Study on the application of an AMR speech codec to VoIP

Authors:
B. Liu
Affiliations:
Sch. of Electron. & Electr. Eng., Kyungpook Nat. Univ., Taegu, South Korea
Venue:
ICASSP '01 Proceedings of the Acoustics, Speech, and Signal Processing, 2001. on IEEE International Conference - Volume 03
Year:
2001

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Cited 4

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Survey on application-layer mechanisms for speech quality adaptation in VoIP

ACM Computing Surveys (CSUR)
Review: VoIP: State of art for global connectivity-A critical review

Journal of Network and Computer Applications

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Abstract

Degradation of speech quality caused by packet loss of voice traffic is still one of critical technical barriers of the VoIP system. We propose a new VoIP system that can adapt transmission bit rate flexibly to network conditions to reduce packet loss. In order to determine the transmission bit rate depending upon the network conditions on a frame basis, we use the time-stamp parameter in the RTP of the H.323 protocol. Experimental results demonstrate that the proposed system is very promising to reduce packet loss that leads to improvement of speech quality.