HUNTSVILLE, Ala., Oct. 31, 2012 /PRNewswire/ -- Digium, Inc., the Asterisk Company, unveiled Asterisk 11 at its annual AstriCon users' conference meeting, a new release that features multiple contributions from the Asterisk developer community. Asterisk 11 includes a number of new features, including support for WebRTC over SIP and native integration with Digium's line of VoIP telephones. It is also a new Long Term Support (LTS) version of Asterisk, the world's most widely adopted open source communications engine.
As a Long Term Support release, Asterisk 11 is primarily focused on stability, performance and security, with a relatively short list of new features. LTS releases receive four years of support, with an additional year of security maintenance. Under this release plan, Asterisk 11 will be supported through 2016.
Significant new features include:
- WebSockets SIP Transport - WebRTC/RTCWEB brings real-time communications to web browsers. The new WebSockets transport for the Asterisk SIP channel allows browser-based SIP clients to connect with Asterisk and establish media sessions.
- DTLS-SRTP Support – A secure transport for RTP media streams used by WebRTC and SIP endpoints.
- ICE, STUN and TURN Support – A set of related technologies for establishing live media streams between software agents running behind network address translators (NATs) and firewalls. ICE, STUN and TURN have been incorporated into the Asterisk RTP engine as part of the effort to support WebRTC.
- Motif - A new channel driver for supporting the Jingle protocol and Google Talk. Motif combines functions previously spread across multiple channels, and makes use of a new and more standards-compliant XMPP implementation.
Asterisk 11 is currently available from the Asterisk.org web site. The Asterisk development community has already begun working on Asterisk 12. For more information and documentation, visit https://wiki.asterisk.org/wiki/display/AST/Asterisk+11+Documentation.
To download the new release, visit: www.asterisk.org/downloads.
Digium®, Inc., provides Asterisk custom communications and Switchvox Unified Communications (UC) business phone systems that deliver enterprise-class features at a price businesses can afford. We are the creator, primary developer and sponsor of the Asterisk project, the world's most widely used open source communications software that turns an ordinary computer into a feature-rich voice communications server. With a community of more than 80,000 members worldwide, Asterisk has been used to create VoIP communication solutions in more than 170 countries. Since 1999, Digium has become the open source alternative to proprietary communication providers, giving people an innovative solution for business communications that increase productivity. Digium's wide range of business communications products is sold through a worldwide network of reseller partners. More information is available at: www.digium.com or www.asterisk.org.
The Digium logo, Digium, Asterisk, Asterisk SCF, Switchvox, AsteriskNOW, Asterisk Appliance and the Asterisk logo are trademarks of Digium, Inc. All other trademarks are property of their respective owners.
SOURCE Digium, Inc.