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Evaluating Quality of Service in Applications: Challenges and Insights

This workshop presentation by Henning Schulzrinne and Wenyu Jiang from Columbia University discusses the application requirements for Quality of Service (QoS) in various domains, including video-on-demand, voice-over-IP, multimedia conferencing, and network games. It highlights the crucial factors for QoS evaluation, the issues with existing tools, and the significance of subjective measures. It also examines the implications of packet loss, latency, and reliability for application performance, emphasizing the need for consistent QoS to support premium services and user satisfaction.

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Evaluating Quality of Service in Applications: Challenges and Insights

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  1. Quality of Service - applications Henning Schulzrinne with Wenyu Jiang Dept. of Computer Science Columbia University NSF QoS workshop, April 2002

  2. Application requirements • What kind of applications? • What matters? • What doesn't? • How do we know?

  3. Applications – the diminishing set (or the cynics view of QoS) • video-on-demand  80 GB disk + P2P-over-TCP • voice-over-IP  where? • multimedia conferencing  still waiting, after 50 years... • network games  latency kills

  4. QoS evaluation • Problem: no tool for reliably evaluating quality end-to-end • network (PSTN & Internet) + hardware + operating system • depends on packet loss, jitter (playout buffer), FEC, loss correlation • need subjective evaluation, but too expensive • applicability of objective measures (PSQM, etc.) • use speech recognition as measure • e-model

  5. Voice quality vs. packet loss

  6. MOS vs. loss for FEC and LBR

  7. Myth: TCP loves lossy networks Lakshman/Madow/Suter ToN 2000

  8. Rrel as Universal MOS Predictor • Mapping from relative recognition ratio Rrel to MOS

  9. Human Recognition Results • Listeners are asked to transcribe what they hear in addition to MOS grading. • Human recognition result curves are less “smooth” than MOS curves.

  10. Research infrastructure • After 10 years, still no low-latency audio research application • existing applications (rat/vat) have erratic latency behavior • no good access to audio queuing information for lip-sync • no good diagnostic tools  blame network

  11. QoS is about reliability • can’t sell premium service that’s unavailable one day a year • auxiliary cost of failure: people scheduling, interruption, embarrassment, ... • consistent 5% packet loss is much better than 5% probability that network is unavailable for seconds • BGP convergence time ~ minutes • conjecture: applications (conferencing, VoIP) don't migrate for lack of predictability

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