Kamailio World 2015 – The Workshops

It is now about one month and a half till the start of Kamailio World Conference 2015. Continuing with the same event structure like in 2014, the afternoon of the first day, the 27th of May, is filled with several technical workshops. These sessions are intended to give a more hands-on perspective on the subjects, with deeper technical content.

Last year, Sipwise showed how to deploy sip:provide CE – the open source out of the box IP Telephony Operator Platform – in a matter of minutes and customize it to fit better your needs. This year, Daniel Grotti, a long time SIP and Kamailio fellow, is going to show how to enable WebRTC for sip:provider CE in order to bridge the communication between the web world and the classic SIP phones. Few other typical use cases will be approached during the session.

Carsten Bock, from NG Voice, is returning with another tutorial to show more of what can be done with Kamailio for IMS and VoLTE deployments. Besides the tutorial, the plan is to have a VoLTE testbed on site for the duration of the entire event, so the participants can test with their own devices.

After presenting at the past editions the concept and the development of CGRateS, a carrier grade open source CDR rating engine, Dan Bogos is now coming with a hands-on session about how to integrate it with Kamailio for prepaid and postpaid billing.

Ability to troubleshoot SIP routing and analyze the flows on the wire is one of the core elements required for VoIP engineers. Lorenzo Mangani, one of the co-founders of Homer SIP Capture project, is going to deliver a session on how to use existing open source tools (including Homer and sipgrep, but not limited to) to make the SIP troubleshooting process easier.

All together are providing an amazing amount of knowledge from the people with first hand experience, those that built the systems. It is a unique opportunity at Kamailio World to get face to face to interact with such people.

The content of conference days is filled with other very interesting sessions, including as well valuable technical details, presenting scalable and secure architectures or other products that can be used to complete the VoIP platforms with new features. Right now you can see details for a sections of presentations in the Schedule page.

Be sure you don’t miss Kamailio World Conference 2015, during May 27-29, in Berlin, Germany – it is the open source real time communications event in Europe!

Secure your participation and register now!

See you in Berlin!

Joint venture on Kamailio to tackle big vendors in telco market

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.

Kamailio Jobs at Sipwise, Vienna, Austria

Sipwise is currently hiring a VoIP System Administrator for an interesting position based in Vienna, Austria.

You have strong skills in Linux system administration (Monitoring and Alerting using SNMP/Nagios/Cacti/MRTG, Scripting in Perl/M4/sh), a deep understanding of highly available system deployments and good knowledge regarding SIP (preferably Kamailio, Sems, Asterisk)?

Sipwise offer you a challenging position to help our team further improving our Kamailio-based Class5 Softswitches, deploying them at customer sites and supporting our customers (large DSL and Cable Providers throughout Europe) and sales teams with technical details.

If you are interested, please send email to Andreas Granig, agranig [at] sipwise.com

OpenSER Admin Course and VoN Rome 2007

I hope it is general opinion that the OpenSER admin course at VoN Rome was successful. The room was full, about 35 attendees. Time was pretty short to get deeper in OpenSER configuration tricks, but should give the base of knowledge to build proper and well-designed VoIP platforms using OpenSER.

It started with introduction of configuration file architecture, going through common components of it, touching the very important aspects of a VoIP environment: NAT traversal, accounting, authentication and authorization, load balancing and high availability. The last topic of the tutorial was integration with Asterisk media server, towards a lot of good steps were done in the past for a straightforward solution.

The day ended with an open discussion, revealing hot questions about DNS balancing and black-listing, NAT traversal, present and future of the project.

VoN Rome itself was pretty well represented by the local market, with nice SIP-enabled video solutions for security and surveillance at Video on the Net section. Otherwise, usual debates SIP-P2PSIP-IMS, etc. Personally I have met for the first time a lot of people using OpenSER, also meeting old friends (e.g., Andreas of sipwise.com, Georgios, Federico, Adrian, ...) from the past events. Photo available at: flickr.com.