摘要:
Repetitive packets in a voice/data stream, are detected and suppressed in the transmitting side of a network, after a predefined number of consecutive repetitive packets have been transmitted. Then, at the receiving side of the network, suppressed repetitive packets are reconstituted by filling the resulting gap in the voice/data stream with the repetitive pattern contained in the last received repetitive packet. When the input voice/data stream is compressed, only non-repetitive packets are compressed so that repetitive patterns are not corrupted by compression.
摘要:
This invention deals with a method and system for dynamically adjusting the communication bandwidth assigned to an audio channel connection in a high speed digital network. More particularly, the invention is made to track the activity of a voice assigned connection (e.g. a PBX or PABX entry to the network), define a so-called activity bit for each block of audio channel signal and then dynamically adjust the assigned network communication bandwidth accordingly. Adjustment of the communication bandwidth is accomplished by integrating the audio channel activity bits through a predefined integration function, the result of which is then compared against predefined threshold values in order to determine an appropriate bandwidth setting for the audio channel.
摘要:
An adaptive apparatus and method for managing a playout buffer (POB) in an edge node of a packet-based data communications network, such as an ATM network, in order to reduce the end-to-end communication delay introduced thereby and to allow the POB filling level to be corrected in the event of clock speed differences. A monitoring mechanism determines if the minimum filling level of the playout buffer over a time period whereby the average filling level of the playout buffer is reduced according to the minimum filling level of the playout buffer over the time period.
摘要:
A method and an apparatus for removing the silence from the digitalized voice signals conveyed through packets or cells switching networks. The silence samples are neither packetized nor sent over the network but are regenerated at the output of the network. The silence samples generated are white noise samples, where the level is adapted to the background noise of the silence samples received at the input node of the network. For long periods of silence, the white noise level is periodically refreshed to be adapted to the last silence samples received at the input node of the network. The method provides also a control of packet or cell loss. The method uses are not control packets; in the later case, it can be used for ATM networks with AAL1. The method is implemented as a program executed in a Digital Signal Processor located on adapter cards dedicated to voice processing in the network access nodes.
摘要:
Voice circuit emulation system in a packet switching network includes a plurality of switching nodes (SW-1, SW-2, . . . ) interconnected by connection lines (TL-1, TL-2, . . . ), enabling voice signals to be transmitted in circuit emulation packets from any source exchange telephone device (PBX-A) to any destination exchange telephone device (PBX-B). Each switching node (SW-1 or SW-2) includes a circuit emulation server (CES-1 or CES-2) including a plurality of connection tables corresponding each to an incoming connection line, each table containing for each incoming connection line, the identification of each outgoing connection line associated to each slot of the circuit emulation packet received from the incoming connection line and the identification of each slot of the packet to be transmitted on the outgoing connection line. A switching module looks up the connection table for each slot contained in each incoming circuit emulation packet received from all incoming lines and thereby transferring the contents thereof to the slot of the outgoing connection line identified in said connection table.
摘要:
A method and an apparatus for reducing the jitter and end-to-end delay on lines of a packet switching network conveying voice or video digitalized data for one or more connections between a local source and a remote source at a constant bit rate.The method and apparatus of the invention are for use in a voice or video processor of a voice or video processing server of a network node; the method and the apparatus provide a way of controlling the remote traffic rate from the remote source before accessing the processor without using an external clocking such as the network clock.The solution proposed by the invention consists in buffering the remote and local traffics to adapt the remote traffic rate to the local traffic rate, which is supposed having a limited jitter, while sequencing of the access of the buffered data to the processor.
摘要:
A data processing method for efficiently transporting multimedia data packets of fixed and/or variable length over an Assynchronous Transfer Mode (ATM) network made to transport fixed length ATM cells including a fixed length user data payload and a fixed length ATM header. The data processing method includes concatenating said fixed and/or variable length user data and appending said concatenated data with a so-called trailer defining the various concatenated user data lengths and identifications, for being further split into ATM cells payloads before being transmitted over said ATM network.
摘要:
A method and an apparatus for transferring structured data of a constant bit rate traffic in an ATM network. The method uses the recurrence of alignment of the structured data field of length N in a 47 octet SA.sub.-- PDU payload. At the transmitting end the CSI bit of the header of the first SAR.sub.-- PDU is set to 1. It is then set to 0 for the next SAR.sub.-- PDUs until the Nth SAR.sub.-- PDU is reached. It is then again set to 1. This process is repeated until the last SAR.sub.-- PDU has been transmitted.
摘要:
The present invention uses the SAR and CPCS functions of the AAL-5 to define an AAL-5 SSCS for performing the AAL-1 and AAL2 functions. The defined AAL-5 format comprises a SSCS trailer of N (preferred value is 8) bytes an CPCS-PDU of N+8 (preferred value is 16) cells (16.times.48 bytes=768 bytes). In the preferred embodiment, the AAL-5 CPCS-PDU is transported inside 16 ATM cells. The CPCS and SSCS trailers provide the the same efficiency as AAL-1 and the global structure based on 16 cells is completely similar and transparent in term of delay and overhead. The payload size is 48 bytes for the 15 first cells, and 32 bytes for the last cell. The CPCS-PDU payload size is always a multiple of 8 bytes.
摘要:
In a loosely coupled multiprocessor environment wherein a plurality of processors (2) are attached to a shared intelligent memory (1), a distributed scheduling mechanism for scheduling of source processors (4) with respective server processes (5) to be executed by the processors (2) upon their attachment to a data message queue (3) contained in the shared intelligent memory (1), the processes (4, 5) using data messages enqueued into, respectively dequeued from said memory (1). According to this scheduling mechanism, an independent scheduler (6) is dedicated to each of the processes of a process group, and all the schedulers monitor the status of the data message queue, and upon receipt of an empty-to-non-empty E-NE signal, the least busy scheduler dequeues shared data from the queue, so that it can be processed by its associated process, without however, loosing fault-tolerance in case of a particular processor failing.