259 results about "Multimedia transmission" patented technology

Private Multimedia Network (PMN) complements, and is an improved alternative to digital videoconferencing and multimedia delivery systems. PMN's desktop and meeting room delivery system is designed to support the exponential growth of enterprise team-based initiatives. PMN provides “one-stop-shopping” for the full multimedia rubric. It delivers user-friendly control and cost/effective TV and broadcast quality videoconferencing and other multimedia services to organizations with “critical mass” campuses and building complexes. Though digital systems dominate the videoconferencing marketplace, PMN's hybrid digital/analog architecture has no digital peer in breadth or quality of service within or between campuses. The novel architecture leverages advances in analog video short-haul technology, digital long-haul technology, and telephony audio and control technology to deliver four-level multimedia services: 1) premise; 2) campus; 3) multi-site; and 4) ubiquitous (any site with ITU compatible multimedia equipment (e.g., videoconferencing) and communication links). On balance, the price/performance afforded by PMN's centralized Telco-based control and audio delivery combined with its decentralized broadcast quality video distribution raise videoconferencing and other multimedia services to a new level of ubiquity. Just as telephones and PC LANs, PMN delivers expensive Boardroom and mobile cart videoconferencing capabilities to every desktop via existing multimedia wall plates. The key phases for this invention are: Lip-synchronization across differing network communication links and protocols; Ubiquitous multimedia service; Cost/effective room and desktop deployment; Telco control and audio; Broadcast quality video; Isochronous Quality; Centralized control and distributed operation; and Interoperable architecture.

The transmission of ATM, Internet, and satellite communications is unified through international standardized protocol embedment. Global and Local performance optimization are achieved through the combination of combinatorial, dynamic, and probabilistic programming. A simultaneous domino effect of bandwidth conservation, efficiency enhancement, reliability improvement, traffic congestion prevention, delay minimization, and speed multiplication are realized for any digital communication system, particular in Internet. With three levels of error coding in ATM cells, the loss cell recovery and bit integrity are preserved. From its mathematical roots, new combinatorial sets are systematically generated, and applications of the sets are identified. Among the applications, optimal sequences can be produced for multi-user and multi-function communication system designs. Optimality is in terms of maximum possible number of sequences with given sequence length and sequence characteristics. The results are unique and theoretically proven. By serial and parallel concatenation of error codecs, reliability of multimedia transmission can be satisfied to any desirable level. As a part of the unified transmission scheme, the acquisition and synchronization method of cascading sequences exhibits significant improvement in correlation properties and detection probabilities. A method of using block designs is demonstrated to derive, to generate, and to construct low-density parity check block codes and threshold decodable convolutional codes. An efficiency evaluation method is formulated for the combination operation of ATM, Internet, and satellites.

A relay apparatus on a network includes an analysis section that analyzes multimedia contents transmitted from a contents provider. The relay apparatus temporarily stores contents identifiers, component information of the contents, and element data pointed by pointers of the component information, and creates a transmission sequence for transmitting the element data. A transmission section transmits the contents identifier, the contents component information and the element data according to the transmission sequence to a service provider. When the relay apparatus receives a request from a terminal or from a service provider even before the service provider completes registration of the entire multimedia contents, the relay apparatus or the service provider can start rendering multimedia transmission services to the terminal.

In one embodiment, the systems and methods utilizes an enciphered permit identification number, called a session certificate, to reference the policy attribute values and session key that are stored in a secured, online reference monitor (SRM). The session key is used to encipher confidential communications, such as voice and audio communications over the Internet (VoIP), between computers. Each computer uses a unique key with a cryptographic transaction protocol for authentication and key agreement (PAKE) to securely communicate with the SRM. A sender computer uses PAKE to get a session certificate and a session key from the SRM. It sends the session certificate to a receiver computer. The receiver computer sends the session certificate to the SRM and gets back the session key. The sender computer encrypts its VoIP message with the session key and transmits it to the receiver computer. The receiver computer decrypts the VoIP message.

