Showing papers on "Telephony published in 2009"

Reliable, long-haul data communications over power lines for meter reading and other communications services

Abstract: A system, method and computer program product provides for power line communications (PLC) over electric power lines includes a device mountable near an electrical distribution transformer (DT) to provide a high speed interface and communicates with one or multiple access devices, which provide low speed interfaces for analog signals or digital signals over RS 232, RS 485, optical, wireless and Ethernet. The device transmits data to/from these access devices over the electric lines to other repeaters over one or more wires of an electrical line or over multiple lines, and serves to strengthen and improve signal quality. Upon detecting a wire or line is having problems carrying data, the data is sent over other wires, and upon power line failures, wireless backup to mobile/GSM and WiMax networks is utilized. The device permits utilities and others to read electric meters, monitor the power quality of the distribution grid and detect power losses/failures/outages, and permits telecom service providers and others to provide a communications link to cell phone towers, WiFi Access Points and enable broadband Internet and telephony in rural, remote or sparely populated areas.

Enhanced telephony computer user interface allowing user interaction and control of a telephone using a personal computer

Abstract: An enhanced telephony (ET) computer user interface that seamlessly integrates features of a personal computer (PC) and a telephone into a coherent user interface. The user is provided with a rich variety of functionality that leverages the fact that the PC has considerably more processing power and greater access to variety of data than the ordinary telephone. This processing power and data access is used to the user's advantage as the telephone's capabilities and functionality are greatly expanded. In general, the ET user interface includes a plurality of environments for the user to choose. These environments include a My Contacts environment, a communication preferences environment, and a Call History environment. Each of these environments contains certain available processes and features for controlling and managing telephones.

System and method for processing telephony sessions

Abstract: In one embodiment, the method of processing telephony sessions includes: communicating with an application server using an application layer protocol; processing telephony instructions with a call router; and creating call router resources accessible through a call router Application Programming Interface (API). In another embodiment, the system for processing telephony sessions includes: a call router, a URI for an application server, a telephony instruction executed by the call router, and a call router API resource.

System and method for processing media requests during telephony sessions

Abstract: In a preferred embodiment, the method of caching media used in a telephony application includes: receiving a media request; sending the media request to a media layer using HTTP; the a media layer performing the steps of checking in a cache for the media resource; processing the media request within a media processing server; and storing the processed media in the cache as a telephony compatible resource specified by a persistent address. The system of the preferred embodiment includes a call router and a media layer composed of a cache and media processing server.

Call redirection for enterprise hosted dual mode service

Abstract: Systems and methods provide a single E.164 number for voice and data call redirection and telephony services such as caller identification, regardless of in which type of network a dual mode mobile device operates. When the dual mode device registers and is active in a GSM network, temporary routing and status updates are triggered and resultant information is maintained in both networks. A mobile terminated call is routed through an enterprise WLAN with call control within the enterprise being handled by SIP or H.323 signaling, and the call is redirected to the mobile device in the GSM network, where call control is assumed by the SS7 network. Services are provided using the protocols native to the active network, and the single E.164 is used consistently along with or lieu of the temporary routing information for subscriber identity specific functions, such as caller identification and voice mail.

Telephony web event system and method

Abstract: An embodiment of the system for publishing events of a telephony application to a client includes a call router that generates events from the telephony application and an event router that manages the publication of events generated by the call router and that manages the subscription to events by clients. The system can be used with a telephony application that interfaces with a telephony device and an application server

Services platform for networked devices that provide telephony and digital media services

Abstract: Telephony and digital media services may be provided to a plurality of locations, such as to a plurality of homes and offices, though the deployment of telephony and digital media services devices to the locations, wherein each device is configured to function as a voice, data and media information center. A services platform in accordance with an embodiment of the present invention enables entities to deploy, manage optimize and monitor a network of such devices in a turnkey fashion. In accordance with one embodiment of the present invention, the services platform is implemented on one or more computers and includes at least a device monitoring subsystem, a device management subsystem and a user interface.

