基于OPNET的多媒体环境下不同编解码器和每包帧数的VoIP业务QoS分析

Muhammad Aamir, Syed Ali Jafar Zaidi
{"title":"基于OPNET的多媒体环境下不同编解码器和每包帧数的VoIP业务QoS分析","authors":"Muhammad Aamir, Syed Ali Jafar Zaidi","doi":"10.1109/INMIC.2012.6511508","DOIUrl":null,"url":null,"abstract":"Voice over IP (VoIP) is nowadays a common component of multimedia converged networks. In this paper, Quality of Service (QoS) analysis of VoIP traffic is presented in a multimedia network having other types of network traffic also deployed i.e. video, database and web. In such traffic occupied network, VoIP is configured with different codecs and frame counts per packet to observe the effect of these variations on the QoS of VoIP in terms of queuing delay, jitter, packet end-to-end delay and packet delay variation (end-to-end jitter) metrics. The significance of this paper is the emphasis on QoS evaluations in a VoIP deployment while observing the effects of variations in voice codecs and packet lengths. Many previous research papers focus on the perceived speech quality and packet losses; whereas queuing delays and end-to-end latency are usually not considered when analysis is done for different codecs and packet lengths at a time. We consider three main voice codecs (G.711, G.729 and G.723.1) and three frame counts per packet values (10, 25 and 50 voice frames per packet) under each codec. The simulation results show that G.723.1 voice encoding experiences higher queuing delays and queue delay variations (jitter) for higher numbers of frames per packet. However, the least packet end-to-end delay is observed in voice packets encoded with G.729 codec for all configured values of frames per packet.","PeriodicalId":396084,"journal":{"name":"2012 15th International Multitopic Conference (INMIC)","volume":"1 1","pages":"0"},"PeriodicalIF":0.0000,"publicationDate":"2012-12-01","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":"12","resultStr":"{\"title\":\"QoS analysis of VoIP traffic for different codecs and frame counts per packet in multimedia environment using OPNET\",\"authors\":\"Muhammad Aamir, Syed Ali Jafar Zaidi\",\"doi\":\"10.1109/INMIC.2012.6511508\",\"DOIUrl\":null,\"url\":null,\"abstract\":\"Voice over IP (VoIP) is nowadays a common component of multimedia converged networks. In this paper, Quality of Service (QoS) analysis of VoIP traffic is presented in a multimedia network having other types of network traffic also deployed i.e. video, database and web. In such traffic occupied network, VoIP is configured with different codecs and frame counts per packet to observe the effect of these variations on the QoS of VoIP in terms of queuing delay, jitter, packet end-to-end delay and packet delay variation (end-to-end jitter) metrics. The significance of this paper is the emphasis on QoS evaluations in a VoIP deployment while observing the effects of variations in voice codecs and packet lengths. Many previous research papers focus on the perceived speech quality and packet losses; whereas queuing delays and end-to-end latency are usually not considered when analysis is done for different codecs and packet lengths at a time. We consider three main voice codecs (G.711, G.729 and G.723.1) and three frame counts per packet values (10, 25 and 50 voice frames per packet) under each codec. The simulation results show that G.723.1 voice encoding experiences higher queuing delays and queue delay variations (jitter) for higher numbers of frames per packet. However, the least packet end-to-end delay is observed in voice packets encoded with G.729 codec for all configured values of frames per packet.\",\"PeriodicalId\":396084,\"journal\":{\"name\":\"2012 15th International Multitopic Conference (INMIC)\",\"volume\":\"1 1\",\"pages\":\"0\"},\"PeriodicalIF\":0.0000,\"publicationDate\":\"2012-12-01\",\"publicationTypes\":\"Journal Article\",\"fieldsOfStudy\":null,\"isOpenAccess\":false,\"openAccessPdf\":\"\",\"citationCount\":\"12\",\"resultStr\":null,\"platform\":\"Semanticscholar\",\"paperid\":null,\"PeriodicalName\":\"2012 15th International Multitopic Conference (INMIC)\",\"FirstCategoryId\":\"1085\",\"ListUrlMain\":\"https://doi.org/10.1109/INMIC.2012.6511508\",\"RegionNum\":0,\"RegionCategory\":null,\"ArticlePicture\":[],\"TitleCN\":null,\"AbstractTextCN\":null,\"PMCID\":null,\"EPubDate\":\"\",\"PubModel\":\"\",\"JCR\":\"\",\"JCRName\":\"\",\"Score\":null,\"Total\":0}","platform":"Semanticscholar","paperid":null,"PeriodicalName":"2012 15th International Multitopic Conference (INMIC)","FirstCategoryId":"1085","ListUrlMain":"https://doi.org/10.1109/INMIC.2012.6511508","RegionNum":0,"RegionCategory":null,"ArticlePicture":[],"TitleCN":null,"AbstractTextCN":null,"PMCID":null,"EPubDate":"","PubModel":"","JCR":"","JCRName":"","Score":null,"Total":0}
引用次数: 12

