【正文】 在一個組號內(nèi)，實現(xiàn)一個主機可以向組內(nèi)的所有主機同時發(fā)起語音通信，使組內(nèi)的可以所有主機都可以進行相互通信。這個程序只需要將電腦連上IP網(wǎng)絡(luò)，根據(jù)要求配置網(wǎng)絡(luò)信息，便可以進行方便的進行相互通話。首先我要感謝我的導(dǎo)師陶洋老師，他在我完成畢設(shè)的過程中，給予了我很大的幫助。這對我今后進行深入研究計算機網(wǎng)絡(luò)以及通信工程是莫大的幫助，特別是通過對C++的學(xué)習(xí)，我深刻的理解了面向?qū)ο蟮某绦蚓幊谭绞剑@能夠幫助我今后能夠?qū)υS多軟件進行優(yōu)化和設(shè)計。s main goal of this project is a set of business interoperability requirements, determine the interface and functional aspects of architecture, the call control procedures, the traffic rules and protocols, study the endtoend service quality parameters and e. 164 conversion between address and IP address, and billing and security issues. TIPHON engineering team want to all kinds of network operators to provide businessoriented solution, its work is largely based on H. 323, a series of Suggestions and the existing circuit switched network standards.(3) IETFIETF formulate new signaling protocol, including the session initial protocol (SIP), Internert and PSTN network. IPTEL working group of IETF is overall responsible for the relevant agreements and framework, including call processing grammar, etc. The working group also writes some business model file, describing the syntax by call processing implementation of business and discussing the use of grammar.(4) IMTCIMTC is made up of more than 150 members from North America, Europe and the asiapacific region, its purpose is to establish an open international standard, to promote the application of interactive multimedia conferencing solutions and implementation. VoIP BBS (Voice over IP Forum) of IMTC aperiodically do some activities to make standard, promote the IP telephone business.Due to the rapid development of Internet, and the great prospects of the business of packet voice, the above organizations actively promote the standardization of IP phone positively driven by each of the manufacturers. According to the current situation and the key work of each standard organization, related to the IP phone protocol can be divided into two main categories: H. 323 protocol and SIP. At present each organization to carry on IP network realtime business (voice, video, etc.) is no different, all is the use of the derived from RTP protocol of IETF, but are different in terms of call setup and control scheme, its representative is the following, H. 323 and SIP protocol.(1) Protocol belongs to the ITU multimedia munication protocol series, , which provide packetbased network voice, video, data and control protocol. H. 323 support point to point munication and support point to multipoint munication protocol with the MCU (multipoint control unit). As a framework agreement, H. 323 provided the system and ponent description, call way description and call signaling program. H. 323 related protocols including H. 263, H. 261 video coding standard and G. 711, G. 723, G. 722, G. 728 audio coding standard, T. 120 multipoint data conference series standard, H. grouping and synchronous standard, H. 245 system control standard, etc. The protocol stack is shown in figure 1. protocol after years of development, the function, is supported by the broad masses of manufacturers. VoIP BBS of IMTC has been used H. 323 as the basis of Internet telephony. system structure including: H. 323 terminal, Gateway, Gatekeeper, and multipoint control unit (MCU). terminal: provide realtime twoway audio, video and data munications. But the H. 323 has not stipulated the audio or video equipment, data application and network interface.Gateway: provide call signaling control, channel information conversion, and h. 323 terminals and other ITU terminals interconnect technology.Gatekeeper: provide management mechanism and management for gateway resources。s Internet changes constantly. The represented content of the bits which flows in the network is continuously evolving from simple data to multimedia. With the growing number of the informational network traffic, all sorts of business based on the Internet are also in rapid development, IP phone is one of them.The promote of the development of IP telephony is attributed to technology promoted and market driven. Since the early 90 s , IP phone has been developed from the early period of IP phone software into the IP telephony gateway. Network equipment and munications equipment manufacturers are racing to a foothold in the market so as to the builtin VoIP function to each network device as much as possible, various products are preempted in the market, develop various operators also carry out the aspect of the business. IP phone is out of the lab and into the network, into the market.At present, VoIP technology has developed from the PC primary products which provide voice services and only limited range inside the IP network, to more business, high reliability and good quality of service including voice, fax, data transmission function of tele business. Now by IP telephony Gateway, the munication between PSTN and Internet has realized, so as to realize the PC to the phone, the phone to PC and telephone to telephone call. Besides, the quality of voice munication is greatly improved, which also can satisfy the requirements of the mercial.The current IP telephony service operators in the world mainly es from large ISP and teles firms, they provide services including IP voice and fax service. Forrester Research estimates that longdistance IP telephone business in 2001 will reach $1 billion, many panies also according to their own development strategy for the experiments of IP phone and deployment, such as ATamp。其次還要感謝指導(dǎo)我完成最終設(shè)計的王婭學(xué)姐，每當我編程出錯時，總是耐心的讀我的程序找出出錯的地方，并提出各種解決方法。是一款與時俱進的網(wǎng)絡(luò)應(yīng)用軟件。所謂群呼，就是指一個主機可以向IP網(wǎng)絡(luò)內(nèi)的所有主機發(fā)起語音通話的請求，使IP網(wǎng)絡(luò)內(nèi)的所有主機都能夠同時進行通信。結(jié) 論本次程序設(shè)計實現(xiàn)了公務(wù)電話通過IP網(wǎng)絡(luò)進行單呼和組呼。主叫摘機：完成撥號，等待連接：連接完成，播放回鈴音，等待被叫端摘機： 連接完成等待摘機通話建立： 通話建立通話結(jié)束，對端掛機顯示： 對端掛機掛機顯示： 掛機被叫端收到單呼請求 收到請求被叫端未摘機，主叫端掛機，被叫端顯示： 被叫端未摘機，主叫端掛機主叫端撥號后對方占線： 對端占線第五節(jié) 組呼通話過程的建立組呼通話與單呼通話的流程比較相似，組呼通話的建立過程與單呼的不同之處在主叫端撥號后連接時沒有回鈴音；被叫端不是空閑狀態(tài)時收到組呼消息，不做處理；被叫端掛機不通知主叫端結(jié)束通話。