Abstract: A system and method that enables broadcasting of data in packets across a network using network coding is described. This system and method enables a network to broadcast information in packets without full knowledge of the network's topology. Further, it enables broadcasting of data in packets with a low probability of failure.

Abstract: Techniques for managing visual compositions for a multimedia conference call are described. An apparatus may comprise a processor to allocate a display object bit rate for multiple display objects where a total display object bit rate for all display objects is equal to or less than a total input bit rate, and decode video information from multiple video streams each having different video layers with different levels of spatial resolution, temporal resolution and quality for two or more display objects. Other embodiments are described and claimed.

Abstract: Techniques to perform rate matching for multimedia conference calls are described. An apparatus may comprise a conferencing server and a rate matching module. The rate matching module may be arranged to adjust bit rates between media communications channels for client terminals in a conference call, with the rate matching module to remove video frames from a set of video information received on a first media communications channel to reduce a bit rate for the video information. Other embodiments are described and claimed.

Abstract: Techniques for variable resolution encoding and decoding of digital video are described. An apparatus may comprise a video encoder to encode video information into a video stream with a base layer and an enhancement layer. The base layer may have a first level of spatial resolution and a first level of temporal resolution. The enhancement layer may increase the first level of spatial resolution or the first level of temporal resolution. Other embodiments are described and claimed.

Abstract: Content synchronization in a file sharing environment is described. A node in a circle share maintains a version vector that represents that node's knowledge of file synchronization events. Version vector data may be transmitted between nodes of the circle share via a presence service in the form of short notification messages. The version vector data is compared to identify one or more nodes to be synchronized. A resource-aware overlay representing the circle share is computed, and is used to initiate communication to determine which of the available nodes with which to synchronize will provide the most synchronization value to other nodes in the circle share.

Abstract: Aspects of the present invention are directed at sending a data item from a sending client to a receiving client. In accordance with one embodiment, a method provides controls for generating an audio-based command to send a data item from a sending client to a receiving client. More specifically, the method includes receiving an audio stream at the sending client from a sending party. As the audio stream is being received, a determination is made regarding whether a command to send a data item to the receiving client was received. If a command to send a data item is included in the audio stream, the method identifies the data item that is the object of the command and then transmits the data item to the receiving client over the network.

Abstract: A method and system for collecting and communicating contextual information relating to a VoIP conversation is provided. Structured hierarchies are utilized for efficient communications of various amounts and types of contextual information relating to a VoIP conversation. Information identifying at least one structured hierarchy, which will be used to carry the contextual information, is transmitted during establishment of a conversation between two VoIP enhanced devices. The structured hierarchy is selected from a set of predefined and declared structured hierarchies. Subsequently transmitted contextual information exchanged between two VoIP enhanced devices is represented in accordance with the identified structural hierarchy. VoIP clients, network infrastructure, and various service providers can collect the contextual information based on the identified structured hierarchy, update the contextual information by adding, deleting and/or modifying the contextual data.

Abstract: A mediation server for controlling contents of incoming and outgoing communication information exchanged as part of a conversation is provided. The mediation server may be a centralized server between an internal (private) network and an external network, utilized for enforcing the internal network's policy and detecting a potential security compromise in the internal network. Predefined evaluation criteria are utilized to enforce internal policy or security policy within the internal network. When communication information is exchanged, the mediation server may monitor potential policy or security breaches in the communication information utilizing the predefined evaluation criteria and execute an appropriate action to prevent potential policy or security breaches.

Abstract: Aspects of the present invention are directed at software for processing a received voicemail message to provide enhanced features in a voicemail system. In one embodiment, a user interface is provided where a callee may generate events to modify audio attributes of a received voicemail message. When an event to modify the audio attributes of the received voicemail message is received, aspects of the present invention obtain a set of event data that describes input generated by the callee. Then, a media player program is used to modify attributes of the data packets in the voicemail message in accordance with the input received from the callee.

Abstract: A local network coding framework and method including techniques to improve efficiency in a wireless network by reducing overhead. The local network coding method includes exchanging data availability between nodes on the wireless network by sending Bloom filters of lists of packets to neighboring nodes. Based on data availability, optimized mixing of pure packets is performed to form mixture packets for output. A separate acknowledgement buffer keeps track of the pure packets transmitted but not acknowledged. If an acknowledgement does not arrive after a certain time period, the packet is assumed to be lost and is retransmitted. An optimized packet mixing process generates mixture packets and decides which nodes to send the mixture packets. The local network coding framework and method also includes methods for representing the composition of a mixture packet and using mixing at a wireless access point to improve the performance of the wireless local area network.

Abstract: A system and method for correcting errors and losses occurring during a receiver-driven layered multicast (RLM) of real-time media over a heterogeneous packet network such as the Internet. This is accomplished by augmenting RLM with one or more layers of error correction information. This allows each receiver to separately optimize the quality of received audio and video information by subscribing to at least one error correction layer. Ideally, each source layer in a RLM would have one or more multicasted error correction data streams (i.e., layers) associated therewith. Each of the error correction layers would contain information that can be used to replace lost packets from the associated source layer. More than one error correction layer is proposed as some of the error correction packets contained in the data stream needed to replace the packets lost in the associated source stream may themselves be lost in transmission.

