540 results about "Voice over IP" patented technology

A method and system to provide push-to-talk from one user to another in a wireless packet data telecommunications network is described. The system may include: a wireless communication network including push-to-talk (PTT) functionality, with a Session Initiation Protocol (SIP) Proxy Server; a SIP Registrar and Location Server operable to store contact addresses of active mobile devices; a Realtime Transport Protocol (RTP) Media Gateway (PTT Server) operable to function as a call endpoint for each of a plurality of mobile devices wherein the plurality of mobile devices are segmented into membership groups, the PTT Server further operable to multicast a communication from one member of the group to the other members of the group; and an Internet Protocol (IP) network interconnecting the SIP Proxy server, the SIP Registrar and Location Server, and the PTT Server.

A transmitter transmits time synchronized data via a pager/WiMax/802.x access to a receiver system, wherein the receiver system is programmed to receive data for specific geographic locations. The geographic locations may be specified by the user or by the receiver system, and includes state, zip codes, towns, counties, towns, or cardinal regions. The receiver is able to find its location when outside its cell region and is able to synchronize to the data transmitted in the new cell region. Further, the receiver system is able to remotely monitor weather data and other information at a different location via wireless Internet or voice over IP. A transceiver may also be used to receive weather or alert data. In response to receiving data, the transceiver transmits the data to low powered devices in a house using a different frequency band than the frequency band it received the data.

A Voice over Internet Protocol (VoIP) network is described in which session state is maintained in access switches, but not signaling gateways which maintain transaction state only during pendency of a related transaction. The signaling gateway further provides transparent inter-operation between the VoIP network and non-IP networks, such as the PSTN, by means of a translator which directly translates messages between the networks.

The present invention provides a system and method for a reliable, low-cost, secure Internet Protocol (IP) based network that provides broadband-based voice communications as well as video and data communications. The IP network is arranged to function with the infrastructure of the Public Switched Telephone Network (PSTN), to control telephone calls in SS7 type networks and to provide the features, applications, and services of the typical SS7 networks in a voice over IP network. The present invention supports large effective call volumes, allows accommodation of a wide range of broadband-based service platforms, provides flexibility to support current and future calling feature services, and provides high quality voice transmission.

A unified communications system enables a user to simultaneously manage real time voice communication with background audio streams such as a music. In one embodiment, the system comprises a plurality of devices connected through a persistent voice over IP channel, where each device is playing background audio independently, and each device is operatively coupled to a microphone. When a connected user speaks, Voice Activity Detection results in the automatic adjustment of background audio on other connected devices, the adjustments based upon user-input settings on each device.

Methods and apparatus for selecting a wireless communication network based on quality of service criteria associated with an application are disclosed. In one illustrative example, one of several different software applications of a mobile communication device is executed. The software application may be, for example, an e-mail application, an Internet data application, a voice-over-IP application, a video player application, an audio player application, or a video game application. Each software application is associated with different quality of service (QoS) criteria for data communications through a wireless network. The quality of service criteria may include, for example, a bandwidth criterion, a delay criterion, a delay variation criterion, and a data loss criterion. A scanning operation is performed to identify a plurality of a wireless networks available in a coverage area of the mobile device. One of the identified wireless networks is selected for communication based on a match between its available quality of service and the quality of service criterion associated with the executed software application. Advantageously, the most suitable network may be chosen and utilized for each software application.

Systems and methods are provided for preserving the privacy of data contained in mirrored network traffic. The mirrored network traffic may comprise data that may be considered confidential, privileged, private, or otherwise sensitive data. For example, the data payload of a frame of mirrored network traffic may include private Voice over IP (VoIP) communications between users on one or more networks. The present invention provides various techniques for securing the privacy of data contained in the mirrored network traffic. Using the techniques of the present invention, network traffic comprising confidential, privileged, private, or otherwise sensitive data may be mirrored in such a manner as to provide for the privacy of such data over at least a portion if not all of the mirrored communications between the mirror source point and the mirror destination point.

An admission control and load balancing system controls admission of packet streams or calls to a network and balances the packet traffic across the network, improving quality of service. The system includes a central database which stores information including cost data associated with individual paths and links across the network. A processor, in communication with the database, coordinates the admission control and load balancing decisions, and updates of the database cost data to reflect the dynamic network conditions, based on input from appropriate data sources. In one embodiment, referred to as the exact algorithm, the database is consulted by the admission control points or gatekeepers prior to admitting each arriving packet stream, and the database contents are updated call-by-call to reflect the allocation of resources to each admitted stream. In another embodiment, referred to as the inexact algorithm, control decision as well as database updates occur on a periodic rather than on a call-by-call basis to promote better scalability. In this embodiment, the processor periodically calculates admission decisions based on cost data in the central database. These admission decisions are then periodically forwarded to a satellite database associated with each gatekeeper, for storage and use in admission decisions until the next update epoch.

A method for implementing Voice over IP through an electronic device including the phase of sampling a voice communication in a corresponding sampled communication; encoding the sampled communication into corresponding encoded data; decoding the corresponding encoded data into decoded data; and connecting the electronic device to a packed switched network for transmitting the corresponding encoded data. The method may include a phase for storing an encoder and a decoder inside a portable memory unit of the electronic device for executing the phase of encoding and decoding the sampled communication. The phase of storing the encoder and a decoder may be executed on a IC Card intended to be inserted in the electronic device.

