Digium today helped open the eyes of attendees at the WebRTC Conference & Expo with breakfast and a presentation. During Digium’s Thursday morning demo in Atlanta, company executives talked about Asterisk, how you can use the open source solution as an app unto itself or as a toolkit or engine, and what the company is doing related to WebRTC.
Asterisk is most commonly used to power traditional PBX and VoIP-only PBX solutions, or hybrid PBX solutions that combine VoIP and traditional telephony. It can also be leveraged as a feature server, by connecting Asterisk to an existing phone switch and simply using Asterisk for its features. Call center/contact center implementations sometimes also use it for automated call distribution; remote agent solution; call monitoring and recording; service on hold; skills-based routing; geographic distribution and routing; and failover and contingency solution; and more.
The company introduced WebRTC with the delivery of Asterisk 11, which it made available last year. New in that version is support for DTLS-SRTP, which is a secure transport for RTP media streams used by WebRTC and SIP endpoints; support for ICE, a framework that tries to find the best path for each WebRTC session; support for STUN and TURN, standards that provide your public IP address and sends data flows, and a cloud fallback for P2P; Motif, a channel driver for supporting the Jingle protocol and Google Talk; and WebSockets SIP transport, which allows browser-based SIP clients to connect with Asterisk and establish media sessions.
At the breakfast gathering, Digium offered what they called a recipe for a free WebRTC demo. It went like this: get a Linux box, add ‘libsrtp’, add Asterisk 11.4.0, follow the instructions to enable WebRTC support (for more on this part, visit wiki.asterisk.org), add a basic dialplan. The company then demonstrated WebRTC running over a Raspberry Pi.
Digium said that Asterisk 12, which it described as “a new animal,” will become available next year. With this release, the company has completely rewritten its SIP implementation and reworked the way APIs are done within Asterisk (Stasis is used for the new interface). This shows that we can expect to see a lot of new WebRTC stuff coming out following that.
Edited by
Jamie Epstein
