{"title":"An Adaptive Receiver Buffer Adjust Algorithm for Voice & Video on IP Applications","authors":"Jing Liu, Z. Niu","doi":"10.1109/APCC.2005.1554146","DOIUrl":null,"url":null,"abstract":"In this paper, a new adaptive receiver buffer adjust algorithm is proposed for voice and video on IP (V2oIP) applications with the consideration of voice and video's characters and network conditions. This algorithm take the voice stream as the main stream, adjust the voice buffer according to the voice characters, and synchronize the video stream with it considering video's characters. This algorithm divides network status into two modes: normal mode and spike mode, according to the delay of the coming packets and the size of buffer. In normal mode, the receiver adjusts the buffer delay at the beginning of every voice's talk-spurt. In spike mode, buffer is adjusted when every packet arrives. The adjust can based on a prediction of the delay according to the delay of former packets, or based on a prearranged way. Simulations and experiments show that this strategy can well conceal the delay jitter and reduce the packet loss rate","PeriodicalId":176147,"journal":{"name":"2005 Asia-Pacific Conference on Communications","volume":"69 1","pages":"0"},"PeriodicalIF":0.0000,"publicationDate":"2005-12-12","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":"2","resultStr":null,"platform":"Semanticscholar","paperid":null,"PeriodicalName":"2005 Asia-Pacific Conference on Communications","FirstCategoryId":"1085","ListUrlMain":"https://doi.org/10.1109/APCC.2005.1554146","RegionNum":0,"RegionCategory":null,"ArticlePicture":[],"TitleCN":null,"AbstractTextCN":null,"PMCID":null,"EPubDate":"","PubModel":"","JCR":"","JCRName":"","Score":null,"Total":0}
引用次数: 2
Abstract
In this paper, a new adaptive receiver buffer adjust algorithm is proposed for voice and video on IP (V2oIP) applications with the consideration of voice and video's characters and network conditions. This algorithm take the voice stream as the main stream, adjust the voice buffer according to the voice characters, and synchronize the video stream with it considering video's characters. This algorithm divides network status into two modes: normal mode and spike mode, according to the delay of the coming packets and the size of buffer. In normal mode, the receiver adjusts the buffer delay at the beginning of every voice's talk-spurt. In spike mode, buffer is adjusted when every packet arrives. The adjust can based on a prediction of the delay according to the delay of former packets, or based on a prearranged way. Simulations and experiments show that this strategy can well conceal the delay jitter and reduce the packet loss rate