VoIP Services Research Articles

Revisiting VoIP QoE assessment methods: are they suitable for VoLTE?

Voice over LTE (VoLTE) emerges as the basis for voice services in future cellular networks. At the same time, Quality of user Experience (QoE) assessment methods have notably evolved in the last years, especially for VoIP services. The question that rises is, are these models ready to be used for VoLTE? In this paper, we contribute to answer this question by: (i) providing a high level exploration of current objective non-intrusive models for VoIP QoE evaluation, (ii) highlighting the review of works aiming to improve the standardized E-model (ITU-T Rec. G.107 (06/2015)) and the examination of non-standardized solutions proposed by the research community, and (iii) bringing forth the challenges identified for the completion of a successful QoE evaluation approach according to modern VoIP services such as VoLTE.

An Enhanced Perceptive Queuing Technique (CBCRTQ) for Traffic Management in VoIP Over MANET

Traffic management in the VOIP transmission is the most important criteria and issue (to be maintained the packet). The traffic maintains packets and data as a source of outcome for VoIP (QoS) over Manet background. Queuing is one of the most important mechanisms in traffic executive. Mobile Adhoc network (MANETs) present a good platform for the fast deployment of VoIP services in many application scenarios. Ad hoc networks consist a set of identical node that moves freely, independently and communicate with other node via wireless links, the main challenge in extreme sensitivity to delay and packet loss in Voip over Manet. In this research a new algorithm proposed which enhance the performance of traffic load in VOIP carryover MANET environment. Through a simulation and mathematical expression we analyze and evaluate QoS parameters, in the existing method there is no load balanced queuing mechanism and lack in cluster selection prototype. Hence researcher proposed a new Class Based Cluster Round Trip Queue (CBCRTQ) algorithm for VoIP (QoS) over MANET. The main objective of this algorithm selection of cluster heads form a virtual backbone and may be used to route packets for nodes in their cluster. There is a need of traffic load balance at Cluster Round Trip methodology in VoIP, Simulation experiment makes clear that the future algorithm does provide longer cluster head selection and duration with traffic load-balancing.

LTE 이동통신 시스템에서 QoS 변화에 따른 VoIP 서비스의 사용자 체감 품질 변화에 대한 연구

LTE 이동통신 시스템에서 QoS 변화에 따른 VoIP 서비스의 사용자 체감 품질 변화에 대한 연구

PERFORMANCE ANALYSIS OF VOIP OVER WIRED AND WIRELESS NETWORKS NETWORK IMPLEMENTATION IN ADEN UNIVERSITY

In this Paper, the objective of simulation models is presented to investigate the performance of VoIP codecs over WiMAX and FDDI networks that specially design for Aden University. To assure if the University IP network is prepared and adequate for this new type of traffic before adding any new components, Aden University IP network will be simulated by using OPNET simulation software then the new VoIP service will be added to the University networks. Different parameters that represent the QoS like end to end delay, jitter, traffic sends and traffic received, MOS are calculated and analyzed in both network scenarios.

Intelligent Adaptive Energy Efficiency & Effective Signal Buffer Management Algorithm for VoIP (QoS) over Manet

Energy Efficient in the VoIP transmission is the major criteria and issues to be maintained the packet and data while processing VoIP (QoS) over Manet environment. Mobile Adhoc network (MANETs) presents a good platform for the fast deployment of VoIP services in many application scenarios. The energy efficiency mechanism for the data packet of voice in the queuing system have not been concentrated which actually consumes the more power while in the queue buffer management. Manet provide a considerable complexity that makes the transmission of real-time applications like VoIP a great challenge due to insufficient energy for packet transmission. In this research a new algorithm is proposed which enhance the performance of energy in manet carrying VoIP traffic. Through a simulation and mathematical expression we analyze and evaluate QoS indicators such as Distance, power, Bandwidth. Several voice codec are studied to determine their effect of insufficient energy. In the existing methods queuing energy have been used depend on number of nodes in certain environment which is drawbacks with mobile nodes. Hence a QoS based Intelligent Adaptive Energy Efficiency & Effective signal buffer management algorithm for voip(QoS) over Manet. The main objective of this paper is to segregate the energy altitude as per detachment the bandwidth is assign by using our IAE 3 signal barrier algorithm in manet environment. In great and dynamic networks, central algorithms are not suitable due to the lack of responsiveness. The bandwidth allocation is done by IAE 3 signal buffer algorithm.