Carrier-wave telephony over power lines: Early history [History of Communications]

Abstract: In this column we discuss the early history of power line voice communication, beginning in 1918 and continuing until the early 1930s. We note that these developments were based on the 1910 demonstration by Major George Squier of the United States Army Signal Corps of his "wired wireless" technique for transmitting multiple telephone channels over one pair of wires. Power companies world-wide picked up on this carrier-wave telephony technique for use over power lines, with electrical manufacturing companies such as GE and Westinghouse in the United States and Telefunken in Germany, among others, then developing a variety of systems for this purpose. By 1930, the technique had reached a period of maturity, with 1000 systems installed throughout Europe and the United States.

System and method for registering an ip telephone

Abstract: A system and method for establishing connection of an IP telephone to a network may include, in response to receiving a registration request from an IP telephone, generating a command to cause network access devices to ping the IP telephone. The command may be communicated to the network access devices. Ping information may be received in response to the network access devices pinging the IP telephone. A network access device may be selected that has the highest quality network access path to the IP telephone. In response to selecting the network access device that has the highest quality network access path, a network address of the selected network access device may be communicated to a network device to enable the IP telephone to communicate with the selected network access device. Credentials may be communicated to the IP telephone to register with the selected network access device.

System and method for replying to voice messages left by callers

Abstract: An exemplary system and associated method for a communications system allowing responses to messages in an electronic environment are disclosed. The communications need not be based on the same platform or carrier. The system comprises a processing center to couple to a telephony network where the processing center provides a sender, not subscribed to the system, with messaging capabilities. The processing center includes a voice mail server to record a message from the sender and retrieve the message for a recipient that is subscribed to the system. A call processing logic module is arranged to access a database to determine whether a telephony device used by the sender has non-voice capabilities. The call processing logic module can notify the recipient of the message and informs the recipient of the capabilities of the sender to allow various response modalities (e.g., by voice mail, e-mail, or text message).

Media based collaboration using mixed-mode PSTN and internet networks

Abstract: A method which allows standard telephone users to audio conference with video conferencing participants over IP networks in a private secure environment. A dial-out is performed from one or more conference client terminals bridging audio between the Internet and the PSTN networks. The process uses a mixed mode hybrid network architecture for call set-up, initialization and teardown including the method to mix audio at the desktop terminal instead of in a general purpose server as in the prior art. The method conferences video and audio between multiple clients and include audio from a standard telephone network within the conference. A virtual private network connects all of the IP clients together including the voice over IP server used to transcode the proprietary audio into the H.323 standard for transport into the telephony network.

Method, apparatus and system for use of presence and location information in intelligent call routing

Abstract: A telecommunication system routes an incoming call to one or more telephony devices associated with a user, including personal digital assistants and other remote devices, based on the called party's historical call pattern created and updated by the system. The system enhances the routing using location or presence information. The call routing is designed to route calls to targeted devices to increase the likelihood that the call is answered by the intended called party while also alleviating the need to place multiple calls in an attempt to locate the called party.

Method and apparatus for pstn-based ip active call recovery and re-routing

Abstract: A method and system for rerouting IP telephony call information. The system includes an existing IP telephony call, the telephony call originating at a first IP telephony phone. A second IP telephony phone receiving the IP telephony call from the first IP phone and a network routing condition, the routing condition causing the IP telephony call to be rerouted. A switchover instruction, the switchover instruction initiating a switchover of the existing IP telephony call to a PSTN.

System and method for call management

Abstract: A method and system for call management is provided. In a system embodiment a call manager server is provided that can managing incoming call requests from certain telephony devices and effect connections to other telephony devices based on those incoming call requests. The call manager server can also maintain a policy that defines permissions as to whether certain telephony devices can request interruption of an ongoing phone call at a destination telephony device.

