fosdem 2012

2012-01-29T23:38:00.000+01:00

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:

http://fosdem.org/2012/schedule/track/telephony_and_communications_devroom

Kamailio & friends dinner at Fosdem 2012

2012-01-27T23:35:00.000+01:00

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 registration@lists.sip-router.org. 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!

New Module – Cassandra DB Connector

2012-01-17T23:33:00.002+01:00

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:

http://kamailio.org/docs/modules/devel/modules/db_cassandra.html

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.”

New module – embedded MSRP relay

2012-01-16T23:30:00.000+01:00

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:

http://kamailio.org/docs/modules/devel/modules/msrp.html

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.

Per socket number of worker processes

2012-01-14T23:28:00.000+01:00

The development branch of Kamailio SIP Server has a new feature that allow setting number of worker processes to handle received traffic per listen socket.

So far there were global parameters that were applied to all sockets (e.g., ‘children’ value set the number of workers for all udp sockets). So far each UDP and SCTP socket had its own pool of workers (e.g., children=4 and 2 udp sockets resulted in 8 processes), while for TCP and TLS was a single pool of workers (e.g., having children (or tcp_children) set 8, resulted in 8 processes no matter how many TCP/TLS sockets).

The new features is based on using a new config parameter, named “socket_workers“, before a “listen” parameter. The value of “socket_workers” will overwrite the value of appropriate “*children” parameter. For UDP and SCTP will result in creating a number of “socket_workers” processes. For TCP and TLS will add an extra set of “socket_workers” processes, that will handle traffic only on those specific sockets.

The value of “socket_workers” is reset with the next listen socket added. If “socket_workers” is not set, the value of “*children” parameter is used in backward compatible fashion.
Some typical scenarios where this feature may become handy:

set a lower number for loopback or internal/replication sockets, as the traffic there is low (e.g, maybe for keepalive monitoring on loopback, or it is only REGISTER requests replication done over the replication sockets)
set a dedicated group of tcp/tls workers for handling HTTP/XMLRPC/XCAP traffic – handling such traffic may be time consuming, in this way you avoid delays on routing SIP over TCP/TLS
fine tune the number of over all forked processes by a SIP server instance, thus controlling better the resources used from the physical server (e.g., overall private memory used by sip server is a matter of how many forked processes are there)

You can see details about the new parameter and examples on the wiki page:

http://www.kamailio.org/wiki/cookbooks/devel/core?&#socket_workers

Fancy time recurrence matching in config

2012-01-11T23:25:00.000+01:00

A new module for Kamailio SIP Server named for now tmrec allow matching of time recurrences based on definitions specified by Internet Calendaring and Scheduling Core Object Specification (Calendar COS – RFC 2445).

It becomes trivial to match current time against rules such as working hours, weekend, up to complex conditions such as the interval from 18:00 to 20:00 of the 98th day of every other year if it is a Thursday.

Here is an example of how to match the working hours 8:30am to 6:30pm on business days:

 if(tmrec_match("20120101T083000|10H|weekly|||MO,TU,WE,TH,FR")
  xdbg("it is within working hours\n");

The rule can be specified via a config variable (e.g., load from user profile stored in database via sqlops). A typical use case is time based routing policies.

You can read more about the new extension in the documentation:

http://kamailio.org/docs/modules/devel/modules/tmrec.html

New Year, New Extension – Embedded execution of managed code (C#)

2012-01-03T23:22:00.001+01:00

The first committed module for Kamailio SIP Server in 2012 is app_mono, which offers embedded execution of manage code (e.g., C#/.NET) via Mono project (http://www.mono-project.com/).
The readme of the new module is available at:

http://kamailio.org/docs/modules/devel/modules/app_mono.html

Current API which exported by SIP server for usage in C# application is documented at:

http://www.kamailio.org/wiki/embeddedapi/devel/mono

Besides C#, app_mono should be able to run managed code compiled from applications written in other languages such as VisualBasic.NET, Java, Java Script, Python, … more are listed at:

http://www.mono-project.com/Languages

A primary use is integration with several widely used Microsoft technologies and APIs, for example C# having up-to-date libraries to connect to Active Directory LDAP service or MS SQL Server.

Happy 2012!

2012-01-01T23:18:00.001+01:00

Same type of activity for me in the past 10 years and still enjoying it! Looking forward to the 11th!

Watch Kamailio SIP Server project news, lot of cools stuff is coming out!

Have a great 2012 friends!

Siremis v3.2.0 Released

2011-12-14T23:48:00.000+01:00

Siremis v3.2.0 is out – the web management interface for Kamailio SIP Server (former Openser) and SIP Express Router (SER). This is a major release, compatible with Kamailio v3.2.x.
This release brings a large set of new features. Among them:

SQL-based CDR rating engine for billing purposes

stored procedure to compute the costs of calls

Management of billing rates

longest prefix rate selections
rating rules can be grouped to allow many sets of values
time unit is configurable per rating rule

Management of remote registration records (uacreg table)
Managment of mtree module (mtree and mtrees tables)
Management of dialog variables table
Update of LCR and SIP Trace views for compatibility with Kamailio 3.2.x
Tools to generate new database table views in a wizard fashion

create new views to database table with a command line tool in 5 steps

Charts drawing statistics of accounting records

graphics to show the evolution of accounting records during the past hours
graphics to show the types of INVITEs (call setup) during the past hours

Tables presenting summary of accounting records

count the number of INVITEs and BYEs in the past hours
present the top activity of accounting records – e.g., top 5 caller and callee
more can be added from configuration file

More SIP server activity charts (e.g., SIP requests traffic load)

e.g., default chart presents how many requests are received in intervals of 10 minutes

Buttons to switch to command pannel to reload Dispatcher or PDT records in SIP server cache

once new records are added, in two clicks they get in the cache of SIP server

Views for managing global black lists table
Many improvements to user interface

selection of local domain is done via select box or picker form (e.g., in aliases, user preferences, pdt, …)
selection of local username is done via picker form (e.g., user black lists, user preferences, aliases, …)
group names can be set in config file and selected from a list box
many static values are given as option to select from a list box (e.g., dispatcher flags, lcr options)

More targets in Makefile to make administration easier

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

Installation Tutorial
Screenshots
Web Demo for Siremis 2.0 - v3.2.0 is built on same core framework, just new features were added (thus more options in the menu)

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

Kamailio Advanced Training Presentation Page

Migrating project from BerliOS to GitHub

2011-12-02T23:51:00.002+01:00

BerliOS, the open source software forge, announced ending of its life by Dec 31, 2011. Although some time later, there was another announcement that the service will continue, to be operated by a non-profit foundation, I thought anyhow of copying two projects I had there to GitHub.

The two small projects I wrote long time ago, while being a researcher at Franhofer Fokus, were not really maintained, but anyhow it would be a pity to lose the code, parts of it may be useful in the future. If anyone wanders, here is short the description of these projects:

pocketsipmsg - this project was used a lot durin 2002-2005 to make demos of SIP instant messaging using iPaq running Windows CE. It is basically a SIP user agent capable of sending and receiving text messages, developed in Visual C for Windows CE
tmrec - this project offers a C library and command line tool for matching time recurrences defined by iCal RFC2445. The code is actually used, being embedded in cpl-c module of Kamailio SIP Server

Initially they were stored in CVS repository of BerliOS, but I switched them to SVN some time ago. Therefore, I was looking how to migrate SVN repository from BerliOS to a GIT repository on GitHub.

Googling gave lot of useful tutorials about migrating SVN to GIT, I am writing here just to show the specific case of migration from BerliOS to GitHub - it could save time for some people interested in same kind of operation.

Personally I used a Mac OS X, so first I installed git-svn from ports:

# port -v -d install git-core +svn

On a Debian/Ubuntu, the command should be:

# apt-get install git-svn

You have to create a file to map BerliOS username (used for SVN commits) to name and email address (to be used by Git). I named it authors.txt, the content:

dcm = Daniel-Constantin Mierla <me@xyz.com>

(no author) = Daniel-Constantin Mierla <me@xyz.com>

root = Daniel-Constantin Mierla <me@xyz.com>

For some reasons, which I didn't want to spend time to investigate why, there were two other authors appearing to have committed in SVN: root and (no author) -- so simply I mapped them to myself as well.

Next step I used git svn to clone the berlios project - here is the command used for pocketsipmsg project:

# git svn clone http://svn.berlios.de/svnroot/repos/pocketsipmsg --no-metadata -A authors.txt -t tags -b branches -T trunk pocketsipmsg

You will have to replace pocketsipmsg with your project ID on BerliOS.

I wanted to get the tags and branches from SVN. In the cloned directory, pocketsipmsg, you can list the branches with:

# git branch -r

You will notice that the SVN tags are now branches, to get them back to tags, you have to execute for each tag (named next as $tagname):

# git tag $tagname tags/$tagname

# git branch -r -d tags/$tagname

On GitHub you have to create the repository for storing the project - see http://help.github.com/create-a-repo/ to learn how to do it, if you haven't done it yet. In my case, I named it also pocketsipmsg.

Then add GitHub as remote repository and push to it:

# git remote add origin git@github.com:miconda/pocketsipmsg.git

# git push origin master --tags

That's all, your project is now stored on GitHub!

In summary, if git-svn is installed, your project does not have tags, you created authors file and added the repository $PROJECTNAME on GitHub under user $USERID, the commands you have to run for migration of $PROJECTNAME from BerliOS to GitHub are:

# git svn clone http://svn.berlios.de/svnroot/repos/$PROJECTNAME --no-metadata -A authors.txt -t tags -b branches -T trunk $PROJECTNAME

# cd $PROJECTNAME

# git remote add origin git@github.com:$USERID/$PROJECTNAME.git

# git push origin master

Enjoy!

Kamailio v3.2.1 Released

2011-12-01T22:30:00.001+01:00

Kamailio SIP Server v3.2.1 stable is out – a minor release including fixes in code and documentation since v3.2.0 – configuration file and database compatibility is preserved.

Kamailio (former OpenSER) 3.2.1 is based on the latest version of GIT branch 3.2, therefore those running previous 3.2.0 versions are advised to upgrade. There is no change that has to be done to configuration file or database structure comparing with v3.2.0.

Resources for Kamailio version 3.2.1

Source tarballs are available at:

http://www.kamailio.org/pub/kamailio/3.2.1/src/

Detailed changelog:

http://www.kamailio.org/pub/kamailio/3.2.1/ChangeLog

Download via GIT:

 # git clone –depth 1 git://git.sip-router.org/sip-router kamailio  # cd kamailio  # git checkout -b 3.2 origin/3.2  # make FLAVOUR=kamailio cfg

Binaries and packages will be uploaded at:

http://www.kamailio.org/pub/kamailio/3.2.1/

Modules’ documentation:

http://www.kamailio.org/docs/modules/3.2.x/

What is new in 3.2.x release series is summarized in the announcement of v3.2.0:

http://www.kamailio.org/w/kamailio-openser-v3-2-0-release-notes/

Kamailio v3.2.0 Developer Guide

2011-11-29T22:26:00.001+01:00

Development guide for Kamailio SIP Server has been updated for v3.2.0 – it goes through internal components, presenting the APIs for pkg/shm memory, locking/synchronization, config file interpreter, database connectors, a.s.o., as well as guiding how to write a new module.

There is a section trying to collect hints about upgrading a module developed for old versions 1.x to newer architecture and APIs in versions 3.x.

The tutorial is available online at:

http://asipto.com/u/kdg

A mirror is hosted at:

http://www.kamailio.org/docs/kamailio-devel-guide/

Looking forward to your contributions to Kamailio SIP Server!

Kamailio Advanced Training, Dec 5-8, 2011, Berlin

2011-11-24T10:04:00.001+01:00

Next Kamailio SIP Server Advanced Training will take place in Berlin, Germany, Dec 5-8, 2011.

Last Kamailio stable series is 3.2.x (Oct 18, 2011, see release notes), continues the work done within SIP-Router.org project. Offering a big lot of brand new features in v3.2.0, starting with an older major version, 3.0.0, you can run mixed Kamailio (OpenSER) and SIP Express Router (SER) modules in the same SIP server instance, giving you the most powerful tools to build stable, very performant and features rich VoIP and Unified Communication platforms.

The class is organized by Asipto and will be taught by Daniel-Constantin Mierla, founder and core developer of Kamailio SIP Server project.

Read more details about the class and registration at:

http://www.asipto.com/index.php/kamailio-advanced-training/

Siremis v2.1.0 Released

2011-11-23T22:02:00.000+01:00

Siremis v2.1.0 has been released – this is an update to previous release v2.0.0, bringing several enhancements and new web pages to manage PUA and RLS. It is still compatible with Kamailio v3.1.x, the last of this kind, next one to be out in the near future will be compatible with Kamailio v3.2.x.

You can find the news about this release, including links to download, screenshots and demos, at:

http://siremis.asipto.com/2011/11/23/siremis-v2-1-0-released/

Alternative download site (tarball or git pull) is from sourceforge project:

http://sourceforge.net/projects/siremis/

Siremis v2.1.0 is working for most of the components with Kamailio 3.2.0, just the few that changed the database structure may not be fully functional (e.g., the modules with tables that have new columns, see http://www.kamailio.org/wiki/install/upgrade/3.1.x-to-3.2.0#sql_commands).

New module - get UA config via presence service

2011-11-19T09:56:00.000+01:00

Kamailio SIP Server has now a new module named presence_profile - the module extends presence server implementation with ability of handling ua-profile event. When an user agent subscribes to ua-profile event for its own AoR, it will retrieve profile data document via body of NOTIFY request.

The profile data document format is usually specific per user agent, such documents have to be built and added to presentity table by the admin or a third party application.

Read more about SIP user agent configuration framework in RFC6080:

http://tools.ietf.org/html/rfc6080.

New module – execute control commands over HTTP

2011-11-15T09:49:00.002+01:00

The development repository of Kamailio SIP Server includes a new module, named xhttp_rpc, that allows execution of RPC control commands via HTTP(S). Just for example, you can list active TCP connections, dump user location records from memory, reload the records for LCR or load balancing rules.

The module reuse existing xhttp module, therefore it has no external library dependencies and the processing rate matches the performances of processing SIP requests.

You can read more about this module, see an example of how to use, at:

http://kamailio.org/docs/modules/devel/modules/xhttp_rpc.html

To use it, once you load the module, point your browser to http://yoursipserverip:5060/rpc and you are ready to go.

