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基于Android的網絡電話軟件設計

發(fā)布時間:2018-03-28 19:17

  本文選題:語音編碼 切入點:電話信令 出處:《哈爾濱工業(yè)大學》2014年碩士論文


【摘要】:隨著科學技術的進步,電子產品的成本和價格不斷下調,智能終端已經深入了人們的生活,成為了日常工作學習中不可缺少的工具,而Android系統(tǒng)依托開放共贏的理念以系統(tǒng)開源的形式快速地占領了智能終端市場的絕大份額,地位越來越重要。針對Android系統(tǒng)良好的應用前景,本課題將Android平臺作為實現平臺。近些年,電話信令技術、網絡技術、多媒體技術、語音編解碼技術、網絡穿透技術不斷進步,同時互聯網應用迅速興起,網絡電話在兼有良好技術基礎及廣闊的市場應用前景下得到了快速的發(fā)展,在工作生活中越來越多的開始取代傳統(tǒng)電信網絡通信。本文的研究內容便為基于Android的網絡語音通話軟件設計。開發(fā)出具有實用價值的通信軟件為實驗室后續(xù)在智能家居領域的進一步拓展具有實質性的意義。系統(tǒng)由客戶端和服務器兩部分組成?蛻舳酥饕ㄕZ音信號處理及傳輸、電話信令SIP、NAT網絡穿透三個模塊,服務器主要完成用戶上線注冊、用戶之間尋址以及用戶注冊的功能。本文工作的主要內容如下:設計實現了語音信號處理及傳輸過程。包括語音的采集、編碼、發(fā)送、接收、解碼及播放六個部分,每部分使用獨立的線程完成。首先調用Android平臺對語音信號進行采集,將采集到的語音信號流交給ILBC編碼庫進行編碼,對編碼完成的語音數據即時交給發(fā)送線程使用socket技術傳輸給指定地址,接收線程則通過socket技術監(jiān)聽指定端口號接收語音數據,并將接收的數據即時交給解碼線程進行解碼,播放線程將已解碼數據即時播放。設計實現了SIP信令客戶端。使用Android系統(tǒng)自帶的API進行SIP信令客戶端的開發(fā),客戶端主要實現有用戶信息注冊及保存、通話的發(fā)起、來電接收的基本功能。設計實現了SIP基本功能服務器。使用開源服務器代碼架設了用于SIP通話的服務器,基于該服務器開發(fā)了用戶管理的軟件界面,實現了對用戶簡易注冊及刪除的基本管理功能。經過實際測試,良好的實現了語音信號的處理及傳輸過程,在局域網之內語音通話質量清晰且延時感不明顯,SIP客戶端實現了SIP服務器的登錄,能夠給指定SIP用戶建立SIP通話,服務器實現了對用戶的注冊及注銷過程。
[Abstract]:With the progress of science and technology, the cost and price of electronic products have been continuously reduced. Intelligent terminals have become an indispensable tool in daily work and learning. The Android system, relying on the concept of open and win-win, has occupied the vast majority of the intelligent terminal market rapidly in the form of open source system, and its status is becoming more and more important. In view of the good application prospects of Android system, In recent years, the telephone signaling technology, network technology, multimedia technology, voice coding and decoding technology, network penetration technology, the rapid rise of Internet applications, Internet telephony has developed rapidly with both good technical foundation and broad market application prospects. More and more people begin to replace the traditional telecommunication network communication in the work life. The research content of this paper is the design of the network voice communication software based on Android. The system consists of two parts: the client and the server. The client mainly includes voice signal processing and transmission. Telephone signaling SIPN Nat network penetrates three modules, the server mainly completes the subscriber on-line registration. The main contents of this paper are as follows: the design and implementation of speech signal processing and transmission process, including voice acquisition, coding, sending, receiving, decoding and playing six parts, Each part is completed by independent thread. Firstly, the Android platform is called to collect the speech signal, and the stream of the collected speech signal is given to the ILBC coding library for coding. On the other hand, the encoded voice data is immediately transferred to the sending thread using socket technology to receive the specified address, while the receiving thread listens to the specified port number to receive the voice data through socket technology, and the received data is immediately handed over to the decoding thread for decoding. The playback thread plays the decoded data instantly. The SIP signaling client is designed and implemented. The SIP signaling client is developed by using the API which comes with the Android system. The client mainly realizes the registration and saving of the user information and the initiation of the call. The basic function of receiving calls. The basic function server of SIP is designed and implemented. The server for SIP calls is set up by using open source server code, and the software interface of user management is developed based on this server. The basic management function of simple registration and deletion of users is realized. After practical test, the process of speech signal processing and transmission is well realized. In the LAN, the voice call quality is clear and the delay sense is not obvious. The client realizes the login of the SIP server, which can set up the SIP call for the designated SIP user, and the server realizes the process of registering and deregistration the user.
【學位授予單位】:哈爾濱工業(yè)大學
【學位級別】:碩士
【學位授予年份】:2014
【分類號】:TN916.5;TP311.52

【參考文獻】

相關期刊論文 前3條

1 謝少武;從PSTN電話向IP電話的演進[J];廣東通信技術;2001年S1期

2 倪維楨;語音編碼綜述[J];數字通信;1998年02期

3 李志軍,華繼釗;中低速率語音波形編碼[J];現代電子技術;2002年01期



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