Abstract: The present invention is directed towards a method and a portable communication device (200) for forming a real time output media stream by mixing a first real time media stream with a second media stream. Provision of mixing of media streams provides a number of attractive functions, for instance. This invention concerns how to increase the usage of a mobile VT application and how to be able to handle integrity issues when used in a portable communication device.

Abstract: A network switch. The network switch includes a processor configured to execute a set of instructions to access a recorded announcement with attendant text and to transmit the recorded announcement with the attendant text to a telecommunications device.

Abstract: A system and methods for providing services through an integrated digital network. The present invention embodies a system and methods for providing telecommunications services through an integrated digital network that provides the superset of all previous network types, and can continue to support being the core for existing networks—now used simply as edge access methods. The method includes connecting a multi-service core to existing telecommunications networks, wherein the core can communicate with conventional network components; interconnecting a gateway between the multi-service core and multiple existing network components, wherein the gateway is configured to share multiple service aspects with each network component and to utilize multiple transport protocols to communicate with different core network components; coordinating communications between the gateway and existing network components; and resolving differences between protocols or communications between existing core network components.

Abstract: A plurality of base station devices are linked together and synchronized to facilitate communication between the respective base station devices and respective downlink subscriber stations. According to one embodiment of the invention, one of the base station devices is designated as a master device, and the other base station devices are designated as slave devices. The respective base station devices are connected to each other by means of a synchronization bus. The master base station device then generates and broadcasts a future time stamp value, which is received by the respective slave base station devices. When the time stamp counter in the master base station device reaches the transmitted value, a control signal is broadcast over the synchronization bus. The slave base station devices then retrieve the time stamp value and reset their respective local time stamp counters to the received value. In this manner, the base station devices are synchronized.

Abstract: Apparatus is used in communicating in a data and voice communication system. The apparatus includes a line card and a crosspoint switch in communication with the line card. The apparatus also includes an accelerator card in communication with the line card and the crosspoint switch. The crosspoint switch is configured to provide a redundant communications path between the line card and the accelerator card.

Abstract: A method for selecting a compression setting to use during a communications session to compress streaming video data includes the steps of determining processor performance data for at least one processor, determining a video data acquisition rate, and using the processor performance data and the acquisition rate to identify a compression setting for the data communications session.

Abstract: Aspects of a method and system for utilization of an reserved and/or out of band channel for maintaining a network connection are provided. In this regard, information relating to training of one or more link partners communicatively coupled to an Ethernet link may be exchanged via a reserved and/or out of band channel on the Ethernet link. The reserved and/or out of band channel may be an auxiliary channel as specified by the IEEE802.1AN standard. The reserved and/or out of band channel may be utilized for scheduling training of one or more of the link partners, determining which of the link partners require training, synchronizing training of the link partners, training the link partners based on changes in environmental conditions. The training may comprise configuring an echo canceller a far-end crosstalk canceller, and/or a near-end crosstalk canceller.

Abstract: A communication device carries out a routing operation of data between a first network and a second network. The communication device comprises: a first interface connected to the first network; a second interface connected to the second network; and a processor that assigns an IP address to a user terminal device connected to the first interface in accordance with a DHCP. The processor changes a first lease period of the IP address set when the processor assigns the IP address to the user terminal device in accordance with the change of a network environment.

Abstract: In a particular embodiment, the disclosure is directed to a method of making a digital subscriber line (DSL) connection to a remote network. The method includes detecting the presence of a powered-on network capable device that is connected to a DSL modem; establishing a network connection over a DSL line to the remote network after detecting the presence of the powered on network capable device; terminating the network connection over the DSL line after detecting an absence of network capable devices connected to the DSL modem; and releasing network resources supported by the remote network after the network connection is terminated.

Abstract: Disclosed are a system and method for providing digital multimedia broadcasting confinement service. The system includes: a relay center side system, which is adopted to the relay center, according to the relay center side packet identification description (PID) key, for modifying the PID of transport stream and generating the distorted program specific information (PSI); and a terminal side system, which is adopted to the DMB terminal, according to the terminal side PID key, for recovering the distorted PSI of the transport stream received from the relay center and decoding it. The method includes the steps of: identifying whether the actual PID of the TS is consistent when the TS is demultiplexed by audio/video PID acquired through the program map table (PMT) analysis of the TS; generating a preset alarm signal; modifying the PID of the audio/video data in the PMT according to the terminal side PID; and demultiplexing the TS according to the modified PID value.