Signal detection and blocking for voice processing equipment

Abstract: A telephone call processor for processing telephone calls comprising voice signals and data signals, the call processor comprising a first telephone interface and a second telephone interface, the call processor being operable in a first mode and in a second mode. In the first mode, the call processor is adapted to receive voice signals and data signals at the first telephone interface and to transmit voice signals and data signals via the second telephone interface. In the second mode, the call processor is adapted to receive voice signals and data signals at the first telephone interface, to block data signals from being transmitted via the second telephone interface and optionally to transmit voice signals via the second telephone interface.

Spectral shaping for speech intelligibility enhancement

Abstract: A speech intelligibility enhancement (SIE) system and method is described that improves the intelligibility of a speech signal to be played back by an audio device when the audio device is located in an environment with loud acoustic background noise. In an embodiment, the audio device comprises a near-end telephony terminal and the speech signal comprises a speech signal received over a communication network from a far-end telephony terminal for playback at the near-end telephony terminal.

Wireless telephony device with breath analysis sensor and methods for use therewith

Abstract: A wireless telephony device includes a memory that stores at least one telephony application. A processor executes the telephony application to process at least one telephone call via a wireless telephony network in response to commands of a user. A breath analyzing sensor analyzes a breath of the user in conjunction with the at least one telephone call and generates breath analysis test data in response thereto. The telephony application generates a breath analysis test message, based on the breath analysis test data and transmits the breath analysis test message via the wireless telephony network.

Telephony fraud prevention

Abstract: A method for guarding against telephony-based fraud that comprises, at a telephony device, identifying a caller ID of an incoming call or a dialled number of an outgoing call attempt or a number to be dialled. The identified caller ID or dialled number or number to be dialled is then compared against a blacklist of telephone numbers. In the event that a match is found, a warning is presented to a user of the device and/or the call or call attempt is terminated.

3GPP LTE Downlink System Performance

Abstract: In this paper we quantify 3rd Generation Partnership Project (3GPP) Long Term Evolution (LTE) Release 8 downlink system performance for a macro cell hexagonal grid scenario. The system performance is analyzed for a closed loop Single User Multi-Input Multi Output (SU-MIMO) mode and compared with Single Input Multiple Output (SIMO) and Multi-User (MU) MIMO modes, for a full buffer scenario and static users. In addition to the full buffer scenario, traffic models are considered for SIMO mode to evaluate impact of partial loading and handover. Voice over Internet Protocol (VoIP) system performance is quantified for static users. Mobility simulations are performed for Video Telephony (VT) users and compared to the static case.

Real-time media broadcasting via telephone

Abstract: Disclosed are systems and methods for live broadcasting of one or more media streams, such as audio streams, via the internet such that it is accessible substantially by a number of listeners at a number of locations. Said system is operable such that the contents of each of said one or more media streams can originate from a call or other communication from a conventional telephony device which does not require internet connection.

Application store and intelligence system for networked telephony and digital media services devices

Abstract: Telephony and digital media services may be provided to a plurality of locations, such as to a plurality of homes and offices, though the deployment of telephony and digital media services devices to the locations, wherein each device is configured to function as a voice, data and media information center. A system in accordance with one embodiment of the present invention includes an application store and an application intelligence subsystem implemented on one or more computers. Each of the application store and the application intelligence subsystem is communicatively connected via a network to a plurality of such telephony and digital media services devices. The application store is operable to provide applications via the network for installation and execution on each of the plurality of devices. The application intelligence subsystem is operable to obtain and report information about applications installed and executed on each of the plurality of devices.

Audio and video channel impact on perceived audio-visual quality in different interactive contexts

Abstract: With the advent of audio-visual IP clients, video telephony becomes a realistic option in many application scenarios. In order to guarantee an adequate quality to its users, providers of audio-visual telephony services need to know the impact of the audio and video transmission channel characteristics on perceived Quality of Experience (QoE) in a realistic interactive setting. For this aim, a conversational video telephony experiment was conducted where the audio and video channel settings were adjusted in a controlled way, and participants were asked about the perceived audio, video and overall quality after carrying out a conversation over the audio-visual channel. We analyze the results with respect to the impact the two modalities have, as well as with respect to the impact of the conversation scenario.

