Setting up Streaming an Asterisk Channel
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Streaming an Asterisk Channel HOW-TO
Introduction
The Asterisk is well known as VoIP service. Obviously, the nature of VoIP technology requires a real time communication, which makes certain requirements to a network connectivity layer in such regards as link speed/bandwidth, IP packet prioritization/control and network latency.
In some cases it is desireable to provide at least one way communication possibility. For example, this idea could be used to give an ability at least "to listen" to a corporate work meeting/conferece, which is organized by means Asterisk MeetMe conference feature to those Remote Locations that still have relatively poor network connectivity layer with a Head Office. Another application is streaming an Asterisk channel to the Internet.
How does it work?
The Asterisk is used with conjunction with sound streaming software such as Icecast2 server. The streaming technology provides "caching/buffering" capabilities. So broadcasted/recieved sound is being buffered, which helps overall sound quality to remain high surviving most of problems in network bandwidth and latencies.
The Asterisk system places an automatic call to connect two extensions. One extension provides an Asterisk channel sound (MusicOnHold, Playback, MeetMe adn etc.), the second one starts the Asterisk Ices application and merges sound to it. The Ices Application depends on Icecast Ices, which is a source of sound to Icecast2 service.
After sound is merged to Icecast Service by means Icecast Ices and Asterisk Ices Applications it could be listened as ogg sound stream by WinAmp, VLC media players. The stream cache parameter should be ajusted according local needs to achieve the acceptable sound quality.