Abstract: Systems and methods for improved streaming of variable bit rate multimedia content are described. The described systems and methods solve some streaming problems by streaming the variable bit rate content at a constant bit rate that is negotiated between a server and a client. As a result, a transmission channel from server to client is utilized in an efficient manner. If the client is able to buffer the streamed content (to a hard drive, for example) the content can be streamed at a rate that is significantly less than its peak bit rate, even at a rate that is less than its average bit rate.

Abstract: A system and process according to the present invention involves tagging prescribed portions of the data of each layer in a layered multicast or layered presentation with an indicator of the importance or utility that the data provides to the receiver. Additionally, the data is tagged with a cost factor involved with sending the data. The aforementioned portions of the data can be an entire data stream of a layer, or some part thereof all the way down to the individual packets making up the stream. The invention also involves determining the optimized scenario for sending the data from the sender to the receiver based on the data tags.

Abstract: A system and method for distributing streaming content data from a server to multiple clients enables the server to handle transient surges of requests by having the clients cooperate with the server and other clients to distribute content, thereby alleviating the load on the server. The server divides the streaming content into multiple sub-streams (e.g., by using multiple description coding), and constructs multiple distribution trees with itself at the root and each client as a node in each of the trees. Each sub-stream is transmitted down a corresponding one of the distribution trees. Clients that receive sub-streams from its parent nodes in the distribution trees in turn forward the sub-streams to their child nodes in the trees.

Abstract: Systems and methods for presenting time-varying multimedia content are described. In one aspect, a lower quality data stream for an initial portion of the multimedia content is received. The lower quality data stream is received at a rate faster than a real-time playback rate for the multimedia content. The lower quality data stream was encoded at a bit rate below a transmission rate. A higher quality data stream of a subsequent portion of the multimedia content is received. The higher quality data stream was encoded at a bit rate that equals the transmission rate. The initial portion and the subsequent portion of the multimedia content are presented at the real-time playback rate. Receiving the initial portion faster than the real-time playback rate provides for a reduction of latency due to buffering by a desired amount.

Abstract: A two-dimensional (2D) image model models the layout structure of a class of document images as an image grammar and includes production rules having explicit layout parameters as data items that indicate information about the spatial relationships among image constituents occurring in images included in the class. The parameters are explicitly represented in the grammar rules in a manner that permits them to be automatically trained by a training operation that makes use of sample document images from the class of modeled documents. After each sample image is aligned with the 2D grammar, document-specific measurements about the spatial relationships between image constituents are taken from the image. Optimal values for the layout parameters are then computed from the measurement data collected from all samples.

Abstract: A transmission method for video image data using an embedded bit stream in a hierarchical table-lookup vector quantizer comprises the steps encoding an image using hierarchical vector quantization and an embedding process to obtain an embedded bit stream for lossless transmission. The bit stream is selectively truncated and decoded to obtain a reconstructed image.

Abstract: To obtain real-time responses with interactive multimedia servers, the server provides at least two different audio/visual data streams. A first data stream has fewer bits per frame and provides a video image much more quickly than a second data stream with a higher number of bits and hence higher quality video image. The first data stream becomes available to a client much faster and may be more quickly displayed on demand while the second data stream is sent to improve the quality as soon as the playback buffer can handle it. In one embodiment, an entire video signal is layered, with a base layer providing the first signal and further enhancement layers comprising the second. The base layer may be actual image frames or just the audio portion of a video stream. The first and second streams are gradually combined in a manner such that the playback buffer does not overflow or underflow.

Abstract: Correction of errors and losses in a receiver-driven layered multicast (RLM) of real-time media over a network is augmented using one or more layers of error correction information. Each receiver separately optimizes the quality of received information by subscribing to at least one error correction layer. Ideally, each source layer in a RLM has one or more associated multicasted error correction data streams. Each error correction layer contains information for replacing lost packets from the associated source layer. More than one error correction layer is proposed to correct for lost packets in other error correction layers. Error correction streams are preferably generated using a pseudo-Automatic Repeat Request (ARQ) wherein a broadcaster sends both the source packets in a primary stream and delayed versions thereof in one or more redundant streams. A hybrid technique combines the psuedo-ARQ method with an adaptation of Forward Error Correction (FEC) techniques.

Abstract: A method and system including an improved generalized reference decoder that operates according to any number of sets of rate and buffer parameters for a given bit stream. Each set characterizes a leaky bucket model and contains three parameters representing the transmission bit rate, buffer size, and initial decoder buffer fullness. An encoder provides at least two sets of these parameters, whereby the decoder selects one or interpolates between them to operate at any desired peak bit rate, buffer size or delay. The generalized reference decoder may select the smallest buffer size and corresponding delay that decodes the bit stream without buffer underflow or overflow, or alternatively may select and operate at the minimum required peak transmission rate, or something between both. In practice, the buffer size, delay and/or the peak transmission rate can be reduced by significant factors, and/or the signal-to-noise ratio (SNR) can be increased.