Abstract: The described implementations relate to distributed network management and more particularly to enhancing distributed network utility. One technique selects multiple trees to distribute content to multiple receivers in a session where individual receivers can receive the distributed content at one of a plurality of rates. The technique further adjustably allocates content distribution across the multiple trees to increase a sum of utilities of the multiple receivers.

Abstract: Described herein is a method that includes receiving multiple requests for access to an exposed media object, wherein the exposed media object represents a live media stream that is being generated by a media source. The method also includes receiving data associated with each entity that provided a request, and determining, for each entity, whether the entities that provided the request are authorized to access the media stream based at least in part upon the received data and splitting the media stream into multiple media streams, wherein a number of media streams corresponds to a number of authorized entities. The method also includes automatically applying at least one policy to at least one of the split media streams based at least in part upon the received data.

Abstract: A “communications rate controller” provides various techniques for maximizing a quality of real-time communications (RTC) (including audio and/or video broadcasts and conferencing) over multi-hop networks such as, for example, the Internet. Endpoints in such networks generally communicate via a segmented path that extends through one or more routers between each endpoint. Maximization of conferencing quality is generally accomplished by providing in-session bandwidth estimation across segments of the network path between endpoints (i.e., communication/conference participants) in combination with a robust non-oscillating dynamic rate control strategy for maximizing usage of available bandwidth between RTC endpoints. Further, the dynamic rate control techniques provided by the communications rate controller are designed to prevent degradation in end-to-end delay, jitter, and packet loss characteristics of the RTC.

Abstract: Difficulties associated with choosing advantageous network routes between server and clients are mitigated by a routing system that is devised to use many routing path sets, where respective sets comprise a number of routing paths covering all of the clients, including through other clients. A server may then apportion a data stream among all of the routing path sets. The server may also detect the performance of the computer network while sending the data stream between clients, and may adjust the apportionment of the routing path sets including the route. The clients may also be configured to operate as servers of other data streams, such as in a videoconferencing session, for example, and may be configured to send detected route performance information along with the portions of the various data streams.

Abstract: An overlay network and scheme for building and using the overlay network are described. As the overlay network is built, new nodes joining the network are connected randomly with other nodes which results in a random graph as the network topology. The network is fully scalable, with each new node providing the same number of network connections for output to other nodes as it consumes when it joins the network. In addition, network coding is implemented at each node to mix packets of data entering each node using random linear functions. The network coding at each node generates new packets that are independent combinations of original data packets entering the node. The new coded packets make the distribution of data to other nodes more efficient and robust.

Abstract: A media recommendation and sharing technique that employs agents on media players/devices to expand the scope of media sharing scenarios. The technique assists a user in discovering media items, such as, for example, music, recordings, play lists, pictures, video games, on nearby media players or devices (devices which are capable of receiving, storing and playing media) which are interesting to the user. The collaborative media recommendation and sharing technique contemporaneously determines a user's media preferences based on media stored on a pair of media devices and recommends media for potential sharing based on these determined user preferences.

Abstract: A system and method that can receive, buffer, and asynchronously combine data from various incoming data packets is disclosed. The system and method can do so by, first, receiving incoming packets of one or multiple generations that have incoming data and incoming metadata, with the incoming data of each of the incoming packets that are of a first generation being a combination of a first, original set of data vectors, and the incoming metadata of each of the first generation of incoming packets including an indicator for the first generation. Second, once these incoming packets are received, the system and method can buffer them based on the indicator and, asynchronously from the receiving and buffering, combine the incoming data in the buffered, first generation incoming packets into outgoing data in an outgoing packet.

Abstract: A “speech onset detector” provides a variable length frame buffer in combination with either variable transmission rate or temporal speech compression for buffered signal frames. The variable length buffer buffers frames that are not clearly identified as either speech or non-speech frames during an initial analysis. Buffering of signal frames continues until a current frame is identified as either speech or non-speech. If the current frame is identified as non-speech, buffered frames are encoded as non-speech frames. However, if the current frame is identified as a speech frame, buffered frames are searched for the actual onset point of the speech. Once that onset point is identified, the signal is either transmitted in a burst, or a time-scale modification of the buffered signal is applied for compressing buffered frames beginning with the frame in which onset point is detected. The compressed frames are then encoded as one or more speech frames.

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 “Media Transmission Optimizer” provides a media transmission optimization framework for lossy or bursty networks such as the Internet. This optimization framework provides a novel form of dynamic Forward Error Correction (FEC) that focuses on the perceived quality of a recovered media signal rather than on the absolute accuracy of the recovered media signal. In general, the Media Transmission Optimizer provides an encoder that optimizes the transmission of redundant frames of electronic media information encoded at different bit rates, and provides optimized playback quality by providing a decoder that automatically selects an optimal path through one or more available representations of each frame as a function of overall rate/distortion criteria.

