Abstract: A modem used for communicating data. The modem may include a packet generator and modulator. The modulator may include an electronic device configured to select at least one frequency from at least three or more candidate frequencies. At least one oscillator may be in communication with the electronic device and configured to generate respective oscillation signals in response to receiving the frequency(s) from the electronic device. A computational element may be in communication with the oscillator(s) and be configured to receive the oscillation signal(s) and produce a modulated audio signal based on the oscillation signal(s).

Abstract: A method for receiving data in a mobile terminal and the apparatus thereof according to the present invention can reduce the areas incapable of data reception by allowing a user's mobile terminal to receive data from a first mobile terminal with a good data reception state that receives data from a base station, rather than the user's mobile terminal directly receiving data from the base station. The user's mobile terminal receives the received data from the first mobile terminal having the good reception state by a wireless communication method.

Abstract: A plurality of communication conference sessions coupled to one another such that when a communication right is requested, a check is made to ascertain whether the communication right has already been allotted to subscribers in a different, coupled communication conference session. In such a case, the communication right is denied, and otherwise, the communication right is granted.

Abstract: A conference engine operates in a multiplex processing scheme and requires a signal hardware summer to process all of the programmed conferences, all of the outputs and all of the input channels. The conference engine is only limited by the number of available input channels, the number of desired conference registers, the speed of the implementation, and the amount of memory available on the target system. The conference engine allows the number of available conference channels, the number of conferences per bridge and the number of bridges to be scaled. Each bridge contains a unique set of conference registers and has full access to the input channel data. Additional conference bridges are added by increasing the operating clock/memory to meet the processing requirements of the additional bridge. Conference registers in each conference bridge can be added/removed depending on the system requirements and are independent of the number of input channels.

Abstract: Presented herein are systems, methods, and apparatus for displaying pictures. In one embodiment, there is presented a decoder system for decoding video data. The decoder system comprises a port and a transport processor. The port receives packets carrying encoded video data from a plurality of video streams. The transport processor adds a header to encoded video data from at least one of the packets. The header identifies a particular one of the plurality of video streams, wherein the at least one packet is from the particular one of the plurality of video streams.

Abstract: A method for joining a virtual talk group includes facilitating communications of a virtual talk group. The virtual talk group includes a plurality of endpoints of different communication networks. The method also includes recording the communications of the plurality of endpoints of the virtual talk group. The method also includes adding a first endpoint to the virtual talk group and presenting the first endpoint with the recorded communications of the plurality of endpoints of the virtual talk group.

Abstract: Transcoders and mixers having reduced algorithmic delay and processing complexity. An improved mixer for signals having encoded speech parameters wherein the parameters obtained through decoding are used by a parameter estimator to improve the encoding by providing a parameter estimate for the mixed signal. In the case of pitch parameters, the mixer uses the principle of strong-pitch-domination. The mixing of wideband signals is simplified by performing mixing of individual lower and upper sub-bands. A transcoder and a mixer that converts a wideband signal into a narrowband signal relies upon high frequency suppression. A transcoder and a mixer that converts a narrowband signal into a wideband signal relies upon filter combination.

Abstract: An apparatus, method, and computer program process audio information in a communication system. The apparatus, method, and computer program identify one or more silent channels of a communication session. The apparatus, method, and computer program also mix audio information from two or more non-silent channels. The mixed audio information is provided to one or more of the channels of the communication session.

Abstract: A method and system for real-time, low latency, high quality audio conferencing are disclosed. The system allows delivering low latency during peer to peer transmission of high quality compressed audio streams between remotely located participants. The system provides transmission of audio as well as any audio data with low latency and high quality. The system solves latency problems to enable participants in different locations to stay in synchronization while performing live over the Internet in multiple locations.

