Bengi Karacali

Assessing network readiness for IP telephony

Networked multimedia applications require stringent real-time QoS guarantees. Successful deployment of such applications closely depends on the performance of the underlying data network. The characteristics and the QoS requirements of these applications are different from traditional data applications. Hence, prior to deployment it is necessary to evaluate a network from a multimedia perspective. In this paper, we focus on IP Telephony and describe a framework for providing tools for IP Telephony readiness evaluation. This framework can be easily generalized to other multimedia applications. Our approach injects synthesized voice traffic and measures perceived end-to-end quality. We present a novel idea of relating voice quality metrics to the performance of data network devices. Following the proposed framework, we developed a prototype tool to evaluate a network and to identify problems, if any, prior to IP Telephony deployment. Our tool automatically discovers the topology of a given network, and collects and integrates network device performance and voice quality metrics. We describe the architecture of our tool and provide sample outputs from a network consisting of 129 devices.

Scalable network assessment for IP telephony

Multimedia applications such as IP telephony are among the applications that demand strict quality of service (QoS) guarantees from the underlying data network. At the predeployment stage it is critical to assess whether the network can handle the QoS requirements of IP telephony and fix problems that may prevent a successful deployment. In this paper we describe a technique for efficiently assessing network readiness for IP telephony. Our technique relies on understanding link level QoS behavior in a network from an IP telephony perspective. We use network topology and end-to-end measurements collected from the network in locating the sources of performance problems that may prevent a successful IP telephony deployment. We present an empirical study conducted on a real network spanning three geographically separated sites of an enterprise network. The empirical results indicate that our approach efficiently and accurately pinpoints links in the network incurring the most significant delay.

Experiences with evaluating network QoS for IP telephony

Successful deployment of networked multimedia applications such as IP telephony depends on the performance of the underlying data network. QoS requirements of these applications are different from those of traditional data applications. For example, while IP telephony is very sensitive to delay and jitter, traditional data applications are more tolerant of these performance metrics. Consequently, assessing a network to determine whether it can accommodate the stringent QoS requirements of IP telephony becomes critical. We describe a technique for evaluating a network for IP telephony readiness. Our technique relies on the data collection and analysis support of our prototype tool, ExamiNet/spl trade/. It automatically discovers the topology of a given network and collects and integrates network device performance and voice quality metrics. We report the results of assessing the IP telephony readiness of a real network of 31 network devices (routers/switches) and 23 hosts via ExamiNet/spl trade/. Our evaluation identified links in the network that were over utilized to the point at which they could not handle IP telephony.

Measurement Techniques in On-Demand Overlays for Reliable Enterprise IP Telephony

Maintaining good quality of service for real-time applications like IP Telephony requires quick detection and reaction to network impairments. In this paper, we propose and study novel measurement techniques in ORBIT, which is a simple, easily deployable architecture that uses single-hop overlays implemented with intelligent endpoints and independent relays. The measurement techniques provide rapid detection and recovery of IP Telephony during periods of network trouble. We study our techniques via detailed simulations of several multi-site enterprise topologies of varying sizes and three typical fault models. We show that our proposed techniques can detect network impairments rapidly and rescue IP Telephony calls in sub-second intervals. We observed that all impacted calls were rescued with only a few relays in the network and the run-time overhead was low. Furthermore, the relay sites needed to be provisioned with minimal additional bandwidth to support the redirected calls.

Network-wide inference of end-to-end path intersections

Network topology information has many uses for networked applications including the design of reliability and performance enhancing schemes. This information is often acquired by network discovery techniques that rely on ICMP and/or SNMP support from the infrastructure. Unfortunately, network infrastructure may support such services at varying levels. In cases where such support is limited, techniques based on strictly end-to-end measurements have been proposed in the literature to infer a logical topology. In this paper, we propose an end-to-end measurement technique that is complimentary to the earlier techniques and relies on multiple-source multiple-destination probing to collect network wide measurements faster. Our technique relies on the interaction between streams of packets at various nodes in the network. Specifically, we aim to measure the reaction of end-to-end probes to a short burst of concentrated traffic (signal) injected to the network. The key assumption is that probes traversing paths that physically intersect with the path of the bursty traffic (signal path) will exhibit extra jitter. The occurrence of extra jitter in the probe packet inter-arrival times matching the signal bursts indicates that the paths of the probe and signal intersect. We conducted an empirical study on an actual production network to evaluate the ability of the technique to detect end-to-end path intersections. Our empirical results indicate that in the network we considered the technique successfully detected path intersections.