VoIP and Kamailio Social Networking Event in Cape Town

2011-11-01T20:37:00.002+01:00

Tuesday, November 8, 2011, it is planned an open dinner in Cape Town, South Africa for people interested in SIP, VoIP, Kamailio or other open source VoIP projects. Everyone is welcome to join, see more details at:

http://www.kamailio.org/w/2011/11/social-networking-event-cape-town-south-africa/

Kamailio v3.2.0 Released

2011-10-18T20:36:00.000+02:00

Kamailio (OpenSER) v3.2.0 is out – a major release with a very large number of new features and improvements.

On October 18, 2011, Kamailio (OpenSER) 3.2.0 has been released – this release is a result of more than 10 months of development and 2 months of testing from the teams of Kamailio (OpenSER) and SIP Express Router (SER) projects.

In the year of the 10th celebration of our project, this version comes with 12 brand new modules in addition to a lot of fresh features in core and old modules. Continue reading full release notes at:

http://www.kamailio.org/w/kamailio-openser-v3-2-0-release-notes/

Enjoy SIP routing in a more flexible and easier way with Kamailio v3.2.0!

Kamailio Developer Seminar, San Francisco, Oct 24-25, 2011

2011-10-03T23:36:00.000+02:00

I (Daniel-Constantin Mierla), co-founder of Kamailio SIP Server project, will provide Kamailio Developer Seminar for one day and a half, Oct 24-25, 2011, in San Mateo, south of San Francisco, California.

Among topics to be approached:

internal architecture of Kamailio/SER SIP server
SIP parser
memory manager
locking manager
database API
configuration file language structure and interpreter
RPC/MI control interface
pseudo-variables and transformations framework
module interface
how to write your own extensions in C as module
inter-module APIs – transaction management, SIP SIMPLE Presence, asynchronous processing, a.s.o.
working with embedded interpreters for high level programming languages: Lua, Perl, Python

Who should attend:

VoIP/Telecom developers intending to write own extensions to Kamailio SIP Server
VoIP/Telecom administrators interested to learn about the internals of Kamailio SIP Server in order to know how to optimize the SIP routing and build proper configuration file
VoIP/Telecom professionals interested to learn about Kamailio SIP Server, how and where can it be used, its current features and future development

The event is a good place for networking with other professionals from VoIP/Telecom and Real Time Communications fields, past similar events having dozens of attendees.

The price per attendee is 250 USD.

Number of seats is limited and access will be granted in first come first served fashion.

Registration or requests for more details can be done via:

registration web form – click here for it
or email to: registration [at] kamailio.org .

Presentations from 10 Years SER Conference

2011-09-22T23:35:00.000+02:00

The slides presented during 10 Years SER Conference, Berlin, September 2, 2011, are made available at:

http://www.kamailio.org/events/2011-10yearsSER/

The summary of usage statistics is available in this presentation. Thank you all that submitted data, we counted over 4 billions of routed call minutes and 500 millions calls per month, from about 35 reports.

Another interesting part of the event was the 10 Years SER Awards, see in these slides who was awarded.

The event was fully booked, everyone is looking forward to the next anniversary. Some pictures from the conference and evening’s social event will be published soon.

Kamailio v3.1.5 Released

2011-09-14T23:33:00.001+02:00

Kamailio SIP Server v3.1.5 stable is out – a minor release including fixes in code and documentation since v3.1.4 – configuration file and database compatibility is preserved.

Kamailio (former OpenSER) 3.1.5 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 3.1.5

Source tarballs are available at:

http://www.kamailio.org/pub/kamailio/3.1.5/src/

Detailed changelog:

http://www.kamailio.org/pub/kamailio/3.1.5/ChangeLog

Download via GIT:

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

Binaries and packages will be uploaded at:

http://www.kamailio.org/pub/kamailio/3.1.5/

Modules’ documentation:

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

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

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

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

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

SEMS v1.4.2 Released

2011-08-26T23:32:00.000+02:00

The SIP Express Media Server (SEMS) version 1.4.2 is available. Users of SEMS 1.4 versions and anyone else is recommended to upgrade to that version. Please find below the relevant changelog, the source tarball is available at:

ftp://ftp.iptel.org/pub/sems/sems-1.4.2.tar.gz

Debian/Ubuntu packages:

https://build.opensuse.org/project/show?project=home%3Ateam-sems

Changelog for SEMS 1.4.2 release:
* auth’ed BYE (wait_for_bye_transaction)
* fixes SIP auth for qop header format
* xmlrpc: fix busyloop with keep-alive
* a few minor SST issues
* builds on Ubuntu 11.4 (build-deps)
* ivr: release GIL on blocking file I/O
* SBC: fix codec filter for unnamed payloads<96
* fixed DSM variables to outgoing call
* some examples and documentation added

This release introduces a new configuration option wait_for_bye_transaction=yes which enables authentication of BYE requests sent by SEMS, which is disabled by default. If this option is not enabled, SEMS behaves like in 1.4.1.

Freezing development for Kamailio v3.2.0

2011-08-08T23:31:00.000+02:00

Based on the last development IRC meeting, the next major release Kamailio v.3.2.0 should happen sometime by end of September, therefore the code should be frozen and go into testing phase about one month before.

The date proposed for freezing is Monday, August 22. By then, every developer has to push to master branch what he/she wants to have in 3.2.0.

The very good side of 3.2.0 is that we didn’t need to touch much the core and main modules, meaning we have a solid tested foundation, further testing should be focused on the new modules and the other enhancements.

What is new in devel version at this time (to be part of v3.2.0) is summarized at:

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

Kamailio at Cluecon 2011

2011-08-04T09:56:00.001+02:00

ClueCon is an annual 3-Day Telephony User and Developer Conference bringing together the entire spectrum of Telephony from TDM circuits to VoIP and everything in between. The location is Chicago, USA, between August 9-11, 2011.

Kamailio project will be touched in at least three presentations:

Daniel-Constantin Mierla, Asipto, co-founder of Kamailio – presenting latest news about development of Kamailio project and how to use it to enable strong security for large VoIP systems
Stefan Wintermeyer, Amooma – presenting Gemeinschaft 4, an IP PBX system built with Kamailio and FreeSWITCH
Alexandr Dubovikov – presenting Homer project, a Kamailio-based SIP capturing system for massive signaling traffic

If you are around Chicago and want to meet, drop a message on our mailing lists.

Kamailio and SER Usage Statistics

2011-08-02T09:55:00.000+02:00

In order to present at 10 years SER conference in Berlin, I (Daniel-Constantin Mierla) am looking to get some statistics about usage of SER-based applications (e.g., SER, OpenSER, Kamailio, OpenIMSCore, …). Everything is going to be like a combined report, not individual listing. Therefore, if you want to participate, your name or company won’t be mentioned.

Here are some stats I thought of:

type of usage: production, evaluation (testing), research
number of subscribers (phone lines)
percentage of phones behind NAT
number of calls per month
number of call minutes per month

I am not looking for exact numbers, but rough estimation (e.g., about 10 000 phone lines, …). Of course, some of the metrics don’t apply always (e.g., if you do termination routed through SIP server, you don’t have subscribers, but just calls traffic). Those doing deployments, can make one summary of statistics for all instances. You may send other statistics you think they worth publishing.

Again, this is voluntary, naming is not required. I hope only those giving real number will write back — I will try anyhow to figure out if someone is just dumping fantasy numbers. If privacy is really a big concern, from case to case, everything can be done under NDA.

Send the statistics or contact me for further details per email at: <daniel [at] kamailio.org>. Many thanks in advance to those that are going to participate in this survey!

10 years SER Conference – Sep 2, 2011, Berlin, Germany

2011-08-01T09:53:00.000+02:00

We are organizing an open conference to celebrate 10 years since the first line of code in SER and Kamailio SIP servers. The event is hosted by FhG Fokus Research Institute in Berlin, Germany, the place where SER project was started. The web page for the event with details about location and program is available at:

http://sip-router.org/10-years-ser/

There is no participation fee for the conference day, registration can be done via web form. At this moment there are about 20 special guests and 14 speakers, with participants from 15 countries.

If you use SER-based applications (like ser, openser, kamailio, openimscore, …) and want to present about it or you want to sponsor the event in exchange of advertising through project channels, contact us via event registration form.

3 years of Kamailio

2011-07-28T21:01:00.009+02:00

Three years ago, OpenSER project had to choose a new name due to trademark infringements claims by an US company and Kamailio was the winner name.

Shortly after, the project faced a new hit, from inside this time, losing the old web domain. The sourceforge.net OpenSER project (where the source code was hosted) was the one showing the relation between the old name and new name, and the truth about whole thing.

Practically, from that moment Kamailio had to build a new brand from scratch, all the old web references being lost at that time. Despite the pessimism of many people and fighting FUD spread by cowards, I believe we succeeded to continue developing a great application, delivering a rock solid open source SIP server, withing a collaborative environment and open community.

I had no doubt about the success, just read (or google) about who is involved in management of the project or where are the people with relevant impact in the development of the project over the year (config file, database API, variables framework ... more to come soon on this blog).

Note that in present time the devel team is completed by the folks at SER project, Kamailio and SER are effectively same application right now, development being done together since autumn 2008 within sip-router.org development framework. To fully complete this picture, we are celebrating soon 10 years since the first line of code, where the major contributors to the project are among the special guests.

Apart of working to remove duplicates (and clarify better the relation and usage of two names Kamailio - SER -- briefly exposed here), the ongoing development of new features blossoms:

IMS extensions being worked to be integrated in main repository
Rich Communication Services enhancements (SIMPLE/OMA/RCS IM & Presence, embedded XCAP server)
embedded interpreters for high level programming languages (such as Lua)
distributed location service and message queues
config file pre-processing
asynchronous processing and more ...

All these facts are visible on the statistics offered by Ohloh service - few stats presented in the next graph.

If you are interested in the achievements of past releases since Kamailio name, see the release notes for versions 1.5.x, 3.0.0 and 3.1.x. Next major release, 3.2.0 is planned sometime by mid of autumn 2011.

Right now Kamailio is sitting now on a solid foundation, with a completely re-tailored architecture since OperSER time, suitable for current and future needs of real time communications, sustained by an increasing development team and backed up by major companies in the market. Its evolution is safer than ever.

My greetings to all Kamailians and thank you for your support over the last years! I am looking forward to meeting as many of you as possible at 10 years SER conference in Berlin, September 3, 2011.

ClueCon 2011

2011-07-20T11:32:00.002+02:00

I am going to speak at Cluecon conference in Chicago, USA, Aug 9-11, 2011.

The speech focuses on how to build strong security for large VoIP operators, a topic where Asipto has broad expertise accumulated over the past 10 years. Some of our experiences were already shared via knowledge base site.

If you attend the conference or you are in Chicago area during these days and want to meet, do not hesitate to drop me a message via contact form.

10yearsSER - things done right - variables

2011-07-03T17:21:00.008+02:00

The veteran users of this project probably remember that day in the summer of 2003 when xlog module was introduced in SIP Express Router (SER) by Elena-Ramona Modroiu.

Starting with that moment, the SER config files were invaded by many xlog(...) and xdbg(...) lines - it was a big step forward to allow SER administrators understand what is going internally during processing of SIP traffic.

Little 'bugs' starting with '%' (named at that time 'specifiers') present all over the config were replaced dynamically at runtime with values taken from SIP traffic or runtime environment, giving back meaningful information about current processing.

Driven by their popularity, the specifiers implemented inside xlog module were moved to core, the marker sign '%' was replaced by '$' and since then they were named 'variables' (aka pseudo-variables). (Note: this evolution happened initially through Kamailio (OpenSER) branch).

The 'little bugs' became free to fly everywhere inside configuration file, in expressions, in assignment operations, module parameters or function parameters. And they did so!

For example, adding the prefix 0049 to dialed number became as simple as:

$rU = "0049" + $rU;

Testing if the dialed number starts with 0049? Here it is:

if($rU =~ "^0049") { ... }

Looking over several configuration files I have these days running on production systems, the operations with variables have the leading role in defining the routing logic. Without them, none of these configs would have been possible, without them, many ideas of new services would have been impossible to implement.

In the year of 10th anniversary, it is the time of sending kudos to Elena-Ramona for her contributions to this project.

PS: to clarify, the terminology 'pseudo-variables' was used due to the fact that SER's configuration language variables concept is different than classic variables. For example: some are read only, some have the value in private memory or shared memory, some are single-value some are multi-value.

10yearsSER - things done right - database api

2011-07-02T16:27:00.004+02:00

Besides throwing a party and being happy about it, a celebration remembers you how old you are. Same applies to SIP Express Router (aka SER) project. It started 10 years ago and the world was so different at that time (imagine: no twitter, no facebook!!!).

Targeting large IP telephony deployments with even millions of subscriber on same instance, using text files to store user profiles was out of discussion. Natural choice by year 2001: SQL database. Which one? Several options out there, so choosing one could have been proven wrong over the years.

The solution was on implement an abstract layer that allows to switch to a different database system without affecting the rest of the code. ODBC is something similar and UnixODBC is one open source implementation of it. The decision not to use unixodbc had to do with the target of going beyond SQL-like database, complexity of the ODBC system and willingness to control the component completely considering that database interaction is a critical factor for performance.

Jan Janak designed and implemented the internal DB API, which exists currently in two versions: v1 implemented mainly in Kamailio-specific modules and v2 implemented mainly in SER-specific modules.

Unlike ODBC, the internal DB API is looking to offer a limited set of features to interact with database, only those required from SER point of view.

These days, the list of database API connectors includes a significant number of implementations:

MySQL
PostgreSQL
SQLite
UnixODBC
DBText
BerkeleyDB
Oracle
LDAP
Flatstore

Practically, SER modules are not affected by what type of database connector you want. Besides that, different modules can connect to different database systems.

The abstract database API was a component that attracted many users to SER over the time. In the year of 10th anniversary, it is the time of sending kudos to Jan for his work.

10yearsSER - things done right - configuration file

2011-07-01T15:46:00.007+02:00

Looking back over the past 10 years since SIP Express Router (aka SER) is developed, I found several things that made it famous. First such thing to talk about is the configuration file.

SER's configuration file is completely different than one may expect, like typical .ini file or, common in telephony, the prefix based routing rules (e.g., Asterisk's dialplan).

SIP promises included flexibility, extensibility and ability to route calls on IP/DNS based networks, with telephony just a subset of its capabilities. Based on that, going for prefix based routing seemed inappropriate for long term goals.

Initially designed by Andrei Pelinescu-Onciul, the structure of SER's configuration file ended to be very like a programming language. That succeeded to get over the time a lot of people to dislike SER itself, mainly those coming from legacy telephony products, very used in routing telephone numbers, based on prefixes.

