To Improve The Voice Quality Over IP Using Channel Coding

Collaborative Editing Systems often use communication channels in order to make the collaboration more effective. In this work, we present a case study of pair communication using audio via VoIP (Voice over Internet Protocol) technology during collaborative modeling sessions. The study provides an analysis of audio and textual chat as communication medium and presents data on usage patterns, user interaction and attitudes when using collaborative editing systems. The qualitative and quantitative analysis suggests that VoIP technology does have advantages over textual chat in collaborative modeling when used for communication.

In order to provide the same or better service quality in the Internet than traditional circuit-switched telephone network, a number of issues have to be solved which have hampered it in the Internet. Therefore, in this paper the VoIP (Voice over Internet Protocol) network design problems are modeled as nonlinear non convex combinatorial integer mathematical programming problems. The optimization problems have great practice value for the network service providers. This problem is a constrained multicast QoS routing and is proved to be a NP complete problem. The total link capacity augmented cost should be minimized when we design a new VoIP network or when the original network could not serve all of the traffic demands. It is also known as a complicated problem. Introduction Originally, the Internet was designed to provide best-effort services for the data generated by computers. The timeliness of data communication is generally delay tolerant. Quality-of-Service (QoS) constraints are not as important as routing flexibility and connectivity. Hence, using IP to transport voice data is contradictory to the basic requirement of the voice service: a timely delivery of voice samples. Although IP was not initially designed to provide services for real-time traffic, recent technical progress has made IP have the capabilities to provide real-time services in near future. In order to provide the same or better service quality in the Internet than traditional circuit-switched telephone network, we must deal with a number of issues that have hampered it in the Internet. Voice service requirements could be discussed from two perspectives: (1) application requirements such as end-to-end delay, jitter, packet loss and overdue probability; (2) user’s perspective such as reliability, availability, and supplementary services [1][2]. Telephony service providers must guarantee the quality of service they provide e.g. maximum one way delay does not exceed 150ms. Internet telephony requires a range of protocols, ranging from those needed for transporting real-time data across to the network e.g. Real-time Transport protocol (RTP) [3], to Quality-of-Service aware routing (QoS routing), signaling protocol, resource reservation, internetworking between IP networks and PSTN, QoS-aware network management and billing protocols. ITU-T defined H.323 to provide multimedia communication in packet networks. From the perspective of network service providers, they want to optimize the network performance such as minimizing the total bandwidth consumption, maximizing throughput or total revenue [4][5] subject to user and application constraints. In this paper, we want to develop the mathematical model for VoIP network. We minimize the total bandwidth consumption under users’ QoS requirements, the network topology and the network capacity. Performance Optimization Model It is generally accepted that Internet telephony and traditional circuit-switched telephony will coexist for quite some time. The VoIP architecture must deal with interworking between IP networks and PSTN, so we need gateways between the two worlds. There are four possible models of VoIP [6]. They are PC-to-PC, Gateway-to-Gateway, PC-to-Gateway, and Gateway-to-PC models. The architecture of VoIP is shown in Fig.1. The first model of VoIP is PC-to-PC architecture, which based 2nd International Conference on Electrical, Computer Engineering and Electronics (ICECEE 2015) © 2015. The authors Published by Atlantis Press 1144 on the assumption that two or more users have access to multimedia computers that are connected to the Internet. Fig.1 VoIP Architecture The VoIP system is modeled as a graph, where the hosts, VoIP gateways, and switches are represented by nodes and communication link sets are represented by links. Let N = {1, 2,..., n} be the set of nodes and L be the set of links in the graph (network). Let G be the set of all user groups. An user group g is a voice communication session requesting for transmission in the network. An user group g may be an unicast from the source to a destination or a multicast from the source to multiple destinations. For each user group g, the traffic is transmitted exactly over one tree. Dg represents the set of destinations of user group g. g λ is the required bandwidth of the voice transmission for each user group g. Below is a verbal description of the VoIP system design problem we considered. Given : Network topology Capacities of network links Equivalent bandwidth of each multicast/unicast group Time threshold and tolerable overdue probability for each multicast/unicast group To determine : The minimum overall bandwidth consumption Routing tree for each multicast/unicast group Overdue probability of each multicast/unicast group Objective : To minimize the total bandwidth consumption Subject to : End-to-end QoS (overdue probability) constraint Tree constraint Multi-commodity flow constraint Capacity constraint Integrality constraint Hop constraint Table 1 and 2 are the notations we use in this paper . Internet VoIP Gateway VoIPGatewa Central office switch or local hub Central office switch or local hub Multimedia PC IP Router Multimedia PC IP Router Gatekeepe r

