Showing papers on "Media Delivery Index published in 2004"

Quality of service concerns in IP-based control systems

Abstract: The popularity of network-based control systems (NBCS) is continuously growing. One of the most intriguing aspects is the transportation of control network data over IP-based networks using accepted standards such as EIA-852. To a large extent the actual quality of control (QoC) in such systems depends on the network timing such as delay and delay jitter. This paper presents a classification of relevant quality of service parameters and identifies application classes. Subsequently, the paper focuses on the effect of delay jitter at a fixed mean delay on the QoC. Two sources of delay jitter are identified in IP-based control systems: 1) network traffic induced and 2) protocol induced. As an example of a simple control loop implemented over an EIA-852-based system we investigate how the induced jitter affects the QoC using a time-discrete simulation model. Conclusions are drawn as to how the findings in the EIA-852 system can be interpreted and extended to a generalized NBCS.

System and method for fast start-up of live multicast streams transmitted over a packet network

Abstract: A system for multicast streaming of programs over a packet network includes a node having a processor that conditions a video bitstream such that packets containing an I-frame are located near program specific information (PSI) packets, the processor marking a random join point (RJP) in the video bitstream immediately preceding the I-frame and PSI packets, the node outputting the conditioned and marked video bitstream across the packet network. An edge device of the network includes a buffer that caches packets of the conditioned and marked video bitstream video starting at the RJP, and sends the cached packets to a client receiver.

Jitter buffer management for mobile communication handoffs

Abstract: A method of providing communication service includes determining a first playback delay based on one or more network characteristics of a first network and one or more network characteristics of a second network. The method also includes storing media received from the first network in a first buffer and playing media received from the first network after the media received from the first network has been stored in the first buffer an amount of time based on the first playback delay. The method further includes detecting a handoff trigger and storing media received from the second network in a second buffer, in response to detecting the handoff trigger. The method also includes playing media received from the second network.

A self-adaptive jitter buffer adjustment method for packet-switched network

Abstract: A self-adaptive jitter buffer adjustment method for packet-switched network is presented in this invention, which involves both positive adjustment to jitter buffer and negative adjustment to jitter buffer. The positive adjustment part performs a positive adjustment to the jitter buffer according to an indication of a sequence number of the packet delaying, to absorb larger network jitters by dynamically detecting sequence numbers of the packets in the jitter buffer. The negative adjustment part performs a negative adjustment to the jitter buffer when the filling level descends, to match smaller network jitters by periodically detecting a filling level of the jitter buffer. This invention realizes the dynamic tracking of network jitter, so that a self-adaptive adjustment to the jitter buffer working parameters can be achieved. Comparing with traditional methods for self-adaptive buffer adjustment, the present method reduces the complicacy and computational costs of the self-adaptive jitter buffer adjustment algorithm. At the same time, the impact of self-adaptive jitter buffer adjustment on the performance of the entire system is greatly reduced.

Empirical scheduling of network packets

Abstract: A method of transmitting packets over a network includes steps of partitioning a packet delivery schedule into discrete time slots; transmitting a plurality of test packets from a first endpoint on the network to an intended recipient in the network using different time slots; evaluating the reliability of the network to transmit the plurality of test packets in each time slot; and selecting one or more time slots in the delivery schedule according to the evaluation step.

Packet-late indication based (PLIB): adaptive jitter buffer

Abstract: Adaptive jitter buffer are being increasingly used in TDM over packet switching networks (TDMoPSN). This paper proposes a simple adaptive jitter buffer adjustment algorithm for efficient jitter absorption. We verify by simulations that the proposed algorithm can absorb and track jitter effectively while guarantee ordered packet delivery even under networks with delay spikes.