摘要

IP语音(VoIP)是当今多媒体融合网络中常见的组成部分。本文介绍了多媒体网络中VoIP业务的服务质量(QoS)分析,该网络中还部署了其他类型的网络业务,如视频、数据库和web。在这种流量占用的网络中,VoIP配置不同的编解码器和每包帧数,从排队延迟、抖动、分组端到端延迟和分组延迟变化(端到端抖动)指标来观察这些变化对VoIP QoS的影响。本文的意义在于强调VoIP部署中的QoS评估,同时观察语音编解码器和数据包长度变化的影响。许多先前的研究论文集中在感知语音质量和丢包;然而,在一次对不同编解码器和数据包长度进行分析时,通常不会考虑排队延迟和端到端延迟。我们考虑三种主要的语音编解码器(G.711、G.729和G.723.1)和每个编解码器下每个包值的三个帧数(每个包10、25和50个语音帧)。仿真结果表明,G.723.1语音编码的队列延迟和队列延迟变化(抖动)随着每包帧数的增加而增加。然而,对于每包帧的所有配置值,用G.729编解码器编码的语音包中观察到最小的包端到端延迟。
本文章由计算机程序翻译,如有差异,请以英文原文为准。
QoS analysis of VoIP traffic for different codecs and frame counts per packet in multimedia environment using OPNET
Voice over IP (VoIP) is nowadays a common component of multimedia converged networks. In this paper, Quality of Service (QoS) analysis of VoIP traffic is presented in a multimedia network having other types of network traffic also deployed i.e. video, database and web. In such traffic occupied network, VoIP is configured with different codecs and frame counts per packet to observe the effect of these variations on the QoS of VoIP in terms of queuing delay, jitter, packet end-to-end delay and packet delay variation (end-to-end jitter) metrics. The significance of this paper is the emphasis on QoS evaluations in a VoIP deployment while observing the effects of variations in voice codecs and packet lengths. Many previous research papers focus on the perceived speech quality and packet losses; whereas queuing delays and end-to-end latency are usually not considered when analysis is done for different codecs and packet lengths at a time. We consider three main voice codecs (G.711, G.729 and G.723.1) and three frame counts per packet values (10, 25 and 50 voice frames per packet) under each codec. The simulation results show that G.723.1 voice encoding experiences higher queuing delays and queue delay variations (jitter) for higher numbers of frames per packet. However, the least packet end-to-end delay is observed in voice packets encoded with G.729 codec for all configured values of frames per packet.
求助全文
通过发布文献求助,成功后即可免费获取论文全文。 去求助
来源期刊
自引率
0.00%
发文量
0
×
引用
GB/T 7714-2015
复制
MLA
复制
APA
复制
导出至
BibTeX EndNote RefMan NoteFirst NoteExpress
×
提示
您的信息不完整,为了账户安全,请先补充。
现在去补充
×
提示
您因"违规操作"
具体请查看互助需知
我知道了
×
提示
确定
请完成安全验证×
copy
已复制链接
快去分享给好友吧!
我知道了
右上角分享
点击右上角分享
0
联系我们:info@booksci.cn Book学术提供免费学术资源搜索服务,方便国内外学者检索中英文文献。致力于提供最便捷和优质的服务体验。 Copyright © 2023 布克学术 All rights reserved.
京ICP备2023020795号-1
ghs 京公网安备 11010802042870号
Book学术文献互助
Book学术文献互助群
群 号:604180095
Book学术官方微信