A system and method for improving communication between a switched network and a packet network is disclosed. In a simplified embodiment, a signaling gateway is utilized for converting signaling in a first protocol into a second protocol, and from the second protocol to said first protocol. At least one media gateway is utilized for converting multimedia provided in a first format into a second format, and from the second format into the first format. A session router for selecting at least one multimedia transmission route to a destination is also utilized, wherein the destination is specified by the switched network. In addition, a media router is also utilized for guiding the multimedia to the destination after conversion by the media gateway.

An apparatus and a method of arbitrarily adjusting the output time of caption information included in a received multimedia transport stream for broadcasting, by user input. The method includes: receiving an input of a time value for adjusting the output time of caption information, from a user through a user interface; by demultiplexing the multimedia transport stream, dividing the transport stream into video, audio, and data packets, decoding each packet, and storing the decoded packets in buffers corresponding to a type of each packet; resetting the output time of the caption information by adding the input time value to or subtracting the input time value from the output time included in the video or audio packet; and according to the output time of the reset caption information, blending the data stored in the buffer with the video information and presenting the blended information on the screen.

A method for transferring a multimedia file from a user's mobile device to user equipment associated with another user is disclosed. The method includes sending an invitation from the user's mobile device to the user equipment to request a transfer of the multimedia file. The method also includes setting up a bearer data channel between the user's mobile device and the equipment to facilitate the transferring of the multimedia file, the transferring is configured to be accomplished using Message Sessions Relay Protocol (MRSP) or Real-Time Transfer Protocol (RTP). The method further includes transferring the multimedia file using the MSRP or RTP and the bearer data channel.

The invention discloses a multimedia transmission system, a receiving terminal and the receiving method. The receiving terminal can receive multimedia service from a mobile multimedia broadcasting service server through a mobile multimedia broadcasting network, also can receive the identical multimedia service from a website server through the internet, monitors the quality of the multimedia service received in two receiving methods during the initial receiving process and the multimedia service playing process of the receiving terminal, and selects a receiving method with the service quality meeting the playing requirements according to the receiving quality to receive the multimedia service. Through the application of the invention, switching between two network transmission methods can be performed, the purpose of network coverage and complement is achieved, the disadvantage that a single network can not be seamlessly covered is overcome, and thus the users can be provided with the multimedia service better.

The invention provides a flow-control-based data transmission method, a flow-control-based exchanger, a node server and a flow-control-based communication system to solve the problem that a current IP network cannot ensure the quality of network transmission. The method comprises the following steps of: aiming at each service application, acquiring flow control information by the exchanger, wherein the flow control information comprises a transmitting time interval and the size of transmitted data; receiving a data packet by the exchanger, and placing the data packet into a packet buffer queue; and polling the packet buffer queue by the exchanger, taking a packet from the packet buffer queue according to the flow control information, and transmitting the packet. The invention can realize accurate flow control and is suitable for multimedia transmission, and the transmission quality is ensured.

The invention discloses a method and equipment for multimedia data transmission. A universal mobile terminal and video equipment are matched, multimedia data is collected by utilizing the mobile terminal, and an appropriate coding scheme is selected through combining the hardware specification of the mobile terminal and a connection manner between the mobile terminal and the video equipment, so that the video equipment is capable of implementing optimal multimedia transmission according to the network condition. The performance of a transmission channel and the performance of the terminal are combined and utilized to the greatest extent, and the multimedia data transmission at high quality is implemented. The performance requirements on a set-top box are reduced, the equipment cost for service development is saved and the utilization rate of the equipment is improved.

The embodiment of the invention provides a method and a device for transmitting stream media and relates to the field of communication technology. The invention provides the technical scheme for improving the acceptance experience of users for H.264 steam media as follows: obtaining data frames from the H. 264 data stream in turn; packing and converting the obtained data frames into a PES (photoelectric scanner) packet; converting the PES packet into a TS (transport stream) packet, and inserting PCR (program control register) information and PSI (program special information) data in the TS packet according to the frame rates of the data frames; converting the TS packet into a RTP (real-time transport protocol) packet; and transmitting the RTP packet according to a RTCP (time transport control protocol) message transmitted by a client. The method and the device are suitable for multimedia transmission in a 3G (The Third Generation Telecommunication) network.

