The joy (and pain) of measuring streaming media |  Software

Everyone knows what it means when you say it takes 5 seconds to download a Web page. When you try to measure streaming audio and video quality on the Internet, a flood of factors arrives. Factors like late and lost packets, rebuffering during playback, bandwidth delivered and bandwidth received add to the complex mix of measurements to evaluate the overall streaming audio and video quality.

Furthermore, streaming content itself is complicated:

  • More data is transferred in real time than with static Web pages
  • The data is much more sensitive to latency.
  • Multiple data types are involved (audio & video).
  • Different default delivery protocols are involved (User Datagram Protocol (UDP) vs. TCP/IP)
  • Multiple technologies are involved (Real, Windows Media, QuickTime).

    By definition, streaming content isn't subjected to the delays associated with downloading very large rich media files to a storage device. More time-sensitive than standard Web pages, streaming media is subject to significant performance problems caused by today's Internet infrastructure, causing jittery video and audio static. Still, online users expect the same smooth, uninterrupted audio and video they get from their television and radio.

    Because you can't improve what you don't measure, streaming content providers need to monitor and measure their streaming performance. What follows is a guide about some of the factors involved in measuring streaming media.

    Three major factors that affect streaming quality, which need to be measured, are startup time, audio quality and video quality. Within these factors are specific elements, such as connect time, redirect time, initial buffer time, video frame rate, recovered, lost and dropped packets, and bandwidth utilization.

    1.Startup time:

    The time it takes from when you press the play button until the clip begins. Startup Time equals the total time required for Initial Connection (including DNS & time to first byte), Redirection Time, and Initial Buffering.

    a.Initial connection: Time it takes to establish a Real Time Streaming Protocol (RTSP) connection between the streaming server and the streaming client (or player).

    b.Redirection: The time it takes to transfer data to the last server from a second and subsequent servers, to which the first server may redirect data.

    c.Initial buffering: The time it takes to start viewing and hearing a streaming media clip from the time the data arrives in the buffer of the client computer.

    2.Audio quality: Derived from audio encoding and audio delivery

    a.Audio encoding includes the number of audio channels, bitrate per channel and quaility of original content.

    b.Audio delivery includes the delivered bandwidth and packet delivery.

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