SKYPE/SILK網路電話的傳輸速率變化對使用者感受之影響的模型建構

如何有效地在端點與端點之間傳輸對延遲較為敏感的聲音資料一直是多媒體網路中的一個課題。其中一個主要的議題即為決定即時網路電話的傳輸速率，使得每單位網路資源的使用對使用者感受能夠最大化。由於網路的頻寬通常是動態變化的，而不斷地提高傳輸速率是否保證比較好的使用者經驗則依然不明瞭。除此之外，單純地在每一個時間點盡量地最佳化傳輸速率，亦可能對使用者的使用經驗造成不好的影響。為了瞭解傳輸速率變化的幅度以及頻率對使用者感受造成的效應，我們招募了127名的受測者針對模擬的Skype通話進行評分，並且有系統的改變這些通話的速率變化幅度以及頻率。因此我們能夠達成 (1) 發現速率變化頻率對使用者感受的影響為一對數關係，並且遵守Weber-Fechner’s Law (2) 速率變化幅度的影響則取決於使用者如何感受不同品質之間的差別 (3) 推導出一封閉模型來描述使用者感受與速率變化之間的關係，其平均錯誤率以及R-square分別為7.5%以及0.86。

關鍵字

網路電話；速率調變；使用者感受；體驗質量；心理物理學

並列摘要

The effective end-to-end transport of delay-sensitive voice data has long been a problem in multimedia networking. One of the major issues is determining the sending rate of real-time VoIP streams such that the user experience is maximized per unit network resource consumed. A particularly interesting complication that remains to be addressed is that the available bandwidth is often dynamic. Thus, it is unclear whether a marginal increase warrants better user experience. If a user naively tunes the sending rate to the optimum at any given opportunity, the user experience could fluctuate. To investigate the effects of magnitude and frequency of rate changes on user experience, we recruited 127 human participants to systematically score emulated Skype calls with different combinations of rate changes, including varying magnitude and frequency of rate changes. Results show that 1) the rate change frequency affects the user experience on a logarithmic scale, echoing Weber-Fechner’s Law [1], 2) the effect of rate change magnitude depends on how users perceive the quality difference, and 3) this study derives a closed-form model of user perception for rate changes for Skype calls.

並列關鍵字

VoIP ； Rate Adaptation ； User Perception ； QoE ； Psychophysics

參考文獻

[2] J. Matta, C. Pepin, K. Lashkari, and R. Jain, “A Source and Channel Rate Adaptation Algorithm for AMR in VoIP Using the E-model,” in Proceedings of ACM Network and Operating System Support for Digital Audio and Video, NOSSDAV, 2003.

[3] Z. Qiao, L. Sun, N. Heilemann, and E. Ifeachor, “A New Method for VoIP Quality of Service Control Use Combined Adaptive Sender Rate and Priority Marking,” in Proceedings of IEEE International Communications Conference, ICC, 2004.

[4] T.-Y. Huang, P. Huang, K.-T. Chen, and P.-J. Wang, “Can Skype be More Satisfying? A QoE-Centric Study of the FEC Mechanism in the Internet-Scale VoIP System,” IEEE Network, Vol. 24(2), pp.42–48, 2010.

[7] R. A. Fisher, “Statistical Methods for Research Workers”, Oliver and Boyd, ISBN 0-05-002170-2, 1925.

[9] RFC 793, “Transmission Control Protocol,” 1981.

國際替代計量

SKYPE/SILK網路電話的傳輸速率變化對使用者感受之影響的模型建構

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