隨著網路頻寬的提升,愈來愈多以前無法普及化的網路服務,在現在的環境下,也已經逐漸實用化及普及,而網路電話正是其中之一,目前最為普遍的網路電話協定,分別是ITU-T(International Telecommunication Union- Telecommunication)提出的H.323協定和IETF(Internet Engineering Task Force)所提出的SIP(Session Initiation Protocol,因為H.323系統整體架構較為嚴謹與完整,且發展時間較久,因此本研究以H.323系統所延伸擴充的客製化H.323系統為基礎來開發監控錄音系統。 本論文的目標是設計一個以客製化H.323協定為基礎之網路電話監控與錄音系統,可以在同時間針對多通電話進行處理。此研究必須要產生H.323網路電話,並且抓取分析網路封包之後,將其控制訊號,例如說Q.931及H.245訊號等進行分析以及對客製化H.323的控制訊訊號進行分析,之後將H.323事件記錄下來,最後將RTP (Real-time Transport Protocol)封包的負載(Payload)取出並且解碼,將聲音內容重組還原。此外本系統還能追蹤監控通話歷程紀錄。
Along with network bandwidth increased by new transmission technology, many applications and services are getting more and more practical and popular. The Internet phone is one of examples. Among various VoIP control protocols, H.323 and Session Initiation Protocol (SIP), proposed by ITU.T and IETF respectively, are the ones being implemented the most. The objective of the thesis is to design a monitoring and recording system for a Customized H.323-based Internet phone. It is expected capable of handing many concurrent calls in processing. The efforts of the research includes generating Customized H.323-based phone calls, capturing and analyzing the packets over networks, parsing the control signal of H.323 related protocols(eg.Q.931,H.225 and H.245) and the control signals of Customized protocol, as well as categorizing and decoding the RTP payloads from voice packets for different type of calls, the work also can record the voice call history. The functions of the intelligent monitoring and recording system has been proved properly. A GUI is designed to verify the system capabilities accordingly. For further examination of a recorded phone, the system stores an audio track of any pair of recorded caller and callee conversation in both .raw and .wav formats. The features provide expansibility of use.