{"title":"On traffic characteristics and user experience of Skype video call","authors":"Jing Zhu","doi":"10.1109/IWQOS.2011.5931328","DOIUrl":"https://doi.org/10.1109/IWQOS.2011.5931328","url":null,"abstract":"In this paper, we study Skype video call in LAN, WAN, and WiMAX. Our main interest is on traffic characteristics and user experience. Using the standard Foreman video sequence (320×240, 30fps, 8 second), we show that a Skype video call can adapt its source rate from < 5kBps up to ∼60kBps. The maximum achievable MOS (Mean Opinion Score) and EFR (Effective Frame Rate) is about 3.5 and 20 fps, respectively. Our experimental results also show that source rate is the dominating factor in determining both traffic characteristics and user experience of a Skype video call; while end-to-end delay or transport layer protocol, e.g. TCP or UDP, plays very little role. We also study the minimum RSSI requirement in a typical WiMAX network for achieving various levels of user experience: good, fair, and low, and derive the explicit logarithmic functions using the curve fitting technique to predict MOS and mean packet inter-arrival time (PIT) based on source rate.","PeriodicalId":127279,"journal":{"name":"2011 IEEE Nineteenth IEEE International Workshop on Quality of Service","volume":"7 1","pages":"0"},"PeriodicalIF":0.0,"publicationDate":"2011-06-06","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"127747715","PeriodicalName":null,"FirstCategoryId":null,"ListUrlMain":null,"RegionNum":0,"RegionCategory":"","ArticlePicture":[],"TitleCN":null,"AbstractTextCN":null,"PMCID":"","EPubDate":null,"PubModel":null,"JCR":null,"JCRName":null,"Score":null,"Total":0}
{"title":"Designing efficient codes for synchronization error channels","authors":"Hao Wang, Bill Lin","doi":"10.1109/IWQOS.2011.5931353","DOIUrl":"https://doi.org/10.1109/IWQOS.2011.5931353","url":null,"abstract":"For communications and networking channels, coding techniques are widely used to correct errors in corrupted messages. The “Quality of Protection” (QoP) that can be provided via error correction directly affects the “Quality of Service” (QoS) experienced by users. The errors are commonly assumed to be substitution or erasure errors. Such systems rely on perfect synchronization so that no bit is deleted and no extra bit is inserted. However, in a system without the presence of perfect synchronization, special coding algorithms may be required to correct potential insertion or deletion errors in transmitted messages. Especially for systems suffering from frequent loss of synchronization, packets may require many retransmissions to guarantee reliable communication. Such schemes may become too expensive to be practical. In this paper, we propose a new synchronization channel error model based on the observations from current communication systems. In this model, the channel introduces at most t synchronization errors in each run of the transmitted sequence. We present run-length limited permutation codes capable of correcting synchronization errors based on this channel error model. Compared to previously developed codes, our codes have the advantage of correcting frequent synchronization errors, and therefore they are suitable for disruptive network channels that suffer severe synchronization failures.","PeriodicalId":127279,"journal":{"name":"2011 IEEE Nineteenth IEEE International Workshop on Quality of Service","volume":"55 1","pages":"0"},"PeriodicalIF":0.0,"publicationDate":"2011-06-01","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"126636887","PeriodicalName":null,"FirstCategoryId":null,"ListUrlMain":null,"RegionNum":0,"RegionCategory":"","ArticlePicture":[],"TitleCN":null,"AbstractTextCN":null,"PMCID":"","EPubDate":null,"PubModel":null,"JCR":null,"JCRName":null,"Score":null,"Total":0}
{"title":"Leveraging statistical multiplexing gains in single- and multi-hop networks","authors":"Amr Rizk, M. Fidler","doi":"10.1109/IWQOS.2011.5931352","DOIUrl":"https://doi.org/10.1109/IWQOS.2011.5931352","url":null,"abstract":"Packet switched networks achieve significant resource savings due to statistical multiplexing. In this work we explore statistical multiplexing gains in single and multi-hop networks. To this end, we analyze performance metrics such as delay bounds for a through flow comparing different results from the stochastic network calculus. We distinguish different multiplexing gains that stem from independence assumptions between flows at a single hop as well as flows at consecutive hops of a network path. Further, we show corresponding numerical results. In addition to deriving the benefits of various statistical multiplexing models on performance bounds, we contribute insights into the scaling of end-to-end delay bounds in the number of hops n of a network path under statistical independence.","PeriodicalId":127279,"journal":{"name":"2011 IEEE Nineteenth IEEE International Workshop on Quality of Service","volume":"7 1","pages":"0"},"PeriodicalIF":0.0,"publicationDate":"2011-06-01","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"125299537","PeriodicalName":null,"FirstCategoryId":null,"ListUrlMain":null,"RegionNum":0,"RegionCategory":"","ArticlePicture":[],"TitleCN":null,"AbstractTextCN":null,"PMCID":"","EPubDate":null,"PubModel":null,"JCR":null,"JCRName":null,"Score":null,"Total":0}