共查询到20条相似文献,搜索用时 31 毫秒
1.
E. Jammeh I. Mkwawa A. Khan M. Goudarzi L. Sun E. Ifeachor 《Telecommunication Systems》2012,49(1):99-111
Network quality of service (NQoS) of IP networks is unpredictable and impacts the quality of networked multimedia services.
Adaptive voice and video schemes are therefore vital for the provision of voice over IP (VoIP) services for optimised quality
of experience (QoE). Traditional adaptation schemes based on NQoS do not take perceived quality into consideration even though
the user is the best judge of quality. Additionally, uncertainties inherent in NQoS parameter measurements make the design
of adaptation schemes difficult and their performance suboptimal. This paper presents a QoE-driven adaptation scheme for voice
and video over IP to solve the optimisation problem to provide optimal QoE for networked voice and video applications. The
adaptive VoIP architecture was implemented and tested both in NS2 and in an Open IMS Core network to allow extensive simulation
and test-bed evaluation. Results show that the scheme was optimally responsive to available network bandwidth and congestion
for both voice and video and optimised delivered QoE for different network conditions, and is friendly to TCP traffic. 相似文献
2.
Harjit Pal Singh Sarabjeet Singh R.K. Sarin Jasvir Singh 《International Journal of Electronics》2013,100(10):1449-1469
Voice over Internet Protocol (VoIP) is a popular communication service nowadays. VoIP reduces the cost of call transmission by passing voice and video packets through the available bandwidth for data packets through Internet protocol. The quality of the VoIP signal is degraded due to the various network impairments. The proposed scheme, interpolated finite impulse response, is implemented as post-processor after decoding the signal in VoIP system. The performance of the proposed scheme is evaluated for various network conditions. The results of the proposed scheme are measured with the objective measurement methods for signal quality evaluation. The performance of the proposed system is compared with the existing techniques for quality improvement in VoIP system. The results show much improvement in speech quality with the proposed scheme in comparison to other similar schemes. 相似文献
3.
随着VoIP技术的发展,VoIP技术结合卫星通信网络的应用越来越广泛。Inmarsat卫星系统是地球同步轨道系统,网络传播时延大,卫星VoIP电话的语音通信是否可行值得研究。结合VoIP关键技术和海事卫星通信语音通信应用场景,探讨了基于Inmarsat卫星网络实现VoIP技术的方案,并分析出此方案下VoIP系统通话过程的单向时延为350 ms,低于ITU G.114的400 ms的要求。在实际使用环境中进行了测试和验证,结果表明,基于Inmarsat网络下实现VoIP的方案是可行的。该方案实现复杂度低,可以方便地实现Inmarsat网络与地面电话网之间的互联互通,也可以为我国自主研制的宽带卫星通信系统实现VoIP技术提供参考。 相似文献
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Ganguly S. Navda V. Kim K. Kashyap A. Niculescu D. Izmailov R. Hong S. Das S.R. 《Selected Areas in Communications, IEEE Journal on》2006,24(11):2147-2158
In the recent past, there has been a tremendous increase in the popularity of VoIP services as a result of huge growth in broadband access. The same voice-over-Internet protocol (VoIP) service poses new challenges when deployed over a wireless mesh network, while enabling users to make voice calls using WiFi phones. Packet losses and delay due to interference in a multiple-hop mesh network with limited capacity can significantly degrade the end-to-end VoIP call quality. In this work, we discuss the basic requirements for efficient deployment of VoIP services over a mesh network. We present and evaluate practical optimizing techniques that can enhance the network capacity, maintain the VoIP quality and handle user mobility efficiently. Extensive experiments conducted on a real testbed and ns-2 provide insights into the performance issues and demonstrate the level of improvement that can be obtained by the proposed techniques. Specifically, we find that packet aggregation along with header compression can increase the number of supported VoIP calls in a multihop network by 2-3 times. The proposed fast path switching is highly effective in maintaining the VoIP quality. Our fast handoff scheme achieves almost negligible disruption during calls to roaming clients 相似文献
6.
High Speed Packet Access (HSPA) Holma H, Toskala A (in HSDPA/HSUPA for UMTS, 2006) is expected to provide enough bandwidth
for voice over IP (VoIP) service. In this article we assess the performance of VoIP over HSPA with different VoIP clients
and voice codecs. The simulations results show that VoIP can have a good voice quality over HSPA if a proper VoIP client and
codec is used. However it is possible that the delay can increase with early HSPA implementations (mobile, network). 相似文献
7.
