現今的通訊領域正吹起一波網路電話(VoIP)的熱潮,而網路電話想要和公眾交換電話網路(PSTN)一樣在通訊系統中歷久不衰,就必須學習傳統電信局的架構,本論文針對傳統電信架構中最底層的總機系統做深入的研究分析,並在Microsoft Visual C++平台上實作出屬於網路電話架構中的語音自動總機(IP Auto-Attendant)。 本論文所提出的IP Auto-Attendant具有以下特色: 1.操作流程完全仿照傳統的總機系統,用戶不必改變以往的使用習慣就可以輕易上手。 2.利用SIP協定的Refer指令讓網路電話做到PBX的電話轉接功能。 3.整合使用TTS技術的語音導覽系統(Audio Guide),只需事先輸入好總機的歡迎詞,就可以實現傳統交換機所搭配的語音總機功能。 4.可搭配語音辨識系統(ASR),讓使用者可以直接用人名或職稱查詢分機號碼並自動轉接。
VoIP grows like crazy in the past 10 years. However, the PSTN system is old, complete and stable enough for public user. Thus, the VoIP system must be designed according to the PSTN’s specification and function. In this thesis, the switch and auto-attendant function of PBX will be researched and implemented in the VoIP system. The IP-based auto-attendant system is focused and integrated into the soft-switch system. Some advantages and features listed as below will be founded in this VoIP system. 1. The call-setup of VoIP system is similar to the PSTN system. The habit of call-setup for public user doesn’t change. 2. The SIP command ‘REFER’ is employed to implement the auto-attendant system. 3. The text-to-speech(TTS) system is employed and integrated into the auto-attendant for the special service such as audio guide. 4. The auto-speech-recognition(ASR) is employed and integrated into the auto-attendant for the auto-transfer machine.