Impact of Background Traffic on Speech Quality in VoWLAN

This paper describes measurements of the impact of background traffic on speech quality in an environment of WLANs (IEEE 802.11). The simulated background traffic consists of three types of current traffics in telecommunication networks such as data transfer service, multimedia streaming service, and Web service. The background traffic was generated by means of the accomplished Distributed Internet Traffic Generator (D-ITG). The impact of these types of traffic and traffic load on speech quality using the test sequence and speech sequences is the aim of this paper. The assessment of speech quality is carried out by means of the accomplished Perceptual Evaluation of Speech Quality (PESQ) algorithm. The proposal of a new method for improved detection of the critical conditions in wireless telecommunication networks from the speech quality point of view is presented in this paper. Conclusion implies the next application of the method of improved detection of critical conditions for the purpose of algorithms for link adaptation from the speech quality point of view in an environment of WLANs. The primary goal of these algorithms is improving speech quality in the VoWLAN connections, which are established in the competent link

Performance and Analysis of Transfer Control Protocol Over Voice Over Wireless Local Area Network

A thesis presented to the faculty of the College of Science and Technology at Morehead State University in partial fulfillment of the requirements for the Degree Master of Science by Rajendra Patil in August of 2008

Experimental Tuning of AIFSN and CWmin Parameters to Prioritize Voice over Data Transmission in 802.11e WLAN Networks

In this paper we experimentally study the impact of two EDCA parameters, namely AIFSN and CWmin, on a mixed voice/data wireless transmission. In particular we investigate how the tuning of these parameters affects both the voice transmission quality and background data throughput. We predict end-to-end voice transmission quality from time varying transmission impairments using the latest Appendix to the ITU-T E-model. Our experimental results show that the tuning of the EDCA parameters can be used to successfully prioritize voice transmission over data in real 802.11e networks. We also demonstrate that the AIFSN parameter more effectively protects voice calls against background data traffic than CWmin. To the best of our knowledge, this is the first experimental investigation on tuning of MAC layer parameters in a real 802.11e WLAN network from the perspective of end-to-end voice transmission quality and end user satisfaction

IMPROVING QoS OF VoWLAN VIA CROSS-LAYER BASED ADAPTIVE APPROACH

Voice over Internet Protocol (VoIP) is a technology that allows the transmission of voice packets over Internet Protocol (IP). Recently, the integration of VoIP and Wireless Local Area Network (WLAN), and known as Voice over WLAN (VoWLAN), has become popular driven by the mobility requirements ofusers, as well as by factor of its tangible cost effectiveness. However, WLAN network architecture was primarily designed to support the transmission of data, and not for voice traffic, which makes it lack ofproviding the stringent Quality ofService (QoS) for VoIP applications. On the other hand, WLAN operates based on IEEE 802.11 standards that support Link Adaptive (LA) technique. However, LA leads to having a network with multi-rate transmissions that causes network bandwidth variation, which hence degrades the voice quality. Therefore, it is important to develop an algorithm that would be able to overcome the negative effect of the multi-rate issue on VoIP quality. Hence, the main goal ofthis research work is to develop an agent that utilizes IP protocols by applying a Cross-Layering approach to eliminate the above-mentioned negative effect. This could be expected from the interaction between Medium Access Control (MAC) layer and Application layer, where the proposed agent adapts the voice packet size at the Application layer according to the change of MAC transmission data rate to avoid network congestion from happening. The agent also monitors the quality of conversations from the periodically generated Real Time Control Protocol (RTCP) reports. If voice quality degradation is detected, then the agent performs further rate adaptation to improve the quality. The agent performance has been evaluated by carrying out an extensive series ofsimulation using OPNET Modeler. The obtained results of different performance parameters are presented, comparing the performance ofVoWLAN that used the proposed agent to that ofthe standard network without agent. The results ofall measured quality parameters hav

Analysis of Packet Throughput and Delay in IEEE 802.11 WLANs with TCP Traffic

The IEEE 802.11 standard is a successfulwireless local area networks (WLAN) technology,because of its easy deployment. With WLAN, theability of the IEEE802.11 standard to supportmultimedia applications with high quality of service(QoS) requirements has increased. This paperevaluates the capability of QoS support in EnhancedDistributed Channel Access (EDCA) mechanism of theIEEE 802.11e standard using TCP protocol. TheEDCA is an enhancement for QoS support in 802.11.EDCA mechanisms allow prioritized medium accessfor applications with high QoS requirements byassigning different priorities to the access categories.The current work discusses the performanceevaluation of 802.11 and 802.11e by simulations usingTCP protocol. A comparative discussion between DCFund EDCA with TCP protocol is reported for differentservices, such as voice, video, best-effort andbackground traffic. Results and simulations show thatthe TCP protocol is usable for transferring audio andvideo data within special programs and applications.Moreover, it is shown that the UDP protocol with itshigher performance is more suitable for this task

