基于编解码的SIP和IAX2 VVoIP协议的QoS性能评估

N. Edan, Ali Al-Sherbaz, Scott J. Turner, S. Ajit
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引用次数: 12

摘要

服务质量(QoS)和基于互联网协议(IP)的多媒体传输的评估一直是人们关注的焦点。本文的主要目标是评估QoS在实际实现中的性能,以及基于基于Asterisk PBX服务器(一个开源通信平台)的SIP和IAX2协议的IP方案上适当的视频和语音(CODECS)的比较。该网络是在各种专用网络中实现的,例如Wire和Wi-Fi,它们作为放置各种呼叫的本地交换。基于带宽、抖动和延迟等QoS参数对VVoIP (video and voice over IP)网络中不同编解码器的质量进行了评价。测量中使用的VVoIP编解码器有:G.711 (ulaw&alaw)、GSM、G.722、Speex、H.263、H.264、H.261和H.263P。对性能结果的评估将使网络规划者和多媒体协议开发人员有机会选择用于VVoIP性能增强的编解码器,从而提高客户满意度。
本文章由计算机程序翻译,如有差异,请以英文原文为准。
Performance evaluation of QoS using SIP & IAX2 VVoIP protocols with CODECS
There has been a strong focus on the evaluation of Quality of Service (QoS) and multimedia transmission over protocols of internet protocol (IP). The main goal of this paper is to evaluate the performance of QoS in real implementation, also a comparison of an appropriate video and voice (CODECS) over IP schemes depending on the SIP and IAX2 protocols based on the Asterisk PBX server(s), an open source communication platform. The network was implemented within various private networks, such as Wire and Wi-Fi, which serves as a local exchange for placing various calls. Quality has been evaluated based on some QoS parameters such as bandwidth, jitter and delay to investigate the performance of different codecs in video and voice over IP (VVoIP) network. The VVoIP codecs used in the measurements are: G.711 (ulaw&alaw), GSM, G.722, Speex, H.263, H.264, H.261 and H.263P. Evaluation of performance results will give network Planners and Multimedia protocols developers an opportunity to select the codec for VVoIP performance enhancement, which can lead to improved customer satisfaction.
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