本文選題：視頻會(huì)議 + SIP��； 參考：《哈爾濱工業(yè)大學(xué)》2016年碩士論文

【摘要】：視頻會(huì)議使處于兩地的人可以更加方便的交流溝通,提高了人類社會(huì)生產(chǎn)的協(xié)作效率,在人類社會(huì)生產(chǎn)中逐漸占據(jù)了重要地位。Internet普及和網(wǎng)絡(luò)基礎(chǔ)設(shè)施全面升級(jí),使得基于Internet網(wǎng)絡(luò)的簡(jiǎn)單、方便且廉價(jià)視頻會(huì)議系統(tǒng)成為可能。但是在Internet中帶寬波動(dòng)較大,網(wǎng)絡(luò)通信質(zhì)量不可控。因此如何保障Internet中視頻會(huì)議傳輸QoS(Quality of Service)問(wèn)題便成一個(gè)重要的研究課題。本課題嘗試從實(shí)時(shí)視頻碼流的碼率控制角度來(lái)提升視頻會(huì)議在不穩(wěn)定的網(wǎng)絡(luò)中的傳輸QoS。核心內(nèi)容是SIP(Session Initiation Protocol)視頻會(huì)議的基礎(chǔ)上加入自適應(yīng)采集(視頻采集)與動(dòng)態(tài)碼流(視頻碼流)提取技術(shù)。主要研究?jī)?nèi)容包括SIP視頻會(huì)議原理和相關(guān)碼流技術(shù)分析,基于RTP/RTCP反饋機(jī)制改進(jìn)的AIMD視頻發(fā)送碼率控制模型的建立,設(shè)計(jì)實(shí)現(xiàn)SIP視頻會(huì)議系統(tǒng)。研究分析SIP視頻會(huì)議協(xié)議結(jié)構(gòu)和SIP視頻會(huì)議視頻碼流技術(shù)。從理論上全面分析一個(gè)基于SIP的視頻會(huì)議的交互流程和架構(gòu)模型。研究分析H.264視頻編碼結(jié)構(gòu)和相應(yīng)碼率控制、視頻轉(zhuǎn)碼方案。根據(jù)視頻會(huì)議場(chǎng)景選擇碼率控制方案并提出一種基于GOP的全編全解視頻轉(zhuǎn)碼方案。研究分析基于RTP/RTCP視頻傳輸?shù)木W(wǎng)絡(luò)擁塞控制方案,將傳統(tǒng)擁塞控制算法AIMD加以改進(jìn)應(yīng)用于自適應(yīng)視頻采集和動(dòng)態(tài)碼流提取的碼率決策部分。其中利用RTP/RTCP反饋控制機(jī)制獲取丟包率、回環(huán)時(shí)間和抖動(dòng)參數(shù)。設(shè)計(jì)一種雙滑動(dòng)窗口機(jī)制用于丟包率的預(yù)測(cè)。對(duì)回環(huán)進(jìn)行低通濾波平滑。根據(jù)丟包率、回環(huán)時(shí)間和抖動(dòng)參數(shù)作為AIMD算法中視頻碼率決策的參數(shù)。在此基礎(chǔ)上設(shè)計(jì)自適應(yīng)視頻采集和動(dòng)態(tài)視頻碼流提取模型。在OpenMCU、OpenSips等開源項(xiàng)目基礎(chǔ)上進(jìn)行二次開發(fā),設(shè)計(jì)并實(shí)現(xiàn)加入自適應(yīng)視頻采集與動(dòng)態(tài)視頻碼流提取技術(shù)的SIP視頻會(huì)議系統(tǒng)。主要是在SIP UA端加入自適應(yīng)視頻采集模塊,在轉(zhuǎn)發(fā)服務(wù)器端添加動(dòng)態(tài)視頻碼流提取模塊。該系統(tǒng)的特點(diǎn)是有更高的網(wǎng)絡(luò)敏感性,能根據(jù)網(wǎng)絡(luò)的差異動(dòng)態(tài)調(diào)整系統(tǒng)中傳輸?shù)囊曨l的碼率,能有效的提高視頻會(huì)議的視頻通信質(zhì)量。本課題充分考慮視頻會(huì)議的應(yīng)用場(chǎng)景以及傳輸碼率的預(yù)測(cè)問(wèn)題。在基于AIMD算法基礎(chǔ)上加以改進(jìn),設(shè)計(jì)出整套能實(shí)現(xiàn)自適應(yīng)視頻采集和動(dòng)態(tài)碼流提取的網(wǎng)絡(luò)敏感性視頻會(huì)議系統(tǒng),有效提升了系統(tǒng)的QoS。
[Abstract]:Video conference makes it more convenient for people in both places to communicate, improve the efficiency of human social production, and gradually occupy an important position in the production of human society. It makes the Internet network based on the simple, convenient and cheap video conferencing system possible. However, the bandwidth fluctuates greatly in Internet, and the network communication quality is not controllable. Therefore, how to guarantee the QoS(Quality of Service in Internet becomes an important research topic. This paper attempts to improve the QoS of video conferencing in unstable networks from the point of view of bit rate control of real-time video stream. On the basis of SIP(Session Initiation protocol video conference, adaptive acquisition (video capture) and dynamic bit stream (video bitstream) extraction are introduced. The main research contents include the principle of SIP video conference and the analysis of related bitstream technology, the establishment of AIMD video transmission rate control model based on RTP/RTCP feedback mechanism, and the design and implementation of SIP video conference system. The structure of SIP video conference protocol and the video bitstream technology of SIP video conference are studied and analyzed. The interaction flow and architecture model of a video conference based on SIP are analyzed in theory. The H.264 video coding structure and the corresponding rate control, video transcoding scheme are studied and analyzed. According to the video conference scene, the rate control scheme is selected and a fully decomposed video transcoding scheme based on GOP is proposed. The network congestion control scheme based on RTP/RTCP video transmission is studied and analyzed. The traditional congestion control algorithm AIMD is improved and applied to the bit rate decision part of adaptive video capture and dynamic bit stream extraction. The RTP/RTCP feedback control mechanism is used to obtain the packet loss rate, loop return time and jitter parameters. A double sliding window mechanism is designed to predict packet loss rate. The return loop is smoothed by low pass filtering. The packet loss rate, loop time and jitter parameters are used as the parameters of video rate decision in AIMD algorithm. On this basis, the adaptive video capture and dynamic video stream extraction model are designed. On the basis of OpenMCU OpenSips and other open source projects, this paper designs and implements a SIP video conference system with adaptive video capture and dynamic video stream extraction technology. It mainly adds adaptive video capture module in SIP UA terminal and dynamic video stream extraction module in forwarding server. The characteristic of this system is that it has higher network sensitivity, can dynamically adjust the bitrate of video transmitted in the system according to the network difference, and can effectively improve the video communication quality of video conference. In this paper, the application scene of video conference and the prediction of transmission bit rate are fully considered. Based on the improvement of AIMD algorithm, a network sensitive video conference system which can realize adaptive video capture and dynamic bit stream extraction is designed, which can effectively improve the QoS of the system.
【學(xué)位授予單位】：哈爾濱工業(yè)大學(xué)
【學(xué)位級(jí)別】：碩士
【學(xué)位授予年份】：2016
【分類號(hào)】：TN948.63

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