Nortel Networks Corporation

Northern Telecom Limited

Paul Vincent Coverdale

Rafi Rabipour

William Navarro

Leo L Strawczynski

Chi Lam Wong

Paul Mermelstein

Chung Cheung Chu

Christopher Michael Forrester

Wayne Raymond Getchell

Paxton Smith

Rafi Rapipour

Sejal Kamani

Ben Kwong

A conference bridge adapter for processing data carried on a media path between, on the one hand, a conference bridge operative to communicate composite packets carrying media information and auxiliary information pertaining to the media information and, on the other hand, an endpoint characterized by an inability to exchange composite packets with the conference bridge. In one direction, a stream of composite packets is received from the bridge. An output media stream without auxiliary information is generated from the media information in each received composite packet. In the opposite direction, a stream of packets carrying only media information is received from the endpoint. Auxiliary information is derived from the media information and a stream of composite packets is generated by combining the media information and the auxiliary information. Thus, a media conference can be established between seemingly incompatible network elements.

Adapter for use with a tandem-free conference bridge

A method of connecting a telephone call through one of a plurality of networks where one of the plurality of networks is an internet protocol network is provided. A first factor for an acceptable quality of service level is received from a user. A second factor responsive to the quality of service for the internet protocol network is determined. The telephone call is connected through the internet protocol network if the second factor is greater than the first factor, otherwise, the telephone call is connected through one of the plurality of networks other than the internet protocol network.

Method and apparatus for providing a configurable quality of service threshold for voice over internet protocol

A method and apparatus for controlling an operative setting of a communications link is provided. The communications link is capable of acquiring a plurality of operative settings. Audio quality in the communications link under different operative settings is compared and an operative setting is selected at least in part on the basis of this comparison. A control signal is sent to at least one component in the communications link to cause the communications link to attempt to acquire the selected setting.

Method and apparatus for controlling an operative setting of a communications link

A conference bridge is provided for managing a conference between media signal sources generating media data packets conveying encoded media information and encoding type information. The media signal sources, in order to enter a conference, generate link messages including respective supported encoding types and transmit the link messages to the conference bridge. The conference bridge receives the link messages and processes them to derive a first common encoding type supported all media signal sources in the conference and a second encoding type supported a subset of media signal sources in the conference. The conference bridge generates link message reply signals conveying the first encoding type and the second encoding type the subset of media signal sources and conveying at least the first encoding type to the media signal sources other than the subset of all media signal sources. Each media signal source is responsive to the link message reply signal to render active at least the first encoding type and selectively render active the second encoding type.

Method and apparatus for packet-based media communication

A high layer protocol organizes Unidirectional Streaming Services (USS) data into frames and said data is transmitted to the wireless user's terminal. The USS data is not delivered until a criteria is satisfied, e.g., the receive buffer is filled to an appropriate value. After the USS data is received, the receiver verifies the received frames. If an error is detected a message is sent from the wireless terminal to the server (providing the unidirectional streaming service) requesting retransmission of the corrupted frame. This is a form of ARQ protocol. If the retransmitted frame arrives prior to the time that frame needs to be delivered, the corrupted frame is replaced by the retransmitted frame. Otherwise, if the retransmitted frame is not received prior to the time that frame needs to be delivered, the corrupted frame is reconstructed. Any retransmitted frame which arrives too late is discarded. An additional benefit of the present invention is that since the frames are buffered prior to delivery, interpolation, as opposed to extrapolation, can be used to improve the reconstruction quality of the corrupted frame. This optional feature of the invention uses both the preceding and succeeding frames, which are available in the buffer, to provide a better estimate of the corrupted frame.

Unidirectional streaming services in wireless systems

This invention relates to a method and an apparatus for processing digital audio signals that may reduce the signal degradation occurring when the signal is exchanged between two communication terminals equipped with vocoders in a communication network. The solution proposed by this invention is to provide a communication terminal with a vocoder including a decoder section provided with a plurality of decoding units. A switch activates a selected one of the decoding units in dependence of the format of the compressed audio signal data frames received from a remote communication terminal. This system allows the communication terminal to support a number of different speech compression formals. In order to achieve simplicity and low cost, the communication terminal is provided with a single encoding unit. This results in an asymmetric arrangement where the communication terminal has a large number of decoding units than encoding units. The great majority of the speech compression algorithms deployed in wireless and Internet telephony standards have the property that their speech decoders are of far less computational complexity than their respective speech encoder. Therefore, a low-cost terminal can be produced which supports a low complexity speech encoder unit and a variety of speech decoder units.

Method and apparatus for asymmetric communication of compressed speech

In recent years, the telecommunications industry has witnessed the proliferation of a variety of digital vocoders in order to meet bandwidth demands of different wireline and wireless communication systems. The rapid growth in the diversity of networks and the number of users of such networks is increasing the number of instances where two vocoders are placed in tandem to serve a single connection. Such arrangements of low bit-rate codecs can degrade the quality of the transmitted speech. To overcome this problem the invention provides a novel method and an apparatus for transmitting digitized voice signals in the wireless communications environment. The apparatus is capable of converting a compressed speech signal from one format to another format via an intermediate common format, thus avoiding the necessity to successively de-compress voice data to a PCM type digitization and then recompress the voice data.

Method and apparatus for improving the voice quality of tandemed vocoders

A method of increasing the terminal coupling loss in a digital half-duplex handsfree terminal is provided. When the terminal receive channel is active an idle code which is effectively zero is transmitted by an idle code generator. When the terminal transmit channel is active a normal speech signal is transmitted. To determine which channel is active, a receive speech signal level is measured at the receive channel input, and a 'virtual' transmit speech signal is computed from a signal in the transmit channel before the idle code generator.

System and method for providing high terminal coupling loss in a

A system for notifying users of digital wireless communications that a mobile unit is moving out of range and communication is about to be lost. The system inserts an audible indication as the received signal strength varies. The audible indication can be in the form of noise which is inserted in such a manner that a digital mobile unit mimics the behavior of an analog mobile unit which is moving out of range, in order to provide a readily identifiable indication to the user that the wireless handset is about to go out of range. Such a system can be used to indicate to both users of wireless units, and to telephone users which are in communication with wireless units, that communication is about to be lost.

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Nepean, Canada

Paul Vincent Coverdale