Abstract: Audio and video frames are synchronized by hashing an audio frame at a sender and combining the resultant hash value with the video frame. The audio frame is transmitted over an audio network, such as a telephone network, and the video frame is transmitted over a digital network, such as an intranet. The audio frame may be combined with additional audio signals from an audio bridge. The receiver receives the audio signal from the audio bridge and performs the same hash function on the mixed signal as was performed on the original signal. The receiver correlates the hash value on the mixed signal with the hash value included with the video frame (wherein the video frame is one of several video frames buffered by the receiver). The receiver can thus identify the video frame that corresponds to the audio frame and render them simultaneously.

Abstract: A single chip integrated circuit wireless data processor demodulates N separate data signals from M separate antennas simultaneously. The multi-antenna processor can be coupled to a baseband processor on the IC, so that it responds to changing channel conditions between two access points, and selectively kicks in if there is noise, interference, frequency fading, a need for an enhanced data rate, a need for an increased operating range, etc. to improve a performance of the baseband processor.

Abstract: Method for dynamically optimizing bandwidth allocation in a variable bitrate conference environment. Conference means with two or more outputs are provided, where each one can output data at different rates, in order to support two or more endpoints which may have different media rates. Two or more endpoints are connected to these conference means for participating in the conference. Whenever more than one video rate is used by participants during the conference, each set of output rates is selected from all possible combinations of output rates in the conference means, wherein the lowest output rate in each selected set is the entry rate of the endpoint joining the conference at the lowest rate. A Quality Drop Coefficient (QDC) for each endpoint that joins the conference is determined for each selected set, wherein the QDC is computed according to the endpoint entry rate and the highest rate, among the output rates of each selected set, that is lower or equal to said endpoints' entry rate.

Abstract: An improved system and method is disclosed for call handling in a packet network such as a CDMA, UMTS, GSM, and other networks similarly configured. The system may include an anchor media gateway that remains in a communication path even if the call has been handed off to another media gateway. The anchor media gateway includes first and second ports for establishing first and second links to transfer packet communications to and from the first and second entities, respectively. First and second packet connections are provided for interconnecting the first and second links, respectively, to an audio server. A third link connected to the audio server and is provided for transferring packet communications to and from the third entity. The audio server manipulates the packet communications received from the first, second, and third entities and provides the manipulated packet communications to the first, second, and third entities.

Abstract: A method and an apparatus for controlling a conference call is a system which comprises a conference bridge for combining several audio signals arriving from terminals and for transmitting the combined signal further to at least one of the terminals using at least one transmitting codec. The conference bridge comprises a spatialization unit for creating a spatial effect. The number of simultaneously active audio components, such as speech signals, is determined from the audio signals of the conference bridge inputs. At least one transmitting codec is selected for use on the basis of the number of simultaneously active audio components and the combined signal is transmitted to at least one terminal using the selected transmitting codec.

Abstract: In one embodiment, the invention provides a method comprising loading a set of transport components into a memory; initializing each transport component of the set of transport components to listen on a particular conferencing interface using a listen string to associate with a network interface, each transport component of the set of transport components listening to a different conferencing interface, the listen string encapsulating parameters to one or more calls to listen; receiving an incoming call signal on the network interface having an incoming conferencing interface; processing the incoming call signal to detect the incoming conferencing interface; and launching an application based on the incoming conferencing interface. Other embodiments of the invention are also described.

Abstract: Techniques are disclosed for conference expansion in a conferencing system. For example, a method of providing for conference expansion in a conferencing system includes the following steps. At least one port is reserved in an original conference allocated in the conferencing system. A hub conference is allocated, when a maximum number of participants for the original conference is reached. A calculated number of ports in the hub conference are reserved for linking to the original conference and to a corresponding number of expansion conferences when allocation of at least one of the number of expansion conferences is called for. The hub conference is linked to the original conference via the reserved port of the original conference and one of the reserved ports of the hub conference.

