Tuning Skype's Redundancy Control Algorithm for User Satisfaction

Huang, Te-Yuan; Chen, K.-T.; Huang, Polly

doi:10.1109/infcom.2009.5062031

Cited by 28 publications

(22 citation statements)

References 9 publications

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“…Huang et al have discovered that Skype did not provide appropriate protections for audio packets and have proposed new schemes to improve MOS [16,17]. They then implemented a new system with audio codecs that Skype has announced to use.…”

Section: Related Workmentioning

confidence: 99%

Exploiting just-noticeable difference of delays for improving quality of experience in video conferencing

Wah

2013

Proceedings of the 4th ACM Multimedia Systems Conference

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This paper proposes a novel approach for improving the quality of experience (QoE) of real-time video conferencing systems. In these systems, QoE is affected by signal quality as well as interactivity, both depending on the packet loss rate, delay jitters, and mouth-toear delay (MED) that measures the sender-receiver delay on audio signals (and will be the same as that of video signals when video and audio is synchronized). We notice in the current Internet that increasing MED as well as reducing packet rate can help reduce the delay-aware loss rate in congested connections. Between the two methods, the former plays a more important role and applies well to a variety of network conditions for improving audiovisual signal quality, although overly increasing the MED will degrade interactivity. Based on a psychophysical concept called just-noticeable difference (JND), we find the extent to which MED can be increased, without humans perceiving the difference from the original conversation. The approach can be applied to improve existing video conferencing systems. Starting from the operating point of an existing system, we increase its MED to within JND in order to have more room for smoothing network delay spikes as well as recovering lost packets, without incurring noticeable degradation in interactivity. We demonstrate the idea on Skype and Windows Live Messenger by designing a traffic interceptor to extend their buffering time and to perform packet scheduling/recovery. Our experimental results show significant improvements in QoE, with much better signal quality while maintaining similar interactivity.

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Section: Related Workmentioning

confidence: 99%

Exploiting just-noticeable difference of delays for improving quality of experience in video conferencing

Wah

2013

Proceedings of the 4th ACM Multimedia Systems Conference

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“…Instead, by reducing the buffer size in order to increase the conversation interactivity, the overall QoE can be enhanced. However, this behavior can be changed if a redundancy control algorithm is introduced [6], so that a VoIP frame can be transmitted several times to cope with high packet loss rates. We will investigate the impact of redundancy control on the optimal playout buffer in a future work.…”

Section: Optimal Buffer Size Derivationmentioning

confidence: 99%

An empirical evaluation of VoIP playout buffer dimensioning in Skype, Google talk, and MSN Messenger

Chen

Huang

et al. 2009

Proceedings of the 18th International Workshop on Network and Operating Systems Support for Digital Audio and Video

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VoIP playout buffer dimensioning has long been a challenging optimization problem, as the buffer size must maintain a balance between conversational interactivity and speech quality. The conversational quality may be affected by a number of factors, some of which may change over time. Although a great deal of research effort has been expended in trying to solve the problem, how the research results are applied in practice is unclear.In this paper, we investigate the playout buffer dimensioning algorithms applied in three popular VoIP applications, namely, Skype, Google Talk, and MSN Messenger. We conduct experiments to assess how the applications adjust their playout buffer sizes. Using an objective QoE (Quality of Experience) metric, we show that Google Talk and MSN Messenger do not adjust their respective buffer sizes appropriately, while Skype does not adjust its buffer at all. In other words, they could provide better QoE to users by improving their buffer dimensioning algorithms. Moreover, none of the applications adapts its buffer size to the network loss rate, which should also be considered to ensure optimal QoE provisioning.

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“…Video calls are more information transmission asking for appear differently in relation to voice-over-IP (VoIP) calls. Data rate for Skype VoIP calls is around 30kbps [2] [3], an extraordinary quality Skype video call can without a lot of a stretch eat up exchange speed up to 950kbps [3]. Video conferencing customer nature of experience (QoE) corrupts by and large if the limited end-to-end video delay goes more than 350ms [4].…”

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confidence: 99%

Untitled

Purohit¹

2017

International Journal of Recent Advancement in Engineering &Amp; Research

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In first paper, Voice over IP (VoIP) is dynamically supplanting modified line telephone structures comprehensive in view of lower cost, call quality change. Overcome the issue we propose a sagacious course of action called Caching Agent an overlay outline involving holding servers set over the Internet Second paper, they propose MOVi (Mobile Opportunistic Video-on-interest), a flexible circulated video-on-interest application in light of pervasive WiFi enabled devices, for instance, phones and Ultra Mobile PCs. The studied model checks the channel pack drop rate, peer movement, and FEC affirmation to get the heterogeneous bundle hardship behavior of individual video sub streams transmitted over the sporadic transmission methods for a cross section framework Grid media sorts out the centers into an unstructured overlay, and grasps a novel push-pull spouting part to get data from the accessory center points. The power mode in the unstructured overlay can work outstandingly with the high beat rate in P2P environment while the push mode can gainfully decrease the gathered dormancy at customer side. We moreover depict our practical game plan of traversal over Network Address Translators (NATs) and firewalls. Keyword: Voice over IP (VoIP), MOVi (Mobile Opportunistic Video-on-demand), peer to peer, Grid media. I. INTRODUCTIONPassing on a better than average organization that can conform to contrasting information exchange limit. Nature of Service is described as the limit of framework to give an average organization by utilizing the possible most compelling degree from the availability of base asserted, give framework limit, rout jitter and deferral [1]. Prerequisites for a Quality of Service (QoS) basic to guarantee an average organization for the learners as the end customers in the framework given circumstances. Video calls are more information transmission asking for appear differently in relation to voice-over-IP (VoIP) calls. Data rate for Skype VoIP calls is around 30kbps [2] [3], an extraordinary quality Skype video call can without a lot of a stretch eat up exchange speed up to 950kbps [3]. Video conferencing customer nature of experience (QoE) corrupts by and large if the limited end-to-end video delay goes more than 350ms [4]. New Direct Link Service [6] (nDLS) recommendation of the 802.11 Standardization effort that inconspicuous components the methodologies for direct correspondence, in base mode, between two stations inside a single BSS. We propose a novel Inter BSS DLS (iDLS) [7] tradition to support a more expansive methodology that licenses direct correspondence to be set up between any two stations paying little regard to the BSS cooperation and repressions.

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Tuning Skype's Redundancy Control Algorithm for User Satisfaction

Cited by 28 publications

References 9 publications

Exploiting just-noticeable difference of delays for improving quality of experience in video conferencing

Exploiting just-noticeable difference of delays for improving quality of experience in video conferencing

An empirical evaluation of VoIP playout buffer dimensioning in Skype, Google talk, and MSN Messenger

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