Abstract: A mobile communications system employs extended channel assignment messaging during the call setup portion of a mobile call. As part of the call setup process, a mobile station sends an access request message to a primary base station that includes a list of alternate base stations. As call setup continues, the mobile station employs diversity reception in monitoring the paging channel of three base stations, each selected as a function of the strength of the signal-to-noise ratios of their respective pilot tones.

Abstract: In a cellular system based on industry standard IS-95 CDMA (code division multiple access), a symbol error count based reverse link outer loop power control technique uses adaptive SER targets. In particular, a base station uses a 2nd order statistic, e.g., standard deviation (variance), to identify, or act as a signature of, a particular cellular (wireless) communications environment. The base station monitors the standard deviation of the symbol error count of a received signal (transmitted from a mobile station). The target signal-to-noise ratio (EbT/N0T) of this received signal is adjusted as a function of the value of the standard deviation and the adjusted EbT/N0T target is used to provide power control.

Abstract: In a packet voice system, a dynamic build-out delay approach in a receiver during the duration of a call. In particular, the build-out delay is applied at least twice during the duration of the call. In one embodiment, a packet voice system includes an ATM Adaptation Layer Type 2 (AAL-2) and Service Specific Convergence Sublayer (SSCS) System. The receiver portion of the SSCS System recovers AAL-2 packets and plays back the compressed audio to a voice decoding element. In providing playback, the receiver applies the build-out delay at the start of each talk-spurt. The voice decoding element provides an uncompressed audio stream. In another embodiment, the receiver portion of the SSCS System applies the build-out delay at the start of the each talk-spurt as a function of the length of the previous silence interval.

Abstract: An AAL2/SSCS packet voice system multiplexes various forms of voice-band traffic including voice packets, fax packets, and data packets into a virtual circuit (VC). This AAL2/SSCS packet voice system executes a dynamic call admission algorithm that takes into account call type in deciding whether to admit a new call to the VC. In particular, this approach takes into account different bandwidth needs for different call types. The AAL2/SSCS packet voice system also performs bit or block dropping on voice packets to mitigate the effects of traffic congestion. The bit or block dropping is done based on the packet queue fill value exceeding at least one queue threshold. Further, the AAL2/SSCS packet voice system also dynamically varies a queue threshold as a function of capacity.

Abstract: In a receiver, a received sync-word is fed into two or more filters, where each filter further comprises a matching filter and a sidelobe reduction filter. Each matching filter is matched to a particular subset of k samples of the known sync-word, e.g., k=3, and generates an m-tap estimate, where m.gtoreq.2. The output signal of each matching filter is applied to a respective sidelobe filter, which reduces sidelobes resulting from the imperfections of matching the received sync-word in the matching filters. As a result, from each received sync-word, each filter forms a channel impulse response estimate of the communications channel at more than one symbol time. Each channel impulse response estimate comprises the amplitude as well as the phase of the channel impulse response.

Abstract: A packet voice system includes an ATM Adaptation Layer Type 2 (AAL-2) and Service Specific Convergence Sublayer (SSCS) System comprising a transmitter and a receiver. A portion of the 32 codepoints of the five bit RES (or UUI) field of the AAL-2 header are predefined to indicate an extended header. This a prior definition is stored in tables in both the transmitter and the receiver. The extended header itself comprises an additional octet appended to the AAL-2 header. A sequence number is normally carried in the RES field for the packet stream of voice calls that require sequence numbers. When the extended header is used, the sequence number is part of the extended header and thus messages are conveyed from the transmitter to the receiver without causing an interruption to sequence number assignment messages The transmitter dynamically uses the extended header and discontinues use of the extended header after a predefined duration of time or upon receiving an acknowledgment from the receiver.

Abstract: A blind convergence technique is restricted to using a subset of equalizer output samples. Illustratively, a receiver implements a windowed MMA approach. In this windowed MMA approach, a sample window overlays the two-dimensional plane representing the set of equalizer output samples. Only those equalizer output samples appearing within the sample window are used during filter adaptation.

Abstract: A decision feedback equalizer (DFE) comprises a feed-forward filter and a feedback filter. Blind training of the DFE is performed using a statistical-based tap updating algorithm for the feed-forward filter, and a symbol-based type of tap updating algorithm for the feedback filter.

Abstract: A user accesses the Internet through a hybrid data communications system architecture in which upstream service is provided via the local loop of the public switched telephone network (PSTN) and downstream service is provided via a cable television (CATV) network. The Cable Modem Data Termination system (CMTS) includes a routing table. Upon detection of a CATV service interruption, the CMTS re-routes IP traffic destined for a destination IP address (IP1) associated with the user to a different IP address, or endpoint, as a function of the routing table. In this instance, the different IP address is associated with equipment that provides downstream IP service to the user via the local loop of the PSTN.

Abstract: In a simultaneous voice and data communication system, a stream of signal points is partitioned into a plurality of symbol blocks, each symbol block including a data segment and a control segment. The data segment carries information from a user, i.e., user data, while the control segment provides control information. A voice signal is then added to at least a portion, or all, of the signal points of each symbol block to provide for simultaneous voice and data transmission to an opposite endpoint. The control information may represent information from a secondary data source, and/or may include information about the characteristics of the succeeding block, e.g., the user data rate, and information pertaining to characteristics of the communications channel.