Scalable network monitoring for multimedia applications in enterprise networks

Networked multimedia applications require stringent quality of service (QoS) guarantees from the underlying data network. Many techniques have been proposed at the network layer to deliver acceptable QoS for such applications. In this paper, we explore providing network monitoring support in the application layer for QoS policies. We study the feasibility of a scalable network monitoring service that real-time applications may access to gather network status information in order to adapt to changing network conditions and take actions such as making application level routing decisions. We propose a scalable, low probing overhead monitoring service based on monitoring utilization on network devices traversed by multimedia traffic. Our motivation is to take advantage of often intersecting end-to-end paths traversed by multimedia traffic in an enterprise setting. We empirically assess the QoS improvements resulting from a service that monitors network utilization for an application involving dynamic server selection on a real production network. Specifically, we compare the performance of various dynamic server selection strategies for multimedia streaming including random selection, round-trip time based selection, proximity based selection, and utilization based selection. Our results indicate that compared to random selection, server selection strategies that rely on our network utilization based monitoring scheme result in significantly lower end-to-end delay and packet loss in streaming sessions. Compared to round-trip time based selection, our scheme results in comparable end-to-end delay and loss figures while incurring significantly lower probing cost

Measuring video quality degradation using face detection

Ensuring end-to-end video quality requires monitoring quality in real-time (in-service) and taking counter-measures in times of adverse network conditions. Such application-layer QoS assurance mechanisms require light-weight video quality metrics that can be implemented with low computational and communication overheads. In this paper, we propose a novel video quality metric for video conferencing-type applications that accurately reflects user opinion and is light-weight for realtime operations. Our motivation is to exploit the characteristics of the video content in such applications, i.e. few speakers with limited motion. Our metric, Simplified Perceptual Quality Region (SPQR), relies on detecting the location of a speakers face in sent and received video frames and comparing the locations between the corresponding frames in the two streams to identify discrepancies as a sign of video quality degradation. Our experiments show that face locations can be determined in realtime by sampling few frames every second. SPQR is a reduced-reference metric that requires minimal transmission overhead between the sender and receiver through a separate channel to communicate the reduced features. In this paper, we present an empirical evaluation of the performance of SPQR using a video phone application. We first show that SPQR effectively detects video quality degradation. Second, we compare our proposed metric to two well-accepted full-reference techniques appropriate for offline analysis, namely PSNR and VQM, and show that SPQR tracks both metrics well. Finally, we show that low grade sampling yields SPQR values comparable to PSNR and VQM scores and thus enabling a light-weight implementation.

Network Instrumentation for End-to-End Measurements

An important consideration when deploying realtime applications such as IP Telephony is how to ensure that the network delivers the stringent QoS requirements of these applications. Prior to deployment, assessing whether an enterprise network can handle IP Telephony ensures a successful deployment. A network assessment approach recently introduced in the ExpertNetTM tool suite involves distributing small measurement devices in the network. These devices inject synthetic RTP traffic and collect end-to-end measurements that represent end-to-end call quality. In this approach, the placement of measurement devices impacts the network coverage and accuracy of the analysis. In this paper, we address the problem of how to place such measurement devices in a network to ensure a comprehensive analysis. Specifically, we focus on two issues: first where to place network measurement devices when the network topology is known and second how to estimate the number of such devices in the absence of network topology information. In this paper, we address the first problem by providing two heuristic algorithms for placing measurement devices in a given network. We evaluate these heuristics using synthetic networks. We address the second problem by proposing a technique to estimate the number of measurement devices using the size of a network. To evaluate the performance of our technique, we conducted a study on actual enterprise network topologies. Using our technique, we predicted the number of endpoints based on network size and compared these figures to the number of endpoints placed by the proposed algorithms using the network topology. Our results indicate that the predictions are comparable to the number of endpoints placed using the actual enterprise network topology.

Path switching and call protection for reliable IP telephony

Application-layer path switching systems ensure the performance and reliability of real-time applications (such as IP Telephony) by exploiting inherent redundancies in the underlying IP network. These systems monitor quality throughout an IP Telephony call, rapidly detect performance problems and, if necessary, re-direct calls around problems and congestion in the network. Under normal network load conditions, these systems can easily find a feasible alternate path and restore call quality within sub-second timeframes. However, at high network load, finding alternate paths is challenging due to inherent inaccuracies in the techniques that attempt to measure an end-to-end path’s ability to accommodate additional calls. These inaccuracies may adversely impact established calls on the path and some of the calls may even oscillate back and forth between various alternate paths. In this paper, we propose some novel techniques that address these problems and ensure system stability and performance at high network load. We introduce the concept of Spatial Slow Start, which allows rapid selection of a feasible path from a large possible set. In addition, we help protect a path’s established calls from other switching calls by using intelligent back-off and probationary admission control techniques. We conducted extensive simulation studies to evaluate the performance of the proposed techniques. The results indicate that our techniques significantly reduce the number of oscillations and the amount of time that calls experience poor quality.

Network assessment for low bitrate video

Assessing the impact of network performance on video quality of experience is critical for successful video deployments in enterprises. Typically, network assessments review network LAN and WAN capabilities, QoS mechanisms, and ensure that adequate bandwidth is allocated for video applications. However, most techniques cannot accurately relate network performance to video quality since typically network measurements are poor indicators of video quality. The impact of network degradation on quality depends on many factors (e.g. the type of video frame impacted by the fault, error concealment capabilities of the video codec, application level QoS control mechanisms etc.) and complicates developing a general model of video quality based on network measurements. In this paper, we address the issues associated with developing such a general model by focusing on a subset of video applications with characteristics better suited to network measurement based quality models. Specifically, we focus on low bitrate video conferencing applications with small picture sizes (CIF, QCIF) and low motion levels. In this paper, we study the feasibility of modeling the quality of such applications using network measurements. We report the results of experiments we conducted using the low bitrate configurations of a video phone application running on a PC. For the configurations we considered, our analysis of video call session traces of this application under varying network conditions indicates that network loss measurements estimate video quality reasonably well.