Digium's 2012 AstriCon users' conference and expo took place last week and, like every other AstriCon before it, the conference helped to expand awareness and knowledge about Digium's flagship product, Asterisk. It's appropriate, then, that Digium unveiled a new version of the software, Asterisk 11, that features a number of contributions from the Asterisk developer community.
New features in version 11 include support for WebRTC over SIP, as well as native integration with Digium's line of VoIP telephones. This version is also Long Term Support (LTS) release. As such, Asterisk 11 is largely focused on stability, performance and security rather than a large batch of new features.
"We offer LTS releases to give our users a stable and well-supported version on which to build applications and solutions," Asterisk marketing director Steve Sokol told Webrtcworld. "We do standard releases to get new features and functions into the codebase. Each LTS release generally includes only a few new features, focusing instead on lessons learned from the previous standard release, which helps to improve the stability, performance and security. Some users prioritize new features and chose to run standard version in production. Others are more interested in stability and support, and chose to base their deployments on an LTS release. It gives us the best of both worlds."
Best of all, being an LTS release, Asterisk 11 features four years of support, with an additional year of security and maintenance, unlike a standard Asterisk release which receives only a single year of bug-fix support with an additional year of security-fix support. This means that Asterisk 11 will be supported through until 2016.
"We generally alternate between standard and LTS releases from year to year," added Sokol.
Delving deeper into the release's new features, its new WebSockets SIP transport allows browser-based SIP clients to connect with Asterisk and establish media sessions, while DTLS-SRTP support ensures secure transport of RTP media streams used by WebRTC and SIP endpoints. Furthermore, ICE, STUN and TURN have been incorporated into the Asterisk RTP engine in order to provide better WebRTC support. Lastly, Asterisk 11 sports a new channel driver, Motif, which supports the Jingle protocol and Google Talk, while combining functions spread across multiple channels in older versions.
Asterisk 11 is available right now from the Asterisk website, while the Asterisk development community is hard at work on Asterisk 12.
Earlier in October, Digium announced changes meant to make it easier for Asterisk integrators, value-added resellers (VARs) and developers to join the Digium channel partner program. Specifically, the company expanded its product portfolio to include IP phones, gateways, redundancy appliances and telephony cards, while also streamlining the partner certification program.
Edited by
Brooke Neuman