Abstract: A terminal for use in a peer-to-peer voice communications system comprising: a base station; and at least one handset arranged to communicate wirelessly with said base station; wherein the base station comprises a first interface to a packet data network and a second interface to a public switched telephone network, said base station being arranged to selectively establish a packet data call from said first interface and a public switched telephone call from said second interface.

Abstract: A method and system for secure Voice over Internet Protocol (IP) (VoIP) communications. The method and system provide secure VoIP voice calls, video, Instant Messaging (IM), Short Message Services (SMS), or Peer-to-Peer (P2P) communications while maintaining privacy over the Internet and other communications networks such as the pubic switched telephone network (PSTN) to and from any network device through a virtual private network infrastructure interconnecting private VoIP network devices. The method and system allow a network device to function as an IP private branch exchange (PBX) or a private VoIP gateway and provide and control VoIP voice communications without using other public or private VoIP gateways or VoIP servers or devices on a communications network such as the PSTN or the Internet.

Abstract: A method for transmitting real time data packets in a cyclic communication system, each transmission cycle having a first partial cycle (IRT-Z) for a real time communication portion and a second partial cycle (NRT-Z) for a non-real-time communication portion. The method includes determination (118) of a cycle number and processing of a transmission sequence of real time data packets in the first partial cycle. The transmission sequence is composed of one or more partial sequences (1, 2, 3, 4, 5), and the composition depends on the cycle number.

Abstract: Methods and apparatus are provided for sending data communications over wireless digital voice communications networks which transmit voice communications in voice frames, each of which contains a digitized segment of a voice communication in a voice frame format. The method including the steps of: encoding the data communication into a plurality of data frames, each of the data frames having the same format as the voice frame format; transmitting the data frames over the wireless digital voice communications network; and decoding the data frames to reconstruct the data communication. The apparatus includes: a processor for encoding the data communication into a plurality of data frames, each of the data frames having the same format as the voice frame format; and a transmitter for transmitting the data frames over the wireless digital voice communications network.

Abstract: A method and apparatus for enabling a local area network wiring structure to simultaneously carry digital data and analog telephone signals on the same transmission medium. It is particularly applicable to a network in star topology, in which remote data units (e.g. personal computers) are each connected to a hub through a cable comprising at least two pairs of conductors, providing a data communication path in each direction. Modules at each end of the cable provide a phantom path for telephony (voice band) signals between a telephone near the data set and a PBX, through both conductor pairs in a phantom circuit arrangement. All such communication paths function simultaneously and without mutual interference. The modules comprise simple and inexpensive passive circuit components.

Abstract: An information handling system comprises a TCP/IP network connected via a modem to an IP telephony device that is then coupled to a network device. Data sent from the network device is addressed for transmission through the IP telephony device to the TCP/IP network. The IP telephony device monitors when an amount of data being received over the network falls below a predetermined threshold. If this occurs, the IP telephony device will throttle data being received by the IP telephony device from the connected network device.

Abstract: A wireless communication system (20) includes a wireless network (30) that selectively modifies an aggregation of voice over internet protocol wireless transmissions. In one example, a mobile station geometry provides an indication whether modification of the aggregation is desirable. For example, a mobile station with a better geometry allows for a higher level of aggregation. Another disclosed example includes determining a current cell load condition and increasing the amount of aggregation and the level of aggregation as cell load conditions increase. Another disclosed technique includes staggering the scheduling of aggregated packets to avoid delays at a scheduler. One or more of the disclosed techniques may be used to efficiently transmit voice over internet protocol communications.

Abstract: A direct way to measure network delay of a packet switched network uses a ping signal injected into the voice stream at a source telephone. The ping signal travels together with the voice stream to a destination telephone. The destination telephone detects the ping signal and returns it to the source telephone through the packet switched network. The source telephone detects the return ping signal sent by the destination telephone and calculates the network delay between the two telephones based on the difference between the time of transmitting the ping signal and the time of detecting the return ping signal. As the ping signal travels in the same path as the voice stream, the network delay between the two telephones is calculated directly and accurately.

Abstract: An IP-based group communication feature in a mobile communication network is based on the pre-established logical connections established between a gateway node (GGSN) and members of a group in the mobile communication network. No separate call establishment procedures using VoIP signaling are used for connecting a user equipment (UE) logically to group(s), but the procedure is preferably based on the normal logical connection establishment signaling (e.g. PDP context creation) as used for data service. The UE implements a host supporting the IP multicasting, and the next IP layer peer for the UE host is located in the gateway node. The gateway node provides an IP interface towards an external multicast router so that the UE host looks like a normal IP host supporting IP multicasting. The IP-based voice traffic is mapped and transferred over pre-established connections between the gateway and the UEs of the specific group.

