以SIP為基礎之點對點多人會議系統設計與實作

Design and Implementation of SIP-based Peer-to-peer Multiparty Conferencing System

童裕淇(Yu-chi Tung)

指導教授 ： 王永鐘

國立臺北科技大學/電資學院/電機工程系所/碩士(2008年)

摘要

本論文實作以SIP為基礎之點對點多人會議系統，應用於即時網路語音會議。本系統分為伺服器端與用戶端兩大部份。在伺服器端部分，採用 Linux Red Hat 9.0 平台，安裝VOCAL(Vovida Open Communication Application Library) server 作為伺服器，此伺服器包含認證、代理及通話管理等之多功能型伺服器。 用戶端部分，本論文提出以SIP為基礎之分散式多人會議，讓用戶端進行多方會談，啟動多方通話方式，採用SIP協定啟始多人會議連線的管理，包含初始化會議、會議成員的加入與離開和終止多媒體通訊的功能。在混音部分，多人語音混音機制主要是讓用戶端採用分散式接收語音封包，自行負責混音，進而達到多方會談，有別於傳統之多點控制單元(Multipoint Control Unit ,MCU)管理多點的通話。還能避免在即按即說（Push To Talk）模式下，用戶端獨佔語音資源的情況發生。 本論文所設計的系統，經由實作測試，證實在多人同時連線之情況下，仍能保持穩定的反應時間，同時提供清晰流暢的語音品質。

關鍵字

SIP協定 ； 分散式多人會議 ； 多人語音混音

並列摘要

We have implemented SIP-based peer-to-peer multiparty conferencing system, applies in the immediate network conference for voice. The system is divided into the server and client side. In server side, the VOCAL server (Vovida Open Communication Application Library) is installed over Linux Red Hat 9.0 platform. The server includes authentication, agents and call management, and other type of multi-function server.
On the other hand, we proposed SIP-based distributed multiparty conferencing for the client to talk multilaterally, the proposed system start multi-way call, by SIP-based over the management of the connection, including the initial Conferencing, the joining council members to, the leaving and termination of multimedia communicational functions. In the mixing, multiparty voice mixing mechanism, used for client distributed receive voice packets, will be responsible for mixing, then to multiparty conferencing. It’s different from the traditional control unit of MCU (Multipoint Control Unit), which is managed the multi-point call. Also happened that to avoid “Push to Talk” mode, the client’s exclusive voice of resources has been occupied.
Our system, through the real testing in network environment, verify that it can still keep steady response time under multiple connections, and furthermore provide clear and smooth speech quality.

並列關鍵字

Session Initiation Protocol(SIP) ； Distributed Multiparty Conferencing ； Multiparty Speech Mixing

參考文獻

[17] 劉至中，分散式多人網路語音會議系統設計與實作，碩士論文，國立台北科技大學，電機工程系所，台北，2007。

[1] P. Mehta and S.Udani, “Voice over IP,” IEEE Potential, 2001, pp. 36-38.

[2] ITU-T Recommendation H.323, “Packet-based Multimedia Communication System,” International Telecommunication Union, 2006.

[3] M. Handley, H. Schulzrinne, E. Schooler and J. Rosenberg, “SIP: Session Initiation Protocol,” RFC 2543, 1999.

[6] F. Andreasen and B. Foster, “Media Gateway Control Protocol (MGCP),” RFC 3435, 2003.