Enhancing Bandwidth Utilization of IP Telephony Over IPv6 Networks

The demand for the telecommunication services,such as IP telephony,has increased dramatically during the COVID-19 pandemic lockdown.IP tele-phony should be enhanced to provide the expected quality.One of the issues that should be investigated in IP telephony is bandwidth utilization.IP telephony pro-duces very small speech samples attached to a large packet header.The header of the IP telephony consumes a considerable share of the bandwidth allotted to the IP telephony.This wastes the network's bandwidth and influences the IP telephony quality.This paper proposes a mechanism(called Smallerize)that reduces the bandwidth consumed by both the speech sample and the header.This is achieved by assembling numerous IP telephony packets in one header and use the header'sfields to carry the speech sample.Several metrics have been used to measure the achievement Smallerize mechanism.The number of calls has been increased by 245.1%compared to the typical mechanism.The bandwidth saving has also reached 68%with the G.28 codec.Therefore,Smallerize is a possible mechanism to enhance bandwidth utilization of the IP telephony.

Effective Hybrid Content-Based Collaborative Filtering Approach for Requirements Engineering

Requirements engineering(RE)is among the most valuable and critical processes in software development.The quality of this process significantly affects the success of a software project.An important step in RE is requirements elicitation,which involves collecting project-related requirements from different sources.Repositories of reusable requirements are typically important sources of an increasing number of reusable software requirements.However,the process of searching such repositories to collect valuable project-related requirements is time-consuming and difficult to perform accurately.Recommender systems have been widely recognized as an effective solution to such problem.Accordingly,this study proposes an effective hybrid content-based collaborative filtering recommendation approach.The proposed approach will support project stake-holders in mitigating the risk of missing requirements during requirements elicitation by identifying related requirements from software requirement repositories.The experimental results on the RALIC dataset demonstrate that the proposed approach considerably outperforms baseline collaborative filtering-based recom-mendation methods in terms of prediction accuracy and coverage in addition to mitigating the data sparsity and cold-start item problems.

Survey and Analysis of VoIP Frame Aggregation Methods over A-MSDU IEEE 802.11n Wireless Networks

The IEEE 802.11n standard has provided prominent features that greatly contribute to ubiquitous wireless networks.Over the last ten years,voice over IP(VoIP)has become widespread around the globe owing to its low-cost or even free call rate.The combination of these technologies(VoIP and wireless)has become desirable and inevitable for organizations.However,VoIP faces a bandwidth utilization issue when working with 802.11 wireless networks.The bandwidth utilization is inefficient on the grounds that(i)80 bytes of 802.11/RTP/UDP/IP header is appended to 10–730 bytes of VoIP payload and(ii)765μs waiting intervals follow each 802.11 VoIP frame.Without considering the quality requirements of a VoIP call,be including frame aggregation in the IEEE 802.11n standard has been suggested as a solution for the bandwidth utilization issue.Consequently,several aggregation methods have been proposed to handle the quality requirements of VoIP calls when carried over an IEEE 802.11n wireless network.In this survey,we analyze the existing aggregation methods of VoIP over the A-MSDU IEEE 802.11n wireless standard.The survey provides researchers with a detailed analysis of the bandwidth utilization issue concerning the A-MSDU 802.11n standard,discussion of the main approaches of frame aggregation methods and existing aggregation methods,elaboration of the impact of frame aggregation methods on network performance and VoIP call quality,and suggestion of new areas to be investigated in conjunction with frame aggregation.The survey contributes by offering guidelines to design an appropriate,reliable,and robust aggregation method of VoIP over 802.11n standard.

Down to Zero Size of VoIP Packet Payload

Voice over Internet Protocol(VoIP)is widely used by companies,schools,universities,and other institutions.However,VoIP faces many issues that slow down its propagation.An important issue is poor utilization of the VoIP service network bandwidth,which results from the large header of the VoIP packet.The objective of this study is to handle this poor utilization of the network bandwidth.Therefore,this study proposes a novel method to address this large header overhead problem.The proposed method is called zero size payload(ZSP),which aims to reemploy and use the header information(fields)of the VoIP packet that is dispensable to the VoIP service,particularly the unicast IP voice calls.In general,these fields are used to carry the VoIP packet payload.Therefore,the size of the payload is reduced to save bandwidth.The performance estimation results of the proposed ZSP method showed a considerable improvement in the bandwidth utilization of the VoIP service.For example,the saved bandwidth in the tested scenario with the G.723.1,G.729,and LPC codecs reached 32%,28%,and 26%respectively.

Multi-indicator Active Queue Management Method

A considerable number of applications are running over IP networks.This increased the contention on the network resource,which ultimately results in congestion.Active queue management(AQM)aims to reduce the serious consequences of network congestion in the router buffer and its negative effects on network performance.AQM methods implement different techniques in accordance with congestion indicators,such as queue length and average queue length.The performance of the network is evaluated using delay,loss,and throughput.The gap between congestion indicators and network performance measurements leads to the decline in network performance.In this study,delay and loss predictions are used as congestion indicators in a novel stochastic approach for AQM.The proposed method estimates the congestion in the router buffer and then uses the indicators to calculate the dropping probability,which is responsible for managing the router buffer.The experimental results,based on two sets of experiments,have shown that the proposed method outperformed the existing benchmark algorithms including RED,ERED and BLUE algorithms.For instance,in the first experiment,the proposed method resides in the third-place in terms of delay when compared to the benchmark algorithms.In addition,the proposed method outperformed the benchmark algorithms in terms of packet loss,packet dropping,and packet retransmission.Overall,the proposed method outperformed the benchmark algorithms because it preserves packet loss while maintaining reasonable queuing delay.