Service providers and enterprises all over the world are rapidly deploying Voice over IP (VoIP) networks because of reduced capital and operational expenditure, and easy creation of new services. Voice traffic has stringement requirements on the quality of service, like strict delay and loss requirements, and 99.999% network availability. However, IP networks have not been designed to easily meet the above requirements. Thus, service providers need service quality management tools that can proactively detect and mitigate service quality degradation of VoIP traffic. In this paper, we present active and passive probes that enable service providers to detect service impairments. We use the probes to compute the network parameters (delay, loss and jitter) that can be used to compute the call quality as a Mean Opinion Score using a voice quality metric, E-model. These tools can be used by service providers and enterprises to identify network impairments that cause service quality degradation and take corrective measures in real time so that the impact on the degradation perceived by end-users is minimal

New methods for voice quality evaluation for IP networks

Voice over Internet protocol (VoIP) is becoming increasingly popular due to its significant advantages regarding cost and support of enhanced multimedia services. Despite of the substantial advantages of using the Internet as the transmission medium for voice calls, we can foresee many undesirable uses, especially in terms of spam. VoIP technology may provide new means for the transmission of bulk unsolicited messages and calls (spam over Internet telephony - SPIT), mainly due to the reduced cost compared to traditional telephony. Therefore, mechanisms are essential to address the SPIT problem, while maintaining the advantages of VoIP technology. In this paper we conduct a comprehensive threat analysis that addresses the ascendant VoIP protocol, the session initiation protocol, in terms of its vulnerability regarding SPIT deliverance. Our analysis introduces the requirements that the mechanisms dealing with the SPIT phenomenon should take into account, in order to be feasible and efficient.

Systems employing the Internet and Voice over Internet Protocol (VoIP) are alternative to the legacy landline telephony system. The services offered by these systems allow users to communicate with their family members, friends, banks and business partners whenever they are online. Particularly, the VoIP service is popular with the Internet users because charges are bound into usually fixed access costs making the price of the long distance calls themselves appear economical and even free. However, VoIP services have several disadvantages. One of the disadvantages of VoIP is, it is exposed to Internet security vulnerabilities, threats and attacks. The attacks come in many forms. One of the methods to attack the VoIP service, is by sending a Spam over Internet Telephony (SPIT). SPIT is normally executed by malicious parties initiating automated, unsolicited, and unwanted communications that used VoIP or video conferencing services, just like an email SPAM. Solutions to mitigate SPIT are still lacking. To defense a VoIP system from SPIT, we have come out with the idea of Automated Intrusion Detection and Prevention System over SPIT (AIDPoS) that can detect and prevent SPIT from external parties to enter the VoIP network. AIDPoS is a combination system of Intrusion Detection System (IDS) and Intrusion Prevention System (IPS) on Voice over IP (VoIP) server. This is an experimental research to find a solution to combat SPIT if it happens in the future. Our result shows that an automated Intrusion detection and Prevention system over SPIT attack is feasible. Our work is on an open source VoIP network. Other vulnerabilities and threats are important, but beyond our research scope and is not covered in this paper.

This paper has evaluated the system performance for the voice over Internet protocol (VoIP) services supported by the IEEE 802.16 systems in terms of the packet loss and delay, and it has also calculated the mean opinion score (MOS) in order to estimate the user's satisfaction for the VoIP services. By the simulation results, the MOS of the VoIP service supported by the unsolicited grant service (UGS) without auto repeat request (ARQ) has been decreased by 1.5 when the channel quality is poor. The reason of the MOS reduction is the increase of the packet loss due to the channel error. For this reason, this paper has considered the ARQ in order to compensate the packet loss when the channel quality is poor. The simulation results have shown that the extended real-time polling service (ertPS) and rtPS with ARQ can compensate the packet loss even if the channel quality is poor, however they can cause the reduction of the VoIP capacity because of the resource used for the retransmission. From the simulation results, the ertPS could save the resource by about 25 % compared to the rtPS with guaranteeing a quality of service (QoS).

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 a technology that transports voice data packets across packet switched networks using the Internet Protocol (IP). Perceived speech quality is the key metric for QoS for VoIP applications. The most popular objective measurement method for VoIP voice is E-Model. The primary aim of this work is to investigate the how the delay affect the VoIP quality and how to combined the affection in to standard E-Model. In this paper, a testing environment to measure the impairment of different delay jitter is setup. The Mean Opinion Scores (MOS) were calculated, Perceptual evaluation of speech quality (PESQ) algorithm is used. We use stand deviation of delay to describe the delay effect on voice quality, and give a new Ij formula, which is added to the standard E-model, as jitter impairment factor. Good accuracy is achieved by our new extended E-model.

Voice over Internet Protocol (VoIP) is a popular communication service nowadays. VoIP reduces the cost of call transmission by passing voice and video packets through the available bandwidth for data packets through Internet protocol. The quality of the VoIP signal is degraded due to the various network impairments. The proposed scheme, interpolated finite impulse response, is implemented as post-processor after decoding the signal in VoIP system. The performance of the proposed scheme is evaluated for various network conditions. The results of the proposed scheme are measured with the objective measurement methods for signal quality evaluation. The performance of the proposed system is compared with the existing techniques for quality improvement in VoIP system. The results show much improvement in speech quality with the proposed scheme in comparison to other similar schemes.

