Audio/video files can be downloaded for future use (streaming stored audio/video) or broadcast to clients over the Internet (streaming live audio/video). The Internet can also be used for live audio/video interaction.

Audio and video need to be digitized before being sent over the Internet.

Audio files are compressed through predictive encoding or perceptual encoding.

Joint Photographic Experts Group (JPEG) is a method to compress pictures and graphics.

The JPEG process involves blocking, the discrete cosine transform, quantization, and lossless compression.

Moving Pictures Experts Group (MPEG) is a method to compress video.

MPEG involves both spatial compression and temporal compression. The former is similar to JPEG, and the latter removes redundant frames.

We can use a Web server, or a Web server with a metafile, or a media server, or a media server and RTSP to download a streaming audio/video file.

Real-time data on a packet-switched network require the preservation of the time relationship between packets of a session.

Gaps between consecutive packets at the receiver cause a phenomenon called jitter.

Jitter can be controlled through the use of timestamps and a judicious choice of the playback time.

A playback buffer holds data until they can be played back.

A receiver delays playing back real-time data held in the playback buffer until a threshold level is reached.

Sequence numbers on real-time data packets provide a form of error control.

Real-time data are multicast to receivers.

Real-time traffic sometimes requires a translator to change a high-bandwidth signal
to a lower-quality narrow-bandwidth signal.

A mixer combines signals from different sources into one signal.

Real-time multimedia traffic requires both UDP and Real-time Transport Protocol

RTP handles time-stamping, sequencing, and mixing.

Real-time Transport Control Protocol (RTCP) provides flow control, quality of data
control, and feedback to the sources.

Voice over IP is a real-time interactive audio/video application.

The Session Initiation Protocol (SIP) is an application layer protocol that establishes,
manages, and terminates multimedia sessions.

H.323 is an ITU standard that allows a telephone connected to a public telephone
network to talk to a computer connected to the Internet.

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