Voice communications over zigbee networks 总被引:3,自引:0,他引:3
Chonggang Wang Sohraby K. Jana R. Lusheng Ji Daneshmand M. 《Communications Magazine, IEEE》2008,46(1):121-127
This article provides an overview of ZigBee-enabled wireless networks and discusses the feasibility of supporting voice communications over ZigBee networks. We begin by providing an overview of the ZigBee technology followed by an evaluation of voice quality and performance over such an impoverished wireless channel. Two types of voice communications, namely full-duplex voice over IP (VoIP) and half-duplex push-to-talk (PTT) are considered. Voice quality of VoIP is measured using the R-factor [1] (a well known objective speech quality metric). The quality of PTT, however, is evaluated based on packet-loss rate, delay, and jitter. The simulation results demonstrate that a low-power, low-rate wireless sensor network can support a limited range of voice services. 相似文献
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针对不同网络间的话音通信设计需求,论述了利用内嵌语音网关模块在网络电话(Voice over Internet Protocol,VoIP)与二线模拟电话之间进行话音脉冲编码调制(PCM)时隙交换的设计方案,给出了硬件实现原理和软件设计流程,对话音通信中内VoIP模块性能、PCM信号接口电平的稳定性以及模块间串口包通信的可靠性进行了详细分析。 相似文献
10.
In a wireless multi-hop network environment, energy consumption of mobile nodes is an important factor for the performance evaluation of network life-time. In Voice over IP (VoIP) service, the redundant data size of a VoIP packet such as TCP/IP headers is much larger than the voice data size of a VoIP packet. Such an inefficient structure of VoIP packet causes heavy energy waste in mobile nodes. In order to alleviate the effect of VoIP packet transmission on energy consumption, a packet aggregation algorithm that transmits one large VoIP packet by combining multiple small VoIP packets has been studied. However, when excessively many VoIP packets are combined, it may cause deterioration of the QoS of VoIP service, especially for end-to-end delay. In this paper, we analyze the effect of the packet aggregation algorithm on both VoIP service quality and the energy consumption of mobile nodes in a wireless multi-hop environment. We build the cost function that describes the degree of trade-off between the QoS of VoIP services and the energy consumption of a mobile node. By using this cost function, we get the optimum number of VoIP packets to be combined in the packet aggregation scheme under various wireless channel conditions. We expect this study to contribute to providing guidance on balancing the QoS of VoIP service and energy consumption of a mobile node when the packet aggregation algorithm is applied to VoIP service in a wireless multi-hop networks. 相似文献
11.
Aiming to introduce voice over IP networks and services in ways that satisfy the voice quality expectations of our customers, we have been conducting laboratory studies of how VoIP transmission affects voice quality while also carefully monitoring and managing several field implementations of VoIP. This article summarizes much of what we have learned in this work, and we hope it provides a useful progress report on the industry's evolution to VoIP. We review our data on the voice quality effects of packet loss, delay, speech coders, packet loss concealment algorithms, and the compression option of suppressing transmission during silence. Because the familiar problem of echo has emerged repeatedly in the VoIP environment, we review this issue in some detail. Packet loss and delay variation measurements made on private VoIP networks are reviewed, and the data here are encouraging. We finish by making our case that the network planning tool known as the E-model is currently an inexact predictor of VoIP network performance. 相似文献
12.
Perceptual QoS assessment technologies for VoIP 总被引:3,自引:0,他引:3
Since quality is not generally guaranteed in an IP network, the proper design and management of networks and/or terminals for high-quality voice over IP services and maintenance of service levels is important. In terms of quality design and management, methodologies for appropriately and effectively evaluating the perceptual QoS of VoIP are indispensable. This article gives an overview of the state of the art of quality assessment technologies for VoIP, including recent work on improving their accuracy. 相似文献
13.