Summarizing the evolution and comparing with what happened to other IP telephony projects, I can definitely conclude that the decision was the right one. Here is a short example of routing SIP INVITE requests to 10.0.0.1 and the other request types to 10.0.0.2:

route {
   if(method=="INVITE") {
       t_relay_to("udp:10.0.0.1:5060");
   } else {
       t_relay_to("udp:10.0.0.2:5060");
   }
}

For example, Asterisk and Freeswitch, although they have their own call routing config language, text or xml based, in most of the cases the routing is decided using programming languages such as Lua, PHP, Perl a.s.o, through embedded interpreters or control interfaces such as AGI.

Probably SER came too early with this idea, at that time there was no fast scripting language that could have been used to match our performance goals, being necessary to define a new one. No matter how many hated it so far, several order or magnitudes more loved it, resulting over the time in a well defined SIP routing language, optimized to be fast for handling SIP traffic. With the power of a logical language in hands, people could easily build their ideas of new services on top of it.

At this moment SER and Kamailio (forked from SER in 2005 and back together since 2008) have embedded interpreters for Lua, Perl or Python, so you can route SIP using applications written in such programming languages, but still the default config language is the internal one.

It is no way I can be sure or claim that using a logical language for configuration file of telephony engines was invented in SER, but for sure it contributed substantially to show the power of such concept in building new and advanced telephony services.

In the year of 10th anniversary, it is the time of sending kudos to Andrei for such idea and his work to implement it.

Maketing - interaction between developers

2011-06-28T22:43:00.001+02:00

[This is part of Maketing blog posts series]

The most important asset for an open source project is without doubt its team of developers. While in a company the monthly income may define the level of loyalty, in open source is about passion in most of the cases.

As a leader of the project, it is critical to find the way to keep the developers motivated and inside the project for long time. There will be many coming in and many going out quickly, if you don't succeed to create a loyal core team which stays in for several good year, you are lost.

At some point, when the project is big, you cannot care about it alone. Physically, you will not be able to handle alone the management, development, advertising and community interaction. Remember that you have also a private life. Therefore you need experienced people to help others coming on board.

How to get new developers? Be open enough. Open enough means not to be very restrictive and not very open. Being very restrictive in accepting contributions induce the false impression that the high quality is protected. Well, indeed, no line of code means no bug as well as nothing to use. On the other hand, accepting everything tha is submitted it is going to create a jungle.

Finding the right solution is a matter of the application. In Kamailio/SER, the design allowed to split in components which are very independent in influencing the quality of the others. Core and main modules are been exigently taken care. New ideas and young code can be very easy just added as independent module, which, if not used has no impact to your running instance.

This modularity solves another potential issue with developers: coding conflicts. One problem may have many solutions, some developers can have different opinions about what is the best one. That is good, allow them to contribute separate modules and users will have the liberty to choose the one they like more.

What about keeping the developers loyal and passionate about the project? First, bring the developers to be part of project's evolution. Involve them in discussions beyond writing code: planning, management, new releases or events a.s.o., everyone will fill safer if they know what is going to happen.

Be aware of language barriers and mentalities. Everyone thinks in mother tongue, writing the thoughts in English may look odd or impolite sometimes. Try to understand beyond words, contact the person privately and figure out more details before judging to sentence.

Don't forget that human interaction make relations stronger. Whenever you have a chance to meet in person a developer, don't hesitate to go out for a drink. An open source project is an social entity.

When you are still leading the project and the other developers (which are not paid by you) contribute substantially more than you to project, then you are succeeded to set the development on the right track and have a self-sustainable evolution.

Ukulele Kamailio Jingle

2011-06-21T23:06:00.000+02:00

Javi Beltran, an open source musician and good friend of Kamailio SIP Server project, recorded a short composition for Kamailio playing Ukulele (Hawaiian instrument, aka Ukelele).

You can listen or download the jingle from:

Javi Beltran – Kamailio Jingle

Feel free to use the jingle as you wish in relation with Kamailio project. You can see more of Javi’s open sourced compositions on his web page.

10yearsSER - the 15000th commit

2011-06-14T13:45:00.002+02:00

In the year celebrating 10th anniversary of SIP Express Router (aka SER) project (from where Kamailio (former OpenSER) forked in 2005 and merged back in 2008), the development branch encountered the 15000th commit in GIT repository.

It was done by the same developer that started the project and committed the 1st one, Andrei Pelinescu-Onciul:

git log --pretty=format:"%h%x09%an%x09%ad%x09%s" --reverse | head -1

512dcd9    Andrei Pelinescu-Onciul    Mon Sep 3 21:27:11 2001 +0000    Initial revision

The 15000th one is:

git log --pretty=format:"%h%x09%an%x09%ad%x09%s" --reverse | head -15000 | tail -1

8a90dd3    Andrei Pelinescu-Onciul    Sat Jun 11 11:24:05 2011 +0200    core: remove unused variables + coding style

This number of commits is counted only for development branch (GIT master), thus it does not take in consideration the branches of stable releases. That will make the number of commits in the project far more higher, but the target was to show only the main stream development evolution, not the fixes in stable branches.

Note that there are personal developer branches holding code commits done in the past and to be merged in GIT master branch in the near future, so the number 15000th commit may be different when checking again later. However, as a marker in the history of the project, Andrei's commit is the first in master branch to hit the 15000. The link to the commit via web GIT viewer is here.

SER's 15000th code commit

You can also check the history of SER/Kamailio commits tracked by Ohloh here.

Be ready to party with us in September at the 10th project anniversary and watch us for the 20 000th commit!

Run your own SIP VoIP service on both IPv4 and IPv6

2011-06-08T14:34:00.005+02:00

Today is World IPv6 Day - since Kamailio SIP server has support for IPv6 since 2002, I thought to contribute to today's celebration showing how to use it to provide a SIP-based VoIP service on both IPv4 and IPv6 networks, also bridging between them, with the same SIP server instance. Following VoIP communication scenarios are possible:

call from IPv4 to IPv4
call from IPv6 to IPv6
call from IPv4 to IPv6
call from IPv6 to IPv4

I used Jitsi SIP softphone to make the calls, Kamailio development version for SIP server and RTPProxy to help bridging the media streams between IPv4 and IPv6.

The How-To do it tutorial is available on the wiki at:

Run your own SIP VoIP service on both IPv4 and IPv6

Happy IPv6 Day Everyone!!!

A look at SIP:Provider CE v2.2

2011-06-07T16:52:00.003+02:00

SIP:Provider Community Edition (SPCE) has recently released the version 2.2. The out-of-the-box VoIP service operating platform added in this version a lot of interesting new stuff.

First is about the upgrade of the operating system to Debian Squeeze. Also Kamailio SIP Server and SIP Express Media Server (SEMS) are integrated with their latest stable branches.

From this point of view, having the latest Kamailio opens the doors to add by yourself any of its features in version 3.1.x directly in the configuration file, such as SIP/SIMPLE presence or secure communication over TLS.

In terms of architecture, the platform was re-sketched from grounds. It runs an instance of Kamailio to guard the other SIP applications, namely the SIP registrar and proxy (another Kamailio instance), the voicemail server (Asterisk) and the back-to-back user agent (SEMS). Besides the role of entry and exit point in the platform, the first instance of Kamailio acts as a load balancer, meaning, for example, that you can add new SIP proxy/registrar servers as you need.

Talking about security, only the Kamailio load balancer is running on public IP, all the rests can run on an internal one, for example 127.0.0.1, making impossible to be accessed from outside, avoiding DoS attacks on them. The load balancer is not using any SQL database, thus is able to absorb impressive amount of SIP traffic, being easy to deal with any kind of attacks. In addition, all the calls are routed through SEMS for SIP signaling topology hiding, protecting the coordinates of core components and the end points.

Caring about security had high priority in this SPCE release, besides those listed above, there are configurable options to protect against scanning and flooding attacks.

A brand new component of the platform is the ngcp-mediaproxy-ng (some notes about it here) which replaces RTPProxy for NAT traversal. The main benefits are in terms of QoS, ngcp-mediaproxy-ng using a kernel module to relay the media packets. The application has been developed in-house, used for many years in production and now released open source under GPLv3 for SPCE.

The web interface got also some fresh air, in particular the administration portal makes more use of web2.0 technologies, improving the user experience.

I am migrating one of the public VoIP services that run Kamailio to SPCE -- then it would be easy to try & feel it quickly. The plan is to go beyond the standard distribution, very likely will have SIP presence and few more features - the targets to be included in the new version of SPCE.

If you want to give it a try by yourself, choose between the APT repository or one of the provided virtual machines images for VMWare of VirtualBox, see details at:

SPCE Quick Install Guide

Stay tuned for more updates!

The race to the new and world wide telco

2011-06-01T09:39:00.008+02:00

Globalization is everywhere. Still one of the most crumbled markets is the telecom world (classic telephony or mobile). But we are tired of changing SIM cards as we land, we are tired of remembering to redirect numbers when we leave the office, we are tired to memorize all our phone numbers, someone's gonna fix it ... lots of money still there and the seat of a world wide telco is yet free.

The race (could be called war as well) started: Apple, Facebook, Google and Microsoft moved troops, stroke and changed the strategies. We know who is top search engine, top gadget maker, top social networking platform or top operating system vendor, but who is going to win the top world-wide telco crown? Time will decide, first for now just a quick look at top fighters.

Microsoft

Being still hot, let's look first at Microsoft and its acquisition Skype.

Deal was agreed, but it is going to take some time to get approved by authorities. This period is a big waste of time. Besides that, the whole eco-system is a mixture, fitting pieces (i.e., Microsoft applications with Skype communication) together to build a puzzle that was not designed for this goal from the beginning it is going to cost more time and quite some money in development and deployment.

Another weak point for this team is poor presence among end user mobile terminals. Windows 7 for mobiles is at its very young age, iOS and Android have far more third addons that significantly balance the end user decisions of what to buy. Moreover Microsoft does not have a branded hardware yet that attracts customers. Buying Nokia can bring them the infrastructure, the knowledge to build mobile hardware and more distribution channels, still the 'device' is not there since Nokia does not have it either. Therefore more lost time.

Many presented their friendly relation with Telecoms as a strong point, I see it completely different, we talk here about taking parts of telecom cake, so it can turn against rather than help them. I expect Telecoms reducing Microsoft's revenue on other channels (desktop and server OS, productivity applications, a.s.o.) once it starts direct competition.

Apple

The company is lone rider. Does cool stuff, good looking devices and rich user experience that everyone want, but for the other companies is hard to do business with them. Apple has a 'good' reputation of changing the rules in the middle of the game as they like.

They are well positioned regarding mobile devices with iPhone and I would count even iPad and iPod Touch. The upcoming iCloud can provide the needed infrastructure to run telecom service.

Facetime service looked interesting, still none of my close contacts switched to use it as primary communication channel. I don't have figures about their subscriber base, but can be the seed for the telecom plans.

Google

By using Gmail ID Google solves quickly the addressing space with unique user ID in a convenient way. The subscriber base is big enough to be very appealing to use the service or interconnect with.

With GPhone and especially with Android OS for mobile phones and pads, Google is extremely well positioned at this moment, recently outselling iPhone.

Google offers GTalk and Google Voice products for voice communication, too separated so far in my opinion.

Facebook

The famous social networking relies primarily on its user base. They realized that lack of a public Facebook unique ID is working against them, so recently they added own email service and try to force users to choose their Facebook address.

End user controlled device is missing completely right now, there were rumors about a Facebook phone, nothing for sale so far. Their messaging system is open for federating through XMPP, an open standard, but internally it is different.

My thoughts

Money is not a big issue for any of these companies, what matter is the time and the immediate gain of taken decisions.

What I would like to see from the new telecom model? Freedom in communication and mobility for users, plus a proper mapping of the service on the Internet architecture and use what drives the Internet and made it famous (the DNS). What I expect?

be able to choose my contact addresses (what used to be phone numbers), where people can call me. I am Daniel, not the only Daniel in this world, but I am the one at Kamailio project, so daniel@kamailio.org can uniquely point to me.
be able to interconnect easily with my own brewed telco system. Look at how email is working - Internet domains (DNS) for routing.
be able to migrate easily my own brewed service to world wide telco's infrastructure (porting my addresses) and the other way around. DNS is again straightforward the solution
be free in the content of the communication, no restrictions on media type imposed by core network

At this moment I see SIP, the open standard protocol very popular in Voice over IP, as the right direction. Yes, I am a SIP guy, but it is hard to argue against the next facts:

SIP has a very large end user equipment base in place - SIP is deployed by many operators and present in form of hardware or software to hundreds of millions of people, probably more than Skype has as Tim Paton stated. That means if one of the companies starts a very appealing SIP service, it costs nothing to all those owners of SIP devices to join the club - a huge market already prepared.
Interconnection still drives a lot of money in telecom - a trusted world wide company that can offer direct SIP-to-SIP interconnect at cheap rates can secure good revenue from existing SIP-based VoIP operators just by offering a bypass of PSTN.
SIP was designed with IP networks in mind, therefore it has email-like addresses and DNS in the core of the protocol
Native extensions for end-to-end (proxy model, see below) Text Messaging and Presence
It works with any real-time media streaming communication - voice, video, desktop sharing, a.s.o.

Using SIP is not easy though, has to be done carefully, the most important aspect is to forget what a classic Telco does and design a new telecom model:

forget about implementing the intelligence in the core network - back-to-back user agent model does not scale to world-wide telco target and imposes restrictions on type of communication. Proxy model scales much more better and requires only the basics of SIP - just interconnect people and operators.
stick to basics of SIP - specifications that worked always and they are very simple to implement. If your engineers cannot design the service using less than 10 SIP related RFCs (3 being the core of the protocol), fire them, is going to be too much of classic telecom.
let the end device be in charge of communication - people like smartphones, they pay a lot for such devices and then want to use them to full features set
charge based on time or volume of data, not type of communication - I may have my new SIP application doing 3D holograms, don't hunt the content type to rise the price, it is a set of bits flowing around for anything, period
TLS as primary transport layer - protect people against ISPs blocking VoIP, ensures encryption of content (texting and presence) and privacy (who is calling who). Moreover, for Interconnect services will provide good authentication of peers, without the overhead of setting new access rules for any new server a company is powering up, removing as well the risk of SPIT
no allocation of random numbers - provide the option for people to choose their contact ID, email-like addresses is something people are familiar with and remember easily

Regarding the technology, if there is nothing developed in-house for the infrastructure part, using open source would bring real advantages. From the applications I am familiar with:

Kamailio SIP server for proxy - hard to beat scalability for routing SIP, including secure TLS communication from end user to core network or for interconnect, plus SCTP as a good choice for within core network communication
a good range of software for dedicated Voice application servers, e.g., Asterisk, Freeswitch or SEMS
gazillions of extensions already in place for add-on services and interaction with social networking or other IP services
access to source code to develop further enhancements in-house for new needs and be able to integrate with the other services offered by third parties or the company itself

How about you? What would you like to get from a new world wide telco? Which technology would be today the best to choose? Do you think of any other company able to run in the race?

