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題名 以SIP為基礎之VoIP電話系統用戶代理者與伺服器的研究
A Research for User Agents and Servers of VoIP Phone System Based on SIP
作者 詹凱富
應尚宏
洪永政
林強生
章博傑
葉生正
關鍵詞 VoIP;SIP;Instant message;User Agent;SER;3G;NAT
日期 2005
上傳時間 17-Oct-2017 12:12:08 (UTC+8)
摘要 SIP(Session Initiation Protocol)除了可建立VoIP網路電話之應用,也可與HTTP、SMTP、RTP...等協定結合,提供整合語音與其它多媒體的通訊服務,如即時訊息(Instant Messaging)與個人現形(Presence)服務。另外,SIP已經被公認為IP網路與傳統PSTN電信網路整合的關鍵技術,而且亦被第三代行動通訊系統(3G)採用為未來無線多媒體通訊的技術標準。故本論文以SIP為標準通訊協定進行用戶代理者UA(User Agent)的研發,完成與Xten的X-lite和工研院的CCL Phone之相容性測試。並對本研究所整合應用的伺服器SER(SIP Express Router)歸納分析其系統設定與操作管理程序,且對穿越NAT(Network Address Translation)的技術提出初步的解決方法。
SIP not only can make session of IP phone, but also can integrate the techniques including HTTP, SMTP, and RTP…. SIP provides integration voice and multimedia communication service, such as Instant Message and Presence, etc. Also, SIP is regarded as resolution to the problem of signal transmission between different types of network. The particular study intending to invite a real UA(User Agent) bases on SIP and the interoperation test of X-lite and CCL Phone. In addition, the study suggests the experiences of the deeper management of SER(SIP Express Router) and the reservation of NAT(Network Address Translation)issue.
關聯 TANET 2005 台灣網際網路研討會論文集
網際網路視訊語音應用
資料類型 article
dc.creator (作者) 詹凱富zh_TW
dc.creator (作者) 應尚宏zh_TW
dc.creator (作者) 洪永政zh_TW
dc.creator (作者) 林強生zh_TW
dc.creator (作者) 章博傑zh_TW
dc.creator (作者) 葉生正zh_TW
dc.date (日期) 2005
dc.date.accessioned 17-Oct-2017 12:12:08 (UTC+8)-
dc.date.available 17-Oct-2017 12:12:08 (UTC+8)-
dc.date.issued (上傳時間) 17-Oct-2017 12:12:08 (UTC+8)-
dc.identifier.uri (URI) http://nccur.lib.nccu.edu.tw/handle/140.119/113683-
dc.description.abstract (摘要) SIP(Session Initiation Protocol)除了可建立VoIP網路電話之應用,也可與HTTP、SMTP、RTP...等協定結合,提供整合語音與其它多媒體的通訊服務,如即時訊息(Instant Messaging)與個人現形(Presence)服務。另外,SIP已經被公認為IP網路與傳統PSTN電信網路整合的關鍵技術,而且亦被第三代行動通訊系統(3G)採用為未來無線多媒體通訊的技術標準。故本論文以SIP為標準通訊協定進行用戶代理者UA(User Agent)的研發,完成與Xten的X-lite和工研院的CCL Phone之相容性測試。並對本研究所整合應用的伺服器SER(SIP Express Router)歸納分析其系統設定與操作管理程序,且對穿越NAT(Network Address Translation)的技術提出初步的解決方法。
dc.description.abstract (摘要) SIP not only can make session of IP phone, but also can integrate the techniques including HTTP, SMTP, and RTP…. SIP provides integration voice and multimedia communication service, such as Instant Message and Presence, etc. Also, SIP is regarded as resolution to the problem of signal transmission between different types of network. The particular study intending to invite a real UA(User Agent) bases on SIP and the interoperation test of X-lite and CCL Phone. In addition, the study suggests the experiences of the deeper management of SER(SIP Express Router) and the reservation of NAT(Network Address Translation)issue.
dc.format.extent 585578 bytes-
dc.format.mimetype application/pdf-
dc.relation (關聯) TANET 2005 台灣網際網路研討會論文集zh_TW
dc.relation (關聯) 網際網路視訊語音應用zh_TW
dc.subject (關鍵詞) VoIP;SIP;Instant message;User Agent;SER;3G;NATen_US
dc.title (題名) 以SIP為基礎之VoIP電話系統用戶代理者與伺服器的研究zh_TW
dc.title (題名) A Research for User Agents and Servers of VoIP Phone System Based on SIPen_US
dc.type (資料類型) article