TCP-friendly VoIP By Scalable Codec over DCCP

碩士 === 國立政治大學 === 資訊科學學系 === 101 === Congestion control is one of the major problems of network management. Most current network applications use either TCP or UDP to transport data. TCP is equipped with a congestion control mechanism but is not suitable for real-time multimedia applications due to...

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Main Author: 林耿誠
Other Authors: 連耀南
Format: Others
Language:zh-TW
Online Access:http://ndltd.ncl.edu.tw/handle/91940874595242706279
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spelling ndltd-TW-101NCCU53940352016-03-18T04:42:09Z http://ndltd.ncl.edu.tw/handle/91940874595242706279 TCP-friendly VoIP By Scalable Codec over DCCP 利用多層編碼配合DCCP形成與TCP友善的網路電話 林耿誠 碩士 國立政治大學 資訊科學學系 101 Congestion control is one of the major problems of network management. Most current network applications use either TCP or UDP to transport data. TCP is equipped with a congestion control mechanism but is not suitable for real-time multimedia applications due to its instability of delay time. On the other hand, UDP fixes its data rate and doesn't change it during the period of transmission even when the network is congested. Under this circumstance, DCCP, which is an unreliable transport protocol but has a congestion control mechanism, is proposed to replace UDP to support real-time network applications such as VoIP. Our previous study showed that a flexible bit rate CODEC to support VoIP over DCCP can effectively control network congestions while maintaining a good voice quality. However, it has an implementation issue yet to be addressed: it requires a bidirectional interaction between DCCP and CODEC. This thesis proposes to use a scalable CODEC approach to support flexible bit rate VoIP over DCCP. The CODEC sends the entire spectrum of input voice stream to DCCP. DCCP then selects the appropriate voice activation level to compose output stream according to the measured network status, which is feedbacked from the receiver side. The interaction between DCCP and CODEC is avoid. The proposed scheme was evaluated in a real local area network against two other protocols under various VoIP environments, CBR over UDP and Flexible Bit-Rate. Experimental results show that the proposed scheme can outperform CBR over VoIP in the most serious network congestion (under our lab configuration) by 40% in packet loss rate and 1.5 in MOS. It can outperform Flexible Bit-rate over VoIP by 8% in packet loss rate and 1.0 in MOS. Finally,the fairness test shows that our scheme can coexist with TCP with a fairness index higher than 95% even when network is congested. 連耀南 學位論文 ; thesis 82 zh-TW
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description 碩士 === 國立政治大學 === 資訊科學學系 === 101 === Congestion control is one of the major problems of network management. Most current network applications use either TCP or UDP to transport data. TCP is equipped with a congestion control mechanism but is not suitable for real-time multimedia applications due to its instability of delay time. On the other hand, UDP fixes its data rate and doesn't change it during the period of transmission even when the network is congested. Under this circumstance, DCCP, which is an unreliable transport protocol but has a congestion control mechanism, is proposed to replace UDP to support real-time network applications such as VoIP. Our previous study showed that a flexible bit rate CODEC to support VoIP over DCCP can effectively control network congestions while maintaining a good voice quality. However, it has an implementation issue yet to be addressed: it requires a bidirectional interaction between DCCP and CODEC. This thesis proposes to use a scalable CODEC approach to support flexible bit rate VoIP over DCCP. The CODEC sends the entire spectrum of input voice stream to DCCP. DCCP then selects the appropriate voice activation level to compose output stream according to the measured network status, which is feedbacked from the receiver side. The interaction between DCCP and CODEC is avoid. The proposed scheme was evaluated in a real local area network against two other protocols under various VoIP environments, CBR over UDP and Flexible Bit-Rate. Experimental results show that the proposed scheme can outperform CBR over VoIP in the most serious network congestion (under our lab configuration) by 40% in packet loss rate and 1.5 in MOS. It can outperform Flexible Bit-rate over VoIP by 8% in packet loss rate and 1.0 in MOS. Finally,the fairness test shows that our scheme can coexist with TCP with a fairness index higher than 95% even when network is congested.
author2 連耀南
author_facet 連耀南
林耿誠
author 林耿誠
spellingShingle 林耿誠
TCP-friendly VoIP By Scalable Codec over DCCP
author_sort 林耿誠
title TCP-friendly VoIP By Scalable Codec over DCCP
title_short TCP-friendly VoIP By Scalable Codec over DCCP
title_full TCP-friendly VoIP By Scalable Codec over DCCP
title_fullStr TCP-friendly VoIP By Scalable Codec over DCCP
title_full_unstemmed TCP-friendly VoIP By Scalable Codec over DCCP
title_sort tcp-friendly voip by scalable codec over dccp
url http://ndltd.ncl.edu.tw/handle/91940874595242706279
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