Abstract: The present solution relates to a telecommunications system where information is maintained of communication groups available to a user terminal (UE). A communication server (PoC server) of the system is able to receive a request message (3-1) sent by the user terminal (UE), the request message (3-1) comprising an indication of the communication group the user wishes to speak to. If the request can be accepted, the communication server (PoC server) transmits a floor granted message (3-2) to the user terminal (UE). According to the present solution the communication group requested by the user is set (3-3) as the priority communication group for said user at least for a predetermined time period.

Abstract: A method and system is disclosed for automatically discovering Cisco Discovery Protocol (CDP) nodes a network in real time. The discover process transmits a signal from a network manager to a first CDP node of the network. The signal requests information regarding additional CDP nodes known to the first node. The network manger receives a response is that identifies additional CDP nodes known to the first CDP node. The transmitting and receiving steps are then repeated for each additional CDP node identified. A list containing information of all identified nodes is then stored.

Abstract: A conferencing system includes an input configured to receive N encoded speech signals from N terminals, and a signal processing arrangement configured to determine L encoded signals, of the N encoded speech signals, each indicative of an amount of sound that is louder than amounts of sound indicated by signals of the N encoded signals other than the L signals, the signal processing arrangement being further configured to produce at least N minus L sets of signals similar to the L signals and to transmit at least a set of the similar signals toward each of the terminals other than the terminals from which the L signals were received.

Abstract: An apparatus is provided for configuring data cells received in a telecommunications process in a continuous stream of data cells of fixed length and each comprising a header and a user data part. The apparatus comprises a processing unit, adapted and configured so that it is able to check data cells for the presence of empty cells, discard the data cells consisting of empty cells and then check the user data parts of the data cells less the empty cells—without the need to buffer the same—as to whether they belong together, and then to assemble the user data parts of the data cells belonging together into a frame. The apparatus may be integrated in, a modem and is also particularly suitable for data cells existing as ATM cells for receiving by an ATM system. The invention relates in addition to a correspondingly sequencing method for configuring data cells.

Abstract: In a wireless code-division multiple access (CDMA) system (100), a talkgroup (101) of subscriber units is provided. A sub-talkgroup (102) of subscriber units, forming a part of the talkgroup is assigned at least one inbound channel (416-417). The talkgroup is assigned outbound channels (415). Members of the sub-talkgroup may simultaneously transmit voice information (410-411) using the at least one inbound channel, which voice information is summed (412) and re-transmitted to the talkgroup using the outbound channels. Voice information is summed so that an individual talker receives summed voice information without the individual subscriber's voice content. Subscriber units in the sub-listengroup are allowed to transmit voice information only after requesting, and receiving, an additional inbound channel.

Abstract: A system and method are disclosed for packet voice conferencing. The system and method divide a conferencing presentation sound field into sectors, and allocate one or more sectors to each conferencing endpoint. At some point between capture and playout, the voice data from each endpoint is mapped into its designated sector or sectors. Thereafter, when the voice data from a plurality of participants from multiple endpoints is combined, a listener can identify a unique apparent location within the presentation sound field for each participant. The system allows a conference participant to increase their comprehension when multiple participants speak simultaneously, as well as alleviate confusion as to who is speaking at any given time.

Abstract: The present invention provides a method and system for distributed conference bridge processing in Voice over IP telephony. A distributed conference bridge multi-casts mixed audio content of a conference call in a way that reduces replication work at the mixing device. A switch is coupled between one or more audio sources and a network interface controller. A conference call agent is used to establish and end a conference call. The switch can be a packet switch or a cell switch. In one embodiment, a distributed conference bridge is implemented in a media server.

Abstract: In a network connected with a plurality of electronic devices 11 to 16, an electronic device 13 as a leader inquires about system resource capacities from electronic devices 11 and 14 which can become controllers, in a step S1. The electronic device 13 further inquires about sizes of DCMs from electronic devices 12, 13, 15, and 16, respectively, in a step S2. The electronic device 13 calculates the memory capacities of the electronic devices 11 and 14 to which DCMs can be assigned, and the size of the DCM of each electronic device, in a step S3. Further, the electronic device 13 determines whether or not the DCM of each electronic device can be assigned to the electronic devices 11 and 14, in a step S4, and then assigns DCMs which can be possessed by the devices 11 and 14, respectively, in a step S5.

