OpenSER/Kamailio


Last weekend I attended a session about Homer Sipcapture at Fosdem in Brussels. Homer is a software that is built in to Kamailio. In Kamailio has there are two modules that is specially adopted for use with this 3rd party Open Source software – Homer SIPcapture.  Homer was awarded a Kamailio Award last year for it’s unique innovation. With Kamailio and Homer, you can capture SIP messages from a running Kamailio production server or from a mirrored port in a switch in your network. What do you get? A searchable database of your SIP traffic and easy to understand visual diagrams of individual SIP sessions. All in a web interface powered by a database of your traffic. I installed it and after a few hours it was a critical tool in my network, helping me to solve problems.

Two components – sender and receiver

Homer has two major components. The packet sender captures traffic and forwards to a receiver. You can use Kamailio in both places, but you can also enable a sender within FreeSwitch as well as use the Homer capture sender that reads off a network interface and sends to the receiver. To use the Homer software with Kamailio, you need a database server (with capacity to handle your traffic) and a web server with PHP support. (more…)

The very first SIP-compatible software I installed myself was SER, Sip Express Router, from iptel.org, a spinoff from the Fokus research institute in Berlin. During the years I have worked with SER and OpenSER/Kamailio, I’ve become friends with many of the developers and have contributed with documentation and training. The team gathers world leading SIP experts and developers, so it’s a joy to meet them and brainstorm and share experiences. In almost all my installations and platforms, a SIP proxy is at the core of the real time multimedia platform. My choice has always been SER/OpenSER/Kamailio.

Starting as a SIP proxy, moving on as an extensible SIP server

Sip Express Router started as an Open Source SIP Proxy. Today we see Kamailio, based on the common sip-router.org core, as an extensible SIP server. With an embedded presence server, http support and interfaces to python and lua for scripting, it’s a platform that can handle many different roles in a large-scale SIP architecture. IPv4 and IPv6 support from the beginning, security integrated and a very active development team. Everything you need to build a standard-compliant cool network for your organization’s real-time communication. 10 years of success.

Jubilee September 2nd, 2011 where it all started

The 3rd of September marks 10 years since the start. Reason to celebrate, don’t you think?

 

We will celebrate on Friday, September 2nd and into the night. FhG Fokus Research Institute hosts a one-day conference session with all the core people. If you have any interest in SIP and these platforms, make sure you will be there.

The event is organized with sponsorships from FhG Fokus Institute, Asipto, Amooma and TeamForrest. At the same time local events will be organized in Barcelona, Spain and Vienna, Austria.  Read the agenda and register today!

Asterisk was originally built as a stand-alone system, a single central point for all telephony communication. In short, a PBX. Nowadays, the Asterisk Open Source telephony server, is used in many ways – many of them not really being PBXs. All kinds of applications are being powered by Asterisk.While building applications with Asterisk, you soon realize that you’re limited to that single server. It’s hard to scale and one limiting factor is that the call state is being held in one server. Many services depend on call states – if an agent in a call center is busy, you need to find an available agent. If a trunk to the PSTN is in use, you might want to find another way out. Call states are important.Of course, the Asterisk project is now working on the long term solution, the Asterisk SCF and the applications that will be built using this framework. But that will take some time. Meanwhile, the Asterisk PBX team has been working on a few ways to distribute the call states between a group of servers. This article will describe a few of the different architectures being worked on. (more…)

I am working with two very different implementation projects this winter. One is about using Asterisk in large call center environments. Another has a goal of building a SIP network for 15.000 phones in a university.

Asterisk for large call centers – full control

The call center platform is focused on a lot of PBX functionality. Every agent stays connected to a conference session, where customers are connected and disconnected, recording is enabled and disabled, dtmf is used to control various services and both AGI and AMI (the manager interface) are both being used heavily to integrate with a controlling application. Asterisk is in full control of each and every call, all the time, but the application controls the flow of the calls. The dialplan is very small for a large-scale Asterisk installation.

Large scale SIP networks – Asterisk on the edge, providing services

For the university, scaling is important. It’s a plain SIP network, with two computer centers in different buildings. All accounts are managed by LDAP, there’s no account defined locally in the VoIP platform. SIP proxys (Kamailio) rule this network and DNS is used for failover. Many services like call transfer, three-party conference calls and call forwarding will be handled by the phones. Asterisk serves as gateways to the old world, Nortel systems, and feature servers for IVR, voicemail and switchboard. No single server is in control of any call.

The power of open standards and open source

Two completely different designs, both made possible by Open Source and Open Standards. Both of them needs to scale. Both of them needs failover, redundancy and stability. And in both cases, we’re replacing expensive legacy telecom equipment with new platforms that will cost less to operate, that has a higher degree of interoperability and much more functionality than the previous solutions. Open telephony wins.

Kamailio - the successor to OpenSER – has been released in a new version, 3.0. This version is based on the merge between SER, SIP Express Router, and OpenSER code bases and developer teams. It’s a SIP server that you can run as a SIP proxy, a session border controller, a SIP load balancer or SIP application server. In version 3.0 you have the power of all the modules and applications in OpenSER and the raw strength of the SER core in one product. Go check it out today!

My friend and co-teacher Daniel-Constatin Mierla has written a series of articles on the soon-to-be released Kamailio 3.0 that highlights new features. The main thing is that this new version of the old OpenSER SIP server is merged with the original SER, as part of the SIP-router project. This gives Kamailio a large set of new features, most notably the rewritten kernel that has more stability, scalability and performance than the old OpenSER. Please visit Miconda’s blogg and learn about all the new features!

Best of New in Kamailio 3.0.0 - #1: include fileI’m starting a series of posts, to highlight the best new features in Kamailio (OpenSER) 3.0.0, of course, from my point of view, hoping to cover most of them before full 3.0.0 is out (RC3 was done yesterday).

During 2008, I worked with my Portuguese business partner Wavecom to build a large installation of Asterisk and OpenSER in Portugal. The network is now running 300 servers, 250 of them running Asterisk, FreePBX and OpenSER. It’s handling 200 connections to legacy PBXs of all kinds and over 10.000 phone numbers. During 2009 we’ve developed a failover solution, to make sure that every FreePBX server always stays up, regardless of the state of the hardware.

We’ve migrated 34 Universities to use Open Realtime Communication with Open Source. They are now migrating their user base from old PBX phones to modern VoIP phones.

Want to learn more?
Come to Amoocom in Germany, May 4-5, 2009 and listen to my talk!   And if you need help with similar large-size projects, contact me on info@edvina.net!