Abstract: There is provided a DSL modem for integrating an analog voice band with DSL bands over a telephone line, where the analog voice band consists of 0-4 kHz frequency range, and the DSL bands consist of frequencies beyond 4 kHz. The DSL modem comprises a splitter configured to receive a telephone line signal from the telephone line and split the telephone line signal into an analog voice band signal and a DSL bands signal; a voice module configured to receive the analog voice band signal, digitize the analog voice band signal to generate digitized voice band signal and packetize digitized voice band signal into packetized voice band signal; a DSL module configured to receive the DSL bands signal and retrieve packetized DSL bands signal from DSL bands signal; and a multiplexer configured to multiplex packetized voice band signal and packetized DSL bands signal to generate multiplexed voice-DSL packets.

Abstract: A method and system is disclosed for interworking between half-duplex and full-duplex conferencing. A half-duplex (HD) conference will be conducted through one conference server and will involve floor-control, a full-duplex (FD) conference will be conducted through another conference server, and the conference servers will exchange bearer traffic so as to cooperatively provide an HD/FD conference. When the HD conference server detects that any of the participating HD client stations holds the floor in the HD conference, the HD conference server directs the FD conference server to put the participating FD client stations in a listen-only mode with respect to the HD conference. On the other hand, when the HD conference server detects that none of the HD client stations holds the floor, the HD conference server directs the FD conference server to lift the listen-only restriction.

Abstract: Provided is a system and method for communicating audio. A method includes transmitting audio information segments on a first signal line, each segment including a format portion representative of audio format modes, and a data portion having audio data corresponding to one or more of the format modes. The method also includes transmitting a number of synchronization markers on a second signal line. Each marker is representative of a timing of one of the audio information segments.

Abstract: An integrated VoIP unified message processing system includes a voice platform that processes data in native VoIP format. There is no use of hardware telephone interface cards (TICs) or software transcoding to transform data to PCM or other formats. Cost reductions are achieved by the elimination of expensive dedicated hardware and scalability is achieved by obviating the need for software transcoding.

Abstract: The present invention provides a method for controlling transmission latency in a communications system, wherein the communications system is subject to a noise signal having at least a first noise phase and a second noise phase. The method includes determining a first bit rate for symbols transmitted during the first noise phase, and a second bit rate for symbols transmitted during the second noise phase, the first bit rate and the second bit rate being constrained such that a transmission latency does not exceed a pre-determined maximum allowed transmission latency; and transmitting symbols at the first bit rate during the first noise phase and at the second bit rate during the second noise phase. In other variants, the invention provides an apparatus, a constrained rate reciever, a transmitter and a signal.

Abstract: A method for communicating utility data includes receiving a voice signal and a data signal via a utility network, combining the voice signal and the data signal to form a communication signal, and transmitting the communication signal to a plurality of utility asset controllers and handheld devices via the utility network.

Abstract: Transmitting media information from a transmitting device to a receiving device. To perform the transmission a media stream is encoded and packets are formed from the encoded media stream, the packets containing application data units, the application data units having a decoding order. At least two blocks are generated from the packets. The packets of the at least two blocks are organized into a first group and a second group, the first group being decodable without reception of said second group. The transmission order of the packets of said first group is arranged succeeding packets of said second group such that the application data units in the transmission order are at least partly in a different order than the decoding order.

Abstract: A broadband wireless access system including: a base station that allocates an uplink bandwidth to each connected subscriber station according to at least one algorithm; and at least one subscriber station that when there is no voice information to be sent through the uplink bandwidth allocated from the base station according to an unsolicited grant service (UGS) algorithm, detects another packet based on another algorithm, encapsulates the detected packet, and sends the encapsulated packet in a payload field of a UGS packet to the base station through the uplink bandwidth. Thus, the broadband wireless access system maximizes utilization of the bandwidth allocated according to the UGS algorithm.

Abstract: An inband signaling modem communicates digital data over a voice channel of a wireless telecommunications network. An input receives digital data. An encoder converts the digital data into audio tones that synthesize frequency characteristics of human speech. The digital data is also encoded to prevent voice encoding circuitry in the telecommunications network from corrupting the synthesized audio tones representing the digital data. An output then outputs the synthesized audio tones to a voice channel of a digital wireless telecommunications network.

Abstract: In conjunction with a data communication network (53) carrying multiple telephony signals and allowing for connection of telephone sets (17), a system and method in which two external feeders (55a, 55b) connect to the data network (53) at two distinct points via two distinct devices. The data network can be based on dedicated wiring or can use existing in premises medium such as telephone, powerlines or CATV wiring. In the latter case, the wiring can still carry the original service for which it was installed. The external telephone connections can be based on the traditional PSTN, CATV network, cellular telephone network or any other telephone service provider network, using specific adapter for any medium used. In the case of connection to a POTS telephone signal, VoIP gateway (or any other converter) is required.