Maketing - from an idea to first announcement

2011-05-31T20:48:00.001+02:00

[This is part of Maketing blog posts series]

Announcing a new open source project is one of the most important steps in its evolution. Whether it is something completely new or an alternative for existing projects, you must be able to communicate properly its target and the benefits.

Talking about software projects, try to do the big announcement when you have something running. People like to feel the products, in software that means they should be able to run something.

Hey, you are at beginning, it may be buggy, incomplete, but there should be something running. It gives the change to engage new people with feedback about issues and desired features.

On the other hand, if you have a cool idea but nothing to present, most of the people will forget about it soon. The first adopters are usually the techies, interested in the same domain and they guide after "show me the code first".

Depending on the market, time to bake a project before announcing it is also very important. With SIP Express Router (SER), baking time was about one year (note: it was actually used in research projects and demos) -- when it was released under GPL, the entire history of development was made public (kept in CVS internally at FhG Fokus Institute and then migrated to berlios.de).

SIP was at its beginning as well, so time was a good friend in this aspect during early 2000. Now it might not be the same. The development done internally during the first year allowed to come to the public with quite rich in features SIP server, the first of its type in open source. In just few months we had a consistent external community involved in the project, as users and developers.

Therefore, here is my list to consider when launching a new project:

release some source code that is working to some extent from day one
don't bake it too much internally - waiting too long you may miss good chances
create basic documentation - how to build and run the application
create a web site to present the project - it will help to easily refer to, spread the word and get listed in web search results
underline the target of the project - what it tries to offer and how it differentiate from similar ones
open communication channels to community - mailing lists, web forums, a.s.o. - keep the archive of discussions public, it will help others learn from old debates
abuse your friends and colleagues for initial feedback and first discussions - there are many watching the archive before joining, a dynamic community attracts new members
use social networking to rich more audience
be sure you are available to answer questions quickly in the early stage - spread the knowledge about the application so that community members can answer similar questions later to new members, lifting some load from you

If the launch is carefully prepared, you are half way done to ensure the success of the project. First impression matters!

Maketing - the art of building open source projects

2011-05-30T12:28:00.007+02:00

An open source project is more than source code - beyond the application itself it is about proper promoting, interactive community and vision.

No matter how good it is the application, if you are not able to promote it and attract a loyal community, then it is not going to succeed. Without transmitting the vision, defining a clear target and development path, community is not jumping on board.

This year is 10 years of SIP Express Router (aka SER) - the reference implementation of open source SIP server. I was involved at the top management of the project from its early beginning, then developing a different path through the fork Kamailio (former OpenSER) for about 3 years and since 2008 I am again full time in it - right now SER and Kamailio share the same source code, same developers, because of database structure constraints, they are still released as independent applications.

Looking back over the years, I realized that building successful open source project is beyond programming skills. With this post I am opening a series of blog posts to share from my experiences.

The posts will be tagged maketing. It is not a typo, just a hash tag resulted from:

make - one of the famous tools used in open source to build your application, very common in Unix/Linux world
marketing - strategies and actions of promoting

Coming in my mind to approach in future posts, not in this order and not limited to:

from an idea to first announcement
key aspects of code development - taking care of own code and contributing to parts from other developers
developers' interaction - the project counted over 80 registered developers from more than 10 countries, over 25 still very active, 5 joined in the past half a year
community - from the group of developers to thousands of users - engaging people beyond programmers
marketing - attending events, organizing events, news and articles
versioning and release policies - what are the important aspects to take care of
branding - how SER, OpenSER and Kamailio went from day one, an unknown name, to become very popular and recognized brand in the market at their time
handling forking - good and bad times
project management - its role, who should do it and how has to be done
penetrating the money market with open source

Expect stories based on my personal experiences, what I considered to be good and bad, how I dealt in special cases, what I think could have been done better. Maybe they will be useful for somebody in the wild to build better open source projects...

Update:

Following posts were released in this series:

Maketing - from an idea to first announcement

Kamailio v3.1.4 Released

2011-05-26T22:25:00.000+02:00

Kamailio SIP Server v3.1.4 stable is out – a minor release including fixes in code and documentation since v3.1.3 – configuration file and database compatibility is preserved.

Kamailio (former OpenSER) 3.1.4 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 3.1.4

Source tarballs are available at:

http://www.kamailio.org/pub/kamailio/3.1.4/src/

Detailed changelog:

http://www.kamailio.org/pub/kamailio/3.1.4/ChangeLog

Download via GIT:

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

Binaries and packages will be uploaded at:

http://www.kamailio.org/pub/kamailio/3.1.4/

Modules’ documentation:

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

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

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

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

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

Energy Efficiency and Performance Testing for Kamailio v3.0

2011-05-25T09:52:00.006+02:00

Jan Janak, one of the core developers of SIP Express Router (SER) - (since version 3.0.0, SER and Kamailio (former OpenSER, which started as fork of SER in 2005) are built from the same source code) conducted a very interesting research project regarding energy efficiency of VoIP systems during 2010, a collaboration between iptel.org and Columbia University.

The team used the source code from sip-router.org GIT repository from January 2010, which corresponds to Kamailio (former OpenSER) and SER v3.0. The latest stable series v3.1 shares the same internal architecture with v3.0.

As part of the research work, he could also gather some figures about capacity and performances of v3.0 with a quite complex configuration file (involving authentication and NAT traversal as well).

You can read the paper about energy efficiency at:

Green VoIP Article

The draft notes about capacity and performances of v3.0 are available at:

Performances and Capacity for v3.0

Some interesting results:

one instance of SIP server with 500 000 online users (mixed users - behind and not NAT routers) - consumed energy 210W
one instance of SIP server with 1 000 000 online users (no NAT involved) - consumed energy 190W
on a 32-bit machine with 4GB of memory and with 2.5GB reserved for SIP server, the server could support 43 000 simultaneous TLS connections - consumed energy 209W
80 000 permanent TCP connections, the SIP server could still handle at least 1000 requests per second and a connection arrival rate of 1000 new connections per second, done for 20 000 new connections. CPU load generated by the SIP server was from 6% to 8%.

I added a new section to the draft notes to list the enhancements done for the latest stable release (v3.1.x) that contribute to performance improvements, like asynchronous TLS, fine tuning of memory for TLS connections and raw UDP sockets.

Run your own Skype-like service in less than one hour

2011-05-23T11:50:00.010+02:00

In the past few weeks, there were couple of articles showing alternatives for Skype (like this one) - applications or/and services.

Why not running your own Skype-like service? Even better, why not do it using open source and free applications and spend 8.5 billions for something else?

The target is to build your own communication service that offers:

peer-to-peer and secure communication (encryption of the content sent between users)
voice calls between two users
video calls between two users
group voice calls (audio conferencing)
screen sharing
chatting - instant messaging between users
contacts list and presence status notifications

The service is going to use the open standard protocol - SIP (Session Initiation Protocol). Two open source application are playing the major roles in the network:

Kamailio SIP Server - for user authentication server and 'super-node' functionality of relaying the voice/video packets when necessary
Jitsi - the client application, a cross platform implementation running on Linux, Mac OS X and Windows

Installation of the SIP server is exemplified on a Debian Squeeze system, but any flavour of Linux can be used.

The step-by-step tutorial is available at:

http://asipto.com/u/2h

It took me about 15 minutes to get the service up and running following the guidelines presented (being very familiar with the two applications), however it should not take more than 1 hour to get everything in action.

Enjoy! Communicate secure and freely!

Remarks about LinuxTag 2011

2011-05-18T12:49:00.004+02:00

This year I attended the last two days of LinuxTag show in Berlin. Kamailio project had a booth at exhibition hall for the four days, but I arrived late for it from the Silicon Valley trip.

However, it was just in time to meet with most of the people that took care of project's booth. In a rare moment, due to spread across the world, there were 5 of the 11 members of Kamailio management team.
From left to right: myself, Elena-Ramona Modroiu (Asipto), Henning Westerholt (1&1), Raphael Coeffic (Tekelec, developer of SEMS project), Andreas Granig (Sipwise) and Carsten Bock (Telefonica/O2).

Henning had a presentation about Linux at 1&1, including remarks about usage of Kamailio. Carsten did the fast track, two slides presentation of Kamailio project.

The number of visitors was balanced during my two days there, Friday and Saturday, with waves of crowds. As usual, many of the well know open source projects were there, such as famous Linux and BSD distributions, CMS projects such as Drupal or Typo3, Mozilla, a.s.o.

Very interesting for us, we learned that the German Federal Office for Information Security (BSI) sponsors an integration project that aims to provide a secure, scalable and easy to use open source PBX platform when they presented the upcoming "Gemeinschaft" PBX version 4 that will use Kamailio and Freeswitch.

Yet another confirmation that secure communications, including SIP over TLS, will have a relevant boost this year in VoIP.

Inside our booth, we offered demos and presentation of various Kamailio use cases:

the telephony system implemented at 1&1
demo of SIP:Provider CE - full featured IP telephony platform
presentation of IMS extensions
how to use SIP Express Media Server (SEMS) for SIP signaling back-to-back user agent with Kamailio or as stand alone performant SIP media application server for features such as IVR, transcoding or prepaid engine

Is this the end of SIP over UDP?

2011-05-11T05:48:00.005+02:00

The recent move in market with the acquisition of Skype is going to impact many things on the VoIP space, in many cases in a positive way.

Most of the SIP traffic these days is over UDP - unreliable and unencrypted protocol - everything is sent in clear text over the internet: who you are calling to, content of the text messages, your presence states.

Skype made a reputation for itself as offering very secure communication. Microsoft is a marketing devil, no secret, any bit they can exploit will become a sales weapon. Secure communication is one of what they've just got.

To be able to fight in the SMB/enterprise market, the Unified Communication (UC)/VoIP solutions providers will have to deliver from now on secure communication systems. SIP has all the meanings of ensuring secure communication, but it was not deployed much in that way. The companies were focused to check the bullets in the list of the 500 old PBX features.

It was a clear increase in the demand for secure SIP communication platforms lately, but the Skype take over by Microsoft will accelerate it a lot. Otherwise it will be extremely hard to compete against the new Microsoft UC solutions.

Fortunately, the open source SIP-based UC applications are ready. On the server side, being a lot deployed in insecure environments (e.g., well known Internet), we directed a lot of efforts at SER/Kamailio open source SIP server project to ensure the security of unified communication sessions - signaling for voice, video or desktop sharing, instant messaging and presence.

Starting with v3.0.0, there is a brand new architecture for TLS communication, designed for scalability (e.g., up to 80 000 active TLS connections on an average server hardware). In v3.1.0 that work was completed with asynchronous processing of TLS connections, increasing substantially the capacity to handle such secure connections.

On the client side, open source softphone applications such as Jitsi can connect to the server through TLS and send the media stream via SRTP (secure RTP) or ZRTP, thus the entire communication is secure.

For client side, there are also many choices of SIP hard phones with TLS and SRTP support, such as Snom or Cisco. Therefore doing secure communication with SIP is possible, very easy. The time to promote and deploy it massively has arrived.

UDP for SIP may stay for some time in many SIP-based UC systems, but this moment can mark the start of its end.

SIP for Skype

2011-05-10T20:46:00.015+02:00

Let's put it straight. Skype was the simplest telephony service ever. Period. And that's perfectly ok.

Frankly, building a Skype-like service is damn easy. Hopefully, now since it is gone and not anymore a potential easy-to-buy target, it will wake up some people in decision positions from the major telephony operators.

It's just lately Skype added video group chat, but what was the rest?

audio conversations directly between two people
instant messaging between two people
audio group chat (audio conferencing)
multi-user chant (group chat)
desktop sharing
contacts list (buddy list) with presence states (offline, online with sub-statuses like away, do not disturb, a.s.o.)
secure communication - encryption for all cases above
interconnect to other networks (PSTN and SIP)

In the core network were just the group of authentication servers with the storage for accounts data (user profiles) and the gateways to break out to PSTN and SIP. There was nothing else done in the Skype core network, everything was done in the client side, offering peer to peer communication model. To localize and make possible to have the conversation with the called person, some instances of the client applications were acting as supernodes, guiding the signaling and media streams.

That's all folks!

Now, let's look at the SIP side. It was designed to be able to establish peer to peer real time communication (RTC) sessions over IP networks. Here is the classic SIP trapezoid:

The difference to Skype implementation is in essence that the core network servers do location services along with authentication and storing the user profiles. But the rest is practically identical as concept. Simple and straight.

What it takes to build a service such as Skype with SIP?

Here is the same bullet list saying if it is possible and how, exemplifying with open source applications (therefore very cost effective to design the service and deploy the components):

audio conversations directly between two people - yes, the very basic concept in SIP. Client applications, lot of them, e.g.,: Jitsi, Ekiga, Linphone, ... Server applications: Kamailio, Asterisk, FreeSwitch, SEMS
instant messaging between two people - yes, the end to end model using SIP MESSAGE request, widely supported. Client applications: Jitsi, Ekiga, Linphone. Server applications: Kamailio, Asterisk, FreeSwitch
audio group chat (audio conferencing) - yes, most client applications can do 3-way conferencing (small conference of 3 people), then Jitsi can do multi-user conferencing. In addition, instances of applications such as Asterisk, SEMS or FreeSwitch can be used as dedicated conference bridges, where anyone with a SIP capable phone can dial in and join group conversations.
multi-user chat (group chat) - yes, server side with Kamailio in a not standardized way, just reusing the SIP MESSAGE to carry the chat content and imc module to control the members of the chat room. Client applications can be also the host of multi-chat room.
desktop sharing - yes, use Jitsi as client and Kamailio as server
contacts list (buddy list) with presence states (offline, online with sub-statuses like away, do not disturb, a.s.o.) - yes, client application: Jitsi. Skype model is SIP end-to-end presence model which is very simple from server side application point of view. The SIP end-to-end presence is implemented in many other SIP client applications. Server side: Kamailio. Contact list can be managed locally or via XCAP. Jitsi is a very good client application for both. When XCAP is used, in server side you can use Kamailio with embedded XCAP server.
secure communication - encryption for all cases above - yes, via TLS for SIP and SRTP for media stream. Using Jitsi and Kamailio can be an example of doing such communication. Servers can run on different ports that will help going through firewalls and the TLS will ensure nobody can detect it is SIP.
interconnect to other networks (PSTN and SIP) - yes, numerous SIP-to-PSTN gateways or termination providers

Actually, looking at the points above, SIP as it was at its early stage of specifications (call setup, instant messaging and presence) is far more closer to Skype communication model than the latest IETF/ITU published specs in regard to SIP.

