Impacts of Buffering of Voice Calls in Integrated Voice and Data Services (original) (raw)
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International Journal of Smart Sensor and Adhoc Network., 2012
Queuing theory and Markov chain analysis plays vital role in analyzing real-life problems. It is applied to wired network, wireless network and mobile communication to analyze the packet traffic in packet switched network. In this simulation and analysis, integrated communication system such as voice and data is simulated with different queue size for voice calls with different arrival and service rate and its results are analyzed to study the impact of buffering of voice and data calls for the proposed integrated wired network using Queuing theory and Markov chain analysis. We also propose to optimize the system characteristics in an attempt to provide better Quality of Service (QoS) for systems with integrated voice and data calls. The proposed models have two traffic flow namely voice calls (real-time traffic like audio) and data calls (data traffic like FTP). A single channel is assigned for voice and data calls. The incoming voice and data calls are queued when the channel is b...
Proceedings - International Conference on Advanced Information Networking and Applications, AINA, 2013
Service and Buffer Management techniques can be used to ensure Quality of Service (QoS) for different traffic flows according to some specific policies. In this study, a single buffer queuing system is considered to model single and multi channel, homogeneous wireless network systems such as wireless local area networks (WLANs) and cellular networks. These systems are now being used to carry both voice and data traffic and hence it is important to optimise these systems in an attempt to reduce the blocking, and minimize the latency to acceptable ranges. Since voice packets are delay sensitive, they have the priority to receive service. Also they require smaller buffering capacities, since the response time to voice requests should be below specific values. In addition, in order to reduce retransmission on reliable data connections, data packets are not usurped by incoming voice packets. In this paper, a mathematical analysis of this scenario is explored. The proposed mathematical model is represented by two dimensions; one for incoming voice packets and one for data packets. The models proposed show that it is possible to store incoming voice packets in the queue in case the channel or channels are busy. Both voice and data packets have finite buffering. Incoming voice packets are blocked when the voice buffer or the common queue is full. Therefore there is an added blocking probability of voice due to the presence of data packets in the system when the common queue is full. The analytical model is validated using simulation. The system proposed attempts to provide minimum delay for voice while reducing the disruption to reliable data connections. Numerical results show that, it is possible to attain these goals with reasonable buffer sizes. This study is useful for understanding the trade-offs and thresholds of single and multi channel systems with voice and data traffic.
Telecommunication Systems, 2016
This study presents models for management of voice and data traffic and new algorithms, which use call admission control as well as buffer management to optimise the performance of single channel systems such as wireless local area networks in the presence of mobile stations. Unlike existing studies, the new approach queues incoming voice packets as well as data packets, and uses a new preemption algorithm in order to keep the response time of voice requests at certain levels while the blocking of data requests is minimised. A new performance metric is introduced to provide uncorrelated handling of integrated services. Queueing related issues such as overall queue capacity, individual capacities for voice and data requests, the probability of blocking, and effects of waiting time on overall quality of service are considered in detail. Analytical models are presented and the results obtained from the analytical models were validated using discrete event simulations.
A unified approach to analyzing models of cellular communication networks with data call buffering
Automation and Remote Control, 2014
A unified approach to the analysis of models of cellular communication networks with queues of data calls was developed. Consideration was given to the models with two types of strategies for admission to the cell channels and the buffer. In one strategy, admission to the voice call channels is based on reservation of channels for handover of calls of the given type, and in the other strategy, on the scheme of truncating the calls of the given type. In both models, the strategy of buffer admission by the data calls relies on the scheme of place reservation for handover of the calls of the given type. The models with finite and infinite data call queues were studied. For different admission strategies, algorithms to calculate the servicing performance indices were given, and the results of their comparative analysis were presented.
Performance analysis for voice/data integration on a finite-buffer mobile system
IEEE Transactions on Vehicular Technology, 2000
Personal communication service (PCS) networks offer mobile users diverse telecommunication applications, such as voice, data, and image, with different bandwidth and quality-of-service (QoS) requirements. This paper proposes an analytical model to investigate the performance of an integrated voice/data mobile network with finite data buffer in terms of voice-call blocking probability, data loss probability, and mean data delay. The model is based on the movable-boundary scheme that dynamically adjusts the number of channels for voice and data traffic. With the movable-boundary scheme, the bandwidth can be utilized efficiently while satisfying the QoS requirements for voice and data traffic. Using our model, the impact of hot-spot traffic in the heterogeneous PCS networks, in which the parameters (e.g., number of channels, voice, and data arrival rates) of cells can be varied, can be effectively analyzed. In addition, an iterative algorithm based on our model is proposed to determine the handoff traffic, which computes the system performance in polynomial-bounded time. The analytical model is validated by simulation. He is an Associate Editor of the IEEE TRANSACTIONS ON MULTIMEDIA. His research interests include internet computing, real-time operating systems, real-time networking, real-time multimedia applications, e.g., video conference and video on demand, computational geometry, combinatorial optimization, VLSI design algorithms, and implementation and testing of VLSI algorithms on real designs. He and his team members have developed several system prototypes including a multimedia digital library, ASIS MDL.
