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Tutorial
Understanding Asterisk
Brian Capouch, Assistant Professor, Computer Science, Saint Joseph's College
Track: Emerging Topics
Date: Tuesday, August 2nd, 2005
Time: 1:30pm - 5:00pm
Location: E142
As the deployment of broadband internet access continues, more and more people will be investigating the use of Voice over IP (VoIP) as a replacement for legacy public switched telephone network (PSTN) services.
This presentation's introductory material covers classic PSTN telephony, speech processing, packetizing speech, and modern VoIP signaling and bearer protocols. The Internet presents many challenges to voice carriage, derived from its packet-switched, unreliable nature. Accomodating packet loss, latency, and jitter are the keys to the transition from the circuit-switched paradigm.
The design, compilation, configuration, and operation of an Asterisk telephony server will be presented in depth. A variety of typical VoIP devices will be used to simulate a SOHO environment. These devices use the Asterisk server to connect to other VoIP systems, as well as PSTN subscribers, using Internet Telephony Service Providers (ITSPs). We will build a working system "from scratch" that illustrates the salient principles.
The last part of the tutorial will review a few "converged" applications that leverage the VoIP platform. A network monitoring tool and a web-based wakeup calling service will be demonstrated as integrated applications with the server built in the session.
Download presentation file
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