Where is the problem in IP telephony? Why no big telecom which had the infrastructure and the financial resources could do any revolutionary step in IP networks? Because they don't think simple and don't look at what is important for the most of the users nowadays. All that matters so far for operators (based on my 10 years in the area) are the roaming/inter-connect fees and the set of 500 old PBX features, which are not used more than 2 percent anyhow in a daily basis, and 95% were never used probably.

The change for a proper evolution would be very simple technically, practically is the hardest: because that's about the change of mentality and concepts.

First, stop pushing new SIP specifications with very complex architecture! They don't help neither implementers nor operators. Roll back to the basics of the SIP, it is really a good protocol for RTC. Forget about compressing signaling, bandwidth issues, gran'ma love to 10 digits dialpad, battery life, a.s.o., they are the false problems for you right now (btw, gran'ma loves to see gran'kids on the screen). Get out simple to use services, bring on board new customers with that. Start listening to young and enthusiastic minds, they know what they like to make their life easier, they have the future ahead. Just as that: build the services for your kids.

If you keep talking only with the classic telephony vendors, then the chances to evolve and develop new ways of communication through your companies are close to zero. It could be the Skype protocol "the one" for the future of telephony, it could be XMPP, or SIP can stay in for that, but that is the least relevant aspect.

The winner will be the proper model, not the technology behind it. Definitely the classic telephony is not that!

Evolution - face it!

2011-05-07T18:46:00.019+02:00

SIP/IP vs PSTN/H323 as concepts in the arena, again. It was flying on twitter a new SIP vs H323 head to head comparison (I assume it is new, there is no publishing date on the page, but they claim it is at least actual).

Just reading the start of the comparison, I had big doubts. They claim that people who did such comparisons in the past were unknowledgeable, students or similar, misleading with their articles. But in this case there isn't even an author (nor list of authors), so we know that those doing it now are proper specialists. For me is more that clear that they have no understanding of SIP and its intended role for real time communications (RTC) over IP networks.

I guess they are telecom people, based on their way of thinking and the reasons they wrote in the article, no matter it is venders or operators. So, first guys, sleep well, SIP is not a replacement protocol to upgrade PSTN network. H323 could have been designed for that, definitely not SIP.

Personally I will refrain from doing technical comments on H323. Simply I don't care about, it is not for the new wave of communication demands. Instead I will punctually tell how wrong the concepts about SIP are in this article. If I was one of the authors, first I would remove the paragraphs from the start of the article, it has ridiculous content: "this is, frankly, the best comparison of H.323 and SIP available anywhere", "comparing apples to apples", ...?!?! Or at least add the authors name so the others can check them and be sure they have right expertise with SIP.

Rarely I could see so much misunderstanding of SIP while claiming to know it. Now, point by point comments about the sections in the comparison article (note: you have to read the article with SIP vs H323 comparison more or less in parallel with the next comments for proper understanding):

Philosophy

First, yes, SIP is for establishing real time communication sessions, an audio call being just a possibility. And that's how it should be. Remember, SIP is not to upgrade PSTN protocol. It is for being able to cope with new communication demands.

Here is your biggest failure guys. SIP has very less to do with the session content. That is advertised in SDP (Session Description Protocol). So, being it audio, video or 3D holograms sessions, SIP does not need to be changed at all.

To rephrase, SIP is to establish the communication session, not to understand the content of the session. That's PSTN where the operator wanted to control the content and have the intelligence on the network, so all pieces around had to understand everything. Lately, we, the end consumers, don't like anymore dummy endpoints, but smartphones. And if these phones are able of transmitting something completely new, then I don't like after spending lot of $$$ on my device to have the operator limiting me. SIP is for that. Operator can charge on minutes, bandwidth, but not on content type -- that's my business. Do you like to be charged differently for a flight because you ate caviar than someone else that ate hot dog? Do you want to be forced to tell to the flight operator what you have eaten?

Now I hope the philosophy is clear: we talk about the "free as in speech" to do with what I own.

Complexity

Can you point the document (RFC) where SIP was extended for video or application sharing? How did you come up with these? Guys, google first...

SIP is really a simple protocol. Even the initial model for instant messaging and presence is very simple and works perfectly with all implementations (it is the end-to-end model, architecture used pretty much the same in Skype or XMPP/Gtalk). Then we got the mobile/telecom operators pushing for so called SIMPLE extensions which is indeed complex. That it is not SIP core, but newer extensions.

Reliability

I haven't understood a term from what it is claimed the H323 does there. But SIP has lot of mechanisms to deal with failure routing, from using DSN SRV (yes, cool down, this is fine, we are over IP networks here, where DNS is the heart) to doing serial forking. Ultimately, if it is the case, the user agent had to do any re-INVITE, perfectly ok because it is the one that knows about the content of the session.

Message Definition

Having university studies in both compilers and network communications, I can put just a smiley here: :-). Really guys, it is where you could get with it? I can start an endless debate here.

Message Encoding

Folks, take a deep breath and give me a break with bandwidth consumption by SIP signaling. The SIP compression concept you pointed were the most useless ever I met, however, pushed by vendors hoping to sell new boxes for few more millions of bucks to their money-milking operators.

Have you computed what is the percentage of the SIP packets in a 5 minutes call with GSM, G729 or G711 audio codecs? How much one could gain compressing less than 10KB? If you search through all SIP related RFCs you can get wrong and complex specifications, does not mean they have to be implemented or they will be ever. First, RFC stands for Request For Comments. People with ideas can write new specs, others can review and approve to become a RFC number, then it is the period to see if it worth a penny or not.

The large number of RFCs related to SIP gets to the conclusion that the protocol is very flexible, easy to extend and many were able to do that, period.

Just to freak you out, Twitter messages are maximum 140 characters long. What does it take to send a Twitter message? An HTTP request with very big JSON body. A HTTP request has more or less same amount (and structure) of SIP message. Then the body for Twitter is far more bigger than SDP for a SIP audio call initiation. Now, does anyone care about compression of Twitter packets? Does any of the users with smart phones twitting over mobile networks complained about? Guys, look around, we are in 2011.

Extensibility - Vendor Specific

I assume you heard that (the risk you claim for SIP) from the cousin of a friend of your wife/husband's colleague from high school meeting first time after graduation at the 30 years reunion. If not, then some links/examples would be good.

Extensibility - Standard

Evolution has some risks, not everything can be backward compatible. But the next phrase I quote from your article blew my mind:

"In addition, several extensions are "mandatory" in some implementations, which cause interoperability problems."

No more comments here.

Scalability - Load Balancing

SIP does not provide any in-protocol standard load balancing "specifications", it is in essence IP based protocol and there are lot of load balancing meanings here, like DNS based ones. Reporting usage can be done very easy by operators/vendors through replies and custom headers.

I am doing SIP load balancing solutions for so many years, never found anything missing in the protocol in this aspect. Lack of vision to apply a protocol is not a weak of the protocol.

Scalability - Call Signaling

Is this again about bandwidth of signaling packets?!?!? Com'on...

Scalability - Statelessness

I'm core developer for 10 years of a SIP Server and didn't know it has to be stateful for TCP. How is that, can you elaborate? We can do it stateless at SIP Express Router (SER) / Kamailio from the first day of TCP support, it's simply working. You say it is actually wrong? Is there any RFC requiring the server to be stateful? Please give the details.

Scalability - Address Resolution

3 is for sure the number making SIP bad. There are either 3 "full messages" or "as many as 30". Pointless what you mention at this section. How did you measure the "processing requirements"? Can you give the sample H323 and SIP packets used in measurements and how you made them? I would like to run the SIP packets through SER/Kamailio to see if the match the "processing" demands.

Come with numbers guys and proper arguments, otherwise is bullshit bingo article.

Addressing

First, about myself, when I'd see no digit-based numbers/IDs in in SIP, I will keep partying one week long. That means SIP got to the level to be used properly.

The part with "unofficial" convention is the best of the article, really, keep it up folks!

Billing

Wow, can't believe it. They are trusting the end device to report proper billing information. Great folks, why don't you start such a business over Internet? How long will take to update an open source softphone to report the call duration the user wants to? Wake up, Internet is open, it is not PSTN and you cannot control customer's devices anymore.

Call Setup

You tried hard guys. You admit H323 may require more elaborate call establishment for complex scenarios, but no example given. You show how is done in a similar case with SIP. Conclusion: it is possible with SIP, not sure it is with H323.

Capability Negotiation

Is it me and potentially my imperfect English, but the H323 side and SIP side have different kind of topic. I see no direct relation between what you say in one side and then compare against in the other.

PSTN Interworking

H323 borrows? I thought H323 is PSTN over IP, sorry if I got it wrong so far. I hope SIP will never become like this, we don't need PSTN over IP and that is the problem now with SIP. Many try to turn it in PSTNoIP protocol. I just pray they will fail.

Services

Yet again a section with random statements. I haven't heard of any new and cool service provider recently via "well suited" H323 protocol. And, fyi, XML, SOAP and CPL are pretty much standards.

Video and Data Conferencing

As mentioned above, you have no clue folks about the purpose of SIP. Does the pilot has any clue of what you ate before boarding the plane? How can he fly the plane then?

Codecs

SIP does not support any codec itself, and never will do. Keep reading the specs guys and understand the purpose of SIP.

Nowadays we call them smartphones those devices that need to create and understand the content of a session that is established via SIP.

Firewall/NAT support

You refer to 5 documents for H323 and 4 for SIP. So we are doing better in SIP then.

Loop Detection

Max-Forward cannot be used at all for loop detection in SIP.

Since H323 can fork, how it will deal with the case in the presentation linked in the SIP side?

If you mean that a gatekeeper is call stateful server, then SIP server can be the same and detect looped calls very easy. If you refer to call stateful H323 server and stateless SIP server, then tell me how is it with H323 stateless server.

Conferencing Entity

What's that in the SIP side: "no, ... conference bridge may provide ... " while in H323 is "yes, a MC is required...". Have you ever heard of SIP audio conferences through SIP with Asterisk, FreeSwitch & co?!?!

Open-source projects

I bet it was hard to choose the one of the SIP open source projects to list, but I am glad you have one for H323. How many H323 open source softphones are out there? What about server applications? Plenty I guess based on your observations here, just stupid Google does not find them.

Conclusions

The biggest fear I have is turning SIP in a protocol to bring PSTN in IP networks. Hopefully it will not happen, although standardization bodies are overwhelmed by guys with old telecom mentality and concepts.

As for the H323 vs SIP article it is written without any understanding of SIP. If the authors want to do a real comparison, bring two guys, one having proper expertise in H323 and the other in SIP, put the problems on the table and let them give the solutions.

The way that article is written shows only frustration and inability to understand and see the demands in the new waves of IP communications, nothing else.

SIP has to be interoperable and integrate easy and seamless with email, web, twitter and what so ever IP based communication service, not with PSTN. It must replace PSTN-type of communications, not just help to upgrade PSTN physical infrastructure.

We need evolution in telecommunications, SIP is the chance for that! But has to be done right. If not, SIP may disappear, however at that time the telco and mobile operators will be already replaced completely in our lives.

Two technologies striking back in VoIP - part II

2011-05-07T18:34:00.006+02:00

Do you remember VoIP/Telephony shows back in 2005-2007? It was IMS (Internet Multimedia Subsystem) all over. Then silence.
What killed the trend? Did everyone deploy it and this was it? No, there were many deploying pre-IMS, half-IMS or IMS-ready solutions (most of them being just some re-labeled existing system by the marketing/sales departments of some vendors) for astonishing amount of money, bringing back to them no other benefit that a SIP server with registrar, location and proxy services could deliver.

Many of us, including me, I was reticent about IMS. It was not because things like being a bad idea, but seeing who is behind the movement (read old telco vendors), organizations with few understanding of new communication demands over IP. They couldn't have it pushed beyond voice. Remember, IMS concept promised a framework to offer voice and new real time applications to mobile networks.

But voice was there, it had a dedicated channel (no IP) even in 3G networks. So an IMS with nothing more than voice brought zero add-value to services. There was no real return of investment for most of those deploying x-IMS.

Things are changing, demands for data over mobile networks is increasing, the operators are forced to go to 4G or what so ever will be next. Here we talk about technologies like LTE. With such technologies, there is no longer a dedicated channel for voice. Everything is IP.
Therefore the operators need a system that has to route the voice over LTE. One of the options, the natural one, is IMS.

IMS is a lot about defining standard interfaces, to handle services such as: interconnect, roaming, billing, access policies a.s.o. Not bad at all. Many concepts map directly, entirely or partially, over the standard SIP concepts, e.g., HSS (Home Subscriber Server) and SIP Registrar/Location Server.

It is almost mid of 2011, for few years there was a gap in (public and high) interest regarding IMS. But now is different. Starting with the end of last year, more and more people showed real interest in bringing IMS extensions in Kamailio. They are in the source code repository for some time, still under heavy work, but you can run it -- see the tutorial pointed at this web page.

First the big advantage for companies looking to use it is trying it at very low cost. Although the IMS extensions were available for some long time for free through OpenIMSCore project (which is part of our SIP-Router.org development eco-system), this time there are made available on top of a very stable VoIP/Telephony engine. All the IMS extensions are modules, so the stability of the other extensions and core is not affected at all.

More over, apart of voice, Kamailio (being the core of the IMS solution) offers add-on services such as presence and instant messaging out of the box. Furthermore, existing embedded programming language interfaces to Lua, Perl or Python offer a straightforwards and easy way to implement RTC goodies (e.g., see some slides with a tutorial for sending notifications to Twitter on missed calls).

I summary, with the IMS extensions on top of Kamailio:

free-as-in-beer system that anyone can try before deciding to pay integrators for professional installation
rock solid underneath system
access to other hundreds of RTC extensions
services beyond voice
embedded scripting languages
open source, vendor-unlocked environment, with many companies able to offer professional consultancy services

Forced by the need of a new voice system for LTE, backed up by cost effective and reach in extra features solutions like you can get with Kamailio, IMS has a good chance to succeed this time.

