International Business Machines Corporation

Cisco Technology, Inc.

Maurice Duault

Claude R Galand

Bernard Y Pucci

Gerard Richter

Jean-Francois Le Pennec

Gerard Lebizay

Aline Fichou

Francois Kermarec

Patrick Michel

Alain Benayoun

Jean L Calvignac

Gerald Lebizay

Jeffrey James Lynch

Jean-Marie L Munier

Laurent Nicolas

Olivier Bertin

Pascal Chauffour

Bernard J Naudin

Denis Chevalier

Pierre Pignal

Ramon E Snyder

Serge Leroy

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.

Adaptive playout buffer and method for improved data communication

A communication system and method for compressing data in a transmission system wherein multiplexed channels are transported over a transmission network of the type comprising a plurality of switching nodes interconnected by connection lines, the exchange of data signals carried out by switching the channels in the network between two exchange telephone devices, and each of the multiplexed channels transporting data bytes representing the data signals from one source exchange telephone device to one destination exchange device during an exchange of information therebetween through the intermediary of a compression/decompression device. The method comprises the steps of comparing, for each multiplexed channel, the signal value associated to each one of a plurality of &ldquo;n&rdquo; consecutive data bytes to a predetermined threshold; deleting, in case said signal value for all said &ldquo;n&rdquo; data bytes is less than the predetermined threshold, all bits which are not necessary to represent the signal value from each of the &ldquo;n&rdquo; data bytes; building a compression frame by concatenating either the &ldquo;n&rdquo; data bytes when they are not modified or the &ldquo;n&rdquo; modified data bytes when bits have been deleted therefrom, and adding to each of said groups an identifier indicating whether said data bytes are modified or not before transmitting said compression frame over said transmission network. Decompressing the frame by determining the identifier value indicating the composition of the bytes; removing the identifier from the bytes; loading the bytes into a buffer and transmitting the bytes to a destination exchange telephone device.

Statistical method of data compression and decompression

Method of processing asynchronous characters in order to transmit them over an ATM network (,or,) of a communication system having a first endpoint (,) and a second endpoint (,), a first group of one or more Data Terminal Equipments (DTE) being connected to the first endpoint and a second group of one or more DTE's being connected to the second endpoint and wherein data are transmitted from the first group of DTE's to the second group of DTE's by the intermediary of the ATM network. Asynchronous characters are processed as they are received at the first endpoint from the first group of DTE's by removing Start, Stop and parity bits. A block is generated consisting of one or more processed asynchronous characters received in sequence from a particular DTE of the first group of DTE's. An ID byte is added to the block to identify the transmitting DTE, yielding a data unit. A plurality of such data units are concatenated to build a data unit frame. The data unit frame is encapsulated into an ATM cell structure in order to transmit the frame over the ATM network.

Method and apparatus for efficiently transporting asynchronous characters over an ATM network

An apparatus and method that determines the end-to-end transit delay at each node of a path, in accordance with the selected probability value indicative of the probability to experience a delay at each node that is smaller than the computed transit delay. Then the computed transit delays per nodes are combined to obtain the end-to-end delay of the path, the combination being either an arithmetic operation or a convolution operation. A method to approximate the convolution operation is also disclosed.

End-to-end delay estimation in high speed communication networks

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.

Repetitive pattern removal in a voice channel of a communication network

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.

AAL-5 SSCS for AAL-1 and AAL-2 in ATM networks

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.

Method and apparatus to reduce jitter and end-to-end delay for

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.

Data processing method for efficiently transporting multimedia packets

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.

Method and system for monitoring traffic to optimize the bandwidth

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.

Method and a system for silence removal in a voice signal transported

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.

Method of transferring structured data of constant bit rate traffic in

A method dynamically changes the bit rate or bandwidth of constant bit rate data structures in an Asynchronous Transfer Mode (ATM) communications environment. The method defines within a data channel a Change indicator (CI) indicative of an end user's request for a bit rate change within said data structures. The Change Indicator is continuously transmitted from a source side to a destination side in the ATM environment, along with the data structures on said data channel. Upon receipt at the destination side of a user's request of a bit rate change, the destination side modifies the value of said Change Indicator. The source side, acknowledges the modification of the value of the Change Indicator, whereby the transmission of data structures continues on the data channel with a new constant bit rate.

Dynamically structured data transfer mechanism in an ATM network

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.

Voice circuit emulation system in a packet switching network

Apparatus for recovering lost buffer contents in a data processing system uses a memory divided into a plurality of buffers provided with buffer control blocks, through which source and destination users exchange information. A buffer management circuit is responsive to requests from users for allocating buffers to source users in order that source users may store the information to be sent to the destination users. This circuit builds buffer queues and dequeues buffers from the queues to send the information contained therein to the destination users and releases the buffers afterwards. A time mark register is settable to n different values in a predetermined order. The value of the time mark register is changed at the expiration of a time period P. Each time a buffer is allocated to one user, the current value of the time mark register is written into a time mark field of the buffer control block and a state field is set to a first value (leased). When the buffer is released, the state field is set to a second value (released). The contents of the buffer control blocks are read at regular time intervals t after period P, and the state field of every buffer control block is tested to determine whether it is set to the second value. If not so set, the time mark field is compared with the value the time mark register had at the time t-xP, where x is a number such as 1&lt;x&lt;n, and the buffer is released in case of equality.

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Saint Laurent du Var, France

Maurice Duault