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dc.contributor.authorWu, Chun-Fengen_US
dc.contributor.authorChiang, Yuan-Chuanen_US
dc.contributor.authorChang, Wen-Wheien_US
dc.date.accessioned2014-12-08T15:29:35Z-
dc.date.available2014-12-08T15:29:35Z-
dc.date.issued2013en_US
dc.identifier.issn0253-3839en_US
dc.identifier.urihttp://hdl.handle.net/11536/21276-
dc.identifier.urihttp://dx.doi.org/10.1080/02533839.2012.726334en_US
dc.description.abstractPacket loss and delay are the major network impairments for transporting real-time voice over internet protocol (IP) networks. In the proposed system, multiple descriptions of the speech are used to take advantage of packet path diversity. A new objective method is presented for predicting the perceived quality of multi-stream voice transmission. Also proposed is a joint playout buffer and forward error control (FEC) adjustment scheme that maximizes the perceived speech quality via delay-loss trading. Experimental results showed that the proposed multi-stream voice transmission scheme achieves significant reductions in delay-and packet-loss rates as well as improved speech quality.en_US
dc.language.isoen_USen_US
dc.subjectplayout bufferen_US
dc.subjectforward error controlen_US
dc.subjectmultiple description codingen_US
dc.subjectvoice quality prediction modelen_US
dc.titleJoint playout and FEC control for multi-stream voice over IP networksen_US
dc.typeArticleen_US
dc.identifier.doi10.1080/02533839.2012.726334en_US
dc.identifier.journalJOURNAL OF THE CHINESE INSTITUTE OF ENGINEERSen_US
dc.citation.volume36en_US
dc.citation.issue2en_US
dc.citation.spage224en_US
dc.citation.epage235en_US
dc.contributor.department傳播研究所zh_TW
dc.contributor.departmentInstitute of Communication Studiesen_US
dc.identifier.wosnumberWOS:000315203100011-
dc.citation.woscount0-
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