隨著網路電話蓬勃發展的趨勢下,現今日新月異的網路視訊電話,可帶給人們跳脫傳統電話不能與人面對面對話的限制。在網路電話的領域中,最常被討論的兩種協定,一是SIP(Session Initiation Protocol),由IETF(Internet Engineering Task Force)制定的網路電話協定。另一則是H.323,由ITU-T(InternationalTelecommunication Union-Telecommunication)制定。本論文實屬應用程式開發與研究,以NeetMeeting網路電話為實例,記錄現行H.323網路視訊電話RTP封包,將其RTP影像封包利用H.263解碼後存至檔案裡,它是以C++程式語言為基礎,配合Winpcap實作此系統,藉由捕捉網卡上的H.323控制封包以及RTP封包後,分析解析封包資訊,將其記錄下來,目前一般的網路視訊電話還沒有此類的記錄功能,本研究成果可提供我們隨時還原每通影像電話傳送過程的功能。對於探討網路電話協定的運作、RTP封包解碼架構狀況有很大的助益。
The development of internet phone has extended beyond the scope of the traditional telephone by allowing people to talk face-to-face using video conference phone. There are two famous protocols designed to support such technology: SIP (Session Initiation Protocol) by IETF (Internet Engineering Task Force) and H.323 by ITU-T (International Telecommunication Union-Telecommunication). In this thesis, we use the Netmeeting package as an example to design a H.323 based video phone monitoring and recording system. The system records the RTP packets of H.323 based video conference. Once the RTP video and audio packets are decoded, they are stored as separate files. The system was developed using C++ language. We use the WinPcap library to capture the control and RTP packets, followed by analyzing their signal flows, and then recording them. Thus the system allows us to retrieve and replay any videos and audios created in the previously established communication session. Such feature is usually not supported by the commercially available internet video phone system. This system may also help us to understand the interet phone protocol operations as well as RTP packet decording structure.