Results 1 - 10
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197
Adaptive FEC-based error control for Internet telephony
- in Proc. IEEE INFOCOM
, 1999
"... www.inria.fr/rodeo/{bolot,sfosse} ..."
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Modeling of Packet Loss and Delay and their Effect on Real-Time Multimedia Service Quality
- PROCEEDINGS OF NOSSDAV '2000
, 2000
"... Internet packet loss and delay exhibits temporal dependency. If packet n is lost, packet n + 1 is also likely to be lost. It leads to bursty network losses and late losses in real-time multimedia services such as Voice over IP (VoIP). This may degrade perceptual quality and the effectiveness of Forw ..."
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Cited by 117 (3 self)
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Internet packet loss and delay exhibits temporal dependency. If packet n is lost, packet n + 1 is also likely to be lost. It leads to bursty network losses and late losses in real-time multimedia services such as Voice over IP (VoIP). This may degrade perceptual quality and the effectiveness of Forward Error Correction (FEC). To characterize this burstiness, we first discuss the modeling of packet loss and delay. We propose the joint use of the extended Gilbert model and the inter-loss distance (ILD) metric to characterize temporal loss dependency. For delay, we introduce a metric called the conditional cumulative distribution function. We have applied these models to some Internet packet traces to validate the necessity and effectiveness of these models. We then evaluate the effect of these dependencies on VoIP by investigating the final loss pattern (FLP) after applying playout delay adjustment and FEC. Our results through a set of simulations confirmed that the FLP is still bursty.
Assessment of VoIP Quality over Internet Backbones
- IEEE Infocom
, 2002
"... 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 Interne ..."
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Cited by 97 (2 self)
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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.
Assessing the Quality of Voice Communications over Internet Backbones
- IEEE/ACM Transactions On Networking
, 2002
"... 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 ..."
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Cited by 74 (0 self)
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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.
Scalable On-Demand Media Streaming with Packet Loss Recovery
- In Proceedings of ACM SIGCOMM
, 2001
"... Previous scalable on-demand streaming protocols do not allow clients to recover from packet loss. This paper develops new protocols that (1) have a tunably short latency for the client to begin playing the media, (2) allow heterogeneous clients to recover lost packets without jitter as long as each ..."
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Cited by 72 (12 self)
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Previous scalable on-demand streaming protocols do not allow clients to recover from packet loss. This paper develops new protocols that (1) have a tunably short latency for the client to begin playing the media, (2) allow heterogeneous clients to recover lost packets without jitter as long as each client's cumulative loss rate is within a tunable threshold, and (3) assume a tunable upper bound on the transmission rate to each client that can be as small as a fraction (e.g., 25%) greater than the media play rate. Models are developed to compute the minimum required server bandwidth for a given loss rate and playback latency. The results of the models are used to develop the new protocols and assess their performance. The new protocols, Reliable Periodic Broadcast (RPB) and Reliable Bandwidth Skimming (RBS), are simple to implement and achieve nearly the best possible scalability and efficiency for a given set of client characteristics and desirable/feasible media quality. Furthermore, the results show that the new reliable protocols that transmit to each client at only twice the media play rate have similar performance to previous protocols that require clients to receive at many times the play rate.
An empirical study of realvideo performance across the internet
- in Proceedings of the ACM SIGCOMM Internet Measurement Workshop
, 2001
"... Abstract—The tremendous increase in computer power and bandwidth connectivity has fueled the growth of streaming video over the Internet to the desktop. While there have been large scale empirical studies of Internet, Web and multimedia traffic, the performance of popular Internet streaming video te ..."
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Cited by 69 (14 self)
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Abstract—The tremendous increase in computer power and bandwidth connectivity has fueled the growth of streaming video over the Internet to the desktop. While there have been large scale empirical studies of Internet, Web and multimedia traffic, the performance of popular Internet streaming video technologies and the impact of streaming video on the Internet is still largely unkown. This paper presents analysis from a wide-scale empirical study of RealVideo traffic from several Internet servers to many geographically diverse users. We find typical RealVideos to have high quality, achieving an average frame rate of 10 frames per second and very smooth playout, but very few videos achieve full-motion frame rates. Overall video performance is most influenced by the bandwidth of the end-user connection to the Internet, but high-bandwidth Internet connections are pushing the video performance bottleneck closer to the server. I.
The Effects of Jitter on the Perceptual Quality of Video
- IN PROCEEDINGS OF THE ACM MULTIMEDIA CONFERENCE
, 1999
"... Today's powerful computers and networks present the opportunity for video across the Internet right to the desktop. However, Internet video often suffers from packet loss and jitter, degrading the user's perceived quality of the video. The effects of packet loss on perceptual quality are w ..."
