Improving VoIP Quality Using Silence Description Packets in the Jitter Buffer
スポンサーリンク
概要
- 論文の詳細を見る
The basic playout scheme (BAS) is designed not to take into account network impairment information during silence periods. We propose a jitter-robust playout mechanism (RST), which uses silence description (SID) packets. The lateness loss percentages are compared between the BAS and the RST algorithms. We report that the accuracy of the playout schedule calculation in the BAS is getting worse as the previous silence interval increases and our proposed RST algorithm is more effective in removing high jitter than the BAS. Under high jitter Internet conditions, the accuracy of the estimates and therefore the resulting of VoIP playout quality can be significantly improved by using the SID packets in the playout schedule recalculation.
- (社)電子情報通信学会の論文
- 2008-11-01
著者
-
Atwood J.
Department Of Computer Science Concordia University
-
Atwood J.
Department Of Computer Science And Software Engineering Concordia University
-
JUNG Younchan
School of Information, Communications & Electronic Engineering, The Catholic University of Korea
-
ZEPERNICK Hans-Jurgen
School of Engineering, Radio Communications Group, Blekinge Institute of Technology
-
Jung Younchan
School Of Information Communications & Electronic Engineering The Catholic University Of Korea
-
Jung Younchan
School Of Information Communications & Electronic Engineering D3 17 The Catholic University Of K
-
Zepernick Hans-jurgen
School Of Engineering Radio Communications Group Blekinge Institute Of Technology
-
Atwood J.
Dep. Of Computer Sci. And Software Engineering Concordia Univ.
関連論文
- Improving VoIP Quality Using Silence Description Packets in the Jitter Buffer
- β-Adaptive Playout Scheme for Voice over IP Applications(Internet)
- Sliding Playout Algorithm Based on Per-Talkspurt Adjustment without Using Timestamps(Multimedia Communication)(Internet Technology IV)
- VoIP Accounting Model : Using the Gap Ratio as a Quality Metric