An empirical study of delay jitter management policies (original) (raw)
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Elsevier eBooks, 2002
This paper presents an empirical study of several policies for managing the effect of delay jitter on the playout of audio and video in computer-based conferences. The problem addressed is that of managing the fundamental tradeoff between display with low latency and display with few gaps. We describe a particular policy called queue monitoring which observes delay jitter over time and dynamically adjusts display latency in order to support low-latency conferences with an acceptable gap rate. Queue monitoring is evaluated by comparing it with two policies from the literature in a study based on measurements from a computer-based conferencing system. Our results show that queue monitoring performs as well or better than the other policies over the range of observed network loads. More importantly, we show that queue monitoring performs better on those network loads for which the other policies exhibit poor performance.
Queue Monitoring: A Delay Jitter Management Policy
Network and Operating System Support for Digital Audio and Video, 1993
This paper describes queue monitoring, a policy for managing the effect of delay jitter on audio and video in computer-based conferences. By observing delay jitter over time, this policy dynamically adjusts display latency in order to support low-latency conferences with an acceptable gap rate. Queue monitoring is evaluated by comparing it with two other policies in an empirical study of
Proceedings of the 4th ACM Multimedia Systems Conference on - MMSys '13, 2013
This paper proposes a novel approach for improving the quality of experience (QoE) of real-time video conferencing systems. In these systems, QoE is affected by signal quality as well as interactivity, both depending on the packet loss rate, delay jitters, and mouth-toear delay (MED) that measures the sender-receiver delay on audio signals (and will be the same as that of video signals when video and audio is synchronized). We notice in the current Internet that increasing MED as well as reducing packet rate can help reduce the delay-aware loss rate in congested connections. Between the two methods, the former plays a more important role and applies well to a variety of network conditions for improving audiovisual signal quality, although overly increasing the MED will degrade interactivity. Based on a psychophysical concept called just-noticeable difference (JND), we find the extent to which MED can be increased, without humans perceiving the difference from the original conversation. The approach can be applied to improve existing video conferencing systems. Starting from the operating point of an existing system, we increase its MED to within JND in order to have more room for smoothing network delay spikes as well as recovering lost packets, without incurring noticeable degradation in interactivity. We demonstrate the idea on Skype and Windows Live Messenger by designing a traffic interceptor to extend their buffering time and to perform packet scheduling/recovery. Our experimental results show significant improvements in QoE, with much better signal quality while maintaining similar interactivity.
2004
Latency and jitter inherently limit the maintenance of consistency in Distributed Interactive Applications such as computer games, distributed whiteboards and real-time, collaborative environments. Although there has been much research into methods for maintaining consistency, there is a distinct lack of research exploring the connection between latency, jitter and the end user experience in Distributed Interactive Applications.
Fast Queuing Policies for Multimedia Applications
2013 22nd International Conference on Computer Communication and Networks (ICCCN), 2013
We present an analytical framework for providing Quality of Service (QoS) using queuing policies that achieves a given target distribution of packets in a network queue. To a large extent, the stationary distribution of packets in the queue resulted from employing a certain queuing policy directly controls the typical QoS metrics for multimedia applications. Therefore, using the packet distribution in the queue as the metric, the proposed framework allows for a more general and precise control of QoS beyond the standard metrics such as bandwidth, jitter, loss, and delay. Moreover, the proposed framework aims to find a fast queuing policy that achieves a given target stationary distribution. This fast adaptation is especially useful for multimedia applications in fast-changing network conditions. As an example, we present a general procedure for obtaining a queuing policy that optimizes for a given arbitrary objective along with the standard QoS requirements. Both theory and simulation results are presented to verify our framework.
Jitter Control and Dynamic Resource Management for Multimedia Communication Over Broadband Network
1998
Abstract Due to the isochronous nature of multimedia streams, jitter management; represents a major challenge to synchronized and timely presentation of multimedia documents. For controlling jitter, we propose the Initial Delay Regulator (IDR). This regulator is based on a network model which abstracts jitter parameters inside a data network. As a result of employing such a regulator, jitter at the destination node can be reduced. and possibly eliminated. depending on resource availability.
Multimedia streaming gateway with jitter detection
IEEE Transactions on Multimedia, 2000
This paper investigates a novel active buffer management scheme, "Jitter Detection" (JD) for gateway-based congestion control to stream multimedia traffics in packet-switched networks. The quality of multimedia presentation can be greatly degraded due to network delay variation or jitter when transported over a packet-switched network. Jitter degrades the timing relationship among packets in a single media stream and between packets from different media streams and, hence, creates multimedia synchronization problems. Moreover, too much jitter will also degrade the performance of the streaming buffer in the client. Packets received by the client will be rendered useless if they have accumulated enough jitter. The proposed active buffer management scheme will improve the quality of service in multimedia networking by detecting and discarding packets that accumulated enough jitter, such as to maintain a high bandwidth for packets within the multimedia stream's jitter tolerance. Simulation results have shown that the proposed scheme can effectively lower the average received packet jitter and increase the goodput of the received packets when compared to random early detection (RED) and DropTail used in gateway-based congestion control. Furthermore, simulation results have also revealed that the proposed scheme can maintain the same TCP friendliness when compared to that of RED and DropTail used for multimedia streams.
In this paper we present the results of an empirical study that examines user's perception -understood as both information assimilation and subjective satisfaction -of multimedia quality, when impacted by varying network-level parameters (delay and jitter). Results showed that whilst, the type of multimedia video content impacts user information assimilation, there was no statistically significant difference in the level as a result of error inclusion. The latter result demonstrates that delay and jitter does not negatively impact information assimilation; an important result for broadcast and distributed educational applications.
Delay Control Using Media Sensitivity in Multimedia Environments
MODARES TECHNICAL AND ENGINEERING, 2010
Increasing real time services in multimedia environments has initiated a new phenomenon in data communication. This paper analyzes delay in multimedia environments focusing on multi-point to point communication [1]. We separate the playtime delay in a point-to-point transmission and propose an optimization scenario for each part. It is proved that sum of the normalized path delay for point-to-point connections is invariant and based on this property, the playtime for delay-sensitive media has been minimized. We have shown that in transmission media, priority queuing is an effective solution where in the receiver side, waiting time for playtime scheduling and queuing discipline are two main factors. It is shown that there is a compromise between packet loss and packet departure time in the receiver side where the acceptable packet loss can adjust the playtime delay adaptively. Theoretical analysis for priority assignment, queuing technique and performance evaluation in different classes of queuing with different playtime scheduling are given.