Abstract: A data communication system includes a plurality of terminal apparatuses #1-#n for periodically transmitting data and retrieving desired data accommodated in at least one time slot in received multiplex data, and a data storage and distribution apparatus for subjecting received data from the plurality of terminal apparatuses to time division multiplexing so as to produce time slots each corresponding to a respective one of the plurality of terminal apparatuses, reproducing resultant multiplex data, and distributing the multiplex data to the plurality of terminal apparatuses.

Abstract: A system and method for improving the intelligibility of a moderator during a multi-party communication session includes receiving a plurality of participant voice streams from a plurality of respective conference participants. An incoming moderator voice stream may be received from a moderator. The plurality of participant voice streams and the moderator voice stream are transmitted such that the intelligibility of the moderator voice stream is improved relative to at least one of the participant voice streams.

Abstract: A method and apparatus for providing unified conferencing services in a telecommunications system is described. A DSP chip in the system contains an internal CPU and on-chip memory storage device. The CPU is programmed such that when a conference has been established, the time slots or channels of PCM information are collected and stored. Channels are sub-sampled at a pre-determined optimal time interval. The energy for each sub-sample, which is a mathematical summation over time utilizing a sum of squares equation, is calculated for each channel. These calculated energies are stored in an associated array. The energies are surveyed to select the N loudest channels. The input signals (the voice information) for those N loudest channels are used to form the conference. More specifically, the sum of the N loudest channels is generated. This sum forms the basis for a broadcast output.

Abstract: An IP (Internet Protocol) conference telephone system feasible for an IP-PBX (Private Branch Exchange) includes a communication control connected to an IP network. When a connection for a conference is set up between a plurality of telephones accommodated in the IP network, each telephone packetizes a voice signal and sends the resulting packet to the network to address it to a conference trunk. An IP bridge interconnected to the network receives packets via the network 10, multiplexes the voice signals of the packets on the channels of a down-going highway each corresponding to a particular telephone, and sends the multiplexed voice signals to the conference trunk. The conference trunk performs addition and subtraction with the above voice signals and returns the results of addition and subtraction on an up-going highway. This allows a conference to be held between the telephones. A multicast router may be substituted for the conference trunk, in which case the network will be adaptive to multicast.

Abstract: A system for reducing the cost of network managment by using a proxy agent and subchannel communications so fewer SNMP licenses and fewer protocol stacks are needed. Subchannel communication is achieved in a plurality of different embodiments. Embodiments having single subchannel transceivers, multiple transceivers, single multiplexer and multiple multiplexers are disclosed. An NMS process using routing table CRC to automatically detect when the NMS topology information is incorrect and automated topology discovery is disclosed. A process for automated discovery of redundant cables during automated topology discovery is disclosed.

Abstract: A time-slot interchanger (TSI 100) is used to control both time-slot access and distribution of signals to physical or virtual signal-processing circuits (106-116). The TSI includes a control table (200) with an entry (202) for each time slot, that specifies the source (204, 205) and destination (230, 231) of signals received during the corresponding time-slot interval and the processing (208-228) of those signals. The TSI is coupled to the signal processing circuits, and sends the received signals to the signal processing circuits for processing along with control signals to effect the processing specified by the corresponding control-table entries. For conferencing, the TSI sends all signals for each conference in sequence to the conferencing circuit (108) before starting and sending any signals for any other conference, thereby allowing a single conference accumulator to support any number of conferences of any size.

Abstract: A receiver manages multiple speakers in a packet network telephone call. A packet gateway receives audio packets from the multiple speakers over the packet network. Memory in each receiver stores the audio packets and information about the multiple speakers in the telephone call. A processor selects which audio packets and what speaker information to retain in memory. The processor determines which of the selected audiopackets to store in memory and mix together to produce an audio output signal by determining from the speaker information which of the multiple speakers are actively talking and which speaker has not actively talked for the longest period of time.

