Improving WLAN VoIP Capacity Through Service Differentiation

Research on Voice over IP (VoIP) in infrastructure wireless local area networks (WLANs) has shown that, because of the access mechanism, the access point severely limits the number of voice calls a WLAN can support. To address this problem we propose a new scheme based on the IEEE 802.11e quality of service (QoS) mechanism where a tradeoff between codec quality and priority is exploited to improve the number of calls that can be supported. In particular, we propose certain priority settings at the access point (AP) to encourage users to switch to a lower quality codec during periods of high contention and thus enable them to maintain the call. We develop a detailed analytical model to show the benefit of this scheme. Our analytical results are validated by extensive simulation and show that the voice capacity in the network can be significantly improved. Furthermore, by using the ITU-T E-model to assess the voice quality, we show that users with a lower quality codec can still maintain an acceptable level of quality using the proposed scheme.

The Voice over Internet Protocol (VoIP) application has observed the fastest growth in the world of telecommunication. VoIP is seen as a short-term and long-trem transmission for voice and audio traffic. Meanwhile, VoIP is moving on Wireless Local Area Networks (WLANs) based on IEEE 802.11 standards. Currently, there are many packet scheduling algorithms for real-time transmission over network. Unfortunately, the current scheduling will not be able to handle the VoIP packets with the proper manner and they have some drawbacks over real-time applications. The objective of this research is to propose a new Voice Priority Queue (VPQ) packet scheduling and algorithm to ensure more throughput, fairness and efficient packet scheduling for VoIP performance of queues and traffics. A new scheduler flexible which is capable of satisfying the VoIP traffic flows. Experimental topologies on NS-2 network simulator wereanalyzed for voice traffic. Preliminary results show that this can achieve maximum and more accurate VoIP quality throughput and fairness index in access point for VoIP over WLANs. We verified and validated VPQ an extensive experimental simulation study under various traffic flows over WLANs.

This paper investigates the resource allocation schemes for Voice over Internet Protocol (VoIP) traffic under high speed train scenario, in which 2-state VoIP traffic model based on Markov chain is considered. Unlike the full-buffer traffic, in which the transmitter buffer is always full of data to be transmitted, VoIP traffic can be characterized by small packets in regular intervals with tight delay constraints. Considering this, two different resource allocation schemes are discussed in this paper. One is an existing scheme adopting fixed Modulation and Coding scheme (MCS) without packet-bundling, which is commonly used for VoIP scheduling in traditional cellular networks; the other one is our proposed scheme, which uses adaptive MCS with packet-bundling and is suitable for broadband wireless communication under high-speed train scenario. System-level simulation is conducted and the simulation results show that the proposed scheme outperforms the conventional resource allocation scheme in terms of the VoIP outage probability and VoIP capacity under high-speed train scenario.

This project is to simulate the Voice over Internet Protocol (VoIP) over IEEE 802.11b and 802.11e WLANs (Wireless Local Area Network) with different Coder-Decoder (CODECs) by using Network Simulator-2 (NS-2). The main objectives of this project are to analyze the VoIP of Mean Opinion Score (MOS), throughput and packet drop rate. The simulation is conducted using different CODECs (G.711, G.729A and G.723.1) and simulated without background traffic (data). The simulation in this project is using 1 wired node and 1 access point connected with mobile nodes in hierarchy. The mobile nodes will send VoIP calls to wired node through access point simultaneously. From the project, it is found that the CODEC G.711 provides the best quality of VoIP calls. However, CODEC G.723.1 can support the most VoIP calls with acceptable quality. In addition, with the first priority given by 802.11e to the VoIP, it can provide a very good perceived audio quality.

Voice over Internet Protocol (VoIP) is no longer only feasible in wired networks, but also applicable through wireless networks as prevalence of wireless networks grows with the technology maturity and rising demand. While making a VoIP call over WiFi wireless networks is applicable, this is not a service with guaranteed quality. With limited bandwidth, an overloaded AP may cause wireless users to experience delayed packets, high packet loss rate, and even low throughput. IEEE 802.11 wireless Local Area Network (WLAN) was originally designed only for data traffic transmission, and therefore deficiency in real-time traffic transmission, such as VoIP applications, is inevitable. Not being able to timely transmit real-time packets not only fails to meet the demand of users, but also increases unnecessary loading in the wireless network. Industries and academies have made lots of efforts on providing QoS for real-time voice traffic which usually coexists with other classes of traffic. In this thesis, a call admission control mechanism is proposed which not only timely monitors the wireless networks, but also dynamically adjusts the number of admissible clients to wireless networks. New VoIP users not only know the network condition before entering it, but also accurately predict transmission quality through sending probing packets. With the mechanism, existing VoIP users in the wireless networks have their service protected, and potential VoIP users can expect good quality of service once admitted.