CPU Load Analysis & Minimization for TCP SYN Flood Detection

Denials of service attacks are well-known as one of the major threats in today's Internet services. Majority VOIP services, DNS servers, online gaming and e-commerce applications are suffering and targeted by hackers using the execution of denial of service attack. Web application attacks and denial of service attacks in distributed architecture is significantly increases day by day. The denial of service attack hampers the load on CPU of web servers during the attack. Therefore, there is a need to minimize the load of CPU after effective attack detection.This paper proposes and implemented denial of service detection framework which consists of packet sniffer, feature extraction, attack detection and output module. The proposed framework detects denial of service attack such as TCP SYN Flood based on threshold and misuse detection. The system is analyzed with the help of CPU load and the load of CPU is minimized after TCP SYN flood attack detection.

Analysis of Burst Ratio in Concatenated Channels

Burst ratio is a parameter that quantifies packet loss patterns in transmission networks. It has been defined for an end-to-end scenario, therefore burst ratio can be determined only if the characteristics of the whole transmission path are known. In this paper, the burst ratio parameter applicability to cases when the transmission path consists of a series of transmission channels with known packet loss rate and burst ratio values is extended. The paper also presents the results of simulations performed with NS2 software, demonstrating the validity of the burst ratio analysis. Consequently, the research makes it possible to determine the value of the burst ratio parameter in concatenated packet networks, which in turn supports delivering higher quality VoIP services.

SEAMLESS VOIP SERVICE USING CENTRALIZED CODEC ADJUSTMENT: A FRAMEWORK IN WIRELESS NETWORK

The evolution of wireless technology leads VoIP application to be a good preferred choice among mobile users. Customers can having a VoIP service while being mobile regardless of location, terminal, and it is independent of network access. It is believed to be an alternative to the PSTN and its service is expected to be as good as PSTN. In essence, QoS plays a crucial part in order to maintain the seamless service during the VoIP session. As wireless networks are prone to channel error and VoIP application is known to be a delay-sensitive application, network traffic congestion could easily affect the voice quality of VoIP. Therefore, the codec selection within the VoIP application must be improved so that the codec can operate at various bit rates in order to cater for different traffic conditions. In this paper, we explain how centralize codec switching can help user to communicate seamlessly. We use WiFi network traffic as an example scenario.

An Approach to Design and Reduce Cost of IP Telephony using SIP TRUNKING

This work gives an over view of IP telephony and explain the existing difficulties in implementing VoIP services & migrate the TDM base telephony to IP telephony. When the architecture is deployed in IP based network. The causes of these problems are explained with three approaches used by carrier to solve interoperability. They are TDM base telephony to IP telephony by using soft switch and voice gateway, using different audio codec’s for better voice quality, band width used in IP telephony. General Terms Routing, Rate, Currentcalls, Server

해상 VoIP 서비스를 위한 통합 커뮤니케이션 기술 개발

해상 VoIP 서비스를 위한 통합 커뮤니케이션 기술 개발

Queuing Disciplines for VoIP and High Resolution Video Application Services Using OPNET Tool

Queuing Disciplines for VoIP and High Resolution Video Application Services Using OPNET Tool

SoftMAC: Layer 2.5 Collaborative MAC for Multimedia Support in Multihop Wireless Networks

In this paper, we present the challenges in supporting multimedia, in particular, VoIP services over multihop wireless networks using commercial IEEE 802.11 MAC DCF hardware, and propose a novel software solution, called Layer 2.5 SoftMAC. Our proposed SoftMAC resides between the IEEE 802.11 MAC layer and the IP layer to coordinate the real-time (RT) multimedia and best-effort (BE) data packet transmission among neighboring nodes in a multihop wireless network. To effectively ensure acceptable VoIP services, channel busy time and collision rate need to be well controlled below appropriate levels. Targeted at this, our SoftMAC architecture employs three key mechanisms: 1) distributed admission control for regulating the load of RT traffic, 2) rate control for minimizing the impact of BT traffic on RT one, and 3) nonpreemptive priority queuing for providing high priority service to VoIP traffic. To evaluate the efficacy of these mechanisms, extensive simulations are conducted using the network simulator NS2. We also implement our proposed SoftMAC as a Windows network driver interlace specification (NDIS) driver and build a multihop wireless network testbed with 32 wireless nodes equipped with IEEE 802.11 a/b/g combo cards. Our evaluation and testing results demonstrate the effectiveness of our proposed software solution. Our proposed collaborative SoftMAC framework can also provide good support for A/V streaming in home networks where the network consists of hybrid WLAN (wireless LAN) and Ethernet