First part of writing about technologies striking back in VoIP was about IPv6. Same like there, things seems to get serious now (repeating the disclaimer from first part: the statements are made based on the activity inside our mailing lists, looking back at the discussions about these subjects in the past 10 years).

On our mailing list, Jason Penton (new developer to work on IMS extensions), revealed his company plans to deploy IMS in several countries across Africa (Uganda, Tanzania, Nigeria & DRC -- click to see the message).

The development of the IMS extensions is coordinated by Carsten Bock, long time developer and member of management team of Kamailio.

If you are interested in IMS, Carsten (along with other developers) will be present at Kamailio booth during LinuxTag 2011, in Berlin. We will be glad to talk with you, just drop by. We have several visitor passes that we can give away, write me an email if you are interested. Of course, flyers and giveaways will be on site, more details about developers availability at:

http://asipto.com/u/21

The Others - We 'Switched' the Seats

2011-05-06T15:37:00.012+02:00

The other day I saw on the web space the launch of a new mailing list about IPv6 by SIP Forum. SIP Forum seems to be an exclusive club of Telco vendors/equipment manufacturers and operators (annual membership fee $7500). Claiming to some extent they help promoting adoption of SIP, the group is well known, also maybe due to organization of various SIP related events such as SIPit (where I participated several times).

Do they help with SIP adoption? I don't think so. Why?

Just short details about what triggered this post now, but the facts are actual since many years. So I subscribed to the new IPv6 mailing list using my address used everywhere on all the mailing lists I am member of. It is from one of the very popular free emails providers. And I got rejected, with the following reason:

Your request to the IPv6 mailing list
Subscription request  has been rejected by the list moderator.  The moderator gave the following reason for rejecting your request:  "As a SPAM control measure, all SIP Forum WG mailing list subscriptions must be from an address (domain name) that has an obvious association with SIP and IP communications technology.  Please re-subscribe under such an email address, or send email to  describing the organizational affiliation with SIP and your interest in participating in the WG mailing list.  I hope you understand and appreciate our desire to keep spam from the list, and will bear with this inconvenience."  Any questions or comments should be directed to the list administrator at: [email address]

To clarify, I don't claim my name or address is something well know in the SIP world, that is the most irrelevant aspect in this context. For the record and the reason I'll point out later, in summary: I have 10 years working with SIP and only SIP, spoke every year at least 3-4 times at SIP related events, participated to many SIP Forum events (including SIPit in Stockholm where I helped setting up the IPv6 SIP testbed, providing several Kamailio instances on IPv6 to be used by all the others for various use cases testing). It was not even SIPv2.0 when I started (RFc3261, June 2002), since then managing and developing SIP Express Router (SER) and/or Kamailio (OpenSER) SIP server projects. The IPv6 support is in our project (May 2002) also before SIPv2.0 was out. My interest was not in learning anything, but helping others to figure out the way to SIP IPv6. We did our work back in 2002.

The issue here is something that's being a reality for many year, nobody is speaking about it. SIP has been hijacked from the purpose of innovating the real time communications over IP networks. It is going to be killed if the situation goes on like this.

Back in early 2000, people working with SIP were open minded, the specification written at that time were for IP networks and new services. It was small companies, enthusiasts, academic institutes and open sources projects really pushing SIP ahead. It was cool, it was fun. Just as an example: KPhone, a very early implementation of SIP softphone, it works even now with audio, video and, surprise, PRESENCE, based on initial specifications (end to end presence) -- and see author's picture, Billy Biggs, when he did it in early 2000.

Now, all organizations that eventually coordinate SIP independent activities, mainly including SIP specifications, are controlled by the old Telco vendors/operators and polluted with Telco mentality. If you are not one of us, fuck off. Besides that, basic concepts of real time communication became nightmares in specifications (did I say SIMPLE?), events and conferences have prohibitive fees, and I can continue...

Hey guys, you do have to retire! With this way of thinking you stopped innovation in SIP, you turned SIP in a protocol to put PSTN architecture on IP. It is not going to work, trust me. We do not longer need PSTN-type of communications, we need new ways of communication. You forced many ideas of services that could have been done very easy via SIP to hack-around solutions through web technologies or similar. Just because it was not a "heavy, well know guy in the gang" doing it.

I may be nobody or somebody in the SIP world now (provided the summary above) and can send an email justifying my interest in SIP and IPv6, that I am not a spammer and beg for acceptance. I'll eventually get it. But if I was just an young guy with good ideas with no experience in the field, just knowing what would make my life easier with VoIP? Should I fax photo ID with SSN and last bill to my Telco?

You, Telco guys, are here for ages. What you brought us lately? Nothing. All SIP desk phones, TDM phones and mobile phones (up to very few years ago) have more or less the interface from 1960 - 10 digits dialpad. You always claimed it is what people want. Right now, you, the vendors, don't how how to get out quickly your touchscreen devices (concept which was brought to the market by a non-Telco-related company) or, you, the operators, how to start social interaction services. Cheaper rates and any kind of innovation in telecommunications lately were because of open source, better and feature rich VoIP switches, plus the new alternative communication streams, don't even think to claim any of those.

You are not capable to change the trend of becoming a bunch of wires and antennas. Soon you won't be even useful for that, as major Internet companies are laying down fiber and power up wifi and wimax networks. I travel a lot, I rather pay 5 bucks for 2 hours internet, check the email, do some voip, video and messaging, than paying you 1 dollar roaming fee per minute.

Your end is coming unless you open your minds and let new, young and enthusiast people coming in and you listen to them. They know what they need, they build their future right now.

The other day experience shows you consider everyone else irrelevant, a bad guy or potential enemy. You could have banned me with my first spam message. You could mark on mailing list settings first (or all) post by 'untrusted' members to be moderated. No, my email was not known to moderator, I am high risk spammer for a 2-hours-ago started list. How many were registered? 10, 20? It was only the welcome message in the archive at that time.

I am not going to try joining again, with less relevance for the benefits of that mailing list. But how many brilliant minds were kicked out from Telco environment because the old dinosaurs in the key positions spending investors and share holders money love their commission with expired vendors? You stay on your private yachts now and need just to call secretary to ship cold champagne, thus, by now smaller, bricks with 10 keys dialpad are more than enough, it would have been even better with ONE KEY. But the rest need to properly interact and communicate.

Fortunately there were and still are THE OTHERS, those rejected with stupid reasons that made no point to retry in the same direction. And they succeeded, my SMS is now Twitter, Facebook & co, my voice (and video and presence which are yet years away to your offerings) is now my own SIP server most of the time, for the rest I have cool options such as Rebtel, Truphone, Gtalk or Skype.

I no longer want to be +123456789, I have a name and cool email address everyone important to me knows and it is easy to remember for them. Btw, if Enum would have made it out, being the unique 'contact address' for all communication meanings, were you able to detect faster that my Enum number belongs to a person with real interest in SIP and IPv6? Would you have been capable to do copy of the number and DNS NAPTR lookup faster than a copy of the email address and web search? Pathetic.

Guess what Telco guys, we were the others, those isolated, but now it is your turn to be. And we will make sure that you change for better or vanish!

PS. The guys behind IPv6 mailing list at SIP Forum might be good people, this is not something personal to anyone. The debate here was about the policies of such organizations, about the management thinking: a walled garden environment, always killing evolution, thing which is present in most of telecom related organizations.

SIP:Provider CE v2.2 RC1 is out

2011-05-02T05:49:00.000+02:00

SIP:Provider CE, a complete open source VoIP provider servicing platform, released v2.2 rc1. It uses Kamailio v3.1 for core SIP routing, Asterisk for voicemail and SEMS for B2BUA applications. See full release notes at:

http://www.sipwise.com/news/announcements/spce-v2_2rc1-release/

This version runs two instances of Kamailio, one acting as load balancer and the second for SIP registrar and proxy services.

Among SIP:Provider CE features: user web management interface, administration web interface and monitoring, call forwarding, click to dial, call blocking, speed dial, postpaid billing engine with individual billing profiles, peering, least cost routing, multi-domain, voicemail, IVR and topology hiding. See more details at:

http://www.sipwise.com/products/spce/features/

San Francisco, Silicon Valley, Trip

2011-04-26T22:16:00.005+02:00

I am going to spend few days in San Francisco, during May 06-11, 2011, visiting businesses and friends around.

If you want to meet and chat about Kamailio, SIP Express Router, SIP:Provider Platform or SIP and VoIP in general, drop me an email at

< miconda [at] gmail.com >.

Two technologies striking back in VoIP - part I

2011-04-25T15:50:00.006+02:00

Living a Deja-Vu?!? Or maybe this time the dreams (of IETF/ITU/ETSI specs writers) come true ... who knows. This is a two parts story, about two technologies which many thought went dead long time ago but they are, again, actual in VoIP: IPv6 and IMS (Internet Multimedia Subsystem).

IPv6, the old story is on first pages again. Back in 2002 I worked at FhG Fokus Research Institute in Berlin, developing SIP Express Router (aka SER), within a majority of European Commission funded projects focusing on IPv6.

One may think this is not strictly related to VoIP, well, it is then at least very related to SIP, the leading routing protocol in VoIP these days. Why? Because SIP was actually designed for IPv6 networks. Why? Don't ask me.

Just slightly off-topic note, to refresh, for the past 10 years we had to invent, smash, re-invent solutions and do workarounds to get VoIP going through IPv4 networks -- or more specific saying, through NAT routers, since NAT was not considered in SIP specifications at all. Mixing STUN, TURN, COMEDIA and RTP relays, slipping on ICE, mangling headers and SDP, we got it working. Not few times, friends know that well, it had to be very specific hack per user agent or even firmware.

Back to IPv6, in Kamailio it has been IPv6 support to the core and main modules since May 2002, during the age of SIP Express Router v0.7.y, that's 9 years ago. Looking at the mailing list archive, it is now obvious the interest in the IPv6 is for the first time serious. Of course, the engine behind is the IPv4 address space shortage. Still there is no significant move towards IPv6 as reported recently. There are many (of us) not that happy to remember hex-based IP addresses, which may not be short at all, by the way.

If I hate (phone, digit-based) numbers, preferring instead SIP addresses (username@domain), think about the love I care for hex-numbers.

Moreover, fixing SIP for IPv4 somehow broke SIP for IPv6. Why? Because now we have to support all the nasty tricks specified to make SIP work for IPv4.

The IPv6 was started about a dozen years ago, initial battle was lost, now it is another one. Will it be the final battle? If IPv6 doesn't win this time, then the war is over for it.

To win, it requires fast and smart moves:

first and most important, is the adoption of IPv6 at the access level - core networks and home routes
then SIP devices (hard and softphones) with proper IPv6 support

If these two are not solved quickly, IPv4 shortage might be workarounded.

The last idea has a strong reason, non-technology related, but at the same time the biggest enemy for IPv6: the cost to switch to it. Internet is way bigger than 10 years ago, replacing/upgrading every piece running IPv4 now is going to be amazing expensive. Time was not the ally for IPv6 at all.

In the next post in this series it will be about IMS, yet another hot topic these days, might not be for VoIP, but surely is for VoLTE...

Maybe in the future we will get to the same situation for Enum? What about mobile IP?!?

Kamailio Booth at LinuxTag 2011

2011-04-21T19:15:00.004+02:00

Kamailio - the Open Source SIP Server project has a booth at LinuxTag, Berlin, Germany, May 11-14, 2011.

Location is Hall 7.2b, booth 112 (near to the Mozilla project).

Come and meet some of us, we will be glad to chat about Kamailio, SIP Router, SEMS, SIP, IMS and VoIP in general. Depending on the day, you can meet at stand:

We will have a demo running latest version and nice flyers to show the capabilities of Kamailio based platforms.

Kamailio v3.1.3 RPMs Available

2011-04-05T19:11:00.000+02:00

The RPM packages for Kamailio recently released v3.1.3 are now available for CentOS, Fedora, RedHat and OpenSUSE, available on the opensuse.org build service:

https://build.opensuse.org/package/show?package=kamailio&project=home%3Akamailio%3Atelephony

Specific download links are:

Kamailio v3.1.3 Released

2011-04-04T21:10:00.000+02:00

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

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

Resources for Kamailio version 3.1.3

Source tarballs are available at:

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

Detailed changelog:

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

Download via GIT:

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

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

Modules’ documentation:

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

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

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

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

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

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

Kamailio at Google Summer of Code 2011

2011-03-27T13:46:00.000+02:00

For the second year, Kamailio project has been selected in Google Summer of Code program, now for year 2011, within Jitsi organization (former SIP Communicator project).

This time the participation is targeting the implementation of a SIP-to-Jingle gateway. Jingle is the voice extension of XMPP. At this moment, Kamailio includes a SIP-to-XMPP gateway for Instant Messaging and Presence, therefore the GSoC project is looking to extend it for voice.

Read more about the project and, if you are a student and interested to implement the project, see how to apply at:

http://www.kamailio.org/w/gsoc-2011/

SIP Communicator Renamed to Jitsi

2011-03-18T13:44:00.000+01:00

Here is a short news just to inform our community that the SIP Communicator SIP softphone was renamed to Jitsi. There are several pages mentioning Kamailio and SIP Communicator out there, among them our participation to Google Summer of Code 2010 as well as the full SIP SIMPLE Presence configuration with XCAP:

http://asipto.com/u/sp

To fetch the latest version of Jitsi SIP softphone, just go to:

http://www.jitsi.org

Note: Kamailio has applied for the second year for participation to GSoC, now for the 2011 edition, again together with Jitsi project. Stay tuned for updates on this topic!

SEMS 1.4.0 Released

2011-03-15T13:43:00.000+01:00

SIP Express Media Server (aka SEMS) version 1.4.0 has been released. SEMS is a high performance media and application server for SIP based VoIP networks started at the same research institute with SIP Express Router (SER), thus having same origins as Kamailio as well as sharing several developers. SEMS interoperability with Kamailio is therefore guaranteed, being the first option for a SIP B2BUA whenever is required in a Kamailio deployment.

This release of SEMS features a powerful Session Border Controller (SBC) module. From completely transparent B2BUA to customized URI/From/To, strictly filtered (messages, headers, codecs) with RTP anchoring, Session Timer enforcement, prepaid and call timer, the SBC facilitates interconnect and core routing in a simple and secure way. Thanks to the new multihoming support, SEMS can now be employed at the border of the networks.