There has been an explosive growth in the use of wireless LANs (WLANs) to support network applications ranging from web-browsing and file-sharing to voice calls. It is difficult to optimally configure WLAN components, such as access points (APs), to meet the quality-of-service requirements of the different applications, as well as ensuring flow-level fairness. Recent work has shown that the widely-deployed IEEE 802.11 MAC Distributed Coordination Function (DCF) is biased against downstream flows. The new IEEE 802.11e standard introduces QoS mechanisms, such as Enhanced Distributed Channel Access (EDCA), that allow this unfairness to be addressed. So far, only limited work has been done to evaluate the impact of these MAC protocols on TCP-based applications. In this paper, through ns-2 simulations, we evaluate the impact of EDCA on TCP application traffic consisting of both long and short-lived TCP flows. We find that the performance of TCP applications is very dependent upon the settings of the EDCA parameters and buffer lengths at the AP. We also show that the performance of the admission control strategy employed depends on the buffer lengths at the AP and the traffic intensity.

IEEE 802.11e provides basic mechanisms for traffic differentiation in WLANs previously unavailable with the IEEE 802.11 standard. The enhanced distributed channel access (EDCA) is probably the most promising subset of such mechanisms, and it is already implemented in a huge number of Network Cards. In this paper we present an algorithm for the provisioning of QoS in EDCA WLANs. The algorithm is based on monitoring and reactive configuration of EDCA mechanisms in scenarios where voice over IP (VoIP) and best effort (BE) applications concurrently access the wireless channel. Results obtained in a real test-bed prove the effectiveness of the algorithm both in protecting VoIP communications and in providing minimal quality levels and fairness to BE applications.

Nowadays, wireless communication technologies are being employed in an ever increasing number of different application areas, including industrial environments. Benefits deriving from such a choice are manifold and include, among the others, reduced deployment costs, enhanced flexibility and support for mobility. Unfortunately, because of a number of reasons that have been largely debated in the literature, wireless systems cannot be thought of as a means able to fully replace wired networks in production plants, in particular, when real-time behavior is a key issue. In this paper, an analysis of the real-time performance that can be achieved in quality-of-service (QoS)-enabled 802.11 networks has been carried out. In particular, a detailed analysis of latencies and packet loss ratios for a typical enhanced distributed channel access (EDCA) infrastructure wireless local area network (WLAN) is presented, obtained through numerical simulations. A number of aspects that may affect suitability for the use in control systems have been taken into account, including the Transmission Opportunity (TXOP) mechanism, the internal architecture of the AP, the use of a time-division multiple access (TDMA)-based communication scheme as well as the adoption of broadcast communications.

Internet protocol (IP) telephony is sensible in the view of providing platform for faster information exchange using voice calls and its demand is increasing. IP telephony in heterogeneous network environments can prove its dominating characteristics over circuit switched network. General packet radio service (GPRS) is mostly used network in India. Interface, using serving GPRS support node (SGSN) gateway for IP telephony, to provide channel to all users using wireless local area network (WLAN) to make voice calls to GPRS mobile user, can provide ultimate solution for the need of cheaper and flexible calling in the areas where service provider's coverage range is quite poor. Interconnecting WiFi and GPRS as mentioned in unlicensed mobile access (UMA) can provide more reliable and single device multi interface platform and add more flexibility to its users having handsets with dual radio interface capability. The performance evaluation has been done for UMA using network simulator. Analysis indicates that the system can perform better with less end to end delay, jitter, packet loss and optimum throughput.

IEEE 802.11 wireless local area networks (WLAN) have been widely deployed, and many applications are implemented on WLAN. As one of the interesting wireless services, wireless voice over internet protocol (VoIP) can provide low-cost real-time voice service to mobile users. But the IEEE 802.11 wireless network cannot satisfy the QoS of VoIP. The delay during handoff process can last up to 1 second. Applications with delay restraint like VoIP will drop the call when the delay is more than 50 ms. Therefore this paper analyzed the delay made by the handoff process, and proposed a pre-scanning fast handoff scheme. This fast handoff scheme is based on selective scanning and caching. It can predict the next AP for STAs and use pre-scanning before the handoff is triggered. Simulations on OPNET show this proposed scheme can reduce the average handoff delay to 23.45 ms and avoid some unnecessary handoffs. The pre-scanning fast handoff scheme is feasible and economical according to its performance in the handoff process.

