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Friday, January 28, 2011

Social Networking Event, Brussels, Belgium

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:

Monday, January 17, 2011

IMS Extensions Available for Testing in Kamailio

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:

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:

Thursday, January 6, 2011

World of SIP in 2011

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?

Wednesday, January 5, 2011

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

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.

Monday, January 3, 2011

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

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:

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!

Saturday, January 1, 2011

It has to be a great year

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!