European Transactions on Telecommunications, 2000
We propose an admission control policy for wireless multimedia networks that is based on the well known threshold-based guard channel method. The new scheme deals with two different types of traffic classes; namely: voice and data. We assume two different thresholds, one for each traffic class. In addition, we propose to buffer the handoff data calls if no free channels are available rather than rejecting them. For handoff voice calls, we propose two methods, namely: a blocking method and a preemptive method. For the blocking method, we reject the handoff voice calls if no channels are available. For the preemptive method, an ongoing data call can be buffered and its channel allocated to the handoff voice call. We study the effect of the thresholds, buffer size and the application of the proposed methods on call blocking probabilities. It is shown that the new call blocking probabilities are only affected by the threshold values. Meanwhile, the data handoff blocking probability exhibited great improvement. For handoff voice calls, when the blocking method is applied, the blocking probability value increases slightly with the increase of buffer size. Meanwhile, for the preemptive method, the handoff voice call blocking probability significantly decreases as the buffer size increases. Based on these results, we develop an algorithm that uses the proposed policy to estimate the appropriate thresholds and buffer size which meet the required call blocking probabilities for each traffic type.
IEEE Transactions on Vehicular Technology, 2006
Resource allocation and call admission control (CAC) are key management functions in future cellular networks, in order to provide multimedia applications to mobiles users with quality of service (QoS) guarantees and efficient resource utilization. In this paper, we propose and analyze a priority based resource sharing scheme for voice/data integrated cellular networks. The unique features of the proposed scheme are that 1) the maximum resource utilization can be achieved, since all the leftover capacity after serving the high priority voice traffic can be utilized by the data traffic; 2) a Markovian model for the proposed scheme is established, which takes account of the complex interaction of voice and data traffic sharing the total resources; 3) optimal CAC parameters for both voice and data calls are determined, from the perspective of minimizing resource requirement and maximizing new call admission rate, respectively; 4) load adaption and bandwidth allocation adjustment policies are proposed for adaptive CAC to cope with traffic load variations in a wireless mobile environment. Numerical results demonstrate that the proposed CAC scheme is able to simultaneously provide satisfactory QoS to both voice and data users and maintain a relatively high resource utilization in a dynamic traffic load environment. The recent measurement-based modeling shows that the Internet data file size follows a lognormal distribution, instead of the exponential distribution used in our analysis. We use computer simulations to demonstrate that the impact of the lognormal distribution can be compensated for by conservatively applying the Markovian analysis results.
Influences of Classical and Hybrid Queuing Mechanisms on VoIP's QoS Properties
VoIP Technologies, 2011
Nowadays we can find many TCP/IP based network applications, such as: WWW, e-mail, video-conferencing, VoIP, remote accesses, telnet, p2p file sharing, etc. All mentioned applications became popular because of fast-spreading broadband internet technologies, like xDSL, DOCSIS, FTTH, etc. Some of the applications, such as VoIP (Voice over Internet Protocol) and video-conferencing, are more time-sensitive in delivery of network traffic than others, and need to be treated specially. This special treatment of the time-sensitive applications is one of the main topics of this chapter. It includes methodologies for providing a proper quality of service (QoS) for VoIP traffic within networks. Normally, their efficiency is intensively tested with simulations before implementation. In the last few years, the use of simulation tools in R&D of communication technologies has rapidly risen, mostly because of higher network complexity. The internet is expanding on a daily basis, and the number of network infrastructure components is rapidly increasing. Routers are most commonly used to interconnect different networks. One of their tasks is to keep the proper quality of service level. The leading network equipment manufacturers, such as Cisco Systems, provide on their routers mechanisms for reliable transfer of time-sensitive applications from one network segment to another. In case of VoIP the requirement is to deliver packets in less than 150ms. This limit is set to a level where a human ear cannot recognize variations in voice quality. This is one of the main reasons why leading network equipment manufacturers implement the QoS functionality into their solutions. QoS is a very complex and comprehensive system which belongs to the area of priority congestions management. It is implemented by using different queuing mechanisms, which take care of arranging traffic into waiting queues. Time-sensitive traffic should have maximum possible priority provided. However, if a proper queuing mechanism (FIFO, CQ, WFQ, etc.) is not used, the priority loses its initial meaning. It is also a well-known fact that all elements with memory capability involve additional delays during data transfer from one network segment to another, so a proper queuing mechanism and a proper buffer length should be used, or the VoIP quality will deteriorate. If we take a look at the router, as a basic element of network equipment, we can realise that we are dealing with application priorities on the lowest level. Such level is presented by waiting queues and queuing mechanisms, related with the input traffic connection interface.
IEEE Transactions on Communications, 2004
This paper proposes a call admission control (CAC) policy for a cellular system supporting voice and data services, and providing a higher priority to handoff calls than to new calls. A procedure for searching the optimal admission region is given. The traffic flow is characterized by a three-dimensional (3-D) birthdeath model, which captures the complex interaction between the on/off voice and best-effort data traffic sharing the total resources without partition. To reduce complexity, the 3-D model is simplified to an exact (approximate) 2-D model for voice (data). The mathematical expressions are then derived for the performance measures and for the minimal amount of resources required for quality-of-service (QoS) provisioning. Numerical results demonstrate that: 1) the proposed CAC policy performs well in terms of QoS satisfaction and resource utilization; 2) the approximate 2-D model for data traffic can achieve a high accuracy in the traffic flow characterization; and 3) the admission regions obtained by the proposed search method agree very well with those obtained by numerically solving the mathematical equations. Furthermore, computer simulation results demonstrate that the impact of lognormal distributed data file size is not significant, and may be compensated by conservatively applying the Markovian analysis results.