標題: 一個同儕網路上基於會議初始協定的語音會議系統
A SIP-based audio conferencing system on P2P network
作者: 呂學增
Shiue-Tzeng Lu
陳耀宗
Yaw-Chung Chen
資訊科學與工程研究所
關鍵字: 網路電話;會議初始協定;同儕網路;語音會議;VoIP;SIP;P2P;audio conferencing
公開日期: 2007
摘要: 隨著網際網路的頻寬不斷地擴展增加,一些要求較高網路頻□的服務愈來愈被廣泛使用,像是網路電話或視訊服務。而在最近十年內,發展了另一種熱門的網路服務架構:同儕式網路(Peer-to-Peer Network),簡稱為P2P網路。在本論文中,我們試圖結合此一網路架構到我們所提出的語音會議系統。 網路電話目前是網際網路上的一種殺手級應用,並且有許多人正享受著它的便利與低廉。以目前網路頻寬而言,一對一網路電話的雙向連線可以達到優良的服務品質(Quality of Service,QoS),然而在更進一步的語音會議來說,三對三、甚至四對四之多方通話連線雖然還是可以達到一定水準的服務品質,但是隨著參與人數增加,語音會議的聲音品質卻會急劇地下降。另一方面,語音會議參與者從要加入會議到真正開始接收他人聲音及自己說話,其中所花費的時間也有可能會過於冗長。因此加速會議建立及優化語音傳送品質,將是一大研究重點。 本篇論文試圖針對語音會議初始建立與P2P網路做結合以及在語音會議進行中媒體串流的控制,做了一個整體性地改進。著重於語音會議建立的加速化及減輕端點使用者網路頻寬瓶頸,使得一個語音會議的參與者可以更快速地開始接聽及說話,而不會有太多的缺點在語音服務品質上。 在本篇論文內容中,我們介紹一些語音會議相關模式及協定,並且提出一個結合多項優點和P2P網路而成的快速語音會議建立機制,另外改進一個相關媒體串流控制協定,進而達到快速進行語音會議和優化語音服務品質的目的。最後利用模擬程式NS-2來驗證我們提出的語音會議系統。
As the Internet steadily broadens the bandwidth scope, some application services such as Internet phone (Voice over Internet Protocol) and video services that required high bandwidth are deployed extensively. In recent decade, a popular architecture of Internet service has been developed: Peer-to-Peer network (P2P network). We will try to integrate the P2P network into our proposed SIP based audio conference system. Nowadays VoIP can be considered as a killer application on the Internet, and there are quite a few users enjoying its ease, comfort and affordable cost. For present network bandwidth, end-to-end bandwidth of the two-way Internet phone connection is able to achieve good quality of service (Quality of Service, QoS). Although the multiparty connection can still reach a certain level of service quality in further audio conference, with the increase in the number of participants, the voice quality of service would quickly decline. On the other hand, the audio conference participants have to wait for quite a long duration from joining the conference to the really speaking and listening. Therefore, acceleration of the establishment and optimization of voice conferencing quality will be our major focus in this thesis. In this thesis, we make an integrated improvement regarding the initial establishment working with the P2P network and the media streaming control protocol for an audio conference. Focusing on the acceleration of the audio conference establishment and reducing the network bandwidth bottleneck of end users make the participants able to quickly start the conversation without too much quality degradation in the voice conferencing services. We proposed a rapid audio conference establishment mechanism that combines advantages of related works on the P2P network and improvement of media streaming control protocol to further achieve a rapid audio conference and good voice quality of service. NS-2 simulation shows that our proposed system features significant improvement on the QoS of voice conferencing.
URI: http://140.113.39.130/cdrfb3/record/nctu/#GT009555554
http://hdl.handle.net/11536/39505
Appears in Collections:Thesis


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