Abstract: A ring-back tone generator system receives an inbound call connection request forwarded from a phone switch (such as an MSC) that receives the inbound call connection request from a caller phone attempting to connect to a target phone (e.g., a subscriber). The ring-back tone generator system initiates generation of a first call connection on which to provide the customized ring-back tone (and/or custom video image data) to the caller phone. The ring-back tone generator system also transmits, based on receiving the inbound call connection request, an outbound call connection request to the phone switch (such as the MSC) to establish a second call connection through the phone switch to the target phone. Consequently, an MSC that normally handles such call processing now relies on the ring-back tone generator system to handle call processing and provide a custom ring-back tone service.

Abstract: Users of network based software applications typically interact with an application via a command line interface or a menu interface. Conventional voice-driven communication threads rely on a dedicated communication line (connection) to maintain a session context, or state, between a user and an application for the duration of a session. Such state information enables the application to reference a session context for the session duration, and employ the session context to maintain a state from previous atomic messages sent and received between the device and the application. By maintaining and referencing a session context indicative of previous messages, an application identifies an atomic message as corresponding to a particular session context, and employs the session context to process the message within the context, or environment, defined by the previous messages in the session.

Abstract: This invention discloses a system for transmission of audio content to a telephone, the system including an audio stream navigation functionality operable by using a telephone user interface and enabling a user to select audio data to be heard by the user via said telephone, an interactive voice response unit operative in response to an input from said telephone user interface operating said audio stream navigation functionality and a transcoder operative in response to an input from said interactive voice response unit for providing audio data, from an audio source remote from said transcoder, to said interactive voice response unit for listening by said user via said telephone. A method for transmission of audio content to a telephone is also disclosed.

Abstract: The present disclosure is directed to systems and methods for routing a message to a communications device. These systems and methods may receive a message via a communications channel, convert the message into a uniform media format, identify at least one of the following: at least one user communications device to receive the message and at least one communications channel for delivering the message, then convert the uniform media formatted message for at least one communications protocol and then forward the message formatted for the at least one communications protocol to the at least one user communications device via at least one communications channel. In accordance with the systems and methods disclosed herein, messages sent in one communications protocol via a communications channel may be delivered to a user in a different communications protocol via a different communications channel.

Abstract: A VoiceXML interpreting system and method including a VoiceXML Interpreter. The VoiceXML interpreter includes a Fetcher operative to retrieve documents, a compiler operative to compile documents retrieved by the Fetcher and a cache which stores compiled documents compiled by the compiler.

Abstract: An IVR operative for voice communication with a telephone network and a data network and including a VoiceXML interpreter The VoiceXML interpreter being operative to provide a streaming audio output to the data network.

Abstract: Wireless network demands continually increase as wireless service providers pursue additional service capabilities. In a cellular communication system, leased lines between remote cell sites and the corresponding Mobile Switching Offices (MSOs) remain a major operating cost. Bandwidth reduction by identification and elimination of payload data and control information which need not be fully replicated because it can be deduced from information accessible or previously transmitted allows fewer lines to support the same bandwidth. A wireless access gateway is operable to aggregate such redundant and regenerable data on a backhaul link between a wireless cell site and the corresponding mobile switching office (MSO) to provide low-latency, type specific lossless bandwidth reduction. The wireless access gateway identifies regenerable information and eliminates portions of the data which the device need not transmit because the data is redundant, or accessible or recreatable, at the receiving side.

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: Via a thin client, a first user communicates with a presence server and establishes a communication channel with a second user through the presence server. The communication channel supports instant messaging in which the first user communicates via use of a first communication protocol and the second user communicates via use of a second communication protocol. When the thin client temporarily disconnects from the presence server, the presence server maintains an active “open” session for the thin client even though the thin client is disconnected from the presence server. The thin client can reconnect to the presence server and re-establish communications over the open session, enabling the first user to again instant message the second user.

Abstract: A method and system for automatic conversion of text to speech including automatically analyzing a text to define at least one vocabulary domain and carrying out a text-to-speech conversion by employing said at least one vocabulary domain.

Abstract: A message notification system and methodology including an on-the-fly message parser which parses data in an incoming message as the message is received and produces on-the-fly parsed information and an on-the-fly message notification generator, producing a message notification in response to the on-the-fly parsed information, at least partially while the on-the-fly message parser parses the data in the incoming message.