Abstract: The invention relates to a private digital communication apparatus for interconnecting a plurality of interface modules which communicate through a common bus. In particular, it concerns a multipoint apparatus which uses a two-wire cable as a bus, and the access to the bus is carried out through the time multiplex technique (TDMA).

Abstract: A multipoint digital simultaneous voice and data (DSVD) conferencing system includes a plurality of local DSVD modems to communicate with a plurality of remote DSVD modems and a bridge to transfer data and digitized voice signals among the plurality of remote DSVD modems by way of the plurality of local DSVD modems. A plurality of simultaneous conferencing connections are maintained between the plurality of local DSVD modems and the plurality of remote DSVD modems. The system support a multipoint DSVD conferencing session connecting the plurality of remote DSVD modems.

Abstract: A multipoint conferencing system communicates with a first remote digital simultaneous voice and data (DSVD) modem and a second DSVD modem. The multipoint conferencing system has a first DSVD modem adapted to communicate with the first remote DSVD modem and a second DSVD modem adapted to communicate with the second remote DSVD modem. Each DSVD modem has an analog to digital converter and a digital signal processor coupled to the analog to digital converter for receiving data from the remote DSVD modems. The system also has a bridge coupled to the first and second DSVD modems for transferring data between the first and second remote DSVD modems.

Abstract: A PC-based home automation system uses a low data-rate transport layer and COM-based software components for control of devices in a home automation network. The home automation system is merged with a messaging-based HAVi-network that uses IEEE 1394 as a high data-rate transport layer. The HAVi-network controls audio/video equipment in a home entertainment system. The home automation services and devices are registered as a HAVi-compliant elements with the HAVi network's FAV or IAV device. The home automation resources (devices and services) have both COM OLE Automation Interfaces and HAVI-compliant interfaces to permit control of the home automation system from the HAVi-network.

Abstract: A quality dispatch service on a CDMA based wireless system. To achieve spectral efficiency and fast access for follow-up calls, in response to a request by a talkgroup member to have a group call established, the non-requesting (listening-only) members are allowed to establish a low-rate (non-voice) signaling link in the reverse direction (inbound) for the limited purpose of (i) providing forward power control information to the fixed infrastructure; (ii) requesting soft hand-off when required; and (iii) maintaining the correct reverse link power control to allow for fast channel access for a follow-on call. In accordance with the illustrative embodiment, the establishment of low-rate inbound signaling links occurs prior to beginning the voice communications.

Abstract: A distributed digital switching system is provided that includes a plurality of service controllers (20) which are interconnected to each other through a network (10). A network arbitrator is provided that controls the flow of traffic on the network (10). Each of the service controllers (20) is operable to provide a switching configuration between inbound data to the service controller (20) and network terminations associated with each of the service controllers (20). A processor (150) on each of the service controllers (20) controls the operations thereof with the voice information stored in a memory (208) which can be configured with a circuit (204). Each of the service controllers (20) has associated therewith a conference circuit for defining and creating a conference that can access all inbound data and provide interconnections therebetween and output this on a single outbound conference channel. There can be multiple conferences created at each service controller (20).

Abstract: A time division multiplexing ("TDM") digital matrix intercom system for connecting intercom stations. The system includes a digital switching system receiving audio input communications signals from at least some of the intercom stations, and sending audio input communications signals to at least some of the intercom stations. The digital switching system includes a TDM system for broadcasting on a TDM bus multiple digital communications signals including such digital audio input signals. The TDM system includes one or more interfaces connecting intercom stations with the TDM bus.

