39 results about "Packet telephony" patented technology

A packet telephone system which employs a packet network that provides virtual circuits. The packet telephone system employs short packets containing compressed speech. The use of the short packets makes possible compression and decompression times and bounded delays in the virtual circuits which are together short enough to permit toll-quality telephone service. The packet telephone system employs an intelligent network interface unit to interface between the packet network and standard telephone devices. The network interface unit does the speech compression and decompression and also responds to control packets from the packet network. Consequently, many telephone system features can be implemented in the network interface unit instead of in the switches. The network interface unit may also be used to provide data connections to devices attached to it. The combination of virtual circuits, with bounded delays, short packets, rapid compression and decompression, and intelligent network interface units makes it possible to build a telephone system with fewer and cheaper switches and fewer links for a given volume of traffic than heretofore possible and also permits substantial savings in provisioning and maintaining the system.

An outlet for a Local Area Network (LAN), containing an integrated adapter that converts VoIP to and from analog telephony, and a standard telephone jack (e.g. RJ-11 in North America) for connecting an ordinary analog (POTS) telephone set. Such an outlet allows using analog telephone sets in a VoIP environment, eliminating the need for an IP telephone set or external adapter. The outlet may also include a hub that allows connecting both an analog telephone set via an adapter, as well as retaining the data network connection, which may be accessed by a network jack. The invention may also be applied to a telephone line-based data networking system. In such an environment, the data networking circuitry as well as the VoIP / POTS adapters are integrated into a telephone outlet, providing for regular analog service, VoIP telephony service using an analog telephone set, and data networking as well. In such a configuration, the outlet requires two standard telephone jacks and a data-networking jack. Outlets according to the invention can be used to retrofit existing LAN and in-building telephone wiring, as well as original equipment in new installation.

A program and a method for a gateway are provided to deny incoming calls to prevent overload. The program provides a maximum CPU utilization threshold CPUT, which is set by the user. When a new incoming call is presented to the packet telephony gateway, the program checks a present CPU utilization CPUP. If the present CPU utilization CPUP is greater than the threshold, the call is refused. This insures that sound quality of the calls currently being handled is maintained, and that existing calls are never dropped.

A system and method for communicating via a packet telephony distribution system is defined that includes a dual-use / analog phone or phone adapter. Any communication device incorporating the adapter can dynamically receive incoming calls and transmit outgoing calls over analog phone lines and over derived phone lines such as VoIP (Voice Over IP). When receiving or transmitting calls, the adapter and an associated gateway perform operations that enable simultaneous communication over the analog phone lines or derived phone lines from any combination of communication devices. In cooperation with the communication devices, the gateway further dynamically controls the incoming and outgoing calls over the analog phone lines and derived phone lines.

We describe a universal Ethernet telecommunications service (UETS) that combines the features of the telephone network, local area networks and the Internet in order to offer integrated broadband services by re-using the infrastructure of the telephone and electrical networks. User-network communication extends the Ethernet local network services to the infrastructure of the operator in the “Ethernet domain”, which is distinguished from the Internet “IP domain”. The access device offers packet telephone service, with power supply over the telephone pairs, which guarantees the emergency call through the exchange being provided with a battery and power control for energy saving. Terminals are also defined with a simple supervisor to support the applications used in the Internet, which communicate via the Ethernet domain with the LLC / MAC or TCP / IP protocols and via the IP domain with the TCP / IP protocols.

A method and apparatus is provided for allocating bandwidth and reducing jitter in an upstream transmission in a shared-media packet-switched access network with multiple upstream channels offering integrated Internet Protocol voice and high-speed data services. To allow multiple simultaneous CBR (constant-bit-rate) voice connections to be supported by a cable modem, when only a single channel can be accessed at a time, a method to select an upstream channel and assign to each connection a time slot in the upstream channel is provided. Jitter, i.e. packet delay variation, results when packets associated with a voice connection are not received at expected intervals. To maintain jitter within predetermined tolerances, the sequence of mini-slots in an upstream channel is divided into frames. Each frame will have a sequence of voice and data-only regions, and the voice region is divided into non-overlapping jitter windows of generally equal length. Even when an upstream channel change is required, each active voice connection is maintained in one of the jitter windows, thereby limiting jitter to the duration of a jitter window.

An adapter is disclosed which allows attachment of a standard telephone to a computer as a standard peripheral. The adapter converts analog signals of the telephone to digital signals of the computer and vice versa. The adapter can also be a stand-alone device allowing the telephone to be attached to a computer network. The adapter, in this case, contains the computing power to support the necessary data network protocols, and PSTN signalling protocols.In this way voice communication is enabled over a computer network without requiring the computers to be equipped with sound cards.

A method for telephony includes receiving at a facility of a telephone service provider a first subscriber request to place a first call to a specified telephone number. A provider request is sent to a remote server (26), which maintains a database (28) of records associating telephone numbers with respective packet network addresses, for a record (72) indicating a packet network address to which calls to the specified telephone number should be placed. Upon receiving the record from the remote server, the record is stored in a local cache (42) at the facility of the telephone service provider. After storing the record, upon receiving at the facility of the telephone service provider a second subscriber request to place a second call to the specified telephone number, the record is read from the local cache and is used in placing the second call to the packet network address.

