本文選題：會話初始協(xié)議 + 實時通信協(xié)議　； 參考：《電子科技大學(xué)》2014年碩士論文

【摘要】：隨著網(wǎng)絡(luò)技術(shù)朝著數(shù)字化、綜合化、智能化的方向飛速發(fā)展,網(wǎng)絡(luò)已經(jīng)深深融入了人們的日常生活、工作中,通過IP網(wǎng)絡(luò)提供語音、視頻、傳真、數(shù)據(jù)等多媒體綜合業(yè)務(wù)已成為實時通信發(fā)展的趨勢和目標(biāo)。這種利用IP網(wǎng)絡(luò)為人們提供語音、數(shù)據(jù)信息交換的技術(shù),伴隨著網(wǎng)絡(luò)的進(jìn)一步普及,已逐漸成為人們相互溝通、聯(lián)絡(luò)的重要手段之一。然而為IP終端設(shè)備提供服務(wù)的IP網(wǎng)絡(luò)存在著各種各樣的安全隱患,網(wǎng)絡(luò)上傳輸?shù)腎P數(shù)據(jù)包極易為黑客所截獲,從而造成關(guān)鍵信息的丟失或被竊聽。本文的目標(biāo)是設(shè)計并實現(xiàn)一臺以系統(tǒng)級芯片(SoC)作為處理器的網(wǎng)絡(luò)電話(VoIP)原理樣機,基于該平臺提供采用SIP協(xié)議的VoIP語音通信功能,并針對網(wǎng)絡(luò)電話通信過程中的實時媒體數(shù)據(jù)RTP報文進(jìn)行加解密處理,通過IP網(wǎng)絡(luò)為用戶提供經(jīng)過了加密保護(hù)的VoIP語音通信。本文首先針對VoIP通信系統(tǒng)中需要應(yīng)用到的相關(guān)技術(shù)(如語音編解碼、呼叫控制協(xié)議及加解密技術(shù)等)進(jìn)行了分析和研究。對當(dāng)前主流的幾種呼叫控制協(xié)議和加解密算法進(jìn)行分析與比較,確定采用會話初始協(xié)議(SIP)作為安全終端的基本信令協(xié)議,以高級加密標(biāo)準(zhǔn)(AES)作為實時通信協(xié)議(RTP)媒體報文的加密算法,再針對SIP協(xié)議和AES算法的基本流程、認(rèn)證機制和編碼規(guī)則進(jìn)行研究。以此為基礎(chǔ)采用標(biāo)準(zhǔn)的SoC芯片為核心設(shè)計并實現(xiàn)安全VoIP語音通信終端。設(shè)計實現(xiàn)的工作分為軟、硬件兩部分:軟件方面主要包括SIP協(xié)議棧的移植,驅(qū)動開發(fā),身份認(rèn)證機制的設(shè)計和調(diào)試;硬件方面則是對主要芯片的選型,原理框圖的設(shè)計,外圍電路的搭建以及AES算法的FPGA實現(xiàn)。本文詳細(xì)介紹了硬件電路和軟件代碼的設(shè)計、調(diào)試過程,并從通話效果、線路抓包等方式針對最終實現(xiàn)的基于SIP協(xié)議的VoIP安全終端進(jìn)行了測試。最后通過在內(nèi)部的測試網(wǎng)絡(luò)中搭建一套開源SIP服務(wù)器,將基于SIP的安全終端接入測試網(wǎng)絡(luò)中,對安全終端的VoIP語音通信功能進(jìn)行測試,并通過網(wǎng)絡(luò)抓包方式對VoIP通信的安全性進(jìn)行了驗證。
[Abstract]:With the rapid development of network technology in the direction of digitalization, integration and intelligence, the network has been deeply integrated into people's daily life and work, providing voice, video, fax through IP network. Multimedia integrated services such as data has become the trend and goal of real-time communication. With the further popularization of the network, the technology of using IP network to provide people with voice and data information exchange has gradually become one of the important means for people to communicate and communicate with each other. However, there are various security risks in IP networks serving IP terminal devices. The IP packets transmitted on the network are easily intercepted by hackers, resulting in the loss of critical information or eavesdropping. The goal of this paper is to design and implement a prototype of VoIP based on system level chip (SoC), and provide VoIP voice communication function based on SIP protocol. The encryption and decryption of real-time media data RTP message in the process of network telephone communication is carried out, and the VoIP voice communication is provided through IP network. This paper firstly analyzes and studies the relevant technologies (such as speech coding and decoding, call control protocol and encryption and decryption) that need to be applied in VoIP communication system. This paper analyzes and compares several popular call control protocols and encryption and decryption algorithms, and determines that the session initiation protocol (SIP) is adopted as the basic signaling protocol for secure terminals. The advanced encryption standard (AES) is used as the encryption algorithm of real-time communication protocols (RTPs). The basic flow, authentication mechanism and coding rules of SIP protocol and AES algorithm are studied. On this basis, the standard SoC chip is used as the core to design and implement the secure VoIP voice communication terminal. The work of design and implementation is divided into two parts: the software mainly includes the transplantation of SIP protocol stack, the development of driver, the design and debugging of identity authentication mechanism, the selection of main chips and the design of schematic block diagram in hardware. The construction of peripheral circuit and the FPGA implementation of AES algorithm. This paper introduces the design and debugging process of hardware circuit and software code in detail, and tests the VoIP security terminal based on SIP protocol from the aspects of call effect, line capture packet and so on. Finally, by building an open source SIP server in the internal test network, the secure terminal based on SIP is connected to the test network to test the VoIP voice communication function of the security terminal. The security of VoIP communication is verified by network packet capture.
【學(xué)位授予單位】：電子科技大學(xué)
【學(xué)位級別】：碩士
【學(xué)位授予年份】：2014
【分類號】：TN915.08

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本文編號：1865646