Abstract: In a wireless code-division multiple access (CDMA) system (100), a talkgroup (101) of subscriber units is provided. A sub-group (102) of subscriber units, forming a part of the talkgroup is assigned at least one inbound code (416-417). The entire talkgroup is assigned an outbound code (415). Members of the sub-group may simultaneously transmit voice information (410-411) using the at least one inbound code, which voice information is summed (412) and re-transmitted to the talkgroup using the outbound code. Subscriber units in the talkgroup, but not included in the sub-group, are allowed to transmit voice information only after requesting, and receiving, an additional inbound code.

Abstract: It is proposed to use several distributed conference units (KE1, KE2) in a conference system (KS) for carrying out conferences, particularly broadband conferences, and to mix the signals (S1 . . . S4). The signals (S1 . . . S4) from subscriber stations (T1 . . . T4) involved in the conference are not necessarily mixed in a single conference unit, rather the mixing is distributed to several conference units (KE1 . . . KE2). In this way it is possible to mix signals that have a partially common transmission path in the communication system into a component signal (12) by means of a first conference unit (KE1). This mixed component signal (S12) is then transmitted to a second conference unit (KE2) and is mixed in this second conference unit (KE2) with one or more other signals (S3, S4). Since the mixed component signal (S12) during the transmission occupies the same bandwidth as a single signal (S1 . . . S4), the transmission capacity of the communication system (KS) needed for the conference can be kept small.

Abstract: A method and a system for audio communication between a plurality of users at a plurality of sites utilizes a set of audio input sensors at each site. Each set of audio input sensors binaurally senses an auditory space in proximity thereto. A metaphorical representation of each of the sites is provided. Each metaphorical representation has a position which is variable within a metaphorical space. The metaphorical representations can be based upon, for example, a physical metaphor, a visual metaphor, an auditory metaphor, or a textual metaphor. The auditory space sensed at each site is combined to form at least one synthetic auditory space. The at least one synthetic auditory space is formed in dependence upon the position of each metaphorical representation within the metaphorical space. A binaurally perceivable auditory environment is produced at one or more sites based upon the at least one synthetic auditory space.

Abstract: A multipoint video conference system includes a plurality of conference terminals, placed at remote points in correspondence with participants in a conference, for transmitting/receiving speech and image signals from/to the participants, and a video conference controller for transmitting/receiving speech and image signals to/from the conference terminals. The video conference controller includes an audio mixer, an identifying section, a frame selecting section, and multiplexers. The audio mixer forms a synthetic speech signal by synthesizing speech signals from the conference terminals. The identifying section detects speech signals from the conference terminals and identifies the conference terminal through which the participant has made utterance. The frame selecting section forms a multi-image signal by selecting image signals equal in number to multiple frames from the image signals from the conference terminals on the basis of the identification result obtained by the identifying section.

Abstract: An audio codec capable of handling complex control and routing of numerous sound inputs is described. The complex control and routing is obtained by weighting various sound inputs in accordance with weighting values and then digitally mixing the weighted sound inputs together. The invention facilitates construction of the audio codec with mainly fixed gain amplifiers, instead of variable gain preamplifiers, thereby saving a large amount of die space and reducing time needed for testing.

Abstract: In an audio communication control unit, audio signals from at least three terminals connected to a switching part are branched to left- and right-channels audio signals, then the left- and right-channel audio signals are processed using sound image control parameters in such a manner as to impart them with target positions different for each terminal, then all left-channel audio signals corresponding to the respective terminals are mixed together into a left-channel mixed audio signal and all the right-channel audio signals are mixed together into a right-channel mixed audio signal, and these left- and right-channel mixed audio signals are distributed and sent to all the connected terminals.

Abstract: According to the principles of the invention, there is provided a conferencing system that dynamically assigns calls to DSP resources. The system may attempt to process each audio conference on a single DSP node, so that information about conference participants does not need to be shared across DSP nodes. Further, the mapping of call channels to resources within a DSP node may be automated so that it is transparent to a conferencing system control application. Where more than one DSP node is required for a particular conference, there is further provided a system for linking DSP nodes. There are also provided methods for managing audio conferencing resources.