This disclosure describes a new method to browse and interact with the World Wide Web and other users while mobile comprising the following components: 1) a mobile communications device including but not limited to Blackberries, personal data assistants, or smartphones; 2) special glasses or viewing surface including but not limited to LCD (liquid crystal display) glasses, 3-dimensional goggles, or Organic LED (light emitting diode) panels; 3) a data input device including but not limited to a data glove, finger sensor, data input pen, or gesturing device; 4) a mobile communications device headset; and 5) a third generation or more powerful wireless carrier, Voice over IP provider, or a wireless LAN technology such as WiFi or, Wi-Max, or Bluetooth.

A method and system for Mobile Internet Protocol (IP) device redundancy. As mobile devices roam away from a home network and change a connective status, mobility binding records are sent to a multicast network address on the home network. The multicast network address multicasts the mobility binding records to other active Mobile IP home agent control nodes, standby home agent control nodes and standby home agents on the home network. The method and system allows standby home agent control nodes or standby home agents to be transparently switched for active home agent control nodes or active home agents that fail without downloading or uploading large numbers of mobility binding records after a failure. The method and system may also help reduce failed calls (e.g., data sessions including Voice over IP (VoIP), H.323, etc.), network congestion and improve user satisfaction in Mobile IP systems.

In order to provide a single common cost-efficient architecture for real time communication services for audio, video, and data over internet protocol, a voice over internet protocol (VoIP) system and architecture is provided by placing border elements (BEs) at the interface boundaries between the access network the user devices use and the VoIP infrastructure. The BEs use SIP protocol as the access call control protocol over any access networking technologies, for example, IP, Ethernet, ATM, and FR, and provides all services transparently to the end users that use SIP-enabled devices. To enable a scalable system, the SIP BEs are decomposed into separate communicating entities that make the SIP BE scalable and provide new capabilities not previously available by a self-contained SIP BE. Further, multiple levels of decomposition of a SIP BE can be provided by the present invention supporting a flexible and scalable SIP BE design that further improves system efficiencies and cost advantages as compared to use of single integrated border or edge elements. Further, a scalable SIP BE, made up of a plurality of physical entities for optimization of a large scale design, acts as a single integrated functional entity to logically execute a set of functions at the border of a VoIP infrastructure.

A soft switch 134 providing wireless PBX voice services to a local area network (WLAN) is used to extend PBX functionality to the cellular domain. A dual mode remote unit is capable of receiving signals both in the cellular system as well as the WLAN. The cellular system is comprised of a data-bearing path and a voice-bearing path. When the dual mode remote unit is within the WLAN, it communicates both voice over IP (VoIP) signaling as well as session initiation protocol (SIP) control signaling over the WLAN. When the remote unit is outside the WLAN, it communicates voice signaling over the voice-bearing path of the cellular network using a standard cellular voice channel. In parallel, it uses the data-bearing path of the cellular network to transmit SIP control signaling.

A Voice over IP (VoIP) or Real Time Messaging (RTM) firewall device is claimed that protects VoIP or RTM network traffic by identifying and controlling the delivery of such network traffic that is unsolicited and undesired by the recipient (i.e. VoIP or RTM spam). The system involves applying a unique marking to RTM messages close to a point of message origination and then at a point close to message termination for the intended recipient examining a reputation store for information on the unique marking and using that information in conjunction with a set of policy rules to decide whether to pass, reject, pass on to an RTM store or otherwise filter the RTM message. The unique marking serves to identify a source characteristic of the message such as the message originator, a corporate affiliation for the originator, or a RTM network characteristic of the originator such as a transmission gateway.

Trust ratings are embedded in certificates of calling devices in a Voice over Internet Protocol (VoIP) communications system. A method of managing trust ratings involves automatically accumulating complaints concerning VoIP calls initiated from calling devices, and comparing respective quantities of accumulated complaints associated with each calling device. When a quantity of accumulated complaints associated with a given calling device exceeds a given threshold, a trust rating of the given calling device is reduced. In subsequent VoIP calls the given calling device attempts to place, the reduced trust rating is included in the call request so that the subsequent calls are subject to more austere call screening operations than calls having the unreduced trust rating. Call recipients thus affect the calling device's trust rating simply by entering complaints associated with received calls. The method effectively combats spam over Internet telephony (SPIT) through participation of called parties.

A signal distribution system to save on the cost of cable modems for cable TV headend operators who wish to deliver broadband digital data services, DSL, video-on-demand or POTS service over their HFC CATV delivery networks. All species within the genus have a shared cable modem and a filtration and combining circuit comprised of a plurality of diplexer filters or junction boxes or both which mix baseband packet data with analog CATV signals onto coax drop lines coupled to the various subscribers and connect POTS or DSL signals onto twisted pair portions of siamese cables. Various species have the cable modem feeding digital data or packets of DOCSIS or other data or digitized DSL signals sent over the HFC to the cable modem or digitized POTS signals send over the HFC to the cable modem to either a packet switch, a DSL concentrator, a voice-over-IP gateway or some combination of the above. This data is either delivered to each subscriber as LAN packets or analog POTS signals or DSL signals or some combination of the above using coaxial cable or siamese cable drop lines.