Voice Communications over Tandem Heterogeneous Networks

串联异构网络上的语音通信

• 批准号: 0429884
• 负责人: Jerry Gibson
• 金额: --
• 依托单位: University of California-Santa Barbara
• 依托单位国家: 美国
• 项目类别: Continuing grant
• 财政年份: 2004
• 资助国家: 美国
• 起止时间: 2004-09-01 至 2008-08-31
• 项目状态: 已结题
• 来源: https://www.nsf.gov/awardsearch/showAward?AWD_ID=0429884&HistoricalAwards=false
• 关键词: Voice; Communications; over; Tandem; Heterogeneous

Voice is the method of choice for real time communications. Voice is so important to human communications that we have constructed entire networks centered around voice, namely, the public switched telephone network (PSTN) and the analog/digital cellular networks. With the emergence of voice over the Internet Protocol (VoIP) and voice over wireless local area networks (Wi-Fi), a future voice call might consist of a digital cellular user communicating through a VoIP backbone network with a voice over Wi-Fi user, resulting in the connection of a digital cellular network through a VoIP network into a Wi-Fi link. Although the digital cellular networks utilize relatively sophisticated error detection and correction, unequal error protection, and error concealment, the error control, packet loss mechanisms, and packet loss concealment methods for VoIP and voice over Wi-Fi are relatively primitive. Furthermore, since each of these networks has different channel behaviors and may use different voice codecs and different network protocols, acceptable, bandwidth efficient voice service may not be achievable over such a connection. This research involves voice communications over tandem heterogeneous networks and voice over Wi-Fi links. We examine (1) improved voice codec tandem performance, by developing speech-mode-adaptive postfilters and perceptual distortion measures, by removing the postfilter and inserting additional filtering within the analysis-by-synthesis loop, and by shaping the excitation within the analysis-by-synthesis loop; and (2) improved packet loss concealment and reduced latency, by jointly designing packet loss concealment methods and protocols for wireless LANs and by developing packet loss smoothing algorithms to use "late" packets to improve later reconstruction. The evaluation of the end-to-end network performance for voice communications uses the NS2 network simulator and voice quality metrics.

语音是真实的时间通信的首选方法。语音对于人类通信是如此重要，以至于我们已经构建了以语音为中心的整个网络，即公共交换电话网（PSTN）和模拟/数字蜂窝网络。随着互联网协议语音（VoIP）和无线局域网语音（Wi-Fi）的出现，未来的语音呼叫可能包括数字蜂窝用户通过VoIP骨干网络与Wi-Fi语音用户通信，从而导致数字蜂窝网络通过VoIP网络连接到Wi-Fi链路。虽然数字蜂窝网络利用相对复杂的错误检测和校正、不等错误保护和错误隐藏，但是用于VoIP和Wi-Fi语音的错误控制、分组丢失机制和分组丢失隐藏方法相对原始。此外，由于这些网络中的每一个具有不同的信道行为，并且可以使用不同的语音编解码器和不同的网络协议，因此在这样的连接上可能无法实现可接受的、带宽有效的语音服务。 这项研究涉及串联异构网络上的语音通信和Wi-Fi链路上的语音。我们研究（1）改进的语音编解码器串联性能，通过开发语音模式自适应后置滤波器和感知失真测量，通过去除后置滤波器并在合成分析环路内插入附加滤波，以及通过在合成分析环路内成形激励;以及（2）改进的分组丢失隐藏和减少的等待时间，通过联合设计用于无线局域网的分组丢失隐藏方法和协议，以及通过开发分组丢失平滑算法以使用“迟来的”分组来改进稍后的重构。语音通信的端到端网络性能评估使用NS 2网络模拟器和语音质量指标。