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

Towards the Quality of Service for VoIP traffic in IEEE 802.11 Wireless Networks

The usage of voice over IP (VoIP) traffic in IEEE 802.11 wireless networks is expected to increase in the near future due to widely deployed 802.11 wireless networks and VoIP services on fixed lines. However, the quality of service (QoS) of VoIP traffic in wireless networks is still unsatisfactory. In this thesis, I identify several sources for the QoS problems of VoIP traffic in IEEE 802.11 wireless networks and propose solutions for these problems. The QoS problems discussed can be divided into three categories, namely, user mobility, VoIP capacity, and call admission control. User mobility causes network disruptions during handoffs. In order to reduce the handoff time between Access Points (APs), I propose a new handoff algorithm, Selective Scanning and Caching, which finds available APs by scanning a minimum number of channels and furthermore allows clients to perform handoffs without scanning, by caching AP information. I also describe a new architecture for the client and server side for seamless IP layer handoffs, which are caused when mobile clients change the subnet due to layer 2 handoffs. I also present two methods to improve VoIP capacity for 802.11 networks, Adaptive Priority Control (APC) and Dynamic Point Coordination Function (DPCF). APC is a new packet scheduling algorithm at the AP and improves the capacity by balancing the uplink and downlink delay of VoIP traffic, and DPCF uses a polling based protocol and minimizes the bandwidth wasted from unnecessary polling, using a dynamic polling list. Additionally, I estimated the capacity for VoIP traffic in IEEE 802.11 wireless networks via theoretical analysis, simulations, and experiments in a wireless test-bed and show how to avoid mistakes in the measurements and comparisons. Finally, to protect the QoS for existing VoIP calls while maximizing the channel utilization, I propose a novel admission control algorithm called QP-CAT (Queue size Prediction using Computation of Additional Transmission), which accurately predicts the impact of new voice calls by virtually transmitting virtual new VoIP traffic

A Novel Packet Aggregation Mechanism for Enhancing VoIP Performance on IEEE 802.11 Wireless Mesh Networks

In 802.11 Wireless Mesh Network (WMN), bandwidth will be wasted much for transferring VoIP flows since each voice frame must contain relatively large amount of protocol data. “Packet Aggregation” mechanism can be applied to merge the voice data of multiple VoIP flows into one frame for transmission. It reduces the waste on bandwidth and increases the maximum number of successful VoIP calls. In addition, the mechanism “MCF controlled channel access” (MCCA) defined in 802.11 standard can be used to obtain better QoS than adopting default EDCA mechanism. In MCCA, mesh stations which wants to transfer VoIP flows can reserve time intervals of the medium for transmission and this reservation will be advertised to their neighbors. It is why MCCA causes less medium contentions than EDCA. In this paper, a mechanism to transfer VoIP flows in IEEE 802.11 WMN by MCCA with packet aggregation scheme is proposed. The effectiveness of the proposed mechanism is shown by simulation results. In addition, the problem named as Routing-Packet Aggregation / De-aggregation-Scheduling optimization problem (abbr. RPADS problem) derived from the proposed mechanism is also studied. A heuristic algorithm for RPADS problem to maximize the total number of supported calls is also proposed

Analysis and Performance Evaluation of IEEE 802.11 WLAN

With fast deployment of wireless local area networks VoIP over IEEE 802.11 wireless local area network (WLAN) is growing very fast and is providing a cost effective alternative for voice communications. WLANs were initially set up to handle bursty nonreal time type of data traffic. Therefore, the wireless access protocols initially defined are not suitable for voice traffic. Subsequently, updates in the standard have been made to provision for QoS requirements of data, especially the real time traffic of the type voice and video. Despite these updates, however, transmitting voice traffic over WLAN does not utilize the available bandwidth (BW) efficiently, and the number of simultaneous calls supported in practice is significantly lower than what the BW figures would suggest. Several modifications have been proposed to improve the call capacity, and recently isochronous coordination function (ICF) was introduced to mitigate the problem of low call capacity. The proposed modified ICF which further improves the performance in terms of the call capacity. The proposed scheme uses multiplexing and multicasting in the downlink to substantially increase the call capacity

Final report on the evaluation of RRM/CRRM algorithms

Deliverable public del projecte EVERESTThis deliverable provides a definition and a complete evaluation of the RRM/CRRM algorithms selected in D11 and D15, and evolved and refined on an iterative process. The evaluation will be carried out by means of simulations using the simulators provided at D07, and D14.Preprin

Evaluating the Performance of VoIP Over Different Wireless Environment

WiFi and WiMAX are the most technologies for broadband wireless access are used nowadays. The excessive demand for providing mobile users with broadband wireless access has attracted tremendous investment from the telecommunications industry in the development and deployment of WiMAX and WiFi networks. Voice over IP (VoIP) over these technologies will be one of the killer applications for rapid deployment of WiMAX and WiFi networks. The legal desire for bundling voice and data will increase the portion of voice traffic in these networks. Therefore, VoIP, as the current technology for making voice calls through packet switch networks, will be a key application in WiMAX and WiFi networks. The increase of Voice over IP (VoIP) applications such as Skype, Google Talk, and MSN Messenger along with emerging deployment of WiMAX networks is making VoIP over WiMAX an attractive market and a driving force for both carriers and equipment suppliers in capturing and spurring the next wave of telecommunications innovation, though challenges remain. However, the enhancement on the hardware and application sides only seems inadequate. The chosen of proper network environment is also crucial in order to deliver the voice communication and multimedia session over the Internet. Optimization of the VoIP call capacity over WiMAX and WiFi networks is one such crucial challenge and remains an open research issue. Hence, in this project, we present the performance evaluation of VoIP in two wireless network protocols, WiMAX and WiFi as a baseline to evaluate the VoIP performance