The invention discloses a video transmission-oriented network bandwidth detection and prediction method and a video transmission-oriented network bandwidth detection and prediction system. The network bandwidth detection and prediction method comprises the following steps of calculating a current available network bandwidth according to a media stream length and receiving time; and predicting an available bandwidth at the next moment according to network bandwidths in the first N time intervals by utilizing a normal-distribution-based bandwidth prediction model. According to the method and the system, the bandwidth is measured in a way of adding information to a media stream, so that the shortcoming of requirements on the transmission of a great number of test data packets in the conventional method is overcome, the available bandwidth of a network can be quickly acquired under the condition of no influence on multimedia transmission, and the method and the system are high in practicability; and the normal-distribution-based bandwidth prediction model is also used for quickly and accurately predicting the available bandwidth of the network at the next moment according to measured data, and quick regulation and accurate prediction can be realized even under the condition of greater bandwidth changes.

The present invention relates to improved methods and systems for transmission of images overlaid on video, particularly images requiring high levels of responsivity (e.g. mouse pointers) relative to the underlying video, from a computing device to a television. In one embodiment, the present invention is directed to a method of transmitting at least one video frame with an overlaid image, comprising the steps of capturing at least one frame of video data separate from the overlaid image at a first rate, capturing the overlaid image separate from the frame of video data at a second rate, packetizing the captured frame of video data, packetizing the overlaid image, wherein the frame video data are in data packets separate from the overlaid image data packets, and transmitting separate video data packets and overlaid image data packets.

The invention discloses a transmission control method for multimedia data in network equipment. In the transmission control method, the buffer memory of exchange equipment is effectively managed; the buffer management of the exchange equipment is divided into a global control part and a queue control part by combining the sensitive characteristics of multimedia data to time delay and packet loss rate and utilizing dynamic threshold control of the transmission level based on voice quality measurement; in the global control part, the threshold is controlled by judging different transmission scenes to guarantee the transmission property under different transmission scenes; in the queue control part, the threshold is divided and regulated based on the transmission level parameters in the voice quality measurement to guarantee the transmission quality of multimedia; and in the dynamic regulation of the threshold, the regulation is performed by adopting a method for controlling according to proportions, and the network multimedia transmission quality is improved by effectively controlling the buffer resources of the exchange equipment.

The invention discloses intelligent gateway equipment based on multi-network integration and a community management intercom system. According to the intelligent gateway equipment based on multi-network integration and the community management intercom system, system operation cost is reduced, installation is convenient and fast, subsequent maintenance and management are convenient, and life experience can be provided for an owner more conveniently and fast. The community management intercom system comprises the intelligent gateway equipment, a community terminal server, remote intelligent equipment, a property management platform, an owner service platform, a common fixed-line telephone, and a common mobile phone, wherein the intelligent gateway equipment comprises a main control board, and a main control unit, a sound interface unit connected with the main control unit through a real-time multimedia transmission unit, a video interface unit, a network interface unit, a phone interface unit and a mobile network communication unit are arranged on the main control board. According to the intelligent gateway equipment based on multi-network integration and the community management intercom system, the community management intercom system uses the intelligent gateway equipment as the core and is based on the owner service platform and the property management platform, the functions of a building intercom system are achieved through the existing internet, a telecom communication network, a mobile network by means of the fixed-line telephone, the mobile phone and the intelligent equipment of the owner, and the system is expanded to achieve movability, flexibility and high efficiency.

The application level multicast module can determine efficient routing configurations for the host node's transmission and reception of multimedia data from a network. Further, the application level multicast module may aggregate, buffer, process, and multimedia data to and from an application such as a multimedia player. The application level multicast module may base routing configuration determinations on many factors. For example, the application level multicast module may base a part of the routing configuration determination on the quality of a multimedia transmission, the performance of the host node, and/or the quality of the performance by other nodes receiving or forwarding multimedia data to the host node.