Abstract:
A system, method, and computer readable medium for remotely migrating from a first service to a second service, comprising sensing a ring wire state, sensing a tip wire state and switching a switch device to a direct current termination based upon a difference between the ring wire state and the tip wire state.
Abstract:
A method, system and computer program product are provided that reduce voice and data over IP (VoIP) packet overhead in an Internet telephony system, and regenerate missing or damaged data in a data packet. A media framer aggregates packets from multiple concurrent calls from several channels into a larger data packet. A transmission control module defines the format for each data packet, and updates and synchronizes header information in the data packets. A single virtual connection transmits data packets and other signals between originating and destination gateways located in the service areas for a caller and called party. System redundancy improves the quality of service by regenerating missing or damaged data in the data packets.
Abstract:
Instead of having to subscribe to multiple telephone lines for multiple devices that a user has, a module of the present invention can connect each of the user's devices to an outside communications network using the same telephone line. Such multiple inside connections to the outside communication network using the same telephone line is achieved by provisioning within the invention module the appropriate telephone and computer interface units for the user's telephones and computers, and an appropriate network interface unit for connection to the telephone line that connects the invention module to the external communications network. The module of the instant invention is further provisioned with an IP routing submodule that communicatively connects the various interface units together by managing the addressing of the data that traverses between the outside network and the devices of the user, by routing the appropriate data packets to the appropriate devices by means of pseudo internal IP addresses assigned to the various devices of the user. Other components within the module convert those data packets that are a part of a voice signal into the appropriate analog signal for output to the telephone of the user. Conversely, such components convert any analog input from the user into a corresponding digital signal that is packetized and output to the external communications network.
Abstract:
In order to make an IP telephone service available simply by connecting a PBX, for example, connected to a PSTN interface to IP telephone network without changing the numbering plan of the PBX if a call arrives at a line IF portion 101, an SIP processing portion 104 refers to a number storage portion 105, identifies the VoIP number corresponding to the PSTN number given to the call as the calling party number, converts the calling party number of the call to the identified VoIP number, and transmits the call through an IP network IF portion 102. Furthermore, if a call arrives at the IP network IF portion 102, the SIP processing portion 104 refers to the number storage portion 105, identifies the PSTN number corresponding to the VoIP number, which is the called party number of the call, converts the called party number of the call to the identified PSTN number, and transmits the call through the line IF portion 101.
Abstract:
To allow a call received from an accommodated device to arrive at the other party by diverting the call from an IP network to a detour network even with different numbering systems between communications over the IP network and over the detour network. A VoIP gateway apparatus 1 includes an ISDN terminal side IF portion 101, an IP network side IF portion 102, an ISDN network side IF portion 103, a condition storage portion 109, a detour determining portion 108, and a number editing portion 106. The detour determining portion 108 determines whether a call arriving at the ISDN terminal side IF portion 101 satisfies a detour condition stored in the condition storage portion 109 or not. If so, the number editing portion 106 edits the called party number of the call under a number editing condition stored in the condition storage portion 109 and then transmits it from the ISDN network side IF portion 103. If not, the call is transmitted from the IP network side IF portion 102 without editing the called party number.
Abstract:
An arrangement which includes a telephone and an interface unit, which interfaces the telephone to both a standard switched telephone communications network and an Internet communications network, is disclosed. The interface unit includes an input coupled to the telephone to receive audio information and two output ports configured to be respectively coupled to the standard switched telephone communications network and the Internet communications network. A processing unit couples the audio information received from the telephone to the first output port when the telephonic communication is to be performed using the standard switched telephone communications network. Alternatively, the processing unit processes the audio information received from the telephone in accordance with standard Internet transfer protocols and couples the processed audio information to the second output port when the telephonic communication is to be performed using the Internet communications network and the standard protocols.
Abstract:
A VOIP gateway that includes a web server. The gateway is configured to accept configuration data over the Internet by way of the web server. In one embodiment, the VOIP gateway includes a telephone line connection interface, able to connect to a telephone line that may carry a plurality of channels. The gateway also includes an Internet connection interface and a channel assigner, user configurable to assign a subset of the channels from the telephone line connection interface to telephone calls incoming from the Internet connection interface that are directed to telephone numbers having a shared characteristic.
Abstract:
A subjective quality monitoring system for packet based multimedia signal transmission systems which determines, during more than one interval of a single call, the level of one or more impairments and determines the effect of said one or more impairments on the estimated subjective quality of said multimedia signal. The quality monitoring system comprises a plurality of quality monitoring functions located at the multimedia to packet conversion points.
Abstract:
A method for providing self-provisioning of VoIP telephony to a subscriber of a VoIP telephony service is disclosed. An un-provisioned residential gateway that is associated with the subscriber is instructed to collect a subscriber numeric identity that uniquely identifies the subscriber and a Personal Identification Number (PIN) information that are associated with the subscriber. An example of a subscriber numeric identity that uniquely identifies the subscriber is the subscriber's E.164 address. The E.164 address and Personal Identification Number (PIN) information is verified. A source IP address that is associated with one or more Media Gateway Control Protocol (MGCP) messages that are sent by the residential gateway is used as a residential gateway IP address for the residential gateway. The residential gateway IP address is then used to provision the residential gateway that is associated with the subscriber. According to certain embodiments, the above steps are carried out by a call agent from a self-provisioning system of a provider of the VoIP telephony services. One or more VoIP connections are established between the residential gateway and an announcement server. The announcement server sends VoIP messages through the VoIP connections to the subscriber via the residential gateway. A protocol server offers the residential gateway, via a Dynamic Host Configuration Protocol (DHCP) server, a limited access IP address, a location of one or more Domain Name System (DNS) servers, and a Media Gateway Protocol (MGCP) endpoint name of the call agent from the self-provisioning system.
Abstract:
A method, apparatus, and communication network system that allows an endpoint to be simultaneously registered with more than one communications server is described. In one embodiment, the communication network system includes a network, a plurality of communications servers that are coupled to the network, and a plurality of endpoints coupled to the network. Each endpoint is capable of being simultaneously registered with more than one communications server. A communication method for an endpoint involves registering a first logical line of the endpoint with a first communications server, and registering a second logical line of the endpoint with a second communications server. Consequently, flexibility is obtained by allowing an endpoint to choose the registering communications server for each logical line of the endpoint.