A multiplexing scheme for H.323 voice-over-IP applications 总被引:1,自引:0,他引:1
Sze H.P. Liew S.C. Lee J.Y.B. Yip D.C.S. 《Selected Areas in Communications, IEEE Journal on》2002,20(7):1360-1368
Voice communications such as telephony are delay sensitive. Existing voice-over-IP (VoIP) applications transmit voice data in packets of very small size to minimize packetization delay, causing very inefficient use of network bandwidth. This paper proposes a multiplexing scheme for improving the bandwidth efficiency of existing VoIP applications. By installing a multiplexer in an H.323 proxy, voice packets from multiple sources are combined into one IP packet for transmission. A demultiplexer at the receiver-end proxy restores the original voice packets before delivering them to the end-user applications. Results show that the multiplexing scheme can increase bandwidth efficiency by as much as 300%. The multiplexing scheme is fully compatible with existing H.323-compliant VoIP applications and can be readily deployed. 相似文献
14.
As the Internet evolves into a ubiquitous communication infrastructure and provides various services including telephony, it will be expected to meet the quality standards achieved in the public switched telephone network. Our objective in this paper is to assess to what extent today's Internet meets this expectation. Our assessment is based on delay and loss measurements taken over wide-area backbone networks and uses subjective voice quality measures capturing the various impairments incurred. First, we compile the results of various studies into a single model for assessing the voice-over-IP (VoIP) quality. Then, we identify different types of typical Internet paths and study their VoIP performance. For each type of path, we identify those characteristics that affect the VoIP perceived quality. Such characteristics include the network loss and the delay variability that should be appropriately handled by the playout scheduling at the receiver. Our findings indicate that although voice services can be adequately provided by some ISPs, a significant number of Internet backbone paths lead to poor performance. 相似文献
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One‐way delay variation (OWDV) has become increasingly of interest to researchers as a way to evaluate network state and service quality, especially for real‐time and streaming services such as voice‐over‐Internet‐protocol (VoIP) and video. Many schemes for OWDV measurement require clock synchronization through the global‐positioning system (GPS) or network time protocol. In clock‐synchronized approaches, the accuracy of OWDV measurement depends on the accuracy of the clock synchronization. GPS provides highly accurate clock synchronization. However, the deployment of GPS on legacy network equipment might be slow and costly. This paper proposes a method for measuring OWDV that dispenses with clock synchronization. The clock synchronization problem is mainly caused by clock skew. The proposed approach is based on the measurement of inter‐packet delay and accumulated OWDV. This paper shows the performance of the proposed scheme via simulations and through experiments in a VoIP network. The presented simulation and measurement results indicate that clock skew can be efficiently measured and removed and that OWDV can be measured without requiring clock synchronization. 相似文献
17.
目前,一些大、中城市的有线电视公司已经拥有庞大的有线电视宽带用户群,而基于HFC网的VoIP系统正是目前在有线电视宽带网条件下综合接入语音、数据和多媒体业务的最佳解决方案之一,其中语音业务可利用中继媒体网关设备完成媒体流转换,并通过标准的E1协议接入PSTN网络,以达到充分利用HFC及PSTN网络资源迅速开展话音业务的目的.同时,作为NGN(下一代网络)中的标准部件,VoIP是面向未来、可持续发展的解决方案之一,在有线电视宽带网所及之处,可以为商业和家庭用户提供质优价廉的IP语音服务. 相似文献
18.
针对在工程应用中如何通过以太网进行话音传输提出了一种设计方案,分析了话音在网络上传输的特点,介绍了一种基于以太网的数字话音传输系统方案。系统以自带网络协议的嵌入式ARM微控制器LM3S9B96为核心平台,采用IP上传送语音(Voice over IP,VoIP)技术实现话音的以太网传输。对系统的话音实际传输效果进行了仿真测试分析,结果表明,话音清晰、失真度和时延小,整体性能满足实际话音通信的要求。 相似文献
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A call admission control framework for voice over WLANs 总被引:1,自引:0,他引:1
In this article a call admission control framework is presented for voice over wireless local area networks (WLANs). The framework, called WLAN voice manager, manages admission control for voice over IP (VoIP) calls with WLANs as the access networks. WLAN voice manager interacts with WLAN medium access control (MAC) layer protocols, soft-switches (VoIP call agents), routers, and other network devices to perform end-to-end (ETE) quality of service (QoS) provisioning and control for VoIP calls originated from WLANs. By implementing the proposed WLAN voice manager in the WLAN access network, a two-level ETE VoIP QoS control mechanism can be achieved: level 1 QoS for voice traffic over WLAN medium access and level 2 QoS for ETE VoIP services in the networks with WLANs as the local access. The implementation challenges of this framework are discussed for both level 1 and level 2. Possible solutions to the implementation issues are proposed and other remaining open issues are also addressed. 相似文献