Abstract: A system and method for delivering digital subscriber line (DSL) service to a subscriber. In a telecommunications network, the signal passing through a protector field is diverted to a cross connect block that can be selectively configured to establish a communication route between the cross connect block and a splitter selectively receiving a DSL signal. The combined signal is then placed back upon the network by the cross connect block, which directs the combined signal back to the protector field.

Abstract: A telecommunications system that supports the provision of video messaging, yet maintains compatibility and operation with user equipment that is not capable of processing, receiving or rendering video content. Based on the type of user equipment, as well as user selected options, network configurations and status, and class-of-service characteristics, the telecommunications systems can revise the manner in which content is provided to the user equipment. Variations in the content provision include providing synchronized audio and video content, audio content only, video content only, video content with closed-captioning and closed-captioning only.

Abstract: An embodiment of a system, method, and apparatus for facilitating wideband audio communication and digital subscriber line data communication includes a filter configured to be coupled to a line conditioning coil. The line conditioning coil includes a frequency response having a first passband extending over a portion of a narrowband voice frequency range, a second passband extending over a portion of a digital subscriber line frequency range. The line conditioning coil further includes a frequency notch having a frequency range between the first passband and the second passband. The frequency notch attenuates signals over a portion of a wideband audio frequency range. The filter has a cut-off frequency below the frequency notch and a passband extending over at least a portion of the frequency notch. The filter is configured to amplify signals within the frequency range of the frequency notch to compensate for the attenuation caused by the frequency notch.

Abstract: A data session with a wireless device ordinarily cannot be initiated from the network side because the wireless device has no predetermined IP or other network address. Methods are disclosed for instead causing the wireless unit to initiate a data session with a target destination, thereby obviating the IP addressing problem. A voice call is first initiated by a requester application to a remote wireless unit. The requester transmits a request message to the remote wireless unit via the wireless voice channel using in-band techniques; preferably including a payload in the polling message that identifies the target destination. In response, the remote wireless unit initiates a packet data session with the target destination.

Abstract: A telecommunications system that supports the provision of video messaging to certain user devices, based on the calling line identifier or ANI associated with the user devices. Based on the CLID or the ANI, as well as user selected options, network configurations and status, and class-of-service characteristics, the telecommunications systems can revise the manner in which content is provided to the user equipment. Variations in the content provision include providing synchronized audio and video content, audio content only, video content only, video content with closed-captioning and closed-captioning only.

Abstract: Provided is an access mediator system for mediating convergence services of communication and broadcasting using a non-communicative appliance. The access mediator system includes: a non-communicative appliance access interface unit performing an interface function so as to converge a non-communicative appliance comprising an interface supporting image and voice multimedia data but not including a communication function; a broadband convergence network access interface unit performing an interface function so as to access a broadband convergence network connected to various service provider servers via wire and wireless interfaces; and a main processor accessing a selected service provider server via the broadband convergence network access interface unit to mediate the selected service provider server and the non-communicative appliance.

Abstract: A local area network (60) within a residence or other building, including both wired (5) and non-wired segments (53). The wired segments are based on new or existing wires (5a, 5b, 5c, 5d, 5e) in the building, wherein access to the wires is provided by means of outlets (61a, 61d), such as a telephone system, electrical power distribution system, or cable television wiring system. The non-wired segments are based on communication using propagated waves such as radio, sound, or light (e.g. infrared). The wired and non-wired segments interface in the outlet, using a module (50) that serves as mediator between the segments. The module can be integrated into the outlet, partially housed in the outlet, or attached externally to the outlet. Such a network allows for integrated communication of data units (24b) connected by wires and data units (24a, 24d) connected without wires.

Abstract: A transmitting terminal creates a data PCM stream in which data is recognized as a PCM stream by placing a prefixed idle code string before data to transmit and a postfixed idle code string after the data to transmit. By switching between a voice PCM stream and a data PCM stream, PCM steam is transmitted to a receiving terminal. The receiving terminal, when detecting the prefixed idle code string from a received PCM stream, receive the data from the transmitting terminal until detecting the postfixed idle code string.

Abstract: Systems and methods for providing aural review of a privacy policy are disclosed. Generally, a first version of a privacy policy is retrieved. A natural language version of the privacy policy is then retrieved based on at least one user preference and an audio representation of the natural language version of the privacy policy is played through an audio system of a device to a user.

