摘要:
Certain exemplary embodiments provide a method, comprising: receiving, at a subscriber interface line card, an analog signal from a POTS subscriber loop circuit; quantizing the analog signal into a plurality of digital samples; encoding the plurality of digital samples via codec instructions running on a digital signal processor installed on the subscriber interface line card; and converting, via conversion instructions running on the digital signal processor, the encoded plurality of digital samples to a plurality of VoATM packets.
摘要:
Certain exemplary embodiments provide a method, comprising: receiving, at a subscriber interface line card, an analog signal from a POTS subscriber loop circuit; quantizing the analog signal into a plurality of digital samples; encoding, via high-quality audio codec instructions running on a digital signal processor installed on the subscriber interface line card, the plurality of digital samples; and converting, via conversion instructions running on the digital signal processor, the encoded plurality of digital samples into a plurality of VoATM packets.
摘要:
A method and apparatus for providing a voicemail notification to a subscriber or customer in a communication network are disclosed. For example, the present method uses a wireless integrated access device (WIAD) to interact with a voicemail server via the Internet. The WIAD determines whether a newly saved voicemail exists for a customer and then interacts with at least one endpoint device that is located at a customer premise to provide a voicemail notification to the customer if a newly saved voicemail exists for the customer.
摘要:
A subscriber link to a central office which employs data compression, forward error correction, and advanced modulation techniques and to connect subscribers to multiple communications networks to provide an array of services. A device provides normal telephone service in the event of an equipment failure. At the subscriber end, a server called an intelligent services director (ISD) provides multiple independent connections for telephones which ordinarily connect to multiple access virtual circuits generated on the subscriber link over a twisted pair. A device called a facilities management platform (FMP) at the central office end of the link, among other things, provides interfacing of the subscriber link to various networks including a digital subscriber loop (DLC) and packet switched networks. Ordinarily telephones connected to the ISD require power and correctly functioning modems and controllers in the ISD and the FMP to have access to the outside world. A fail-safe mechanism, however allows at least one chosen phone to function in the event of a failure. The chosen phone must be capable of pulse or DTMF dialing. The connection through which it operate can be switched directly to the twisted pair media connecting to the FMP. At the FMP, the twisted pair is switched to tie the connection directly to a line card of the DLC. Alternatively, the ISD contains an interface to a specialize phone designed for the ISD environment. The interface provides the appearance to the DLC of a regular POT.
摘要:
A power amplifier includes a first amplifier, a second amplifier, a transfer switch and a controller. The first amplifier amplifies a first frequency bandwidth, and the second amplifies a second frequency bandwidth. The second frequency bandwidth is less than the first frequency bandwidth. The transfer switch includes a first input port, a second input port, a first output port and a second output port. The first input port is connected to a primary power source, while the second input port is connected to a battery. The first output port is connected to the first amplifier and the second output port is connected to the second amplifier. The controller controls the transfer switch to connect the first primary power source to the first amplifier in a primary mode of operation. The controller also controls the transfer switch to connect the battery to the second amplifier in a second mode of operation.
摘要:
Providing high quality voice/sound communications over a local loop of a telephone network is disclosed. The method includes receiving a voice signal, digitizing the voice signal into a high quality voice signal, utilizing sampling rates and/or sizes above the threshold, negotiating voice processing characteristics between a customer premises equipment and a network element, receiving speech at a customer premises equipment according to the negotiation, converting the received speech into high bandwidth signal and transmitting the high bandwidth signal to a telephone local loop, transmitting the high bandwidth signal from the local loop to wideband node that packetizes the high bandwidth signal for transmission to a packet network and receiving the packetized signal from the packet network at a switch that switches between an on-network or off-network status. A voice over IP platform can route packetized signals from the packet network to the telephone network or another packet network.
摘要:
Providing high quality voice/sound communications over a local loop of a telephone network is disclosed. The method includes receiving a voice signal, digitizing the voice signal into a high quality voice signal, utilizing sampling rates and/or sizes above the threshold, negotiating voice processing characteristics between a customer premises equipment and a network element, receiving speech at a customer premises equipment according to the negotiation, converting the received speech into high bandwidth signal and transmitting the high bandwidth signal to a telephone local loop, transmitting the high bandwidth signal froth the local loop to wideband node that packetizes the high bandwidth signal for transmission to a packet network and receiving the packetized signal from the packet network at a switch that switches between an on-network or off-network status. A voice over IP platform can route packetized signals from the packet network to the telephone network or another packet network.
摘要:
Disclosed is a system and method for routing a call to a dual mode wireless device. In accordance with an embodiment of the invention, a network node receives a call. The network node determines that the call is associated with a dual mode wireless device. Once this determination is made, the network node then selects one of a plurality of networks (e.g., cellular network or packet-based network, such as a VoIP network) for use in connecting the call to the dual mode wireless device. The network node then routes the call to the dual mode wireless device via the selected network. During the call, the network node re-routes the call to another network if the network node determines that this other network is now better suited for the call.
摘要:
A system and method are disclosed for providing high quality sound communications in an IP Centrex environment. The method aspect of the invention comprises, from a network switch, negotiating between a first customer premises equipment (CPE) and a second CPE, the negotiation being related to a possible quality of a call between the first CPE and the second CPE. Next, the switch controls mapping between a dialing plan and a network address, determining a network address of the first CPE and the second CPE and connecting the call between the first CPE and the second CPE. In this manner, the highest quality and broadest bandwidth possible between the first CPE and second CPE through the IP Centrex environment may be used for the call.
摘要:
Certain exemplary embodiments comprise a method comprising a plurality of activities, comprising: via a CPE telephony device simultaneously connectable to a POTS connection and to a connection to a data network: facilitating, via the POTS connection, a PSTN call, and facilitating, via the data network connection, a display of data to a call participant of the PSTN call, the data provided the data network.