A statistical framework for artificial bandwidth extension exploiting speech waveform and phonetic transcription

Abstract: In the past, artificial bandwidth extension (ABWE) has primarily been investigated to enhance transmitted narrowband speech signals at the receiving side. State-of-the-art schemes show improved quality versus narrowband speech; however, a clear gap to wideband speech is still reported. This is largely due to the insufficient ABWE performance on fricatives, particularly /s/. We asked ourselves to what extent the speech quality could be improved, if we knew the currently spoken phoneme. In this paper we present a framework using phonetic transcriptions as a-priori knowledge besides the speech waveform. Possible applications are high-quality offline ABWE of telephone, pilot, or historic speech recordings, memory efficient narrowband speech synthesis followed by ABWE, and extension of narrowband telephone databases to train wideband acoustic models for automatic speech recognition. For the classical conversational telephony application, an improved ABWE scheme is also proposed making use of transcription information only during training.

Complex-Network Modeling of a Call Network

Abstract: Recently, real-life data have revealed that the number of calls originating from or received by a telephone number in a network follows a power-law distribution. They show that a few telephone numbers make or receive a very large number of calls, whereas a large number of telephone numbers make or receive very few calls. The data have overthrown the general assumption that all telephone numbers are similar in generating telephone traffic. The first objective of this paper is to therefore construct a telephone call network (TCN) with connection properties following power-law distributions. With a more realistic TCN, researchers and engineers will be able to evaluate the telephone traffic behavior more accurately. Having constructed the aforementioned TCNs, we then consider the scenario when there is a sudden surge in the number of telephone calls, for example, during natural or man-made disasters. Under such a condition, the telephone network is usually overloaded and cannot operate properly. To mitigate the problem, we propose a preferential call blocking (PCB) scheme, aiming at blocking calls to target telephone numbers which have large numbers of incoming calls (in-strengths). We will investigate the effect on the carried traffic intensity when the PCB scheme is applied. We will compare the results with a benchmark, which corresponds to the case when all calls are blocked with equal probability. For the sake of completeness, we will also study the effectiveness of the blocking schemes when applied to a traditional TCN, in which all telephone numbers can call one another with equal probability.

Systems for providing telephony and digital media services

Abstract: A system, method and apparatus for providing telephony and digital media services to a location, such as a home or office, is described herein. In one embodiment, the system includes a telephony and digital media services device that is configured to function as an all-in-one voice, data and media information center. The device provides telephony functionality both directly and through associated handsets. The device pairs a user-friendly touch-screen interface with a high-performance hardware/software architecture capable of delivering advanced media applications and graphics combined with landline quality telephony service all in one integrated system.

ITU-T coders for wideband, superwideband, and fullband speech communication [Series Editorial]

Abstract: This section of the magazine presents recent algorithms developed by the ITU to provide high quality coding beyond traditional narrowband telephony. Speech coders can be characterized by their bit rate, quality, complexity, and delay. Typical applications fall into one of two categories, one-way and two-way. The first includes storage applications such as telephone answering systems, streaming, multimedia delivery, and push-to-talk calls. The second includes realtime communications such as two person phone calls and conference calls. In this latter category, if the delay is too large - exceeding 300 ms round-trip - humans have difficulty communicating, while for storage and playback operations delay is not a factor. The complexity of a speech coder is one of the main contributing factors to its cost and energy usage. Complexity is most often measured in terms of memory usage (both RAM and ROM) and the number of instructions executed per second. All applications are sensitive to cost, and many are sensitive to energy usage as well. The desired bit rate is determined by channel capacity or storage capacity, depending on the application.