Please use this identifier to cite or link to this item: https://doi.org/10.1109/ICME.2005.1521388
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dc.titleOptimization of source and channel coding for voice over IP
dc.contributor.authorHuang, Y.
dc.contributor.authorKorhonen, J.
dc.contributor.authorWang, Y.
dc.date.accessioned2013-07-04T08:25:28Z
dc.date.available2013-07-04T08:25:28Z
dc.date.issued2005
dc.identifier.citationHuang, Y.,Korhonen, J.,Wang, Y. (2005). Optimization of source and channel coding for voice over IP. IEEE International Conference on Multimedia and Expo, ICME 2005 2005 : 173-176. ScholarBank@NUS Repository. <a href="https://doi.org/10.1109/ICME.2005.1521388" target="_blank">https://doi.org/10.1109/ICME.2005.1521388</a>
dc.identifier.isbn0780393325
dc.identifier.urihttp://scholarbank.nus.edu.sg/handle/10635/41350
dc.description.abstractVoice over Internet Protocol (VoIP) applications must typically choose a tradeoff between the bits allocated for Forward Error Correcting (FEC) and that for the source coding to achieve the best speech quality at a given packet loss rate. In this paper, we present a new scheme to optimize the speech quality subject to the bandwidth constraints and the packet loss rate. The scheme adopts Adaptive Multi-Rate (AMR) speech codec along with a FEC scheme based on Exclusive OR (XOR) operations. Retransmission is also taken into account if the Round Trip Time (RTT) is within a certain limit. We use a simplified E-Model as objective metric. Subjective listening tests show that our scheme improves the perceptual speech quality significantly compared to the non-adaptive baseline speech transmission system. © 2005 IEEE.
dc.description.urihttp://libproxy1.nus.edu.sg/login?url=http://dx.doi.org/10.1109/ICME.2005.1521388
dc.sourceScopus
dc.typeConference Paper
dc.contributor.departmentCOMPUTER SCIENCE
dc.description.doi10.1109/ICME.2005.1521388
dc.description.sourcetitleIEEE International Conference on Multimedia and Expo, ICME 2005
dc.description.volume2005
dc.description.page173-176
dc.identifier.isiutNOT_IN_WOS
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