Abstract: In one aspect of the present invention, a network gateway is configured to facilitate on line and off line bi-directional communication between a number of near end data and telephony devices with far end data termination devices via a hybrid fiber coaxial network and a cable modem termination system. The described network gateway combines a QAM receiver, a transmitter, a DOCSIS MAC, a CPU, a voice and audio processor, an Ethernet MAC, and a USB controller to provide high performance and robust operation.

Abstract: Analog HPNA hub including at least one group of coils, the coils inducing HPNA signals there between, a plurality of filters, each of the filters coupled with a respective one of the coils and further coupled, via respective telephone wiring, with at least a respective HPNA node, wherein each of the filters enables transmission of HPNA data signals there through, and wherein each of the filters prevents transmission of conventional telephony signals there through.

Abstract: A portable tester for testing mutual effects of multiple services received via a shared network access link includes service signaling means for simulating user premises devices. In one embodiment, the tester is a triple play services tester supporting three IP addresses at the same time.

Abstract: A system and method for delivering digital subscriber line (DSL) service to a subscriber from a remote terminal of a telephone network. Included is a first route for delivering a telephone signal to the subscriber, and a second route for passing the telephone signal through a DSL system before delivering the signal to the subscriber. A disrupter is then used to selectively activate either the first or second route.

Abstract: The present invention provides a system and method for prioritizing delay-sensitive packets relative to each other for transmission from a router in a network, based on a delay variation estimated for each delay-sensitive packet by a quality of service monitor. In accordance with the present invention, late delay-sensitive packets are assigned a higher priority than, and are transmitted before, other delay-sensitive packets, thereby reducing the level of jitter and improving the quality of the packet streams that comprise late delay-sensitive packets.

Abstract: A management server includes a determination unit and a notification unit. The determination unit determines a communication network to which a mobile terminal connectable to a plurality of the communication networks by one telephone number is connecting. The notification unit issues a notice on the communication network connecting to the mobile terminal based on a result of the determination by the determination unit.

Abstract: The present invention provides a method for controlling transmission latency in a communications system, wherein the communications system is subject to a noise signal having at least a first noise phase and a second noise phase. The method includes determining a first bit rate for symbols transmitted during the first noise phase, and a second bit rate for symbols transmitted during the second noise phase, the first bit rate and the second bit rate being constrained such that a transmission latency does not exceed a pre-determined maximum allowed transmission latency; and transmitting symbols at the first bit rate during the first noise phase and at the second bit rate during the second noise phase. In other variants, the invention provides an apparatus, a constrained rate receiver, a transmitter and a signal.

Abstract: In conjunction with a data communication network carrying multiple telephony signals and allowing for connection of telephone sets, a system and method in which two external feeders connect to the data network at two distinct points via two distinct devices. The data network can be based on dedicated wiring or can use existing in-premises medium such as telephone, powerlines or CATV wiring. In the latter case, the wiring can still carry the original service for which it was installed. The external telephone connections can be based on the traditional PSTN, CATV network, cellular telephone network or any other telephone service provider network, using specific adapter for any medium used. In the case of connection to a POTS telephone signal, VoIP gateway (or any other converter) is required.

Abstract: In order to satisfactorily transmit audio data of various formats through a transmission line, when data having a predetermined data length as a unit is transmitted by a predetermined format between devices connected to a predetermined transmission line, label data indicating a system of transmitted audio data is located at a starting portion of the data having the predetermined data length, sub-label data is located at an interval behind label data if necessary and data accompanying with audio data is located and transmitted, whereby a side which receives this data can easily identify a system of audio data or the like.

Abstract: A method for dynamically handling the data path of a wireless IP phone is provided. The method comprises establishing a first communication link between the wireless IP phone and a wireless local area network (WLAN), establishing a second communication link between the wireless IP phone and a general packet radio service (GPRS) network, establishing a conversation session between the wireless IP phone and a second IP phone via a voice over internet protocol (VoIP) through the first or second communication link, determining whether the data path of the conversation session passes through the first or the second communication link according to a signal strength of the first communication link, and exchanging a plurality of packets of the conversation session through the data path.

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: A system in a Voice Over Internet Protocol (VOIP) network eliminates false voice detection in voice band data service (e.g., modem or facsimile transmission) by sequentially enabling silence detection then voice detection. In Voice Band Data (VBD) mode, the system initially enables silence detection and disables voice detection. If silence is detected in a VOIP signal associated with a VOIP call, the system enables voice detection. If voice is detected, the system switches from VBD mode to voice mode and enables processing (e.g., echo cancellation and/or data compression) that improves voice communications.