Unlike radio frequency broadcasts and other
networks such as phone systems, the Internet was not designed to handle
time-dependent data, such as audio material. Because the Internet
spans a large geographical
area and provides dependable stability, vast engineering resources
have been spent to expand the possible capabilities.
Applications such as
NetPhone, an Internet
telephony
application,
CU-SeeMe, a desktop
videoconferencing application,
and
RealAudio, a
broadcast-on-demand audio system, transfer audio/video information
in, or near, real time. The bandwidth available is usually not adequate for high-quality sampling and bit rates; compromise and compression are standard. The systems do rival
their traditional counterparts in functionality and usefulness.
The advantage of Internet-based telephony and videoconferencing
is economic: for the same relatively
low price one pays for Internet access, long-distance phone rates are bypassed.
As some institutions already implement integrated voice and data services,
leveraging those telecommunications lines for additional uses may be a prudent
use of bandwidth.
Audio systems such as RealAudio use a buffering system to give the
illusion of real-time transfer. The software will delay reproducing
a transmission for up to two seconds in order to build a buffer of data,
ensuring smooth reproduction in case of a delay in the link. Video
systems will go even further by dropping video frames in order to
relay undisturbed the often more vital audio element.
The RealAudio server now has the ability to broadcast live
audio feeds. The function requires much more server capacity, and
the buffering time may increase from 2 to about 10
seconds.
Next Section:
Transmission of Digital Audio Files:
File Compression