...
  • 主题
高级搜索  >
历史搜索 清空历史
首页> 外文期刊>Electronics and Electrical Engineering >Comparative Analysis of the Performance of Different Codecs in a Live VoIP Network using SIP Protocol
【24h】

Comparative Analysis of the Performance of Different Codecs in a Live VoIP Network using SIP Protocol

机译：使用SIP协议的实时VoIP网络中不同编解码器性能的比较分析

获取原文
获取原文并翻译 | 示例
           
页面导航
  • 摘要
  • 著录项
  • 相似文献
  • 相关主题

摘要

Voice over IP (VoIP) is the ability to transmit speech over packet-switched IP networks ant it is the common term for telephone service over the Internet. If the user has a satisfactory quality Internet connection voice signals can be delivered over an Internet connection instead of local phone company. Establishing a phone call using VoIP will create a digital signal from the analog input, place those digital signals into packets with source and destination network addresses and finally send the information over the Internet or internal company IP networks, thus bypassing the need for the PSTN lines. However, it is important to note that the source and destination terminals must support the codec for the correct encoding and decoding. Today, most popular VoIP standard is Session Initiation Protocol (SIP). This standard provides direct call between VoIP (SIP) terminals, or the use of media gateways, which can be used to negotiate connections between TDM and/or other endpoints. In VoIP, quality of service (QoS) simply means being able to listen and speak in a clear and continuous way, without unwanted noise or other distortions [1]. QoS depends on the technical parameters: delay, jitter and packet loss. In VoIP a delay of 150ms is acceptable, while a higher value results in degradation of voice quality, which becomes unacceptable at values higher than 300 ms [2].
机译：IP语音（VoIP）是通过分组交换IP网络传输语音的能力，这是Internet上电话服务的通用术语。如果用户具有令人满意的Internet连接质量，则可以通过Internet连接而不是本地电话公司传递语音信号。使用VoIP建立电话将通过模拟输入创建数字信号，将这些数字信号放入具有源和目标网络地址的数据包中，最后通过Internet或内部公司IP网络发送信息，从而无需PSTN线路。但是，请务必注意，源终端和目标终端必须支持编解码器才能进行正确的编码和解码。如今，最流行的VoIP标准是会话启动协议（SIP）。该标准提供了VoIP（SIP）终端之间的直接呼叫，或使用了媒体网关，可用于协商TDM和/或其他端点之间的连接。在VoIP中，服务质量（QoS）只是意味着能够以清晰连续的方式收听和讲话，而不会产生不必要的噪声或其他失真[1]。 QoS取决于技术参数：延迟，抖动和数据包丢失。在VoIP中，150ms的延迟是可以接受的，而较高的值会导致语音质量下降，而在高于300ms的值上则变得不可接受[2]。

著录项

相似文献

  • 外文文献
  • 中文文献
  • 专利
获取原文

客服邮箱：kefu@zhangqiaokeyan.com

京公网安备：11010802029741号 ICP备案号：京ICP备15016152号-6 六维联合信息科技 (北京) 有限公司©版权所有

  • 客服微信

  • 服务号