WiFi Network Virtualization to Control the Connectivity of a Target Service

This paper proposes a WiFi network virtualization technique to control the connectivity of a target service. The packet-level delay violation ratio can be reduced even in a congested situation by provisioning dedicated base station (BS) resources (a set of dedicated BSs) to the target service and allowing only the corresponding terminals to associate with the BSs. The proposed technique is novel in that BSs are specially configured to use the same MAC address, and thus all the decisions on BS selection and handover are separated from those BSs and terminals and are put together into a centralized controller, while consistent layer-2 data paths in a backhaul network are also cooperatively configured. Simulation results show that the proposed technique can control the delay violation ratio of a target VoIP service and keep the ratio extremely low and comparable to that under IEEE 802.11e. A proof-of-concept prototype including two multi-channel virtualization-capable WiFi BSs and a BS switch is developed using off-the-shelf WiFi modules and a commercial OpenFlow switch. Experimental results show that the terminals can make a handover to a dedicated BS in less than 65 ms without any packet drop and association break, and confirm that the effect of the managed handover is limited even in a VoIP application.

Yagi Antenna Design and Analysis 11 Folded Dipole Feeding Elements for VoIP Networks

The problem that often occurs in VOIP networks using wireless media is the weak signal received and transmitted by the transceiver device at a long distance from the wireless router, so that the quality of VOIP service is often disconnected. The purpose of this research is to design and realize an 11 element yagi antenna with folded feed elements. dipole as an antenna for a transceiver device whose gain is better than a built-in antenna, so that it can support VOIP services. Yagi antenna is an antenna consisting of 3 elements, namely a reflector element, a driven element and a director element. The reflector element and the director element cause the yagi antenna to be directional so that a large gain is obtained from a certain point, while the driven one is used is a folded dipole antenna which has an impedance of 300 ohms so that the matching technique to the 50 ohm transmission line becomes easier. Ghz is less than 1.5 and RL is less than -14 dB. The bandwidth obtained is 620 MHz, the gain is 16.08 dBi with directional radiation and linear polarization. In implementation, the VOIP service is stable at a distance of 50 m using a yagi antenna, compared to the less stable built-in antenna.

Implementation and Analysis of Wireless VoIP Networks with Routerboard Management at State Polytechnic of Malang

In this article, the implementation of wirelessVoIP at State Polytechnic of Malang uses microtic management with the aim of knowing the QoS (Quality of Services) performance between systems without microtic management and systems using microtic management. In the implementation of this system, two tests were carried out, namely QoS testing when without proxy management and when using proxy management. From the two tests, the performance will be compared by doing data compilation using the VQ manager software. The QoS parameters to be taken are delay, jitter, packet loss and throughput. From the measurement it is known that when the VoIP server serves <= 3 calls simultaneously, the MOS value between the managed system (MOS = 3.7) and the system without management (MOS = 3.7) is almost the same because the value of delay and packet loss in the system without management and systems with management did not differ much, namely 107 ms and 83 ms, and the packet loss value was the same, namely 5%. And when serving> 3 simultaneous calls, there is a difference of 0.18 from the MOS value between the managed system (MOS = 3.48) and a system without management (MOS = 3.3) with a value of delay and packet loss for systems without management and systems with management, namely 527 ms and 340 ms, and the packet loss values ??were the same, namely 8% and 7.2%.

Capture and Analysis of Malicious Traffic in VoIP Environments Using a Low Interaction Honeypot

The number of users of VoIP services is increasing every year. Consequently, VoIP systems get more attractive for attackers. This paper describes the implementation of a low interaction honeypot for monitoring illegal activities in VoIP environments. The honeypot operated during 92 days and collected 3502 events related to the SIP protocol. The analysis of the results allows understanding the modus operandi of the attacks targeted to VoIP infrastructures. These results may be used to improve defense mechanisms such as firewalls and intrusion detection systems.