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Cited by 69 (13 self)
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Today's powerful computers and networks present the opportunity for video across the Internet right to the desktop. However, Internet video often suffers from packet loss and jitter, degrading the user's perceived quality of the video. The effects of packet loss on perceptual quality are well-understood, but to date there have not been careful user studies measuring the impact of jitter on perceptual quality. The major contributions of this work are carefully designed experiments that measure and compare the impact of both jitter and packet loss on perceptual quality of packet video. We find that jitter degrades perceptual quality nearly as much as does packet loss, and that perceptual quality degrades sharply even with low levels of jitter or packet loss as compared to perceptual quality for perfect video.
Joint Source/FEC Rate Selection for Quality-Optimal MPEG-2 Video Delivery
- IEEE Transactions on Image Processing
, 2001
"... This paper deals with the optimal allocation of MPEG-2 encoding and media-independent forward error correction (FEC) rates under a total given bandwidth. The optimality is defined in terms of minimum perceptual distortion given a set of video and network parameters. We first derive the set of equati ..."
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Cited by 58 (9 self)
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This paper deals with the optimal allocation of MPEG-2 encoding and media-independent forward error correction (FEC) rates under a total given bandwidth. The optimality is defined in terms of minimum perceptual distortion given a set of video and network parameters. We first derive the set of equations leading to the residual loss process parameters. That is, the packet loss ratio (PLR) and the average burst length after FEC decoding. We then show that the perceptual source distortion decreases exponentially with the increasing MPEG-2 source rate. We also demonstrate that the perceptual distortion due to data loss is directly proportional to the number of lost macroblocks, and therefore decreases with the amount of channel protection. Finally, we derive the global set of equations that lead to the optimal dynamic rate allocation. The optimal distribution is shown to outperform classical FEC scheme, thanks to its adaptivity to the scene complexity, the available bandwidth and to the network performance. Furthermore, our approach holds for any standard video compression algorithms (i.e., MPEG-x, H.26x).
Adaptive Playout Scheduling and Loss Concealment for Voice Communication over IP Networks
- IEEE TRANSACTIONS ON MULTIMEDIA
, 2002
"... A new receiver-based playout scheduling scheme is proposed to improve the trade-off between buffering delay and late loss for real-time voice communication over IP networks. The scheme estimates the network delay from past statistics and adaptively adjusts the playout time of the voice packets. In c ..."
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Cited by 56 (2 self)
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A new receiver-based playout scheduling scheme is proposed to improve the trade-off between buffering delay and late loss for real-time voice communication over IP networks. The scheme estimates the network delay from past statistics and adaptively adjusts the playout time of the voice packets. In contrast to previous work, the adjustment is not only performed between talkspurts, but also within talkspurts in a highly dynamic way. Proper reconstruction of continuous playout speech is achieved by scaling individual voice packets using a time-file modification technique based on the WSOLA algorithm. Results of subjective listening tests show that this operation does not impair audio quality, since the adaptation process requires infrequent scaling of the voice packets and low playout jitter is perceptually tolerable. The same timescale modification technique is also used to conceal packet loss at very low delay,i.e.,one packet time. Simulation results based on Internet measurements show that the trade-off between buffering delay and late loss can be improved significantly. The overall audio quality is investigated based on subjective listening tests,showing typical gains of 1 on a 5point MOS scale.
Analysis of On-Off Patterns in VoIP and Their Effect on Voice Traffic Aggregation
- In Proc. of ICCCN 2000
, 2000
"... We present an experimental analysis of on-off patterns in Voice over IP (VoIP), where we study the talk-spurt/gap distribution produced by two modern silence detectors: ITU G.729 Annex B Voice Activity Detector (VAD) and NeVoT Silence Detector (SD). The results indicate that spurt/gap distributions ..."
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Cited by 44 (2 self)
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We present an experimental analysis of on-off patterns in Voice over IP (VoIP), where we study the talk-spurt/gap distribution produced by two modern silence detectors: ITU G.729 Annex B Voice Activity Detector (VAD) and NeVoT Silence Detector (SD). The results indicate that spurt/gap distributions are fairly sensitive to both the sound volume and the type of silence detectors, but all of them showed that the traditional assumption of exponential distribution does not always fit well with the audio sessions we recorded. Both the spurt and gap distributions are more "heavy-tailed" than the exponential curve. In particular, the gap distribution deviates much more strongly from the exponential model, even when "hangover" is applied.
