Asterisk, the open source communications solution used in a wide number of internet protocol (IP) telephony and business VoIP (voice over internet protocol) applications across the globe, has been the beneficiary of some key enhancements according to Digium, the software company behind the platform.
In an announcement made at AstriCon, the recent Asterisk users’ conference in the US, Digium said that the brand new version of Asterisk (Asterisk 11) does not deviate too far from the tried and trusted version already deployed by a great many business VoIP providers and resellers around the world but, at the same time, contains some notably worthwhile features.
Among these features is a facility enabling users to connect to Asterisk and make phone calls directly via web browsers. This functions via the new version’s support for Web Real-Time Communication (Web RTC) links.
Digium says that Asterisk 11 will secure these Web RTC-enabled communications for users via the additional features it has included of Datagram Transport Layer Security (DTLS) and Stream Control Transmission Protocol (SRTP).
A further enhancement of the new version designed to help effect successful Web RTC communications through Asterisk is the incorporation of STUN, ICE and TURN protocols.
Digium says that Asterisk 11 also includes a new single channel support drive for both Jingle and Google Talk – called Motif.
Available via Digium’s main Asterisk website, the new version is backed by four years’ worth of support.
Commenting on the release of Asterisk 11, Internetnews.com’s senior editor, Sean Michael Kerner, said:
“Web RTC is something that holds amazing promise, and I strongly suspect that Asterisk 11 will be one of the defining infrastructure technologies that enables that promise to become a reality.”
In the meantime, development is also reported to have started on the next version of Asterisk – Asterisk 12.