Abstract: A system and method that can receive, buffer, and asynchronously combine data from various incoming data packets is disclosed. The system and method can do so by, first, receiving incoming packets of one or multiple generations that have incoming data and incoming metadata, with the incoming data of each of the incoming packets that are of a first generation being a combination of a first, original set of data vectors, and the incoming metadata of each of the first generation of incoming packets including an indicator for the first generation. Second, once these incoming packets are received, the system and method can buffer them based on the indicator and, asynchronously from the receiving and buffering, combine the incoming data in the buffered, first generation incoming packets into outgoing data in an outgoing packet.

Abstract: An energy based technique to estimate the positions of people speaking from an ad hoc network of microphones. The present technique does not require accurate synchronization of the microphones. In addition, a technique to normalize the gains of the microphones based on people's speech is presented, which allows aggregation of various audio channels from the ad hoc microphone network into a single stream for audio conferencing. The technique is invariant of the speaker's volumes thus making the system easy to deploy in practice.

Abstract: A technique is disclosed that can efficiently control congestion, while supporting heterogeneity for streaming data among multiple computers in a network. A plurality of nodes is divided into a plurality of distribution trees within a computer network, wherein the data is divided into a plurality of prioritized layers. When a node experiences packet loss, the location of the congestion is inferred. If the congestion is at or near the outgoing link, outgoing traffic is shed to alleviate the congestion by shedding child node(s) receiving descriptions in the least important layer of data that the child node(s) are receiving. Similarly, if the congestion is at or near the incoming link, incoming traffic is shed by shedding parent nodes that are sending descriptions in the least important layer of data that the node is receiving. Nodes with available bandwidth are further instructed to subscribe to additional descriptions.

Abstract: A technique for providing and receiving personalized e-coupons is presented. In general, the technique involves an e-coupon provider sending e-coupons to a user of a mobile communication device, such as a cellular telephone or PDA, which are personalized in various ways so as to make them attractive to the user. In one embodiment, the e-coupons are provided based on location information received from the mobile communication device. In another embodiment, the e-coupons are provided based on the user's purchasing history. The mobile communication device that receives e-coupons from the provider includes an e-coupon handler program to facilitate the procurement and receipt of the e-coupons. In general, the e-coupon handler receives e-coupons and displays them to the user on a display of the mobile communication device. The e-coupons could have been requested by the e-coupon handler, or pushed to it by the e-coupon provider.

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 method and system provides the ability to share access information for external data over a digital voice communication channel. The access information of external data may be exchanged instead of the external data itself. More specifically, a recipient device may receive contextual information which relates to the access information of external data. The contextual information may be processed to identify the source of the external data and other information necessary to access the external data. For example, a hyperlink directed to the external data in a Web server may be exchanged while the recipient device and the sending device are involved in a digital conversation. The recipient device can access the external data by activating the hyperlink.

Abstract: Techniques to perform distributed caching for multimedia conference calls are described. An apparatus may comprise a conferencing server and a frame management module. The conferencing server may receive a sequence of video frames from a sending client terminal and send the sequence of video frames to multiple receiving client terminals. The frame management module may receive a client frame request for one of the video frames from a receiving client terminal, retrieve the requested video frame, and send the requested video frame in response to the client frame request. Other embodiments are described and claimed.

Abstract: Techniques for managing output bandwidth for a conferencing server are described. An apparatus may include a receiver to receive input video streams at first bit rates from multiple client terminals. The apparatus may include a rate allocation module to allocate an output bit rate for an output video stream corresponding to each input video stream based on distortion rate information where a total output bit rate for all output video streams is equal to or less than a total output bit rate budget for a conference server. The apparatus may include a video transrating module to reduce the first bit rate to a second bit rate for one or more input video streams in accordance with the allocations to create the output video streams. Other embodiments are described and claimed.

Abstract: Implementation of hybrid peer-to-peer streaming with server assistance is described. In one implementation, a media source is selected from amongst a plurality of media sources for retrieval of streaming media content. The selection might be based, for example, on an amount of the streaming media content received at respective time units. In one scenario, if the amount received at a time unit is less than a target amount, the streaming media content is retrieved from at least one streaming media server. Conversely, if the amount received at a time unit is more than the target amount, the streaming media content is retrieved from at least one peer-to-peer network. In another embodiment, a playback buffer is monitored to determine an amount of streaming media content at the respective time units. The media source is then selected based on the amount of the streaming media content in the playback buffer.

Abstract: Techniques for virtual conferencing servers are described. An apparatus may comprise a conferencing server to manage a conference call with multiple client terminals. The conferencing server may have a virtual conference manager to select a first client terminal to operate as a first virtual conferencing server, and transfer conference call operations for a second client terminal from the conferencing server to the first virtual conferencing server. Other embodiments are described and claimed.