At AstriCon in Denver this week, Digium (www.digium.com) announced it had added a “wide-band media engine” into the latest release of Asterisk. The new support is hailed as the biggest upgrade to the software, including support for “an unlimited number” of codecs and support for SILK, 32kHz Speex, and pass-through support for CELT. Plus there’s a new “HD capable” conferencing bridge application and text messaging routing support.
Digium has replace what it describes as Asterisk’s “telephony-grade” media engine with a more advanced one. Support is now available for “studio-quality” audio and a “nearly unlimited” number of codecs.
New codecs added/supported include Microsoft’s (Skype’s) SILK codec in full-blown superwideband mode, the wideband version of Speex, and pass-through support for “several” CELT variants.
Sampling rates have been improved. Previously, 8 kHz and 16 kHz sampled audio was supported. Asterisk now supports 8, 12, 16, 24, 32, 44.1, 48, 96, and 192 kHz rates.
Digium has dumped, er “replaced” the MeetMe conference bridge with an HD-capable “intelligent” bridge application called ConfBridge. The app supports all codecs and conference rates and works on any Asterisk 10 system. “Intelligent” mixing algorithms provide each participation with the optimal audio quality for their connections. It is also fully customizable so sys admins and integrators can configure call-in menus on a caller-by-caller basis.
Other new features include videoconferencing support in ConfBridge, text message routing for SIP MESSAGE and XMPP – so you can get both a messaging server and a bridge between different messaging protocols, and “significant new fax” capabilities, including T.38 gateway capabilities.
From an HD voice perspective, about the only thing missing out of this release would be official support for AMR-WB and some of the higher-end Polycom-based G.7xx codecs.
Go visit www.asterisk.org for more details.