Novel Method of Improving Quality of Service for Voice over Internet Protocol Traffic in Mobile Ad Hoc Networks

In recent years, the application of Mobile Ad-hoc Network (MANET) with Voice over Internet Protocol (VoIP) has been increased.  However, the level of Quality of Service (QoS) for VoIP traffic in MANET, while there is no infrastructure, will reduce when dealing with a large number of calls. In this type of dynamic environment, the developing of a new infrastructure becomes more costly and time-consuming. In this paper, we proposed an efficient method, called the Quality of Service-Nearest Neighbor (QoS-NN), to improve the QoS level for VoIP in order to manage the huge number of calls over MANET network. We utilized the Ad-hoc On-demand Distance Vector (AODV) protocol as the underlying routing protocol to implement our proposed method. We evaluated the proposed QoS-NN method using Network Simulator version 2 (NS2). The performance of the proposed QoS-NN method was compared with Lexicographic order method. The comparison was evaluated in terms of R-factor, end-to-end delay, packet loss ratio, and packet delivery ratio performance metrics. In addition, the proposed method evaluated under different network parameters such as VoIP CODECs, node mobility speed, number of calls and number of nodes. The comparison results indicate that the proposed QoS-NN outperform the Lexicographic order method

Performance Evaluation of VoIP in Mobile WiMAX; Simulation and Emulation studies

Worldwide Interoperability for Microwave Access (WiMAX) is an acronym for IEEE 802.16 family which is a leading contemporary broadband wireless Access (BWA) technology. IEEE 802.16e is intended for mobile WiMAX, which supports vehicular mobility with the stringent quality of service (QoS) parameters for various data traffics. Voice over IP (VoIP) provides low cost, modern telephony which can become a better alternative for classical telephony; however there are some issues need to be addressed prior to the deployment of any new technology. Significance of simulation study results can be verified and assessed by emulation testbed results. It is expected that both the results should match closely with each other. This paper makes an effort to study the performance evaluation of VoIP for a mobile user and how the QoS parameters vary for different speeds. The simulation and emulation of a mobile WiMAX system using EXata 2.0.1 are performed. The effectiveness of the comparison of results is discussed

Voice Traffic over Mobile Ad Hoc Networks: A Performance Analysis of the Optimized Link State Routing Protocol

This thesis investigates the performance of the Optimized Link State Routing (OLSR) protocol on Voice over Internet Protocol (VoIP) applications in Mobile Ad hoc Networks (MANETs). Using VoIP over MANETs takes advantage of the mobility and versatility of a MANET environment and the flexibility and interoperability a digital voice format affords. Research shows that VoIP-like traffic can be routed through an ad hoc network using the Ad hoc On-demand Distance Vector routing protocol. This research determines the suitability of OLSR as a routing protocol for MANETs running VoIP applications. Representative VoIP traffic is submitted to a MANET and end-to-end delay and packet loss are observed. Node density, number of data streams and mobility are varied creating a full-factorial experimental design with 18 distinct scenarios. The MANET is simulated in OPNET and VoIP traffic is introduced using one source node to send traffic to random destinations throughout the network. Simulation results indicate delay is between 0.069 ms to 0.717 ms, which is significantly lower than the recommended 150 ms threshold for VoIP applications. Packet loss is between 0.32% and 9.97%, which is less than the 10% allowable packet loss for acceptable VoIP quality. Thus OLSR is a suitable routing protocol for MANETs running VoIP applications

Voice over Internet Protocol over Wireless Local Area Network: A Review

The use of Voice over Wireless Local Area Network is seeing a meteoric rise in popularity as a result of its simplicity, non-intrusiveness, and cheap cost of implementation, as well as its low cost of maintenance, universal coverage, and fundamental roaming capabilities. Nevertheless, deploying Voice over Internet Protocol (VoIP) over Wireless Local Area Network (WLAN) is a challenging task for many network managers, architects, planners, designers, and engineers. Because of this, there is a need for a guideline to design, model, and simulate the network before it is deployed. In this work, a variety of models, including mathematical, theoretical, statistical, and graphical models, that are used to measure the quality and features of VoIP are discussed

A Survey of Bandwidth Optimization Techniques and Patterns in VoIP Services and Applications

This article surveys the various techniques adopted for optimising bandwidth for VoIP services over the period 1999-2014. The improvement of bandwidth can be realized through; silence suppression measure of repressing the silent portions (packets) in a voice conversation using Voice Activity Detection algorithm; by so doing, the transmission rate during the inactive periods of speech is reduced, and thus, the mean transmission rate can be reduced. A second measure is packet header reduction which defines a process of multiplexing and de-multiplexing packet headers to curb excesses. Voice/ Packet Header compression is considered the most productive of all the techniques, offering a scheme where VoIP packets are compressed from the 40 bytes of size to a smaller byte size of 2 bytes. When combined with aggregation, compression potentially yields a compressed size of up to 1 byte. In either case, bandwidth save is reached using compression and decompression codecs of varying data and bit rates. It is envisaged that an improvement in the performance of codecs would yield a better result in terms of enhancing results favourably in Voice over broadband networksComment: 8 pages, 7 figures. ISSN (Print): 1694-0814 | ISSN (Online): 1694-078

Quality of Service optimisation framework for Next Generation Networks

Within recent years, the concept of Next Generation Networks (NGN) has become widely accepted within the telecommunication area, in parallel with the migration of telecommunication networks from traditional circuit-switched technologies such as ISDN (Integrated Services Digital Network) towards packet-switched NGN. In this context, SIP (Session Initiation Protocol), originally developed for Internet use only, has emerged as the major signalling protocol for multimedia sessions in IP (Internet Protocol) based NGN. One of the traditional limitations of IP when faced with the challenges of real-time communications is the lack of quality support at the network layer. In line with NGN specification work, international standardisation bodies have defined a sophisticated QoS (Quality of Service) architecture for NGN, controlling IP transport resources and conventional IP QoS mechanisms through centralised higher layer network elements via cross-layer signalling. Being able to centrally control QoS conditions for any media session in NGN without the imperative of a cross-layer approach would result in a feasible and less complex NGN architecture. Especially the demand for additional network elements would be decreased, resulting in the reduction of system and operational costs in both, service and transport infrastructure. This thesis proposes a novel framework for QoS optimisation for media sessions in SIP-based NGN without the need for cross-layer signalling. One key contribution of the framework is the approach to identify and logically group media sessions that encounter similar QoS conditions, which is performed by applying pattern recognition and clustering techniques. Based on this novel methodology, the framework provides functions and mechanisms for comprehensive resource-saving QoS estimation, adaptation of QoS conditions, and support of Call Admission Control. The framework can be integrated with any arbitrary SIP-IP-based real-time communication infrastructure, since it does not require access to any particular QoS control or monitoring functionalities provided within the IP transport network. The proposed framework concept has been deployed and validated in a prototypical simulation environment. Simulation results show MOS (Mean Opinion Score) improvement rates between 53 and 66 percent without any active control of transport network resources. Overall, the proposed framework comes as an effective concept for central controlled QoS optimisation in NGN without the need for cross-layer signalling. As such, by either being run stand-alone or combined with conventional QoS control mechanisms, the framework provides a comprehensive basis for both the reduction of complexity and mitigation of issues coming along with QoS provision in NGN

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