VoIP Codec Selection for Digital PLC Systems

IP-based networks developments highly integrate the communication systems and significantly reduce the installation cost. Digital PLC systems with the capability of Ethernet and VoIP services are designed because of the demand of new communication services in the electric power industry. Also due to the limited bandwidth in digital PLC systems, there are several techniques for optimal use of transmission capacity. In this paper, the possibility of using VoIP services in digital PLC systems has been investigated. Many conventional standards are studied for several parameters such as total algorithm delay, voice payload size/duration, packets per second, and bandwidth to select the best codec. Finally, a proper codec is recommended to employ in the digital PLC's VoIP service according to the two important factors in quality of voice services, i.e. available bandwidth and latency.

VoIP service model for multi-objective scheduling in cloud infrastructure

Voice over IP VoIP is very fast growing technology for the delivery of voice communications and multimedia data over internet with lower cost. Early technical solutions mirrored the architecture of the legacy telephone network. Now, they have adopted the concept of distributed cloud VoIP. These solutions typically allow dynamic interconnection between users on any domains. However, providers face challenges to use infrastructure in the best efficient and cost-effective ways. Hence, efficient scheduling and load balancing algorithms are a fundamental part of this approach, especially in presence of the uncertainty of a very dynamic and unpredictable environment. In this paper, we formulate the problem of dynamic scheduling of VoIP services in distributed cloud environments and propose a model for bi-objective optimisation. We consider it as the special case of the bin packing problem, and discuss solutions for provider cost optimisation while ensuring quality of service.

데이터 트래픽 중심의 LTE망에서 VoIP를 위한 스케줄링 알고리즘 성능 분석

최근 국내외에서는 LTE가 도입 초기단계를 넘어, LTE-A로의 도약이 이루어지고 있다. 이를 기반으로 고속의 서비스를 제공함으로써, 이동 전화로 시작한 휴대폰의 역할은 데이터 중심의 스마트폰으로 바뀌고 있다. 시스코에 의하면 2018년 전세계 모바일 데이터 트래픽은 지난해 대비 11배 증가 전망할것으로 내다봤다. 한국에서 모바일 비디오가 전국 모바일 트래픽에서 차지하는 비중은 2013년 66%에서 2018년 75%로 증가할 전망이다. 하지만 여전히 음성 서비스가 가장 중요하며, 높은 VoIP 통화 품질을 제공하기 위해서는 다양한 데이터 트래픽을 사용하는 가운데 LTE 망 상황에 따라 전송 속도를 조절하고, 낮은 BLER을 제공해야 한다. 실제 통화를 위한 전송율이 최대 23.85 Kbps인 AMR-WB 코덱을 사용하지만, 초당 수십~수백 MB까지 전송할수 있는 LTE일지라도 셀 가장자리의 사용자에게 일정 수준의 전송률을 떨어뜨리지 않고 유지하는 것은 어려운 일이다. 본 논문에서는 이러한 점을 개선하기 위해 특히 문제가 많이 발생하는 셀 가장자리의 품질을 개선하고자, 다양한 스케줄링 알고리즘의 성능을 분석하고 개선 방안을 제시하였다. 제시된 방안은 2계층 스케줄링 알고리즘을 사용하며, 셀 가장자리에 위치한 VoIP 사용자를 일정 비율 타 어플리케이션에 비해 우선 처리하는 스케줄링 알고리즘을 제공하는 기술이다. 본 논문에서는 throughput과 BLER 측면에서 VoIP 성능이 향상되었음을 확인하였다.

On-line Monitoring of VoIP Quality Using IPFIX

The main goal of VoIP services is to provide a reliable and high-quality voice transmission over packet networks. In order to prove the quality of VoIP transmission, several approaches were designed. In our approach, we are concerned about on-line monitoring of RTP and RTCP traffic. Based on these data, we are able to compute main VoIP quality metrics including jitter, delay, packet loss, and finally R-factor and MOS values. This technique of VoIP quality measuring can be directly incorporated into IPFIX monitoring framework where an IPFIX probe analyses RTP/RTCP packets, computes VoIP quality metrics, and adds these metrics into extended IPFIX flow records. Then, these extended data are stored in a central IPFIX monitoring system called collector where can be used for monitoring purposes. This paper presents a functional implementation of IPFIX plugin for VoIP quality measurement and compares the results with results obtained by other tools.