This addition also allows to overcome the bottleneck of one NIC – giving the possibility to fully exploit SEMS’ great performance. In the app development area, the DSM language has matured to become a viable candidate also for implementing complex application logic, thanks to language constructs like for, if and functions. SEMS can be downloaded in source from its ftp site at:

http://ftp.iptel.org/pub/sems/1.4/1.4.0/

Podcast and article on Techistan about Kamailio

2011-03-10T13:42:00.000+01:00

From its website, Techistan, an online technology magazine, defines itself as: “Techistan is a time and place where peaceful debate and discourse on technology rule. It is the gathering of the brightest in the world, Techistan and you.”

Courtesy of Suzanne Bowen (DIDx) , Techistan has published an article about Kamailio, along with a podcast with Daniel-Constantin Mierla, co-founder of Kamailio project:

click here to go to the Techistan article

The discussion begins with the origins of the project, that started back in 2001 as SIP Express Router (SER), forked in 2005 to OpenSER, then it was renamed in 2008 to Kamailio. Shortly after the rename, Kamailio and SER projects started the work of integration, at this time being again same software application – development being hosted at sip-router.org.

The year 2010 is summarized during the podcast (see also Kamailio 2010 Awards), marking the release of two major versions based on the new architecture, v3.0.x and 3.1.x – development done together by the two teams of Kamailio and SER:

Today’s world wide hot topics such as reach communication services or security were debated, underlining Kamailio’s advanced implementations for such services, from a SIP SIMPLE presence server with RLS and embedded XCAP server to asynchronous TLS communication able to handle several ten thousands active connections on a single instance running on a laptop.

The presentation approached the recently launched SIP:Provider CE project, an out-of-the-box VoIP servicing platform with Kamailio for SIP routing and management interfaces for administration, monitoring and user portal. It ended with a look at the future, the current work to integrate IMS extensions and where we will be in real time communications in few years.

Joint venture on Kamailio to tackle big vendors in telco market

2011-03-01T19:44:00.002+01:00

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

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

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

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

You can read the announcement here.

Silicon Allee, Berlin

2011-02-28T00:04:00.000+01:00

Silicon Allee - an initiative by Schuyler Deerman & friends tries to setup a monthly meeting for people living or visiting Berlin area to exchange ideas and discuss the future of technology.

Even I am unable to make it to the first meeting (March 7, 2011), I am looking forward to seeing the evolution of this initiative and participate to the next ones.

UC Expo 2011

2011-02-25T23:55:00.000+01:00

UC Expo 2011 takes place in London, UK, between March 08-09, 2011. I 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, feel free to contact us:

Go to Contact Page

CeBIT 2011

2011-02-20T23:48:00.000+01:00

CeBIT 2011, the biggest digital show, takes place in Hanover, Germany, March 01 – 05, 2011. I am visiting the event the event on Thursday, March 03, 2011.

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 2011, don’t hesitate to contact me:

Go to Contact Form

Kamailio 2010 Awards

2011-02-20T21:50:00.013+01:00

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

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

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

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

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

Let the show begin...

Blogging:

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

Related Projects:

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

Technical Support:

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

New Contributions:

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

Developer Remarks:

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

Advocating:

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

VoIP Services:

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

Business Initiatives:

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

Events:

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

Academic Environment:

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

Innovation in communications:

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

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

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

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

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

Call Center World 2011

2011-02-17T23:53:00.000+01:00

Call Center World 2011 is taking place in Berlin, Germany, Feb 21-24, 2011. Localized in the same city with Asipto, the event is a great opportunity to meet and discuss our latest offerings and developments.

Strictly related to the topic of the congress, our load balancing and security solutions are ideal for scaling and guarding IVR systems.

We will be around as usual during these days, feel free to contact us if you want to meet:

Go to Contact Form

Kamailio Presentations at Fosdem 2011

2011-02-10T22:23:00.001+01:00

Fosdem 2011 edition was again very successful. The only problem that comes (naturally) to this event: it started to be very crowded, making a challenge to attend any kind of speech there.

For me was the opportunity to meet with many folks around Kamailio project and VoIP, most of them known from past editions, but every time I return from Brussels with a bag of new connections.

Saturday evening we had the traditional Kamailio dinner (we should turn it next year in Open Source Telephony dinner), this time in a fancy restaurant, but with very tasteful menu choices.

During the days we tried to use the breaks at maximum and discuss a bit about the future development in Kamailio and what is new in the VoIP world.

In the Open Source Telephony Devroom, Kamailio had two presentations:

Henning Westerholt and Marius Zbihlei - Scaling Location Services in Large SIP Networks with Kamailio - slides available here
Myself - Unifying SIP and Web Worlds with Lua - slides available here

Kamailio v3.1.2 Released

2011-02-08T20:58:00.000+01:00

Kamailio SIP Server v3.1.2 stable is out – a minor release including fixes in code and documentation since v3.1.1 – configuration file and database compatibility is preserved.

Kamailio (former OpenSER) 3.1.2 is based on the latest version of GIT branch 3.1, therefore those running 3.1.0 or 3.1.1 are advised to upgrade — there is no change required to be done to configuration file or database.

Resources for Kamailio version 3.1.2

Source tarballs are available at:

http://www.kamailio.org/pub/kamailio/3.1.2/src/

Detailed changelog:

http://www.kamailio.org/pub/kamailio/3.1.2/ChangeLog

Download via GIT:

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

Binaries and packages will be uploaded at:
http://www.kamailio.org/pub/kamailio/3.1.2/

Modules’ documentation:

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

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

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

Social Networking Event, Brussels, Belgium

2011-01-28T02:58:00.000+01:00

In about one week the annual open source developer conference FOSDEM (http://fosdem.org/2011/) takes place in Brussels. Like in the years we would like to meet there for a social event, a dinner on Saturday evening, 6th of February.

There is already a bunch of confirmations, so far are coming:

Daniel-Constantin Mierla
Marius Zbihlei
Henning Westerholt
Timo Reimann
Stefan Sayer
Raphael Coeffic
Emil Kroymann
Olivier Taylor

So if you want to participate, please speak up now! Send an email to Henning Westerholt to reserve your set: henning.westerholt [at] 1und1.de

By the way, the schedule for the Open Source Telephony developer room is now final, two talks about Kamailio, drop by do learn about our latest developments:

http://fosdem.org/2011/schedule/track/open_source_telephony_devroom

IMS Extensions Available for Testing in Kamailio

2011-01-17T21:36:00.001+01:00

Courtesy of Carsten Bock and his team, several IMS extensions from the OpenIMSCore project are ported and directly available for testing built on top of latest development version of Kamailio project. Right now, the IMS extensions are held by carstenbock/ims branch in GIT repository.

For testing at this moment is better to use the Debian repository made available for this purpose:

- install the Key:

wget http://repository.ng-voice.com/ngvoice-debian-gpg.key
apt-key add ngvoice-debian-gpg.key

- add our repository to your “/etc/apt/sources.list”:

deb http://repository.ng-voice.com lenny main contrib non-free
deb-src http://repository.ng-voice.com lenny main contrib non-free

- to install Kamailio & IMS Modules:

apt-get install kamailio kamailio-ims-modules

In order to update the packages, you need to remove any previous installation of kamailio from Debian repository (apt-get remove kamailio), clean your local deb-cache (apt-get clean) and reinstall the packages.

The repository also contains other packages, such as:

- the RTP-Proxy (for the P-CSCF, package rtpproxy, latest stable version)
- the Fokus FHoSS HSS-Server (package openimscore-fhoss)

More installation notes here:

http://www.ng-voice.com/our-solution/installation/

Here is a summary of available IMS modules:

CDP / CDP-AVP

The CDP (C-Diameter-Peer) modules provide an Diameter-interface, which are used by several components of the OpenIMS-core: They are used as Cx-Interface for the I-/S-CSCF and for the Rx-Interface for the P-CSCF. The modules may be used in other ways, too (e.g. for an Sh-Interface for an Kamailio-based aplication server)

P-CSCF

The Proxy-CSCF in the IMS Architecture acts as an entry point to the network. The pcscf module of the original OpenIMS-core aggregates many functions required at this component: Header manipulation/verification, RTP-Relay and presence-support for the “reginfo”-event. Optional, the Rx-Interface for Billing may be enabled.

I-CSCF

The Interrogating-CSCF is a kind of “Loadbalancer” or a entry Proxy for the “home-network” of an IMS setup. The I-CSCF will retrieve the location for a user from the HSS, it will check, where a user is registered or where it should register (based on user-settings, required capabilities later maybe even load). The icscf-module implements the according interfaces towards the HSS (Cx) and according header manipulation/verification methods.

S-CSCF

The Serving-CSCF is acting as a registrar and as decision engine regarding the routing of the Request. It retrieves the user-data and routing rules from the HSS and applies them to the processed requests. The scscf-module implements the according interfaces towards the HSS (Cx), the interfaces towards application-servers (Isc) and according header manipulation/verification methods.

MGCF

The MGCF-Module of the OpenIMS core implements header and content manipulation for interconnections towards Class 4 networks.

E-CSCF & LRF

The Emergency-CSCF and the Location Resource Function (LRF) implements IMS compliant emergency call routing. The modules provide required content aggregation methods.

An open source Home Subscriber Service project is under development as well:

http://sourceforge.net/projects/openhss/

World of SIP in 2011

2011-01-06T23:00:00.022+01:00

Not really predictions, but more what is driven by current situation in SIP world, here I present my thoughts about this market from the perspective of Kamailio Project co-founder and leader these days.

2011 is my 10th year in the SIP world. Officially I started on 1st of January 2002, few months after I graduated Computer Science University I joined the development team of SIP Express Router (SER) at Fraunhofer FOKUS Institute in Berlin, Germany. Since then I was involved in top management and development of this line of open source SIP servers: SER-Kamailio (formerly named OpenSER). All my living revenue was and still is generated from SIP world. Lot of things I would have liked to be reality by now, haven't happened yet, but it is about the time for some of them.

Therefore is more about feeling the directions where things are moving right now and trying to comment a bit and show where Kamailio can help at very best.

SIP over TLS will take off to the masses

2010 exploded in terms of VoIP attacks and fraud. Weak protection provided by most widely used transport protocol in SIP, namely UDP, along side with digest authentication and raw text based IP communication are making the life too easy for attackers.

TLS makes the phone-to-server SIP signaling channel fully encrypted. Major SIP phone vendors such as Snom showed a clear focus on security as well and many other SIP phones support TLS. As SIP is now more and more the first telephony line for many of us, we like to keep everything about it private. I don't think the ISP needs to know who I am calling and how long (yes, yes, I know they (say they) don't monitor it, right, yeah), it has to be only between the me and my ITSP because I have confidentiality as part of the contract with them for telephony services.

As for attackers, ITSPs can use free of cost self-signed certificates which will put quite some tough barrier to any kind of attack. For peering, ITSP-to-ITSP, I will expect many will start to allow traffic from other ITSP even they don't have preset peering agreement, but only based on a trusted certificate signed by recognized certificate authorities.

Regarding Kamailio, since version 3.1.0, it has support for asynchronous TLS communication (before this enhancement, Kamailio 3.0. with its new core architecture, was able to scale up to 80 000 active connections on an usual server hardware for residential user type of SIP traffic, now the numbers should go much higher). That rules out any alternative of SIP server out there by orders of magnitude.

IPv6, the time has arrived

My good friend Olle is on the barricades for several good months by now, trying to wake up in time the SIP world to be ready and start the switch to IPv6. At the SIPit in May 2010, I and Olle set up a Kamailio test bed to help implementers advance with their IPv6 support.

Kamailio had IPv6 support before TCP, so IPv6 is approaching as well 10 years. In the early 2000, IPv6 was a hot topic at least in the research area, so we added it and we improved over the years. Core and main modules are fully compliant with IPv6 on all transports layers, very well tested. With the main developer of IPv6 support in SER-Kamailio member of our development core team, you are in the safe boat for the future.

You like it or not, time for IPv6 is now -- and we are ready.

Re-empowering the SIP proxy model

Back in 2002, when I got involved in SIP, the communication model was 'the SIP endpoint has the power'. But somehow that was changed by Telecom and Mobile Operators adopting SIP and trying to replace legacy telephony services not by adapting communication model, but by enforcing legacy architecture to SIP.

The old telephony model was based on very dummy endpoints (the less by now 5 bucks microphone+speaker+dialpad wired to the switch or pbx, i.e., can and wire) and smart core network, but where are we today?

Look at the market trends, everyone is buying (hey Telcos, heads and (both) ears up!!!) SMARTPHONES. Do you think I will keep using your service with my $$$ device and cannot benefit of it for video, 3D holograms or what so ever fancy things may show up tomorrow because you stick to old Back-to-Back User Agent (B2BUA) model and those are not supported YET (if ever) by the two UA you placed in the middle, face-to-face to me and my callee?

You better reconsider that, let me and my caller negotiate the communication type, I am paying to you already IP connectivity fee, if you want my money for telephony, I should be able to decide my communication type. And I WANT to talk with my caller, not with a half of your dummy B2BUA.

I am not a B2BUA believer at all, I admit it has a good role for certain cases (e.g., gatewaying), but for the core of communication, that has to be completely removed. Look out there: Facetime, Facebook, GoogleTalk, Yahoo, Skype - the customer (client) is the smart side and that is reflected in service growth.

ITSP will have to adapt and let communication capabilities in hands of customers, or die. They will offer the communication infrastructure (like location services, routing failover, call hunting, hosted routing preferences, a.s.o.) and server side add-on services (that will act like another end point).

Not much to underline about Kamailio in this story, it is by definition transparent to UA capabilities. Do you have two SIP phones able to do 3D holographic calls, just wire them to Kamailio, dial and ENJOY!

SIP and Web

If we talk about telecommunications, today or in the future, it is SIP. But that is not all to satisfy current communications needs and the most important complementary service is the Web. Still to decide which one leads the real time communication, the fact is we want both and we need both.

Blending the both systems in one service is for sure a win bet. API defines the interaction for and between various very popular services these days, and these APIs are carried mainly via HTTP and Web2.0 technologies. Smart ITSPs will start offering such interfaces as well.

The latest version of Kamailio has an embedded HTTP server and a very scalable XMLRPC interface. Therefore you can jack directly into Kamailio via HTTP(S). Do you want to initiate a click-to-dial, check you missed calls or call history, verify your credit level? It is straightforward for you as ITSP to implement that with Kamailio and offer new attractions to your customers, with a single sign-on password for telephony and plus-value web-sip services.

