A. Mukhopadhyay, T. Chakraborty, S. Bhunia, I. S. Misra, S. Sanyal
{"title":"拥塞无线网络场景下增强VoIP性能的研究","authors":"A. Mukhopadhyay, T. Chakraborty, S. Bhunia, I. S. Misra, S. Sanyal","doi":"10.1109/COMSNETS.2011.5716509","DOIUrl":null,"url":null,"abstract":"Stringent QoS maintenance in a wireless environment for VoIP communication is a major challenge. VoIP inherently generates constant bit rate traffic and is highly sensitive to network delay. However, unpredictable network congestion makes a VoIP session so degraded that its QoS goes below a tolerable limit. Accordingly, a suitable solution is needed to adapt varying network conditions satisfying minimum QoS while maintaining transparency to the end user. We have extensively studied the effect of variable voice packet payload size with changing number of voice sample frames in the payload of the RTP packets. For this, we have used OPNET Modeler 14.5.A to observe the performance in terms of MOS, End-to-end delay through extensive simulations. Results are provided with valid discussions. Based on the observation, we propose an Adaptive VoIP (AdVoIP) algorithm that may be used in any real life VoIP network for further enhancement of the performance. The adaptive algorithm may use RTCP Receiver Reports to assess the network conditions in real wireless scenarios.","PeriodicalId":302678,"journal":{"name":"2011 Third International Conference on Communication Systems and Networks (COMSNETS 2011)","volume":"200 1","pages":"0"},"PeriodicalIF":0.0000,"publicationDate":"2011-02-17","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":"11","resultStr":"{\"title\":\"Study of enhanced VoIP performance under congested wireless network scenarios\",\"authors\":\"A. Mukhopadhyay, T. Chakraborty, S. Bhunia, I. S. Misra, S. Sanyal\",\"doi\":\"10.1109/COMSNETS.2011.5716509\",\"DOIUrl\":null,\"url\":null,\"abstract\":\"Stringent QoS maintenance in a wireless environment for VoIP communication is a major challenge. VoIP inherently generates constant bit rate traffic and is highly sensitive to network delay. However, unpredictable network congestion makes a VoIP session so degraded that its QoS goes below a tolerable limit. Accordingly, a suitable solution is needed to adapt varying network conditions satisfying minimum QoS while maintaining transparency to the end user. We have extensively studied the effect of variable voice packet payload size with changing number of voice sample frames in the payload of the RTP packets. For this, we have used OPNET Modeler 14.5.A to observe the performance in terms of MOS, End-to-end delay through extensive simulations. Results are provided with valid discussions. Based on the observation, we propose an Adaptive VoIP (AdVoIP) algorithm that may be used in any real life VoIP network for further enhancement of the performance. The adaptive algorithm may use RTCP Receiver Reports to assess the network conditions in real wireless scenarios.\",\"PeriodicalId\":302678,\"journal\":{\"name\":\"2011 Third International Conference on Communication Systems and Networks (COMSNETS 2011)\",\"volume\":\"200 1\",\"pages\":\"0\"},\"PeriodicalIF\":0.0000,\"publicationDate\":\"2011-02-17\",\"publicationTypes\":\"Journal Article\",\"fieldsOfStudy\":null,\"isOpenAccess\":false,\"openAccessPdf\":\"\",\"citationCount\":\"11\",\"resultStr\":null,\"platform\":\"Semanticscholar\",\"paperid\":null,\"PeriodicalName\":\"2011 Third International Conference on Communication Systems and Networks (COMSNETS 2011)\",\"FirstCategoryId\":\"1085\",\"ListUrlMain\":\"https://doi.org/10.1109/COMSNETS.2011.5716509\",\"RegionNum\":0,\"RegionCategory\":null,\"ArticlePicture\":[],\"TitleCN\":null,\"AbstractTextCN\":null,\"PMCID\":null,\"EPubDate\":\"\",\"PubModel\":\"\",\"JCR\":\"\",\"JCRName\":\"\",\"Score\":null,\"Total\":0}","platform":"Semanticscholar","paperid":null,"PeriodicalName":"2011 Third International Conference on Communication Systems and Networks (COMSNETS 2011)","FirstCategoryId":"1085","ListUrlMain":"https://doi.org/10.1109/COMSNETS.2011.5716509","RegionNum":0,"RegionCategory":null,"ArticlePicture":[],"TitleCN":null,"AbstractTextCN":null,"PMCID":null,"EPubDate":"","PubModel":"","JCR":"","JCRName":"","Score":null,"Total":0}
Study of enhanced VoIP performance under congested wireless network scenarios
Stringent QoS maintenance in a wireless environment for VoIP communication is a major challenge. VoIP inherently generates constant bit rate traffic and is highly sensitive to network delay. However, unpredictable network congestion makes a VoIP session so degraded that its QoS goes below a tolerable limit. Accordingly, a suitable solution is needed to adapt varying network conditions satisfying minimum QoS while maintaining transparency to the end user. We have extensively studied the effect of variable voice packet payload size with changing number of voice sample frames in the payload of the RTP packets. For this, we have used OPNET Modeler 14.5.A to observe the performance in terms of MOS, End-to-end delay through extensive simulations. Results are provided with valid discussions. Based on the observation, we propose an Adaptive VoIP (AdVoIP) algorithm that may be used in any real life VoIP network for further enhancement of the performance. The adaptive algorithm may use RTCP Receiver Reports to assess the network conditions in real wireless scenarios.