IMPROVING QoS OF VoWLAN VIA CROSS-LAYER BASED ADAPTIVE APPROACH

Voice over Internet Protocol (VoIP) is a technology that allows the transmission of voice packets over Internet Protocol (IP). Recently, the integration of VoIP and Wireless Local Area Network (WLAN), and known as Voice over WLAN (VoWLAN), has become popular driven by the mobility requirements ofusers, as well as by factor of its tangible cost effectiveness. However, WLAN network architecture was primarily designed to support the transmission of data, and not for voice traffic, which makes it lack ofproviding the stringent Quality ofService (QoS) for VoIP applications. On the other hand, WLAN operates based on IEEE 802.11 standards that support Link Adaptive (LA) technique. However, LA leads to having a network with multi-rate transmissions that causes network bandwidth variation, which hence degrades the voice quality. Therefore, it is important to develop an algorithm that would be able to overcome the negative effect of the multi-rate issue on VoIP quality. Hence, the main goal ofthis research work is to develop an agent that utilizes IP protocols by applying a Cross-Layering approach to eliminate the above-mentioned negative effect. This could be expected from the interaction between Medium Access Control (MAC) layer and Application layer, where the proposed agent adapts the voice packet size at the Application layer according to the change of MAC transmission data rate to avoid network congestion from happening. The agent also monitors the quality of conversations from the periodically generated Real Time Control Protocol (RTCP) reports. If voice quality degradation is detected, then the agent performs further rate adaptation to improve the quality. The agent performance has been evaluated by carrying out an extensive series ofsimulation using OPNET Modeler. The obtained results of different performance parameters are presented, comparing the performance ofVoWLAN that used the proposed agent to that ofthe standard network without agent. The results ofall measured quality parameters hav

Effect of Free Bandwidth on VoIP Performance in 802.11b WLAN Networks

In this paper we experimentally study the relationship between bandwidth utilization in the wireless LAN and the quality of VoIP calls transmitted over the wireless medium. Specifically we evaluate how the amount of free bandwidth decreases as the number of calls increases and how this influences transmission impairments (i.e. delay, loss and jitter) and thus degrades call quality. We show that the amount of free bandwidth is a good indicator for predicting VoIP call quality

VOIP WITH ADAPTIVE RATE IN MULTI- TRANSMISSION RATE WIRELESS LANS

“Voice over Internet Protocol (VoIP)” is a popular communication technology that plays a vital role in term of cost reduction and flexibility. However, like any emerging technology, there are still some issues with VoIP, namely providing good Quality of Service (QoS), capacity consideration and providing security. This study focuses on the QoS issue of VoIP, specifically in “Wireless Local Area Networks (WLAN)”. IEEE 802.11 is the most popular standard of wireless LANs and it offers different transmission rates for wireless channels. Different transmission rates are associated with varying available bandwidth that shall influence the transmission of VoIP traffic

Modelling the IEEE 802.11 wireless MAC layer under heterogeneous VoIP traffic to evaluate and dimension QoE

PhDAs computers become more popular in the home and workplace, sharing resources and Internet access locally is a necessity. The simplest method of choice is by deploying a Wireless Local Area Network; they are inexpensive, easy to configure and require minimal infrastructure. The wireless local area network of choice is the IEEE 802.11 standard; IEEE 802.11, however, is now being implemented on larger scales outside of the original scope of usage. The realistic usage spans from small scale home solutions to commercial ‘hot spots,’ providing access within medium size areas such as cafés, and more recently blanket coverage in metropolitan. Due to increasing Internet availability and faster network access, in both wireless and wired, the concept of using such networks for real-time services such as internet telephony is also becoming popular. IEEE 802.11 wireless access is shared with many clients on a single channel and there are three non-overlapping channels available. As more stations communicate on a single channel there is increased contention resulting in longer delays due to the backoff overhead of the IEEE 802.11 protocol and hence loss and delay variation; not desirable for time critical traffic. Simulation of such networks demands super-computing resource, particularly where there are over a dozen clients on a given. Fortunately, the author has access to the UK’s super computers and therefore a clear motivation to develop a state of the art analytical model with the required resources to validate. The goal was to develop an analytical model to deal with realistic IEEE 802.11 deployments and derive results without the need for super computers. A network analytical model is derived to model the characteristics of the IEEE 802.11 protocol from a given scenario, including the number of clients and the traffic load of each. The model is augmented from an existing published saturated case, where each client is assumed to always have traffic to transmit. The nature of the analytical model is to allow stations to have a variable load, which is achieved by modifying the existing models and then to allow stations to operate with different traffic profiles. The different traffic profiles, for each station, is achieved by using the augmented model state machine per station and distributing the probabilities to each station’s state machine accordingly. To address the gap between the analytical models medium access delay and standard network metrics which include the effects of buffering traffic, a queueing model is identified and augmented which transforms the medium access delay into standard network metrics; delay, loss and jitter. A Quality of Experience framework, for both computational and analytical results, is investigated to allow the results to be represented as user perception scores and the acceptable voice call carrying capacity found. To find the acceptable call carrying capacity, the ITU-T G.107 E-Model is employed which can be used to give each client a perception rating in terms of user satisfaction. PAGE 4 OF 162 QUEEN MARY, UNIVERSITY OF LONDON OLIVER SHEPHERD With the use of a novel framework, benchmarking results show that there is potential to maximise the number of calls carried by the network with an acceptable user perception rating. Dimensioning of the network is undertaken, again compared with simulation from the super computers, to highlight the usefulness of the analytical model and framework and provides recommendations for network configurations, particularly for the latest Wireless Multimedia extensions available in IEEE 802.11. Dimensioning shows an overall increase of acceptable capacity of 43%; from 7 to 10 bidirectional calls per Access Point by using a tuned transmission opportunity to allow each station to send 4 packets per transmission. It is found that, although the accuracy of the results from the analytical model is not precise, the model achieves a 1 in 13,000 speed up compared to simulation. Results show that the point of maximum calls comes close to simulation with the analytical model and framework and can be used as a guide to configure the network. Alternatively, for specific capacity figures, the model can be used to home-in on the optimal region for further experiments and therefore achievable with standard computational resource, i.e. desktop machines