基於AFEC之嵌入式網路電話裝置語音品質控制機制

A Bandwidth-efficient AFEC-based Voice Quality Control Mechanism for Embedded Systems

李開振(Kai-Zhen Li)

指導教授 ： 黃育綸 ； 邵家健

國立交通大學/資訊學院/資訊科學與工程研究所/碩士(2009年)

https://doi.org/10.6842/NCTU.2009.00584

摘要

隨著網路電話的蓬勃發展，近年來網路電話的聲音品質也逐漸變成眾所矚目的焦點。影響網路電話聲音品質的因素除了所使用的編碼方式之外，其還包括了網路狀況，特別是封包遺失。目前的研究有提出了一些如何來減少封包遺失對封包交換網路的影響方法。盡管這些方法在很多情況下表現都不錯，但並不是很適合使用於如網路電話等高度即時互動的應用上。因為網路電話的聲音品質除了封包遺失，還會被延遲及抖動等因素所影響。本論文提出一個可以用來探討封包遺失及錯誤修復等跟網路電話聲音品質之間的關係的模型。基於這個模型，我們設計出一個控制機制，可以在擁擠的網路裡，增加最少的頻寬而有效提升網路電話的聲音品質。我們所提出的機制建構在自適應前向錯誤更正系統(AFEC)上，應用Reed Solomon Code來完成封包級的錯誤修復， 而這樣的設計也更能符合真實的丟包情況。為了可以實現在嵌入式系統上，我們有對所需要的運算複雜度進行簡化。我們把所提出的機制實作在一個有網路電話功能的家庭路由器上，同時也完成了一系列的實驗。透過分析實驗結果得知，在這個機制的控制下，即使封包遺失率達到30%左右時， 網路語音品質還能維持在設定的標準之上。相較於其他方法，我們的機制能更有效率地使用網路頻寬。

關鍵字

網路電話 ； 網路丟包 ； 錯誤更正 ； 語音品質

並列摘要

As the massive deployment of packetized voice over the Internet, the voice quality has become an important topic in recent years. Besides the codec type, the voice quality can also be affected by network conditions, especially the network loss. To remedy this issue, many mechanisms were proposed to model or recover the loss in packet networks. Despite these mechanisms perform well in many cases, however, they are not suitable for real-time and interactive applications such as VoIP, where the voice quality is sensitive not only to network loss, but also to jitter and end-to-end delay. In this thesis, we propose a model to address the quality of packetized voice under network loss and redundancy for loss recovery. Based on this model, we present a control mechanism to recover the voice quality of VoIP in a lossy network with less bandwidth overhead. Our mechanism is built on Adaptive Forward Error Correction (AFEC) with Reed- Solomon codes and performed at the packet level to better reflect the real world loss pattern. We try to reduce the computing complexity of our mechanism so that it can be easily realized on embedded systems. As a proof of our work, we implement the proposed mechanism on a home gateway with VoIP capability. We also perform some experiments to verify the effectiveness of our model. The experiments show that by using the proposed mechanism, under the loss ratio up to 30%, the voice quality can still be maintained above the designated threshold with the best bandwidth efficiency compared to the traditional AFEC mechanisms.

並列關鍵字

voip ； packet loss ； Adaptive forward error correction ； voice quality

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