Watch this video of a presentation by Robin Rodriguez of Ifbyphone at Cluecon 2010 just to get how simple is to add new SIP-Web services to your ITSP offering with latest Kamailio.

That's all folks! I still keep few ideas I would like to be able to use in my pockets for later on, more likely suitable for 2012. What is your take regarding the evolution of SIP world in 2011?

Kamailio Development Training, Barcelona, Feb 10-11, 2011

2011-01-05T21:30:00.000+01:00

Next Kamailio Development Training takes place in Barcelona, Spain, during February 10-11, 2011, organized by Asipto and Voztelecom.

The coordinator of the training is Daniel-Constantin Mierla. The goal of the event is to teach how to write your own code for Kamailio SIP Server. Note that it is not a training for VoIP administrators looking to learn how to configure and operate Kamailio-based SIP-VoIP platforms.

During the two days in the class, following topics will be approached:

internal architecture
SIP parser
memory manager
locking manager
database API
config file language interpreter
RPC interface
pseudo-variables and transformations framework
module interface – write your own extensions in C as modules
documentation docbook format

The price per attendee is 160 Euro.

Number of seats is limited and access will be granted in first come first served fashion. Registrations or requests for more details must be done via email at registration [at] kamailio.org .

For people that happens to be in the area but not interested in programming C and/or not participating to the class, there will be a Social Networking Event during the evening of Feb 10, where we meet for food, drinks and discussions about SIP, VoIP and more.

Participation to this event is open and free for anybody, very likely to be a dinner or a pub session where every participant pays for himself/herself. Registration is still required so we know how many seats to reserve and send you the location of the event. Among people you will meet there are going to be Daniel-Constantin Mierla, Jesus Rodriguez and Inaki Baz-Castillo.

Social Networking Event, Irvine, CA, USA, Jan 25, 2011

2011-01-03T22:10:00.001+01:00

Since I am traveling to California for the Kamailio Advanced Training, I am taking the opportunity to host the first Social Networking Event for Kamailio and SER Projects in 2011.

It is the usual free event, very likely to be a dinner or a pub session, where every participant pays for himself or herself. It is going to happen on Jan 25, 2011, in Irvine, CA, exact location will be emailed to participants. If you want to participate, send an email to registration [at] kamailio.org. As usual, the event is open for anyone interested in our projects, SIP or VoIP in general.

To get a feeling about how is going to be like, you can visit the photo gallery from past similar events:

http://www.asipto.com/gallery/v/social_meeting_barcelona2008/

http://www.asipto.com/gallery/v/FOSDEM2009/

Prepare yourself for tough discussions about present and future of SIP and VoIP along side some good food and drinks.

Looking forward to meeting many of you during 2011!

It has to be a great year

2011-01-01T21:50:00.002+01:00

In autumn of 2011, the first line of code in Kamailio & SIP Express Router (SER) projects is celebrating 10 years of existence. That being said, we have to make 2011 a special one.

On personal records, I am starting the 10th year of working with SIP and only SIP. Either as a researcher or as private consultant, starting with 1st of January 2002 I made a living out of SIP and VoIP.

Stay connected and watch the project closely, this year will be amazing!

Happy New Year!

2010-12-31T19:27:00.001+01:00

Looking back to 2010, it was an amazing year Kamailio project - two major releases v3.0.x and v3.1.x, lot of new features, all in top of a better and more scalable core we have now after the integration of Kamailio with SER.

Just to mention few here: asynchronous TCP and TLS, embedded Lua, XCAP server, HTTP server, pre-processor directives, full SCTP implementation with multi-homing, geoip API, asynchronous message queues, over 20 new modules -- for sake of completion, here are the links with the announcements:

http://www.kamailio.org/w/kamailio-openser-v3.0.0-release-notes/
http://www.kamailio.org/w/kamailio-openser-v3.1.0-release-notes/

As for 2011, we will continue to deliver new features, but maintain as well our line of scalable, secure and rock solid SIP server. New projects related to Kamailio already announced their launch, participation to VoIP and Open Source events already booked, new thinks are baked and getting ready to launch soon -- 2011 is going to be for sure fascinating as well as challenging.

Thanks for your support for the project! I wish you a great 2011 in personal life and business, enjoy tonight party!

Happy New Year!!!

Merry Christmas!

2010-12-24T13:40:00.000+01:00

I could safety say that 2010 was the greatest year by achievements in the history of Kamailio and SER SIP proxy server projects so far. It will not be short to summarize it, so I will do a separate post for that.

Now I just wast to thank everyone making that possible and wish a Merry Christmas to all my friends, Asipto customers, developers and users of Kamailio and SER!

Great winter holidays to all Kamailians and SERians!!!

SIP:Provider CE

2010-12-13T17:11:00.000+01:00

Andreas Graning of Sipwise announced today the release of sip:provider community edition – home page link:

http://www.sipwise.com/products/spce/

It is a open-source SIP based Class5 VoIP soft-switch leveraging the capabilities of Kamailio, SEMS and Asterisk, combined with custom components in order to provide consistent and easy-to-use provisioning, billing and configuration maintenance.

SEMS 1.3.1 Released

2010-12-13T12:55:00.000+01:00

SIP Express Media Server (SEMS) released 1.3.1 with several bug fixes and improvements since 1.3.0, among them:
- fixed CMake build scripts
- fix for architectures w/o atomic built-in functions
- add lost accept_fr_without_totag sample cfg option
- fixed missing CRLF in transfer header

The tarball can be downloaded from:
http://ftp.iptel.org/pub/sems/sems-1.3.1.tar.gz

SEMS is a lightweight media server that can be used along with Kamailio to provide back-to-back user agent functionality, voicemail, IVR, audio conferencing, a.s.o.

Kamailio v3.1.1 Released

2010-12-02T22:41:00.000+01:00

Kamailio v3.1.1 is out – a minor release including fixes in code and documentation since v3.1.0 – configuration file and database compatibility is preserved.

Kamailio (OpenSER) 3.1.1 is based on the latest version of GIT branch 3.1, therefore those running 3.1.0 are advised to upgrade — there is no change required to be done to configuration file or database.

Resources for Kamailio version 3.1.1

Source tarballs are available at:

http://www.kamailio.org/pub/kamailio/3.1.1/src/

Detailed changelog:

http://www.kamailio.org/pub/kamailio/3.1.1/ChangeLog

Download via GIT:

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

Binaries and packages will be uploaded at:
http://www.kamailio.org/pub/kamailio/3.1.1/

Modules’ documentation:

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

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

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

FreeSwitch as Media Server and SBC for Kamailio 3.1

2010-11-30T21:16:00.000+01:00

There is a new version of step by step tutorial about using FreeSWITCH and Kamailio together for large VoIP platforms.

Besides upgrade to use latest Kamailio major stable release, v3.1.0, there are couple of new features added in the architecture of the VoIP platform:

along with providing media services (voicemail, conferencing, a.s.o.), FreeSwitch is used now also as SBC for topology hiding and media relay (this helps for transconding needs, playing audio messages during the early session or NAT traversal)
Kamailio config has added features to detect DoS/DDoS and scanning attacks, secure SIP communication over TLS, IP authentication and a bunch of neat things that help for an easier maintenance and update of parameters such as database connectivity details, local IPs, etc.

The link to tutorial is:

http://asipto.com/u/kfsb

Asterisk 1.6 and Kamailio 3.1 Realtime Integration Tutorial

2010-11-28T21:54:00.003+01:00

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

http://asipto.com/u/c

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

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

Experiences from 18 Hours of SIP Scanning Attack

2010-11-18T14:05:00.001+01:00

During the testing period of Kamailio 3.1.0, while running it at voipuser.org, I had the chance to watch live and analyze a SIP scanning attack.

Yesterday I noticed another one by looking at Siremis 2.0 charts, therefore I wrote an article with some hints about what you can use to protect your SIP services within Kamailio configuration file.

You can read it at:

http://asipto.com/u/i

SIP Routing Logic in Lua with Kamailio

2010-11-17T20:38:00.001+01:00

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

http://asipto.com/u/h

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

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

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

2010-11-06T14:23:00.004+01:00

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

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

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

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

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

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

Next is an example of usage:

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

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

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

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

Here you find the online documentation for mqueue module:

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

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

2010-11-02T19:18:00.001+01:00

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

Last stable series of Kamailio SIP Server, the 3.1.x (Oct 06, 2010, see release notes), continues the work done within SIP-Router.org project. Among brand new features in v3.1.0, starting with the previous major version, 3.0.0, you can run mixed Kamailio (OpenSER) and SIP Express Router (SER) modules in the same SIP server instance, giving you the most powerful tools to build stable, very performant and features rich VoIP and Unified Communication platforms.

The class is organized by Asipto in collaboration with Flowroute and will be taught by Daniel-Constantin Mierla, co-founder and core developer of Kamailio SIP Server project.

Read more details about the class and registration at:

http://www.asipto.com/index.php/kamailio-advanced-training-usa/

Siremis v2.0.0 Released

2010-11-01T19:16:00.000+01:00

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

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

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

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

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

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

2010-10-29T20:04:00.000+02:00

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

Last stable series is 3.1.x (Oct 06, 2010, see release notes), continues the work done within SIP-Router.org project. Among brand new features in v3.1.0, starting with the previous major version, 3.0.0, you can run mixed Kamailio (OpenSER) and SIP Express Router (SER) modules in the same SIP server instance, giving you the most powerful tools to build stable, very performant and features rich VoIP and Unified Communication platforms.

The class is organized by Asipto and will be taught by Daniel-Constantin Mierla, co-founder and core developer of Kamailio SIP Server project.

Read more details about the class and registration at:

http://www.asipto.com/index.php/kamailio-advanced-training/

Kamailio Business Directory Launched

2010-10-28T20:00:00.000+02:00

Today was announced the launch of Kamailio Business Directory.

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

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

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

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

Asterisk, Kamailio, Openfire and Social Media Integration

2010-10-27T19:59:00.000+02:00

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

You can download the slides from:

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

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

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

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

Debian Based Kamailio Distribution

2010-10-25T19:58:00.000+02:00

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

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

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

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

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

Kamailio v3.0.4 Released

2010-10-20T22:10:00.000+02:00

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

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

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

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

Resources for Kamailio version 3.0.4

Source tarballs are available at:

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

Detailed changelog:

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

Download via GIT:

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

Binaries and packages will be uploaded at:

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

Modules’ documentation:

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

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

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

Kamailio v1.5.5 Released

2010-10-20T22:09:00.000+02:00

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

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

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

Source tarballs are available at:

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

Detailed changelog:

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

Download via SVN:

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

Tag for this release can be browsed at:

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

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

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

Binaries and packages will be uploaded at:

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

Modules’ documentation:

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

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

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

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

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

2010-10-14T09:58:00.002+02:00

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

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

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

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

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

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

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

The documentation for uac module is available at:

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

Apt Deb Repositories for v3.1.0

2010-10-12T12:23:00.001+02:00

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

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

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

Supported OSes are Debian Lenny, Squeeze and Ubuntu Lucid.

Check for more details the wiki page:

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

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

2010-10-11T22:43:00.005+02:00

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

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

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

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

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

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

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

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

You can browse module's documentation online at:

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

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

2010-10-07T23:02:00.001+02:00

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

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

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

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

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

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

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

Kamailio v3.1.0 Released

2010-10-06T12:37:00.001+02:00

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

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

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

Continue reading the release notes at:

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

Best of New in Kamailio 3.1.0 - ToC

2010-10-05T11:29:00.007+02:00

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

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

Here is the table of content:

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

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

Best of New in Kamailio 3.1.0 - #7: Pre-processor Directives

2010-10-04T22:02:00.006+02:00

Previous major version, 3.0.0, introduced for first time configuration file pre-processor directives that allowed to include files, define IDs and test their definition.

It was one of the most appreciated additions in 3.0.0, allowing to structure better the configuration file, to enable or disable features with one single line change.

Kamailio v3.1.0 added new features to these pre-processor directives:

defined IDs can take values, being either strings, integer or even statements
defined IDs with values are replaced across configuration file
new subst directive that does perl-like substitutions inside string values within configuration file

For example, the default configuration file for v3.1.0 defines DBURL token to be used for db_url parameters:

#!define DBURL "mysql://openser:openserrw@localhost/openser"
...
modparam("auth_db", "db_url", DBURL)

Changing access to database requires now to update a single place, the define of DBURL.

If you have pieces of config that repeat, you can define them and reuse:

#!define MYLOG xlog("[$Tf]: message $rm from $fu to $ru\n")
...
route {
 ...
 MYLOG;
 ...
 MYLOG;
 ...
}
...

The new subst allow replacement inside string values - example: replace DBPASSWD token with 123qaz in all strings inside configuration file:

#!subst "/DBPASSWD/123qaz/"
modparam("acc", "db_url", "mysql://user:DBPASSWD@localhost/db")

Note that defines and subst apply to all parts of configuration file, not matter are global parameters, module settings or routing blocks.

Another useful features related to config pre-processor directives is the ability to define IDs from command line using -A parameter:

kamailio -A MYVALUE=abc

You can see as well the core cookbook section for pre-processor directives:

http://www.kamailio.org/dokuwiki/doku.php/core-cookbook:3.1.x

Best of New in Kamailio 3.1.0 - #6: Asynchronous TLS

2010-10-03T12:14:00.001+02:00

Demand for secure communication increased lately. Caused or not by more and more attacks on VoIP systems lately, as well as better penetration of Instant Messaging and Presence in SIP networks, IP telephony providers look now seriously to offer full content protection as a service.

Kamailio 3.1.0 is the first version that can be truly used for large scale full encrypted SIP communication engine. OpenSER (up to Kamailio 1.5.x) had more like a beta implementation, it was added for prototyping purposes and never improved, maybe also a result of missing TLS clients at that time.

Previous version, Kamailio 3.0.x, had a completely re-architectured TLS implementation, inherited from SIP Express Router (SER) v2.1.0, which was designed for massive scalability. Still, the missing piece in 3.0.x was asynchronous support. This was added in 3.1.0.

Asynchronous TLS support in v3.1.0 stands out through its simplicity to configure. It is on by default, transparent to configuration file. You can disable it via TCP connection control parameter:

tcp_async=no

The tls module has a comprehensive documentation:

http://kamailio.org/docs/modules/stable/modules/tls.html

TLS being a layer on top of TCP, many tunings can be done via global TCP parameters:

http://www.kamailio.org/dokuwiki/doku.php/core-cookbook:3.1.x