Simulation and Analysis of Impact of Buffering of Voice Calls in Integrated Voice and Data Communication System (original) (raw)
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Impacts of Buffering of Voice Calls in Integrated Voice and Data Services
2011
In this study, we aim to analyse the relationship between various characteristics of a communication system with data and voice call requests. Queuing theory and Markov chain analysis are effectively used for this purpose. Such a study is useful for understanding how the proposed mathematical models behave which represents a system with integrated voice and data calls in homogenous wireless networks. We also propose to optimise 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 dimensions; one for voice calls and one for data calls. A channel is assigned for two input traffic call, namely, voice and data calls. The incoming voice and data calls are queued when the channel is busy. Since voice calls are delay-sensitive, priority is given to voice calls. Also, since there is only one channel, data calls are only serviced if there are no voice calls in the system. For such systems, it is important to analyse the impact of buffering the voice calls as well as data calls for various mean rates of call requests, and mean service times. The analytical models presented are generic which is applicable for various systems with similar characteristics. Numerical results are also provided. The results show that the proposed models can be used for optimisation of the performance of a given network.
Improving Quality of Service Through Buffer Optimization in VoIP Network
Voice over Internet Protocol (VoIP) is developed for voice communications system based on voice packets transmitted over IP network with real-time communications of voice across networks using the Internet protocols. Quality of Service (QoS) mechanism is applied to guarantee successful voice packets transmitted over IP network with reduced delay or drop according to assigned priority of voice packets. In this paper, the goal of simulation models is present to investigate the performance of VoIP buffer size for improving quality of service (QoS) through buffer optimization with the simulation results by using OPNET modeler version 14.5. The performance of the proposed algorithm is analyzed and compared the quality of service for VoIP. The final simulated result shows that the VoIP service performance best buffer size 128 Kb over WiMAX network.
A comparative study of different queuing techniques in VOIP, video conferencing and file transfer
Daffodil International University Journal of Science and Technology, 1970
Today's Internet only provides Best Effort Service. Traffic is processed as quickly as possible, but there is no guarantee of timelines or actual delivery. With the rapid transformation of the Internet into a commercial infrastructure, demands for service quality have rapidly developed. People of the modern world are very much dependent on various network services like VOIP, Videoconferencing and File Transfer. Different types of Traffic Management systems are used in those services. Queuing is one of the very vital mechanisms in traffic management system. Each router in the network must implement some queuing discipline that governs how packets are buffered while waiting to be transmitted. This paper gives a comparative analysis of three queuing systems FIFO, PQ and WFQ. The study has been carried out on some issues like: Traffic dropped Traffic Received and packet end to end delay and the simulation results shows that WFQ technique has a superior quality than the other techniq...
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.
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.
Performance of VoIP applications in a simple differentiated services network architecture
2001
IP networks were designed to support non-real time applications, such as file transfer or e-mail. These applications are characterised by their bursty traffic and high bandwidth demands at burst times, but they are not highly sensitive to delay or delay variation (jitter). On the other hand VoIP application requires timely packet delivery with low delay, jitter and packet loss values. Integration of voice and data onto a single network is becoming a priority for many network operators. To achieve that goal IP networks must be enhanced with mechanisms that ensure the quality of service required to carry real-time traffic such as voice. Three parameters emerge as the primary factors affecting voice quality within networks that offer VoIP technologies: clarity, end-to-end delay and echo. To support interactive voice application on an IP network we must be able to control four QoS categories: bandwidth, delay (latency), jitter and packet loss.
This paper showed an analysis and comparison between new buffer design with both concept buffers in the PQ algorithm and SPBA algorithm. In the PQ algorithm, there are four buffering packet are low, normal, medium and high. The buffering packet in PQ algorithm is greedy. However, fourth the buffering is not optimal used. It is caused by PQ algorithm is based on the priority, whereas this buffering just always serviced is the highest priority. While under priority are rarely or never serviced will cause other buffering rarely used. While SPBA algorithm is architecture easiest, and it does not need any resource reservation or threshold dropping, but only makes use of priority scheduling. SPBA algorithm, where incoming packets are placed into the two priority traffic classes is the high class and low class. On the SPBA algorithm is there are not available reservation sources to save the remaining packets when the explosion (burst) traffic occurred, that could result in packet drop and packet loss. Then, with the efficiency of PQ buffering algorithm, can provide greater impact to reduce delays. In the new buffering algorithm, simplify four buffering into three packets (High, Medium and Low) is proposed. In the analysis and comparison new buffering algorithm could be known problems and weaknesses of both algorithms.
Call request buffering in a PCS network
Proceedings of INFOCOM '94 Conference on Computer Communications, 1994
In a personal communication services (PCS) network, a set of channels is assigned to every cell. When a phone call arrives, it consumes a channel until the end of the conversation. If no channel is available, the call is dropped. In many cases, channels may return shortly after a call is dropped. Thus, if some buffering mechanism is introduced to the channel allocation algorithm, a cell may accommodate more phone calls. This is referred to as call request buffering. This paper proposes both analytical and simulation models to study the impact of two call request buffering schemes. Our results indicate the potential to greatly improve network performance (i.e.] reducing the call blocking probability) by increasing the call-setup period by a reasonably small amount. For the offered load engineered at 1 % blocking probability, call request buffering carries 5%-50% more load compared with the system without buffering.
Review of Different Queuing Disciplines in VOIP, Video Conferencing and File Transfer
Now a day Internet only put up best effort service. Traffic is transmitting as earliest as possible, but during transmission, there is no guarantee of timelines or real delivery of packets. With the swiftly transformation of the Internet into a commercial infrastructure, demands for a quality of service have developed in rapid rate. People of the present world are very much depending upon the various network services like VOIP, Video conferencing and File Transfer [6,7]. Various categories of Traffic Management systems are used in those services. Queuing is one of the very important mechanisms in traffic management system. Each router in the network must implement some queuing discipline that control how packets are buffered while waiting to be transmitted. The main aim of this paper is to highlight quality of service (QoS) analysis using different queuing disciplines.
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.