The interference effects on voice over Internet protocol (VoIP) applications over wireless local area networks (WLANs) are here dealt with. In particular, attention is paid to the IEEE 802.11 g standard, with Bluetooth signals, additive white Gaussian noise (AWGN) and competitive data traffic acting as interferer. The analysis is conducted experimentally following a cross-layer approach and through the deployment of a proper testbed within a semi-anechoic chamber. The purpose is to investigate on the effects of some common types of interference on a WLAN supporting VoIP applications, and to deduce from the experiments valuable hints for improving the setup of a WLAN in the presence of interference. Cross-layer measurements are performed at network/transport, physical and application layer.

The changing usage of a mature campus-wide wireless network

IEEE 802.11e EDCA(Enhanced Distributed Channel Access)는 서로 다른 AC(Access Category)에 따라서 CW(Contention Window)를 이용하여 Quality-of-Service(QoS)를 지원한다. 그러나 무선 네트워크 환경이 혼잡 (Congested)할 경우 패킷의 충돌 발생 확률을 높일 수 있다는 문제점이 여전히 존재한다. 이를 해결하기 위해 여러 방법들이 제시되었으나 혼잡 네트워크상에서는 동일 우선순위를 가지는 패킷 전송을 위해서 전송 채널을 이용하는 경쟁 과정에서 패킷 충돌(Collision)이 발생할 확률이 여전히 존재한다. 따라서 본 논문에서는 EDCA 포화 상태에서 전송 효율을 높이는 APCA(Advanced Priority Collision Avoidance) 알고리즘을 제안한다. 제안된 알고리즘은 IEEE 802.11e 표준을 기반으로 RTS/CTS (Request to Send / Clear to Send)를 이용하여 패킷 전송 FC(Frame Control) 필드의 예약된 필드(Reserved Field) 비트를 이용하여 데이터 패킷 충돌을 회피하는 것이다. 시뮬레이션의 결과로 제안된 알고리즘이 기존의 EDCA에 비해 패킷 전송 실패율이 감소했음을 보였다. Jain's fairness index를 이용하여 제안된 APCA 알고리즘이 네트워크 혼잡 상황에서 EDCA 방식 보다 공정성을 유지하는 것을 확인하였다. The Enhanced Distributed Channel Access (EDCA) function of IEEE 802.11e standard defines contention window (CW) for different Access Category (AC) limits to support Quality-of-Service (QoS). However, it have been remained the problem that the collision probability of transmission is increasing in congested network. Several different solutions have been proposed but the collision occurs among same priority queue within the same station to compete the channel access. This paper presents an APCA (Advanced Priority Collision Avoidance) algorithm for EDCA that increases the throughput in saturated situation. The proposed algorithm use reserved field's bits of FC(Frame Control) using IEEE 802.11e standard's RTS/CTS (Request to Send / Clear to Send) mechanism to avoid data collision. The simulation results show that the proposed algorithm improves the performance of EDCA in packet loss. Using Jain's fairness index formula, we also prove that the proposed APCA algorithm achieves the better fairness than EDCA method under network congested condition.

In IEEE 1609, it uses IEEE 802.11 Enhanced Distributed Channel Access (EDCA) mechanism to access the channel. IEEE 802.11 EDCA is a new wireless technology for wireless access in the vehicular environment (WAVE). It defines a new supplement to the existing IEEE 802.11 MAC protocol. In IEEE 802.11 EDCA, the aim is providing a QoS support. While the system serves different access categories (ACs), EDCA does not perform well under high load conditions. In order to improve the efficiency, we pay attention to the EDCA with transmit opportunity (TXOP) mechanism. We first proposed a Markov chain model and studied the behavior. We extend the model to support IEEE 802.11 EDCA and presented a more accurate analysis under nonideal channel environment. We also compared it with that without TXOP mechanism under channel error environment.

The popular IEEE 802.11 WLAN today does not provide any quality of service (QoS) because of its contention‐based channel access nature of the medium access control (MAC). A simple software upgrade‐based solution, called a dual queue scheme, can be used to provide QoS to real‐time services such as Voice‐over‐IP (VoIP). The dual queue scheme operates on top of the legacy MAC. In this paper, we present a modified dual queue (MDQ) scheme by considering a practical implementation limitation. On the other hand, the emerging 802.11e MAC includes a contention‐based channel access mechanism, called enhanced distributed channel access (EDCA). Prioritized channel accesses are provided to different traffic classes under EDCA.In this paper, we comparatively evaluate the performances of MDQ and EDCA via simulations in terms of delay, jitter and throughput for various scenarios. We also compare the two access schemes when multiple transmission rates are used via rate adaptation mechanism for reliable transmission over erroneous wireless channel. The simulation results show that EDCA apparently provides a better performance in most situations. However, MDQ provides an acceptable performance close to that of EDCA so that it can be used when the EDCA is not available and/or the hardware upgrade to 802.11e is not desirable for the cost reason. Copyright © 2006 AEIT