Recently, plenty of voice quality measuring devices of VoIP(Voice over Internet Protocol) service is developed to support high quality VoIP service for communication. However, the existing voice quality measuring devices select the unique measurement method, which uses PESQ(Perceptual Evaluation of Speech Quality) or R-factor only. However, the each voice quality measurement method differs from its usage and assumed range, which leads to the slightly different results at the same testing area for voice services. At the actual VoIP service business and related regulatory organizations, they request to use the above two types of algorithm at the same time for measuring voice quality. However, the development expenses are very high to use the above two types of algorithm at the same time for measuring voice quality due to additional resources and subsidiary devices. VoIP business and regulatory agencies also request the hardware specification of the voice quality measurement device is similar to that of VoIP service user terminal, considering VoIP service supply at the wireless mobile internet such as WiBro(Wireless Broadband). In this paper, we present the voice quality measurement method by using of the above two algorithms at the one physical measurement device at the same time for more accurate and reliable VoIP voice quality measurement. In addition, we introduce the development method of the voice quality measurement device with relatively low cost.

Cable telecommunication technology allows high-speed bi-directional transfer of Internet protocol (IP) traffic, between the CATV headend system and customer locations, over all-coaxial or hybrid fiber-coax (HFC) cable network. With the rapid progress of VoIP (voice over IP) technology, IP telephony service via cable networks has become a preferred strategy for cable data service providers. This paper presents an implementation of VoIP cable modem, which is named DCM-100. Design issues, hardware architecture, software architecture, and performance testing results are described.

Old telecommunication systems are gradually being replaced by a new system that works over IP networks, which is known as voice over internet protocol (VoIP). VoIP has several merits (e.g., very cheap call rate), which make it increasingly popular in the telecommunication world. However, VoIP faces numerous impediments that decelerate its promotion. One of the top impediments is the wasted bandwidth caused by VoIP systems. Numerous methods have been proposed to handle this impediment, including packet aggregation methods. This paper proposes a novel aggregation method, called packet aggregation and carrier header (PA-CH), to reduce the amount of the large bandwidth caused by VoIP. As the name suggests, PA-CH saves in bandwidth by aggregating the packets in a header and using the redundant fields in the packet header to carry a portion of the packet voice data. The performance of the introduced PA-CH method was investigated based on three main metrics, namely, link capacity, allocated bandwidth reduction, and voice data shortening. Simulation results indicate that the proposed PA-CH method outperforms the comparison methods in three factors. For instance, the proposed method’s allocated bandwidth reduction ratio reaches 51% when the number of calls running concurrently reaches 100. Therefore, the proposed PA-CH method achieves its goal of reducing the wasted bandwidth caused by VoIP.

Visual and Vocal communication can be transferred through Circuit switched Network or Packet Switched Network. Public Switched Telephone Network (PSTN) is not an affordable option therefore over existing packet switched network, Voice over Internet Protocol (VoIP) has become a preferable alternative due to its reduced cost. However, despite its reduced cost it has so many challenges which affect its successful deployment. This is because; the quality of VoIP is mainly affected by jitter, delay, packet loss and some other parameters. This research was carried out to evaluate voice quality in VoIP experimentally under different scenarios using OPNET network simulator. A VoIP network was simulated using Riverbed modeler academic edition 17.5 and the behavior and quality of VoIP was studied and analyzed under different scenarios. The results of the analysis and the performance evaluation are presented in this paper. This work can guide researchers and designers to design a network for VoIP services and its deployment. It can also guide the operators to choose speech compression technique for better voice quality.

Voice over IP (VoIP) is a developing technology and a key factor in both the emerging cyberspace engineering and also an accomplishment to set up its position in the telecom industry. VoIP technology is based on internet technology; where data packets switching system is used rather than circuit switching. Whereas, an analog signal is changed over into the digital signals in the full-duplex transmission. VoIP technology is replaced with the conventional public switched telephone network (PSTN) system due to the high flexibility and low-cost. The purpose of this research work is to deliberate experimental and computational performance in the view of quality of service (QoS) parameters of VoIP over local area network (LAN) network. The VoIP systems implementation is based on two different operating system framework (Linux and Windows), whereas, Linux-based and Windows-based private branch exchanges (PBXs), such as Asterisk (Linux-open source) and Axon (window-close source) are configured, installed and verified. QoS factors (such as packet loss, delay, jitter, etc.) are observed over the Asterisk and Axon PBXs in a LAN domain with the assistance of Paessler switch activity grapher (PRTG) monitoring tool. The validations of results are looked at for QoS parameters crosswise over both PBXs with data load (i.e., file transfer and HTTP traffic) during VoIP calls. The productivity and execution of Axon and Asterisk have been equated and analyzed over experimental based outcomes.