As the Internet evolves into a ubiquitous communication infrastructure and provides various services including telephony, it will be expected to stand up to the toll quality standards set by traditional telephone companies. Our objective in this paper is to assess to what extent today's Internet is meeting this expectation. Our assessment is based on delay and loss measurements taken over wide-area backbone networks, considers realistic VoIP scenarios and uses quality measures appropriate for voice. Our findings indicate that although voice services can be adequately provided by some ISPs, a significant number of paths lead to poor performance even for excellent VoIP end-systems. This makes a strong case for special handling of voice traffic on those paths. Even on the good paths, rare loss events can occasionally cause perceptible degradation of voice quality. Finally, the appropriate choice of the playout buffer scheme for each path was found to be of critical importance for the perceived quality.

As the Internet evolves into a ubiquitous communication infrastructure and provides various services including telephony, it will be expected to meet the quality standards achieved in the public switched telephone network. Our objective in this paper is to assess to what extent today's Internet meets this expectation. Our assessment is based on delay and loss measurements taken over wide-area backbone networks and uses subjective voice quality measures capturing the various impairments incurred. First, we compile the results of various studies into a single model for assessing the VoIP quality. Then, we identify different types of typical Internet paths and we study their VoIP performance. For each type of path, we identify those characteristics that affect the VoIP perceived quality. Such characteristics include the network loss in a path and the delay variability that should be appropriately handled by the playout scheduling at the receiver. Our findings indicate that although voice services can be adequately provided by some ISPs, a significant number of Internet backbone paths lead to poor performance.

High transmission speeds, increased burstiness of traffic, and statistical multiplexing of traffic render traditional approaches to network management and control ineffective. This thesis develops insight into the operation and performance of high-speed networks by developing tractable models and approximations. The insight gained is utilized to propose ways of enhancing the efficiency of network resources and facilitating ease of network management and control. Dynamic routing algorithms for routing Virtual Circuits (VCs) in Asynchronous Transfer Mode (ATM) must take into account their heterogeneous bandwidth characteristics and quality of service requirements. We classify ATM networks according to the network characteristics which have the greatest bearing on the performance of dynamic routing algorithms and discuss appropriate routing algorithms for each cla...

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In a typical real-time voice application, voice packets are produced at deterministically-spaced time intervals. In the network they encounter a variable amount of delay that changes the deterministic time intervals. A receiving host can employ a buffer to delay the playout of the voice packets in order to reconstruct the original timing. Adaptive techniques can perform continuous estimations of the network delays and dynamically adjust the buffering delay at the beginning of each talkspurt. Such adjustments are usually undetectable by the human listener. This research develops a new, adaptive "gapbased " algorithm that can be tuned for both end-to-end delay and packet loss to satisfy a user-desired tolerance. This new gap based algorithm adapts the buffering delay based on historical information of arrival and playout times of received voice packets in the previous talkspurt. A simulation study shows that the new gap based algorithm can reduce delay by 10% when compared with existing ...

In this paper, we present new adaptive playout scheduling (POS) and loss-concealment (LC) schemes for delivering high and consistent conversational voice communication quality (CVCQ) perceived by users in real-time VoIP systems. We first characterize the delay and loss conditions of an IP network and a human conversation in a VoIP system. We then identify the attributes that affect the human perception of CVCQ, which include listening-only speech quality (LOSQ), conversational interactivity (CI), and conversational efficiency (CE). We investigate their trade-offs with respect to system-controllable mouth-to-ear delays (MEDs) and the amount of redundant piggybacking. Finally, we evaluate our adaptive POS and redundancy-based LC schemes by packet traces collected in the PlanetLab. Categories and Subject Descriptors H.4.3 [Information Systems Applications]: Communications

This paper addresses the problem of quantitatively evaluating the quality of a speech stream transported over the Internet as perceived by the end-user. We propose an approach being able to perform this task automatically and, if necessary, in real time. Our method is based on using G-networks (open networks of queues with positive and negative customers) as Neural Networks (in this case, they are called Random Neural Networks) to learn, in some sense, how humans react vis-a-vis a speech signal that has been distorted by encoding and transmission impairments. This can be used for control purposes, for pricing applications, etc.

This paper presents an experiment comparing the relative impact of different types of degradation on subjective quality ratings of interactive speech transmitted over packet-switched networks. The experiment was inspired by observations made during a largescale, long-term field trial of multicast conferencing. We observed that user reports of unsatisfactory speech quality were rarely due to network effects such as packet loss and jitter. A subsequent analysis of conference recordings found that in most cases, the impairment was caused by end-system hardware, equipment setup or user behavior. The results from the experiment confirm that the effects of volume differences, echo and bad microphones are rated worse than the level of packet loss most users are likely to experience on the Internet today, provided that a simple repair mechanism is used. Consequently, anyone designing or deploying network speech applications and services ought to consider the addition of diagnostics and tutorials to ensure acceptable speech quality. Keywords Internet audio, speech, media quality assessment, subjective assessment, multicast conferencing. 1.

This paper is a review of the quality requirements for multimedia network services and some means of ensuring them.

One of the most interesting improvements of the Internet today is the provisioning of services for interactive audio--visual applications. Such applications have quality requirements which place limits on both transfer delays and information losses. Our goal is to support interactive video with only small modifications to network routers and control systems. The approach we favor is to ensure the quality for a session by forward--error correction. The code strength is dynamically tuned to meet a user's quality expectation, given the experienced loss process of the transfer. To make the tuning feasible, the network state must be predictable. We accomplish this by regulation of the load through sender--based admission control. It uses end--to--end probes of the network state and a self--imposed blocking if then sender determines that a sensible transfer cannot be made. The paper outlines this procedure along with a review of source coding considerations and a description of tunable error...