A method for packet telephony includes receiving over a packet communication network (22) a first message originating from a first end-user (32A) in a first aggregating organization (28) to place a call to a second end-user (32D) in a second aggregating organization (30). Responsively to the first message, respective values are assigned to one or more security attributes that are indicative of a likelihood that the call is a spam call. A second message is transmitted over the packet communication network toward the second end-user, the second message containing at least one of the respective values that was assigned to at least one of the security attributes.

An outlet for a Local Area Network (LAN), containing an integrated adapter that converts VoIP to and from analog telephony, and a standard telephone jack (e.g. RJ-11 in North America) for connecting an ordinary analog (POTS) telephone set. Such an outlet allows using analog telephone sets in a VoIP environment, eliminating the need for an IP telephone set or external adapter. The outlet may also include a hub that allows connecting both an analog telephone set via an adapter, as well as retaining the data network connection, which may be accessed by a network jack. The invention may also be applied to a telephone line-based data networking system. In such an environment, the data networking circuitry as well as the VoIP / POTS adapters are integrated into a telephone outlet, providing for regular analog service, VoIP telephony service using an analog telephone set, and data networking as well. In such a configuration, the outlet requires two standard telephone jacks and a data-networking jack. Outlets according to the invention can be used to retrofit existing LAN and in-building telephone wiring, as well as original equipment in new installation.

A network monitoring agent is disclosed that monitors network conditions, such as traffic volume, and determines when to dynamically adjust the encoding scheme for one or more connections. The network monitoring agent can select an encoding standard based on, for example, current network traffic volume, network error characteristics, time of day or day of week. In the illustrative network traffic implementation, an encoding standard that provides a lower degree of compression and a higher quality level is selected at times of lighter network traffic. Likewise, as network traffic increases, an encoding standard that provides a higher degree of compression, although at a lower quality level, is selected in order to maximize the network utilization. The network monitoring agent notifies one or both of the devices associated with each connection of changes in the encoding scheme. Generally, both devices must change the compression algorithm at the same time, to ensure proper decoding of received packets. The initiating device inserts a notification in a field of a predefined number of packet headers to inform the recipient device that subsequent packets will be encoded with a different specified encoding algorithm, until further notice. Thereafter, the recipient device can load the appropriate codec to properly decompress and decode the received packets.

Speech at the beginning of a talkspurt in a discontinuous transmission (DTX) packet telephony system is speeded up to help make up for an access delay incurred during channel allocation. Incoming speech frames are buffered, a pitch period for a current portion of the signal is estimated, and then a pitch period=s worth of the signal is cut from that portion. This is continued until the original access delay, as estimated from the time lag between the commencement of voice input for the talkspurt, and notification that a channel is available, is eliminated. The remainder of the talkspurt is then transmitted without such compression.

A queue server is established in a packet telephony environment, wherein the queue server is defined as a callable entity, separate from the H.323 workstation for which it is being used. The queue may be used with any piece of H.323 software without modification to the software. All that need be done is the configuration that is typically accomplished when H.323 software is installed on the workstation. For example, the queue may be identified as a callable entity to receive call forwarding from the busy device or endpoint. The H.323 software of the endpoint needs to be able to handle only one call at a time, thereby reducing memory requirements for the equipment.

A method for packet telephony includes receiving over a packet communication network (22) a first message originating from a first end-user (32A) in a first aggregating organization (28) to place a call to a second end-user (32D) in a second aggregating organization (30). Responsively to the first message, respective values are assigned to one or more security attributes that are indicative of a likelihood that the call is a spam call. A second message is transmitted over the packet communication network toward the second end-user, the second message containing at least one of the respective values that was assigned to at least one of the security attributes.

A dual mode packet phone is provided, comprising a first connector to connect the phone with a data network, and a second connector to connect the phone with a backup network. The phone automatically switches between the data and backup networks in the event that the data network fails to process the call in a timely manner. In a preferred embodiment, the phone is a Voice-over-IP phone, and comprises a voice processing unit, a control unit, a line interface, a data network interface, and a backup switch.

A mechanism provides for communication of “keep-alive” messages from clients to servers in a packet telephony network environment. The servers may be call agents and the clients may be gateways or MGCP-controlled IP phones. A client (e.g., gateway) registers a virtual endpoint. Upon a period of inactivity in which the client does not receive any commands or acknowledgments from an assigned server (e.g., call agent), the client starts to send keep-alive messages periodically to the server. The keep-alive message may include an endpoint identifier that identifies the registered virtual endpoint. If the server fails to respond to the keep-alive messages after a period of time, the client initiates a fallback mechanism from a first call control protocol (e.g., MGCP) to a second call control protocol (e.g., H.323 or SIP), to provide call control handling using a default application. The keep-alive messages may be sent to other servers among the plural servers that are configured to operate with the client before determining whether to initiate the fallback mechanism. During the fallback state, the keep-alive message may be sent periodically until a response is received from the server. Communications with the assigned server may be reestablished thereupon based on the first call control protocol.

