The Contributory Effect of Latency on the Quality of Voice Transmitted over the Internet

Deployment of Voice over Internet Protocol (VoIP) is rapidly growing worldwide due to the new services it provides and cost savings derived from using a converged IP network. However, voice quality is affected by bandwidth, delay, latency, jitter, packet loss e.t.c. Latency is the dominant factor that degrades quality of voice transfer. There is therefore strong need for a study on the effect of Latency with the view to improving Quality of Voice (QoV) in VoIP network. In this work, Poisson probability theorem, Markov Chain, Probability distribution theorems and Network performance metric were used to study the effect of latency on QoS in VoIP network. This is achieved by considering the effect of latency resulting from several components between two points in multiple networks. The NetQoS Latency Calculator, Net-Cracker Professional® for Modeling and Matlab/Simulink® for simulating network were tools used and the results obtained compare favourably well with theoretical facts

Relay path selection approaches in peer-to-peer VoIp systems

Multipath overlay routing technologies are seen as alternative solutions for VoIP because they inherit path diversity from peer-to-peer overlay networks. We discuss and compare the performances of two relay path selection approaches proposed for VoIP overlay systems through extensive simulations. We propose a new method for relay path computation that takes into account both path disjointness and other network quality factors (such as packet delay or loss). We further apply our method in different overlay network scenarios by varying the supernode distribution. It is found that there is a considerable improvement of path performance when relaying traffic through highly connected ASs using the new method

Analisis Codec dan Payload pada Micronet dan CISCO pada Jaringan VPN­MPLS

VoIP seperti halnya aplikasi real­time lainnya menuntut jaminan layanan minimum yang harus dipenuhi, jaringan berbasis IP biasa / jaringan internet saat ini tidak mampu memberikan jaminan itu karena sifatnya yang hanya memberikan Best­effort Service. Masalah pokok kualitas VoIP yang jelek sebenarnya tergantung pada kestabilan bandwith bagi traffic VoIP dan sensitifitasnya terhadap degradasi performansi jaringan seperti delay, jitter dan packet loss.(Shu Tao ; et al ,2005). Permasalahan mengenai kualitas VoIP yang jelek di atas dihadapi pula oleh PT.Aplikanusa Lintasarta, keterbatasan bandwidth akses yang dimiliki pelanggan telah menjadi masalah utama kualitas VoIP pada saat ini, upgrade bandwidth jaringan akses tidak menjadi referensi solusi. Proses pembahasan ini meliputi : analisa jaringan akses, pengukuran utilisasi bandwidth jaringan akses, analisa traffic data dan uji korelasi Perubahan setting codec dan payload pada perangkat VoIP terhadap konsumsi bandwidth. Hasil ini adalah sebuah model sistem pengujian perangkat VoIP gateway dalam hal pengujian konsumsi bandwidth VoIP dan referensi standar setting codec dan payload yang paling efesien dalam konsumsi bandwidth

Improved voice quality with the combination of transport layer & audio codec for wireless devices

Improving voice quality over wireless communication becomes a demanding feature for social media apps like facebook, whatsapp and other communication channels. Voice-over-internet protocol (VoIP) helps us to make quick telephone calls over the internet. It includes various mechanism which are signaling, controlling and transport layer. Over wireless links, packet loss and high transmission delay damage voice quality. Here VoIP quality will be measured by three main elements which are signaling protocol, audio codec and transport layer. To improve the overall voice quality, we need to combine these three elements properly to get the best score. Otherwise perceptual speech quality will not be the right tool to measure the voice quality. Here we will use Mean Opinion Score (MOS) for calculated jitter values and end to end delay. At the end, best combination of audio codec & signaling protocol produced the quality speech

Building Cooperation in VoIP Network through a Reward Mechanism

In this paper, for solving the moral hazard problem of super nodes in VOIP network and achieving better communication quality, we establish a reward mechanism based on classical efficiency-wage models. In the reward mechanism, the function of reward is to encourage super nodes to contribute their bandwidth and cover their effort costs, whereas the function of fine is to prevent opportunistic super nodes from shirking. We consider that network quality and idle bandwidth are the essential criterions for selecting qualified super nodes. Once all super nodes can satisfy specific conditions, the required reward can be derived so as to improve the VoIP platform\u27s revenue. Moreover, we also suggest several targets both in technical and economic view that the platform provider can strive in order to boost his/her market share. In addition, the case of Skype is discussed in this study and we also examine its current pricing strategy

Assessing and Redesigning Enterprise Networks through NS-2 to Support VoIP

AbstractIn the recent past, Voice over IP (VoIP) deployments over data networks are gaining popularity due to the massive growth in the broadband internet access. Successful deployment of these applications depends directly on the performance of the underlying data network. Based on this and the fact that today's data networks are operated to perform many significant applications, network administrators seek out a way to measure the impact of these applications on the existing network performance before deploying them. Occasionally, network redesign is necessity; considering redesign's alterations should preserve most of the existing network characteristics to reduce overall impact of deploying new applications in the network performance. In this paper, we evaluated readiness of the existing enterprise network through NS-2 to support VoIP and based on findings a solution for redesigning the enterprise network is proposed to enhance the new network performance and leaving sufficient capacity for future growth. We applied our approach on a medium size enterprise network as a case study, the results prove improvements in network performance after redesigning the existing enterprise network

