以SIP為基礎之VoIP電話系統用戶代理者與伺服器的研究

Abstract

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.