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

Overview of a media convergence centre (MC2)

Organizational alliances are rapidly being formed as a means for effective cooperation with a common goal within a targeted value chain. The combination of such communication, coordination and cooperation leads to new organisational forms and scenarios within the Digital Ecosystem space that require technological support. Convergence refers to the move towards the use of a single united interaction medium and media. Such a solution enables telecommunications services that are concurrently coupled with enterprise and internet data. Due to the versatile nature of today's extended enterprise, a flexible, feature-rich, adaptive and widely accessible converged solution is required.This paper proposes a Media Convergence Centre (MC:) solution that allows users to participate in a converged multimedia collaboration network using a variety of interaction devices in an easy and convenient manner

IMPROVING QoS OF VoWLAN VIA CROSS-LAYER BASED ADAPTIVE APPROACH

Voice over Internet Protocol (VoIP) is a technology that allows the transmission of voice packets over Internet Protocol (IP). Recently, the integration of VoIP and Wireless Local Area Network (WLAN), and known as Voice over WLAN (VoWLAN), has become popular driven by the mobility requirements ofusers, as well as by factor of its tangible cost effectiveness. However, WLAN network architecture was primarily designed to support the transmission of data, and not for voice traffic, which makes it lack ofproviding the stringent Quality ofService (QoS) for VoIP applications. On the other hand, WLAN operates based on IEEE 802.11 standards that support Link Adaptive (LA) technique. However, LA leads to having a network with multi-rate transmissions that causes network bandwidth variation, which hence degrades the voice quality. Therefore, it is important to develop an algorithm that would be able to overcome the negative effect of the multi-rate issue on VoIP quality. Hence, the main goal ofthis research work is to develop an agent that utilizes IP protocols by applying a Cross-Layering approach to eliminate the above-mentioned negative effect. This could be expected from the interaction between Medium Access Control (MAC) layer and Application layer, where the proposed agent adapts the voice packet size at the Application layer according to the change of MAC transmission data rate to avoid network congestion from happening. The agent also monitors the quality of conversations from the periodically generated Real Time Control Protocol (RTCP) reports. If voice quality degradation is detected, then the agent performs further rate adaptation to improve the quality. The agent performance has been evaluated by carrying out an extensive series ofsimulation using OPNET Modeler. The obtained results of different performance parameters are presented, comparing the performance ofVoWLAN that used the proposed agent to that ofthe standard network without agent. The results ofall measured quality parameters hav

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

Quality of media traffic over Lossy internet protocol networks: Measurement and improvement.

Voice over Internet Protocol (VoIP) is an active area of research in the world of communication. The high revenue made by the telecommunication companies is a motivation to develop solutions that transmit voice over other media rather than the traditional, circuit switching network. However, while IP networks can carry data traffic very well due to their besteffort nature, they are not designed to carry real-time applications such as voice. As such several degradations can happen to the speech signal before it reaches its destination. Therefore, it is important for legal, commercial, and technical reasons to measure the quality of VoIP applications accurately and non-intrusively. Several methods were proposed to measure the speech quality: some of these methods are subjective, others are intrusive-based while others are non-intrusive. One of the non-intrusive methods for measuring the speech quality is the E-model standardised by the International Telecommunication Union-Telecommunication Standardisation Sector (ITU-T). Although the E-model is a non-intrusive method for measuring the speech quality, but it depends on the time-consuming, expensive and hard to conduct subjective tests to calibrate its parameters, consequently it is applicable to a limited number of conditions and speech coders. Also, it is less accurate than the intrusive methods such as Perceptual Evaluation of Speech Quality (PESQ) because it does not consider the contents of the received signal. In this thesis an approach to extend the E-model based on PESQ is proposed. Using this method the E-model can be extended to new network conditions and applied to new speech coders without the need for the subjective tests. The modified E-model calibrated using PESQ is compared with the E-model calibrated using i ii subjective tests to prove its effectiveness. During the above extension the relation between quality estimation using the E-model and PESQ is investigated and a correction formula is proposed to correct the deviation in speech quality estimation. Another extension to the E-model to improve its accuracy in comparison with the PESQ looks into the content of the degraded signal and classifies packet loss into either Voiced or Unvoiced based on the received surrounding packets. The accuracy of the proposed method is evaluated by comparing the estimation of the new method that takes packet class into consideration with the measurement provided by PESQ as a more accurate, intrusive method for measuring the speech quality. The above two extensions for quality estimation of the E-model are combined to offer a method for estimating the quality of VoIP applications accurately, nonintrusively without the need for the time-consuming, expensive, and hard to conduct subjective tests. Finally, the applicability of the E-model or the modified E-model in measuring the quality of services in Service Oriented Computing (SOC) is illustrated

A hybrid and cross-protocol architecture with semantics and syntax awareness to improve intrusion detection efficiency in Voice over IP environments

Includes abstract.Includes bibliographical references (leaves 134-140).Voice and data have been traditionally carried on different types of networks based on different technologies, namely, circuit switching and packet switching respectively. Convergence in networks enables carrying voice, video, and other data on the same packet-switched infrastructure, and provides various services related to these kinds of data in a unified way. Voice over Internet Protocol (VoIP) stands out as the standard that benefits from convergence by carrying voice calls over the packet-switched infrastructure of the Internet. Although sharing the same physical infrastructure with data networks makes convergence attractive in terms of cost and management, it also makes VoIP environments inherit all the security weaknesses of Internet Protocol (IP). In addition, VoIP networks come with their own set of security concerns. Voice traffic on converged networks is packet-switched and vulnerable to interception with the same techniques used to sniff other traffic on a Local Area Network (LAN) or Wide Area Network (WAN). Denial of Service attacks (DoS) are among the most critical threats to VoIP due to the disruption of service and loss of revenue they cause. VoIP systems are supposed to provide the same level of security provided by traditional Public Switched Telephone Networks (PSTNs), although more functionality and intelligence are distributed to the endpoints, and more protocols are involved to provide better service. A new design taking into consideration all the above factors with better techniques in Intrusion Detection are therefore needed. This thesis describes the design and implementation of a host-based Intrusion Detection System (IDS) that targets VoIP environments. Our intrusion detection system combines two types of modules for better detection capabilities, namely, a specification-based and a signaturebased module. Our specification-based module takes the specifications of VoIP applications and protocols as the detection baseline. Any deviation from the protocol’s proper behavior described by its specifications is considered anomaly. The Communicating Extended Finite State Machines model (CEFSMs) is used to trace the behavior of the protocols involved in VoIP, and to help exchange detection results among protocols in a stateful and cross-protocol manner. The signature-based module is built in part upon State Transition Analysis Techniques which are used to model and detect computer penetrations. Both detection modules allow for protocol-syntax and protocol-semantics awareness. Our intrusion detection uses the aforementioned techniques to cover the threats propagated via low-level protocols such as IP, ICMP, UDP, and TCP

Creation of value with open source software in the telecommunications field

Tese de doutoramento. Engenharia Electrotécnica e de Computadores. Faculdade de Engenharia. Universidade do Porto. 200