基于Speex语音引擎的VoIP系统设计与实现  

Design and implementation of VoIP system based on Speex codec

作　　者：谢晓钢[1] 蔡骏[1] 陈奇川[1] 欧建林[1] 

机构地区：[1]厦门大学计算机科学系,福建厦门361005

出　　处：《计算机应用研究》2007年第12期320-323,共4页Application Research of Computers

基　　金：厦门大学"985工程"二期信息创新平台资助项目(0000-X07204)

摘　　要：论述了一套基于Speex语音引擎和RTP的VoIP系统设计和开发,介绍了该系统服务器端和客户机端的软件实现。该系统具有点对点通信、算法延时小、丢包补偿和延时补偿性能好等特点,并具有多方通话功能。性能对比实验表明,该系统的通话质量优于几套流行的开源VoIP软件,能满足实际应用的要求。The paper proposed the design and software implementation of a VoIP system, both on the server side and the client side. The VoIP system was written with Speex and RTP, and deployed for Windows platform. It featured peer-to-peer communication, low algorithmic latency, low packet loss, and effective compensation for jitter. It had a full range of practical functions, such as conference calling. Speech quality assessment has been done by comparing and contrasting the performance of the system to those of several other VoIP systems. Testing results show that the system has a satisfactory voice quality for practical use.

关 键 词：基于IP网络的语音传输 Speex 实时传输协议 多方通话 

分 类 号：TP393[自动化与计算机技术—计算机应用技术]