Open Source Asterisk Telephony Goes High-Def
Digium expands Asterisk's capabilities to handle HD audio and video, as well as improved audio and video conferencing.
Asterisk, the leading open source platform for telephony, is adding high definition (HD) media capabilities, including video, as well as better voice and videoconferencing, to make sure it will be more than a footnote in the next wave of communications applications.
Asterisk 10, currently in late beta testing, introduces support for a "virtually unlimited number of Codecs" (the algorithms for encoding and decoding media), according to Digium, the main corporate sponsor of the open source project. Digium announced Asterisk 10 at the end of October. Among the Codecs included with the platform are the wideband version of Speex, Skype's super-wideband SILK, and pass-through support for several CELT variants, according to Digium.
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Asterisk previously supported 8- and 16-kHz audio, but the new platform provides for higher quality sampling rates of up to 192 kHz. In addition, the MeetMe conferencing software included in previous releases has been replaced with an HD-capable bridge called ConfBridge that also works with videoconferencing endpoints that support session initiation protocol (SIP). Asterisk 10 also improves fax handling and adds text message routing, with support for SIP messaging and XMPP chat, allowing it to act as a text message server and bridge between protocols.
"It handles virtually any kind of media you'd want to pass through it," said Steve Sokol, Digium's director of Asterisk. In conferencing applications, Asterisk 10 will be able to deliver HD audio to participants on the corporate network, even if some participants are dialed in from phone lines that can't support that level of quality, he said. The idea is to send each participant the best quality the equipment is capable of receiving, he said.
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The developer community is also in the process of defining Asterisk SCF, a scalable communications framework aimed at larger enterprise and carrier applications, which will be architected as a set of distributed components that can either be clustered on a single server or spread across many servers, making it suitable for cloud computing. Sokol said Asterisk SCF 1.0 should be out in the first quarter of 2012.
Asterisk has long been recognized as the leading open source PBX and VoIP technology--a Linux-based alternative to proprietary networking equipment. Digium sells a small business server product based on the software and also supports other companies who create products that include Asterisk technology.
"Asterisk provides a rock solid technology stack--it's very similar to what Apache and MySQL are to the Web development world," said Jeronimo Romero, chief technology officer at EUS Networks, which has used Asterisk to create applications for its customers such as stock-floor trading systems with integrated telephony. Using Asterisk "allows us to put the requirements to the test in a way that is really transparent," making sure compliance applications are addressed, he said. With proprietary architectures, only the vendor can fix bugs or change the behavior of embedded software in networking equipment, he said.
In general, it's often easier and cheaper to use Asterisk as the basis of an enterprise application, Romero said. He acknowledged other platforms tend to have the upper hand when it comes to features like tight integration with Microsoft Office, but added that extensions for Microsoft integration are available through the Asterisk developer community.
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