A packet telephony appliance includes a Euphony network processor that integrates networking and DSP functions to provide a low cost and efficient solution in building a networked appliance. In particular, a Euphony ATM Telephone (EAT) is built around the Euphony network processor. The EAT uses a real-time operating system to provide predictable processing and networking support. The EAT implements IObufs, which provides a unified buffering scheme that allows zero-copy data movement. Furthermore, the EAT uses an Event Exchange (EVX), which provides a flexible mechanism for event distribution, allowing software modules to be composed together in an extensible manner. EVX and IObufs are used together to provide highly efficient intra-appliance communication. The EAT provides a platform that can evolve gracefully to support new protocols, advanced telephony services and enhanced user interfaces.

The present invention provides an automatic restoration architecture for packet telephony applications. Each communication endpoint has access to a pair of edge routers, which interface with a core network through which communication paths are established. The communication paths effectively provide for the delivery of packets between a first edge router for a first endpoint and a first edge router for a second endpoint. When a communication session is established, a communication path is set up between the first edge router for the first endpoint and the first edge router for the second endpoint. Upon detecting a failure along the communication path, an attempt to restore the communication path between the first edge routers may be provided. If such restoration is not effective, an alternate communication path may be established with a second edge router for either of the first or second endpoints.

A method for telephony includes receiving at a facility of a telephone service provider a first subscriber request to place a first call to a specified telephone number. A provider request is sent to a remote server (26), which maintains a database (28) of records associating telephone numbers with respective packet network addresses, for a record (72) indicating a packet network address to which calls to the specified telephone number should be placed. Upon receiving the record from the remote server, the record is stored in a local cache (42) at the facility of the telephone service provider. After storing the record, upon receiving at the facility of the telephone service provider a second subscriber request to place a second call to the specified telephone number, the record is read from the local cache and is used in placing the second call to the packet network address.

A framework system serves to translate calls between dissimilar networks. More specifically, the framework system serves to translate calls between dissimilar packet networks and / or between a packet switch network and the public switched telephone network. The system includes a media engine module for terminating and handling channels of data transmitted between the networks in real time. An administrator module provides control and management interfaces for controlling processing applied to each channel. A conduit module serves to enable the distribution of software throughout the framework system allowing the distributed software to use interprocessor communication mechanism in the system. Optionally, a signaling engine module serves to provide access to call control information for connection between two networks.

Authentication systems and methods for increasing the security of online account access and transactions by leveraging the use of customer equipment provided by VoIP service providers. A method includes registering a packet telephony device with a packet telephony service provider for subsequent packet telephony communication, where the registration is based at least on an encoded encryption key. On a subsequent request to access an account, instructions are transmitted which require physical access to the packet telephony device to perform. Upon receipt of an indication that the instructions were successfully performed, the request is authenticated and access to the account is granted. Authentication may require a secure connection be automatically established between a web-enabled device and a packet telephony device. The instant disclosure leverages the security in the customer equipment hardware such as a Terminal adaptor (TA) or router so that a compromised account may be recovered.

In a communication system, a packet telephone obtains an internet address from an internet service provider and discovers a location of the packet telephone. The packet telephone transfers a message indicating the internet address and the location. A service node receives the message and stores the internet address and the location of the packet telephone in a location database. The packet telephone initiates an emergency call using the internet address. The packet network service node processes the emergency call by processing the internet address of the packet telephone in the location database to obtain the location of the packet telephone and by transferring the emergency call to an emergency service along with the location of the packet telephone.

This method includes that: captures user text, matches it with the character of RTP data text, according to the match result, confirms if it is RTP data text, records the 5-element group info of RTP data text, according to the change situation of other 4 items info with the same source IP address in the recorded 5-element group info, recognize the illegal VoIP user. The device recognizing illegal grouping phone user includes FPGA transmission logic chip and content addressing storage. The system recognizing illegal grouping phone user includes the device involving FPGA transmit logic chip and content addressing storage. This method, device and system can recognize simply illegal grouping phone user without interrupting text transfer.

A software ACD is described that is interposed between a switch or packet telephony environment, and an applications computer running a call processing application. The ACD may pass through primitive commands from the applications computer for direct execution by the switch or packet telephony environment, or in response to complex commands from the applications computer it may generate one or more different commands for execution by the switch or telephony environment.

An improved computer telephony (CT) server which utilizes a standard message structure to communicate with applications, but may interface such applications to a variety of different telephony environments by translating the standard message structure used by such applications into the specific message structure required by each such telephony environment. The aforementioned telephony environments may include PBXs, ACDs, packet telephony environments and public switched telephone networks. The desired telephony environment may be chosen by the application, or by the computer telephony server, some combination of both, or entirely automatically.

A method and apparatus are provided for recording a call between a client and an agent through an automatic contact distributor. The method includes the steps of the automatic contact distributor receiving a recording request from the agent, the automatic contact distributor activating a recording program, the recording program sniffing traffic within a packet telephone of the agent to detect packets exchanged between the agent and client and the recording program recording the detected packets between the agent and customer.