An adaptive receiver buffer adjust algorithm for VoIP applications considering voice characters

Abstract: A new adaptive receiver buffer adjust algorithm is proposed for VoIP applications with the consideration of voice characters and network conditions. It divides network status into two modes: normal mode and spike mode, according to the delay of the coming packets and the size of buffer. In normal mode, the receiver adjusts the buffer delay at the beginning of every talk-spurt according to the delay jitters collected during the last talk-spurt. In spike mode, two buffer adjust methods are proposed. One is to adjust the buffer delay for every packet based on a prediction of the delay according to the delay of former packets; the other is to adjust the buffer delay based on a prearranged way. Simulations and experiments show that this strategy can well conceal the delay jitter and reduce the packet loss rate.

Method of dynamic adaptation for jitter buffering in packet networks

Abstract: A method of controlling a buffer for reducing jitter in a packet network is provided, with a fast attack and a slow decay time to track delay changes in the network. The principal function of the method for controlling the jitter buffer is to minimize the delay within the buffer and use packet loss compensation in the event that the buffer enters an underflow condition.

Enhanced video streaming using dual network mode

Abstract: A method and system for video data delivery for 3G mobile systems is disclosed. The throughput and quality of video data delivery in real-time is enhanced by using a combination of circuit-switched and packet-switched network and using optimal video frame dispatching patterns that are calculated based on transmission feedback, status of the network and QoS parameters.

System and method for processing rx packets in high speed network applications using an rx fifo buffer

Abstract: A system and method are provided for processing packets received via a network. In use, data packets and control packets are received via a network. Further, the data packets are processed in parallel with the control packets.

Adaptive playback buffer for wireless streaming media

Abstract: The wireless streaming media communications are fragile to the delay jitter because the conditions and requirements vary frequently with the users' mobility. Buffering is a typical way to reduce the delay jitter of media packets before the playback, however, it will incur a longer end-to-end delay. Our motivation in this paper is to balance the elimination of delay jitter and the decrease of end-to-end delay. We propose a novel adaptive playback buffer (APB) based on the probing scheme. By utilizing the probing scheme, instantaneous network situations are collected and then using with the delay margin and the delay jitter margin, the playback buffer is adaptively adjusted to represent the continuous and real-time streaming media at the receiver. Unlike the previous studies, the novelty and contributions of the paper are: a) accuracy: by employing the instantaneous network information, the adjustment to the playback buffer correctly reflects the current network situations, which makes the adjustment effective; b) efficiency: by utilizing the simple probing scheme, APB achieves the current network situations without the complex mathematic prediction and makes it efficient to adjust the playback buffer. Performance data obtained through extensive simulations show that our APB is effective to reduce the delay jitter and to decrease the buffer delay.

Method and apparatus to efficiently transmit streaming media

Abstract: A method and system to efficiently transmit streaming media are described. The system includes a network protocol engine configured to receive a request for a data stream from a client system and to transmit data packets from the data stream to the client system; and a packet pacing sub-system, responsive to the request for the data stream. The packet pacing sub-system is configured to wait to receive a plurality of data packets from the data stream, and schedule delivery events for the plurality of data packets. The system may further include a streaming media protocol engine to determine a delivery time for data packets in the data stream. The packet pacing sub-system may be configured to receive an associated delivery time for the data packets from the streaming media protocol engine and schedule the delivery events for the data packets according to the associated delivery times for the plurality of packets.

Automatic adaptation of quality of service over packet-switched networks

Abstract: An end node in a packet-switched network is proposed wherein the end node is arranged to exchange packets consisting of a packet header and payload data with a server node connected to the packet-switched network. The end node includes at least one module adapted to extract quality of service parameter values from the headers of packets received over the packet-switched network (402) and to insert the extracted quality of service parameter value in the headers of packets destined for transmission to the server node over the packet-switched network (405). In this manner the end node automatically adapts the quality of service parameters received over the packet-switched network to correspond with the type of quality of service defined in received packets. Such a system means that only the server node need be configured with the quality of service parameters applicable on the particular packet switched network, so greatly reducing the installation and configuration overheads in a multi-user network.