Analysing the characteristics of VoIP traffic

In this study, the characteristics of VoIP traffic in a deployed Cisco VoIP phone system and a SIP based soft phone system are analysed. Traffic was captured in a soft phone system, through which elementary understanding about a VoIP system was obtained and experimental setup was validated. An advanced experiment was performed in a deployed Cisco VoIP system in the department of Computer Science at the University of Saskatchewan. Three months of traffic trace was collected beginning October 2006, recording address and protocol information for every packet sent and received on the Cisco VoIP network. The trace was analysed to find out the features of Cisco VoIP system and the findings were presented.This work appears to be one of the first real deployment studies of VoIP that does not rely on artificial traffic. The experimental data provided in this study is useful for design and modeling of such systems, from which more useful predictive models can be generated. The analysis method used in this research can be used for developing synthetic workload models. A clear understanding of usage patterns in a real VoIP network is important for network deployment and potential network activities such as integration, optimizations or expansion. The major factors affecting VoIP quality such as delay, jitter and loss were also measured and simulated in this study, which will be helpful in an advanced VoIP quality study. A traffic generator was developed to generate various simulated VoIP traffic. The data used to provide the traffic model parameters was chosen from peak traffic periods in the captured data from University of Saskatchewan deployment. By utilizing the Traffic Trace function in ns2, the simulated VoIP traffic was fed into ns2, and delay, jitter and packet loss were calculated for different scenarios. Two simulation experiments were performed. The first experiment simulated the traffic of multiple calls running on a backbone link. The second experiment simulated a real network environment with different traffic load patterns. It is significant for network expansion and integration

Joint path and resource selection for OBS grids with adaptive offset based QOS mechanism

Ankara : The Department of Electrical and Electronics Engineering and the Institute of Engineering and Sciences of Bilkent University, 2007.Thesis (Master's) -- Bilkent University, 2007.Includes bibliographical references leaves 71-76It is predicted that grid computing will be available for consumers performing their daily computational needs with the deployment of high bandwidth optical networks. Optical burst switching is a suitable switching technology for this kind of consumer grid networks because of its bandwidth granularity. However, high loss rates inherent in OBS has to be addressed to establish a reliable transmission infrastructure. In this thesis, we propose mechanisms to reduce loss rates in an OBS grid scenario by using network-aware resource selection and adaptive offset determination. We first propose a congestion-based joint resource and path selection algorithm. We show that path switching and network-aware resource selection can reduce burst loss probability and average completion time of grid jobs compared to the algorithms that are separately selecting paths and grid resources. In addition to joint resource and path selection, we present an adaptive offset algorithm for grid bursts which minimizes the average completion time. We show that the adaptive offset based QoS mechanism significantly reduces the job completion times by exploiting the trade-off between decreasing loss probability and increasing delay as a result of the extra offset time.Köseoğlu, MehmetM.S

On algorithms, system design, and implementation for wireless mesh networks.

Yuan, Yan.Thesis submitted in: November 2007.Thesis (M.Phil.)--Chinese University of Hong Kong, 2008.Includes bibliographical references (leaves 84-87).Abstracts in English and Chinese.Chapter 1 --- Introduction --- p.1Chapter 1.1 --- Wireless Mesh Network --- p.3Chapter 1.1.1 --- Architecture Overview --- p.3Chapter 1.1.2 --- Routing Protocols --- p.5Chapter 1.2 --- Contribution of this Thesis --- p.7Chapter 1.3 --- Organization of this Thesis --- p.8Chapter 2 --- Background and Literature Review --- p.9Chapter 2.1 --- VoIP on Wireless Mesh Networks --- p.9Chapter 2.1.1 --- Performance of VoIP on Wireless Mesh Networks --- p.9Chapter 2.1.2 --- Optimizations for VoIP over Wireless Mesh Networks --- p.12Chapter 2.1.3 --- Path and Packet Aggregation Scheme --- p.14Chapter 2.2 --- Network Coding on Wireless Mesh Networks --- p.15Chapter 2.2.1 --- The Concept of Network Coding --- p.15Chapter 2.2.2 --- Related Work --- p.16Chapter 3 --- Adaptive Path and Packet Aggregation System --- p.19Chapter 3.1 --- Overview --- p.19Chapter 3.2 --- The Adaptive Path Aggregation Routing Algorithm --- p.20Chapter 3.2.1 --- Protocol Overview --- p.20Chapter 3.2.2 --- Data Structure --- p.21Chapter 3.2.3 --- The Concept of Link Weight and Path Weight --- p.26Chapter 3.2.4 --- APA Operations --- p.27Chapter 3.3 --- The Packet Aggregation System --- p.39Chapter 3.3.1 --- Overview --- p.39Chapter 3.3.2 --- Packet structure --- p.40Chapter 3.3.3 --- Local Compression --- p.41Chapter 3.3.4 --- Packet Aggregation/Disaggregation --- p.42Chapter 3.4 --- Performance Analysis --- p.44Chapter 3.4.1 --- Integration of the path aggregation routing protocol and the packet aggregation system --- p.46Chapter 3.5 --- Performance Evaluation --- p.48Chapter 3.5.1 --- Testbed Setup --- p.48Chapter 3.5.2 --- Packet aggregation --- p.48Chapter 3.5.3 --- Combined scenario: path and packet aggregation --- p.58Chapter 3.6 --- Summary --- p.65Chapter 4 --- Network Coding System in wireless network --- p.67Chapter 4.1 --- Overview --- p.67Chapter 4.2 --- System Architecture --- p.68Chapter 4.2.1 --- Packet Format --- p.68Chapter 4.2.2 --- Encoding and decoding --- p.69Chapter 4.3 --- Performance Evaluation --- p.71Chapter 4.3.1 --- Experiment Setup --- p.71Chapter 4.3.2 --- Performance Metric --- p.72Chapter 4.3.3 --- Experiment Results --- p.72Chapter 4.4 --- Summary --- p.79Chapter 5 --- Conclusions and Future Directions --- p.8