Astricon, the Asterisk conference, celebrates big this year with its tenth edition, event to take place in Atlanta, GA, USA, during October 8-10, 2013. Using Kamailio and Asterisk is something very common, therefore our presence at the show is going to be very consistent.
Daniel-Constantin Mierla, co-founder of Kamailio project
Klaus Darilion, member of Kamailio management board
Olle E. Johansson, main contributor of SIP channel in Asterisk and co-founder of Astricon
Peter Dunkley, main author of WebSocket support for WebRTC in Kamailio
Other developers, friends and community members of Kamailio will be around, among them JR Richardson, Nir Simionovich, Eric Klein (all three taking care of Kamailio presence at last year edition), Alex Balashov, Fred Posner, James Body, Randy Resnick.
With a set of very interesting parallel tracks, we want to make sure you will be able to learn as much as possible about Kamailio and Asterisk, on topics such as scalability, security or rich communication services. We are eager to talk to you, thus you can contact us on mailing lists (user or devel communities) at any time to get in touch. You will find some of us in the exhibition area as well during the event.
Considering that next Asterisk release comes with a refactored SIP channel and based on quick check of the agenda, it is obvious that the event is full of novelty content you don’t want to miss. No matter you are new to Kamailio or new to Asterisk, this is the right event to attend for starting to use the two together.
Looking forward to meeting many of you in Atlanta!
The GIT master branch of Kamailio includes now a new module – rtpproxy-ng. It is designed to be next generation RTP relay control protcol, using bencode as the base for formatting control command. It can be used as a drop-in replacement for old rtpproxy module, but you have to use mediaproxy-ng as RTP relay. Both rtpproxy-ng module and mediaproxy-ng application were developed by Sipwise, main author in the Kamailio devel team being Richard Fuchs.
One of the very appealing features when using rtpproxy-ng and mediaproxy-ng is the ability to bridge WebRTC endpoints to classic SIP phones without any dedicated SBC or media gateway. Mediaproxy-ng is able to decode/encode SRTP to RTP back and forth.
Kamailio SIP Server v4.0.3stable is out – a minor release including fixes in code and documentation since v4.0.0 – configuration file and database compatibility is preserved.
Kamailio (former OpenSER) v4.0.3 is based on the latest version of GIT branch 4.0, therefore those running previous 4.0.x versions are advised to upgrade. There is no change that has to be done to configuration file or database structure comparing with older v4.0.x.
Kamailio SIP Server v3.3.5stable is out – a minor release including fixes in code and documentation since v3.3.4 – configuration file and database compatibility is preserved.
Kamailio (former OpenSER) v3.3.5 is based on the latest version of GIT branch 3.3, therefore those running previous 3.3.x versions are advised to upgrade. There is no change that has to be done to configuration file or database structure comparing with older v3.3.x.