Abstract: Apparatus and accompanying methods for use in a long-haul echo control device to automatically measure sufficient changes in path length and compensate a long-haul echo canceller for resulting variations in echo occurring over a path. The apparatus measures a round-trip path delay interval of, e.g., a predefined signaling tone(s) sent between international switching centers while establishing an outbound call connection over the path. Once the delay is measured, the apparatus sets echo delay of a corresponding long-haul echo canceller to accommodate that delay. For a multi-channel facility (e.g., T1 or E1), path delay is separately measured for each channel on that facility, and then the apparatus changes the echo delay to the same amount for the corresponding echo canceller associated with each and every channel on that facility.

Abstract: This invention discloses a voice communication system including a telephone network including a multiplicity of telephones interconnected by telephone network interconnections, a computer network having a multiplicity of nodes and enabling e-mail communication between said nodes, a multiplicity of voice response computers, each voice response computer being connected to a node of the computer network and being actuable by an input received from one of the multiplicity of telephones via the telephone network for communicating voice received via said one of the multiplicity of telephones via e-mail over the computer network. A method of voice communication including a telephone network including a multiplicity of telephones interconnected by telephone network interconnections is also disclosed.

Abstract: A method and system for interactive message communication, including the steps of receiving a marked up e-mail by an interactive voice response system, the marked up e-mail containing a menu, converting the menu from text speech, producing a speech menu, reading the speech menu from the interactive voice response system to a telephone unit, and transmitting a response to the marked up e-mail based on the speech menu.

Abstract: A method for management of electronic mail, comprising the steps of converting an e-mail message from text to speech, receiving an input request for a selected e-mail message, reading the selected e-mail message, recording a reply to the selected e-mail message, producing an audio file, and sending the audio file as an attachment to a reply e-mail.

Abstract: A signal system 7 (SS7) processing system for use in an SS7 network includes a primary interface configured to process communications according to at least one layer of the SS7 protocol stack, the primary interface being configured to provide checkpoint messages relating to the communications and the at least one layer of the SS7 protocol stack, each layer of the at least one layer of the SS7 protocol stack having a corresponding set of information associated with each communication, and a backup interface configured to process communications according to the at least one layer of the SS7 protocol stack, the backup interface being coupled to the primary interface, and configured, to receive the checkpoint messages from the primary interface, wherein the checkpoint messages contain sufficient information for the backup interface to properly process communications that are transferred from being processed by the primary interface to being processed by the backup interface and contain less than all information i

Abstract: Apparatus and accompanying methods for use in a long-haul echo control device to automatically measure sufficient changes in path length and compensate a long-haul echo canceller for resulting variations in echo occurring over those paths. Specifically, the inventive apparatus measures a round-trip path delay interval of, e.g., a predefined signalling tone(s) sent between international switching centers during the course of establishing an outbound call connection over that path. Once this delay interval is measured, the apparatus commensurately sets echo delay of a corresponding long-haul echo canceller to accommodate this path delay. For a multi-channel facility (e.g., T1 or E1), path delay is separately measured for each channel on that facility. Should the measured path delay sufficiently change for any one such channel, then the apparatus changes the echo delay to the same amount for the corresponding echo canceller associated with each and every channel on that facility.

Abstract: Apparatus for an echo canceller, and an accompanying method for use therein, for improving voice performance in handling mobile-to-mobile calls in a GSM environment, specifically with a base station controller (BSC) at each end of a call path where one BSC supports tandem free operation (TFO) and the other does not. Here, each canceller has a capability, through unique in-band messaging carried by robbed-bit signaling, to communicate information with its peer (distant) echo canceller across the call path so as to permit each canceller to coordinate its operation with the other, and particularly to enable a specific voice enhancement feature set optimal to a particular type of call, e.g., mobile-to-mobile, and without adversely affecting the TFO protocol.

Abstract: A method and apparatus for providing admission control, queue management, and shaping/scheduling of flows in a communication device is described. In one embodiment, the communication device is a quality of service access communications device having a central processing unit and a memory module. The communication device includes a queuing module having a receive segment, a control segment, and a transmit segment. The transmit segment includes a fly-by flow admission control block that performs admission control of flows. The transmit segment further includes a two-tiered hierarchical shaper/scheduler block having a level-1 shaper/scheduler and a plurality of level-2 shaper/schedulers for shaping and scheduling of flows.

Abstract: Methods and apparatus for providing data communication between modems coupled over digital channels are provided. In one aspect, the present invention features an apparatus used in a system for providing communication over a digital channel to couple a first modem on a first network to a second modem on a second network. The apparatus includes a modem module that couples to the first modem at a data rate determined in part by a modulation rate of the modem module. The apparatus further includes a digital channel interface module coupled to the modem module, the digital channel interface module including a buffer that receives data from the digital channel and provides data to the modem module. The apparatus also includes a control circuit that detects a level of data in the buffer and modifies the modulation rate of the modem module based on the level of data detected.