{"title":"Sender based adaptive VoIP quality improvement using constructive feedback","authors":"Ehsan Faghihi, M. Behdadfar, M. E. Sadeghi","doi":"10.1109/IKT.2015.7288752","DOIUrl":null,"url":null,"abstract":"QoS analysis of VoIP traffic is inevitable in quality improvement of multimedia converged networks. Therefore, adaptive speech quality improvement is vital for managing the VoIP communication quality in order to meet users' quality expectations and decrease utilized resources. This paper proposes a new approach for augmenting the real-time VoIP communication quality and tries to explore its impact on VoIP QoS parameters. The scheme includes different voice codecs and different voice payload sizes for each codec. Simulation results show that the proposed scheme leads to superior VoIP quality compared to some existing algorithms.","PeriodicalId":338953,"journal":{"name":"2015 7th Conference on Information and Knowledge Technology (IKT)","volume":"38 1","pages":"0"},"PeriodicalIF":0.0000,"publicationDate":"2015-05-26","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":"4","resultStr":null,"platform":"Semanticscholar","paperid":null,"PeriodicalName":"2015 7th Conference on Information and Knowledge Technology (IKT)","FirstCategoryId":"1085","ListUrlMain":"https://doi.org/10.1109/IKT.2015.7288752","RegionNum":0,"RegionCategory":null,"ArticlePicture":[],"TitleCN":null,"AbstractTextCN":null,"PMCID":null,"EPubDate":"","PubModel":"","JCR":"","JCRName":"","Score":null,"Total":0}
引用次数: 4
Abstract
QoS analysis of VoIP traffic is inevitable in quality improvement of multimedia converged networks. Therefore, adaptive speech quality improvement is vital for managing the VoIP communication quality in order to meet users' quality expectations and decrease utilized resources. This paper proposes a new approach for augmenting the real-time VoIP communication quality and tries to explore its impact on VoIP QoS parameters. The scheme includes different voice codecs and different voice payload sizes for each codec. Simulation results show that the proposed scheme leads to superior VoIP quality compared to some existing algorithms.