Abstract:
An apparatus and method for communicating electronic data via a network infrastructure (101) having a unicast mechanism and a multicast mechanism. Said apparatus comprises a server (100), which contains electronic data and is capable of using said unicast and multicast mechanisms for communicating said electronic data to one or more clients (102), the apparatus comprises means (103) adapted to make a decision, taking into account a predetermined set of parameters, whether said server (100) shall use said unicast mechanism or said multicast mechanism for communicating said electronic data to said clients (102) and said server (100) is arranged to communicate said electronic data to said clients (102) in accordance with said decision.
Abstract:
Methods and apparatus are provided for reserving resources in a wireline network from a wireless network. A resource reservation request is made by a service client to a service broker (SB). The service broker (SB) contacts a bandwidth broker (BB) to determine the available resources in the wireline network. The service broker (SB) also contacts a geographical domain server (GDS) for information related to wireless access for the requested reservation. Using information provided by the geographical domain server (GDS), the service broker (SB) contacts a radio bearer broker (RBB) to determine the resources available in a particular wireless network. Based upon the information provided by bandwidth brokers (BB) and radio bearer brokers (RBB), the service client can reserve the requested resources.
Abstract:
The present invention relates to methods for improving speech quality in e.g. an IP-telephony system. The invention reduces audio artefacts being due to overrun or underrun in a playout buffer caused by the sampling rates at a sending and receiving side not being at the same rate. The inventive solution modifies an LPC-residual on a sample-by-sample basis. The LPC-residual block comprising N samples is converted to a block comprising N+1 or N-1 samples. A sample rate controller 400 decides whether samples should be added to or removed from the LPC-residual. The exact position where to add respective remove samples is either chosen arbitrarily or found by searching for low energy segments in the LPC-residual. A speech synthesiser module 430 then reproduces the speech. By using the proposed sample rate conversion method the playout buffer 440 can be continuously controlled. Furthermore, since the method works on a sample-by-sample basis the buffer can be kept to a minimum and hence no extra delay is introduced.
Abstract:
A decoder improves delayed packet concealment in a packet network by using two decoder sections. A first decoder section (30) bases its decoding during the concealment phase on erroneous filter states and a set of speech parameters, whereas a second decoder section bases its decoding on saved (36) and updated filter states and the same speech parameters. The outputs of the two decoder sections are thereafter combined (34) to form the final speech signal. This decoding strategy produces a speech signal with smooth transitions from delayed to non-delayed packets and uses information from the most recent packets for speech generation.
Abstract:
In packet communication paths (18) that include header compression, header fields of packets to be communicated are altered. The alteration of the header fields (14) does not disturb their functionality, and its transparent to the header compression scheme of the packet communication path (18). The altered header fields (14) are provided for compression by the header compression scheme, resulting in improved performance of the header compression scheme. Performance improvements can also be achieved in packet communication paths (18) that do not use header compression, by violating the integrity of header fields in packets to be transmitted over the packet communication path (18).
Abstract:
The soft state of a header compression scheme in a communication system carrying packet traffic including a real time communication signal can be updated (63) during periods of communication signal inactivity (62), during which there is no need to transmit the communication signal. The header compression soft state can also be updated by stealing bits (83, 84) from the communication signal to carry the header update information (73). If the communication signal includes source encoded data, the header compression soft state can be updated selectively (126) based on the bit rate (122, 124) of a codec that produced the source encoded data.
Abstract:
Un aparato para mejorar el rendimiento de las comunicaciones por paquetes sobre un camino de comunicación por paquetes (18), que comprende una entrada (11) para recibir los campos de cabeceras de paquetes para ser comunicados sobre el camino de comunicación del paquete, dicho aparato caracterizado por: un procesador del campo (26) acoplado a dicha entrada para violar la integridad de uno de dichos campos de la cabecera para producir un campo de la cabecera violada (25) seleccionando, en base a una comparación umbral, entre un valor de dicho campo de la cabecera y un valor previo de un campo de la cabecera recibido previamente o entre un valor de dicho campo de la cabecera recibida y un valor de cero, y una salida (14) acoplada a dicho procesador del campo (26) para poner a la salida dicho campo de la cabecera violada con el valor seleccionado para el camino de comunicación del paquete.
Abstract:
In packet communication paths that include header compression, header fields of packets to be communicated are altered. The alteration of the header fields does not disturb their functionality, and is transparent to the header compression scheme of the packet communication path. The altered header fields are provided for compression by the header compression scheme, resulting in improved performance of the header compression scheme. Performance improvements can also be achieved in packet communication paths that do not use header compression, by violating the integrity of header fields in packets to be transmitted over the packet communication path.
Abstract:
An improved forward error correction (FEC) technique for coding speech data provides an encoder module which primary-encodes an input speech signal using a primary synthesis model to produce primary-encoded data, and redundant-encodes the input speech signal using a redundant synthesis model to produce redundant-encoded data. A packetizer combines the primary-encoded data and the redundant-encoded data into a series of packets and transmits the packets over a packet-based network, such as an Internet Protocol (IP) network. A decoding module primary-decodes the packets using the primary synthesis model, and redundant-decodes the packets using the redundant synthesis model. The technique provides interaction between the primary synthesis model and the redundant synthesis model during and after decoding to improve the quality of a synthesized output speech signal. Such "interaction," for instance, may take the form of updating states in one model using the other model.