Abstract:
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.
Abstract:
The invention provides a mechanism for allowing a shared memory/parallel processing architecture to be used in place of a conventional uni-processing architecture without requiring code originally written for the conventional system to be rewritten. Exclusive Access and Shared Read Access implementations are provided. A rollback mechanism is provided which allows all the effects of a task to be undone.
Abstract:
Multi-media networks will require that a data flow be given certain quality-of-service (QOS) for a network connection but pre-negotiation of this sort is foreign to the current data networking model. The real time traffic flow in the data network requires distinct limits on the tolerance to delay, and the variations in that delay. Interactive voice and video demand that the total delay does not exceed the threshold beyond which human interaction is unacceptably impaired. The present invention allows the network to discover the nature of the service for each traffic flow, classifies it dynamically, and exercises traffic conditioning by means of such techniques as admission control and scheduling when delivering the traffic downstream to support the service appropriately.
Abstract:
Multi-media networks will require that a data flow be given certain quality-of-service (QOS) for a network connection but pre-negotiation of this sort is foreign to the current data networking model. The real time traffic flow in the data network requires distinct limits on the tolerance to delay, and the variations in that delay. Interactive voice and video demand that the total delay does not exceed the threshold beyond which human interaction is unacceptably impaired. The present invention allows the network to discover the nature of the service for each traffic flow, classifies it dynamically, and exercises traffic conditioning by means of such techniques as admission control and scheduling when delivering the traffic downstream to support the service appropriately.
Abstract:
The invention provides a method and a system for selectively delivering information to callers in an AIN environment. In accordance with the method the service switching point to which is connected the terminal equipment of the caller monitors the condition of the telephone line to detect an AIN trigger, such as an off-hook event among other possibilities. When such event occurs, the service switch point formulates a message query. The query is routed to a service control point via one or more service transfer points, that holds knowledge of the service subscribed by the caller and the information to be displayed for this service. The service control point analyzes the request and assembles the appropriate response. That response is then returned to the service switch point that formats it properly and delivers it to the terminal equipment of the caller. In one embodiment the information delivery is effected during call establishment. In a variant, information is delivered while no call is being attempted.
Abstract:
A method and system is disclosed for triggering handoff of a call from a CDMA network to an AMPS network on which it is overlaid responsive to a determination that the CDMA RF link has degraded to such an extent that the call quality will be degraded significantly or the call is likely to be dropped is disclosed. A handoff of a call from a CDMA network to an AMPS network if, at a given time: (1) the number of forward link erasures is greater than a first maximum number of forward link erasures; or (2) the number of reverse link erasures is greater than a first maximum number of forward link erasures; or (3) the number of forward link erasures is greater than a second maximum number of forward link erasures and the forward traffic channel gain is greater than a maximum forward traffic channel gain; or (4) the number of reverse link erasures is greater than a second maximum number of reverse link erasures and the ratio of energy per bit to noise power spectral density on the reverse link ("EB/NO") is greater than a maximum EB/NO. In alternative embodiments, the present invention may be used to trigger handoff of a call from a first CDMA cell site to a second CDMA cell site or to an AMPS cell site serving a cell on which the cell served by the first CDMA cell site is not overlaid.
Abstract:
A robust method for determining the boundaries of cells and the associated reliability of the RF coverage within these boundaries is presented. The invention accurately determines the average range from the base station to the cell edge from RF signal strength measurements with a linear regression approach. The accuracy of this estimate is quantified both as a range uncertainty (e.g. +/- 100 meters) and as a cell coverage reliability (i.e. area/edge) through 1) simulation, 2) analysis of real data, and 3) theoretical analysis. It is shown that if the estimate of the cell radius meets the desired accuracy, then the corresponding estimates of coverage reliability (both area and edge) are more than sufficiently accurate. It is recommended that radio survey analyses incorporate this test as part of the coverage validation process.
Abstract:
A fixed wireless access system comprising a base station for bidirectional communication with a switching center, a private branch exchange (PBX) for establishing bidirectional communication with a plurality (K) of user stations, and a wireless trunk interface (WTI) for bidirectional wireless communication with the base station over an air interface and for bidirectional communication with the PBX over a trunk line is disclosed. The invention also proposes to deliver the station directory number (DN) over the fast associated control channel (FACCH) or the slow associated control channel (SACCH) of the IS-54B air interface. This is achieved using DTMF signalling or by providing a new set of messages in the extended protocol (EP) capability of the IS-54B air interface.
Abstract:
An apparatus and method for maintaining records in a telephone directory involves maintaining a personal directory of at least one record associated with at least one party with which a telephone call is conducted. The record has a field for identifying the party and a frequency of use field for storing a number representing the number of times a call is conducted with the party. In response to a call conducted with a party, the personal directory is searched for a record associated with the party and upon finding such a record, the contents of the frequency of use field of the record are incremented and the records of the personal directory are sorted in an order dependent upon the contents of the frequency of use fields.