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研究生: 羅堅誌
CHIEN-CHIH LO
論文名稱: 利用Linphone設計與實作一個在異質網路下的語音即按即說廣播服務
Design and Implementation of a Voice Push-to-Talk Service using Linphone in a Heterogeneous Network Environment
指導教授: 呂政修
Jenq-Shiou Leu
口試委員: 孫敏德
Min-Te Sun
阮聖彰
Shanq-Jang Ruan
鄭瑞光
Ray-Guang Cheng
學位類別: 碩士
Master
系所名稱: 電資學院 - 電子工程系
Department of Electronic and Computer Engineering
論文出版年: 2012
畢業學年度: 100
語文別: 中文
論文頁數: 43
中文關鍵詞: 異質網路廣播系統排程嵌入式系統
外文關鍵詞: process scheduling
相關次數: 點閱:201下載:3
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數位化的行動通信技術逐漸使人群之間的交流更為頻繁,在過去Open Mobile Alliance(OMA)發展出即按即說(Push-to-talk)的行動數據服務,即按即說的使用方式類似傳統AM/FM類比式無線電對講機,不過更方便且成本更低,並能將語音同時清晰的廣播給許多人聽到。雖然目前付費語音數據服務已經普及,但是近幾年使用資料網路來傳送即時語音資料的技術VoIP(Voice over IP),已經應用在人們的生活中,讓更多人期待能用低成本的設備與技術來解決遠距離語音通信的問題,來滿足一對一或一對多人的通話需求。新一代即按即說服務使用VoIP技術,結合不同性質的網路,將不再局限於一個網路之間互通,可將語音通話等多媒體服務擴展到更大的區域,依特性來延伸各種多元化服務,並可與公眾交換電話網路PSTN(Public Switched Telephone Network)及公眾陸地行動網路PLMN(Public Land Mobile Network)內的裝置相互溝通廣播。本文用開放原始碼軟體Linphone在Linux系統下實現語音即按說廣播平台,並且針對語音廣播服務在異質網路做探討與分析,討論如何在相同的軟硬體及網路條件下,有效的運用系統去控制發送的排程及調整接收裝置上緩衝區,對語音傳輸服務做優化,並得到更清晰的語音品質。


The evolving mobile technologies have made communications between people easier. Such as Push-to-talk over Cellular(PoC) developed by Open Mobile Alliance in 1990s, similar to analog AM/FM handheld walkie-talkie, but more convenient and cheaper, and can converse over half-duplex communication lines. And besides the wide spread cellular voice services, there are technologies like Voice over Internet Protocol (VoIP) using data network to transmit voice for long distance communication at lower cost and fulfilling the increasing demands, either one-on-one of one-to-many. The next generation of Push-to-talk technologies using VoIP, can provide voice communication and multimedia services over heterogeneous networks, such as WiFi, 3G and LAN, and can connect with devices from Public Switched Telephone Network(PSTN) and Public Land Mobile Network(PLMN). In this study, we use open sources software Linphone with Linux system to implement a Push-to-talk platform in order to increase the voice quality and decrease transition delay by optimizing OS scheduling and device buffer, and discuss its performance within homogenous and heterogeneous networks.

論文摘要 I ABSTRACT II 誌謝 III 目錄 IV 圖片索引 VI 第 1 章 緒論 1 1.1 研究動機 1 1.2 研究目的與方法 2 1.3 論文架構 3 第 2 章 背景知識探討 4 2.1 VoIP(voice over IP) 4 2.1.1 Session Initiation Protocol (SIP) 5 2.1.2 即時傳輸協定 Real-time Transport Protocol (RTP) 6 2.1.3 G.711語音壓縮演算法 8 2.1.4 隨按即說(Push-to-talk, PTT) 9 2.2 Linux系統架構 11 2.3 Linux系統排程應用 14 第 3 章 廣播服務架構 16 3.1 語音即按即說廣播服務架構 16 3.2 Linphone 軟體 18 3.3 廣播發送端系統架構 20 3.3.1 Network Stack Buffer 21 3.3.2 Advanced Linux Sound Architecture 21 3.4 廣播接收端系統架構 22 第 4 章 語音品質評估方法 23 4.1 實驗設備介紹 23 4.2 語音品質測量工具 24 4.3 VoIP 語音品質評估標準 25 第 5 章 廣播語音品質評估與效能分析 27 5.1 單點廣播探討 27 5.1.1 G.711 A-law 基本語音品質評估 27 5.1.2 VoIP Jitter Buffer與Audio Buffer 29 5.1.3 Jitter Buffer與語音品質關係之實驗 30 5.1.4 Audio Buffer與語音品質關係之實驗 31 5.1.5 Jitter Buffer與固定Audio Buffer 影響之實驗 32 5.2 多點廣播探討 33 5.2.1 同質區域網路廣播 33 5.2.2 異質廣域網路廣播 36 5.2.3 異質跨區域網路廣播 39 5.2.4 廣播即時性的模擬與分析 41 第 6 章 結論 43 參考文獻 44

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