System and method for collecting and presenting service level agreement metrics in a switched metro ethernet network

Abstract: A method for monitoring a network includes injecting a plurality of data packets into the network. The data packets are transmitted between a source device and a destination device. A plurality of reflected data packets is collected. The plurality of reflected data packets are reflected from the destination device to the source device. Also, the plurality of reflected data packets includes at least a portion of the data packets injected into the network. The method further includes determining a total number of the reflected data packets. A packet delivery rate, a latency value, and a jitter value can be calculated based at least partially on the total number of reflected data packets. Further, the packet delivery rate, the latency value, and the jitter value can be reported to a user.

End-to-end jitter minimization with TCM for real-time multimedia traffic in DiffServ networks

Abstract: The quality of multimedia services over Internet could be seriously degraded due to unexpectedly increased end-to-end jitter. We propose an innovative end-to-end jitter minimization mechanism with three-color marking (TCM). An early-arrival packet is assigned for a delayed-FIFO queue to consume extra credits. For a late-arrival packet, the highest-priority queue of weighted-round-robin (WRR) scheduling is allocated for exerting on reducing delay deficit to avoid exceeding end-to-end delay constraints. We conduct experiments by simulation and the results reveal that the proposed scheme can achieve very satisfactory end-to-end delay jitter, especially for longer-length packets under heavy traffic load. The impact of using different buffer sizes at a router to the packet loss rate and the end-to-end delay jitter is also investigated.

Quality measuring system for network provisioning and customer service in content delivery

Abstract: In Japan, medium- to high-speed access networks using technologies such as ADSL have recently been expanding rapidly. We propose a quality management system enabling efficient, accurate measurement of streaming video quality to meet the requirements of network provisioning and customer service. Streaming video is delivered through a content delivery network (CDN) consisting of a content server, network, and customers' personal computers (client PC). In the digital content delivery business, it is thus necessary to manage the overall quality for all these elements.

A user-centered approach to enhance QoS for networked video

Abstract: Buffer management plays an important role for enhancing quality of service (QoS) for video streaming over IP networks. However, most existing buffer management techniques have been developed according to the network's point of view; consequently, QoS requirements from users' perspectives are not well satisfied. The paper proposes a novel buffer management scheme based on user's expectations. Simulation results show the advantages of the scheme in terms of video quality, service fairness and network efficiency.

System and method with adjustment for delivery of packets

Abstract: A system and method for delivery of packets is provided. In an embodiment, a client is operable to query a first layer of the protocol stack used to provide a link that carries packets for said client. Based on the query, the client is operable to adjust how those packets are delivered over another layer of the protocol stack in order to help improve the likelihood of successful delivery of those packets.

Control transmission pace at IP layer to avoid packet drop

Abstract: Avoiding packet loss is critical for time sensitive network applications, such as multimedia streams for video/voice. Delaying and dropping low priority packets to ensure high priority and time sensitive data stream deliver during network congestion is a basic QoS (quality of service) mechanism over current network infrastructure. This mechanism works if time sensitive data stream is the minority of the network traffic and if the network is not very congested. The methodology of dropping low priority data does not scale when time sensitive data stream uses high percentage of network bandwidth. This is because bandwidth required by video/audio applications can vary in very wide range when real-time data becomes majority network traffic, that is, television (TV), telephone, visual telephone, videoconferencing, gaming, and other video/audio based applications are all deployed on Internet. Then, what is the proper percentage of bandwidth to reserve? and which packets should be dropped if available bandwidth is less than demanding? A major issue is that letting bottleneck routers drop packets is not a proper methodology to guarantee quality of service. If packets cannot be delivered due to exhausted network bandwidth, these packets should be tossed as earlier as possible to reduce bandwidth waste or should be delayed at transmission hosts for later transmission. Also, applications should have right to selectively toss data for enhancing service quality, rather than let routers randomly drop packets. Therefore, mechanisms to avoid packet drop need to be deployed in Internet infrastructure. This paper studies how well priority (class) based traffic shaping can help time sensitive data delivery, addresses technology of packet drop avoidance (PDA), and shows how packet drop avoidance mechanism improves real-time applications' performance by reducing bandwidth waste, packet delay and loss. This paper then addresses why PDA should be deployed in Internet protocol (IP) layer.