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dc.contributor.authorYou, Liang-Fu
dc.date.accessioned2009-08-23T04:46:34Z
dc.date.accessioned2020-05-29T06:19:30Z-
dc.date.available2009-08-23T04:46:34Z
dc.date.available2020-05-29T06:19:30Z-
dc.date.issued2006-10-18T09:33:09Z
dc.date.submitted2001-12-20
dc.identifier.urihttp://dspace.fcu.edu.tw/handle/2377/1915-
dc.description.abstractThe computer telephony gateway (or called internet telephony gateway)is the bridge between telecommunication and internet. Usually using a computer added a telephony interface card becomes a computer telephony gateway. The paper first offers two architecture of the internet telephony gateway using modems,a sound card and a network interface card that every personal computer already has. With the internet telephony software "SpeakFreely" we experimented and found out that the packet loss took place in insufficient bandwidth and the spike took place in network jam. The paper issued an algorithm for detection and solution of the spike or bandwidth insufficiency. With such algorithm to improve the voice quality of internet telephony.
dc.description.sponsorship中國文化大學,台北市
dc.format.extent10p.
dc.format.extent479236 bytes
dc.format.mimetypeapplication/pdf
dc.language.isozh_TW
dc.relation.ispartofseries2001 NCS會議
dc.subjectReal time protocol
dc.subjectComputer telephony gateway
dc.subjectHybrid transformer
dc.subjectAdaptive playout buffer
dc.subjectSpike
dc.subjectRTP
dc.subject.otherInformation Communications,Society and Laws
dc.titleThe Study of Telephony based on Real Time Protocol
dc.title.alternative以即時協定為基礎的網際網路電話研究
分類:2001年 NCS 全國計算機會議

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