183 results about "Voice over internet protocol voip" patented technology

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.

A wireless communication system is provided that has at least three nodes arranged in a multi-hop ultra wide band (UWB) communication network such that communications from a first node destined for a third node pass through a second node. Each of the devices in the system includes a radio and a media access control (“MAC”) module that is configured to establish multi-hop UWB wireless communications between the three or more wireless communication devices that enables high bandwidth applications such as Voice Over Internet Protocol (“VoIP”); multiplayer gaming; Wireless High Definition Television; and Internet Protocol Television (“IPTV”) among others. The MAC module is configured to avoid bandwidth reservation conflicts so that network performance does not degrade as the number of hops or the number of nodes in the wireless communication system increases. The MAC also facilitates utilization of multiple channels to maximize the available spectrum and is further configured to dynamically switch between channels to maximize throughput and meet or exceed quality of service (“QoS”) requirements such that QoS is guaranteed and network resources are efficiently utilized.

A power line communication (PLC) hub incorporates multiple network communication media and services for devices. Implementations can be plugged into an electrical outlet to furnish a single point for both electrical power and network connectivity. Each of the electrical outlets is tied through an electrical distribution system to a network through a conventional PLC bridge to provide network connectivity through the electrical distribution system using PLC technology. The PLC outlet hub can include various combinations of applications and / or services including Voice Over Internet Protocol (“VoIP”) gateway, media server, Internet router / gateway, Local Area Network (“LAN”), both wired and wireless voice and video conferencing capability, including a VoIP, Voice over WiFi (“VoWiFi”), Power Over Ethernet (“PoE”), Wireless 802.11a / b / g / n capability and (Wireless Universal Serial Bus “WUSB” / Ultra Wide Band “UWB”) wireless connectivity, and Blue Tooth. Versions of the PLC outlet hub can have an Ethernet IEEE 802.3 family (such as IEEE 802.3af) compliant power supply to furnish power to PoE enabled devices.

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 method and system for fast setup of group Voice over Internet Protocol (VoIP) communications. The method and system provide a lower VoIP signaling overhead and a faster setup-time of group calls such as N-way audio and / or audio-video conference calls. Multi-party VoIP connections are quickly set-up over one or more virtual communications channels established between plural network devices using a session layer application, one or more group transport layer ports and a broadcast destination network address.

The present invention takes advantage of Voice over Internet Protocol (VoIP) technology by introducing VoIP-based call center telephony equipment that is software-based and runs on inexpensive off-the-shelf personal computer (PC) systems. With the VoIP-based call center system of the present invention, the traditional Public Switched Telephone Network (PSTN) is coupled to a Voice over Internet Protocol (VoIP) gateway in order to convert all incoming traditional telephone communication into VoIP based telephony telecommunication. This is performed using the well-known SIP telephony protocol set forth in RFC 3261. Once converted to the VoIP format, the incoming VoIP-based calls are directed to the VoIP based Call Center Server system. The Call Center Server system provides all the sophisticated call center features that were formerly only available in large call centers created with specialized expensive telephone equipment

An approach providing the efficient use of speech synthesis in rendering text content as audio in a communications network. The communications network can include a telephony network and a data network in support of, for example, Voice over Internet Protocol (VoIP) services. A speech synthesis system receives a text string from either a telephony network, or a data network. The speech synthesis system determines whether a rendered audio file of the text string is stored in a database and to render the text string to output the rendered audio file, if the rendered audio is determined not to exist. The rendered audio file is stored in the database for re-use according to a hash value generated by the speech synthesis system based on the text string.

Systems and methods are provided that facilitate distributed multichannel wireless communication and provide the highest level quality of service (“QoS”) guarantee and support extremely high bandwidth applications such as voice over internet protocol (“VOIP”) streaming audio and video content (including high definition), and multicast applications and also supports convergent networks, ad hoc networks, and the like. A modular MAC architecture provides a group of nodes with the ability to simultaneously communicate with each other using multiple separate communication channels during the same timeslots. The additional throughput gained by employing multiple communication channels is amplified by dynamically mapping the communication channels and timeslots in a network so that multiple channels can be reused simultaneously throughout the network during the same timeslot in a fashion that does not create collisions.

According to the invention, a network interface device for processing a telephone call is disclosed. The network interface device includes a first and second communication interfaces, a telephone switch and a controller. The first communication interface is coupled to at least one of a wireless phone network, a public switched telephone network (PSTN), and a voice over Internet protocol (VOIP) network. The first communication interface receives the telephone call from any of a number of callers that are remote to the network interface device. The second communication interface coupled to one or more phones at a user location where the one or more phones are associated with a telephone number that any of the callers can use to call the one or more phones. The telephone switch is coupled to both of the first communication interface and second communication interface and optionally routes an incoming phone call to the second communication interface if one or more access control rules permit routing the incoming phone call to the second communication interface. The controller analyzes the one or more access control rules and either routes the incoming phone call from the first communication interface to the second communication interface or prevents the incoming phone call from reaching the second communication interface. The one or more phones ring when the incoming call is routed to the second communication interface.

A system for transferring a call from one teleagent to another in response to a loss of voice quality in a voice over Internet protocol (VoIP) communication system. Network parameters that affect voice quality over broadband connections are detected manually or automatically and an alert is generated. In response to the alert, a supervisor can transfer the call in progress to a second agent so that a better communication link can be provided and voice quality restored. Such transfer can be seamless to the caller. In different embodiments, transfer, or intercept, can occur manually or automatically (e.g., without supervisor intervention).

An ISUP / SIP gateway is implemented to encapsulate SS7 location requests from a switched telephone network into an Internet protocol SIP message including one or more location objects. The ISUP / SIP gateway is implemented between a mobile switching center (MSC) and a VoIP positioning center (VPC) to provide support of unlicensed mobile access (UMA) voice over Internet Protocol (VoIP) call routing, e.g., for E9-1-1, 4-1-1, 2-1-1, etc. The disclosed location objects that are encapsulated into a SIP message includes calling party number, called party number, CGI, International Mobile Subscriber Identity (IMSI), mobility indicator, and / or access point MAC address. At the SIP / ISUP gateway, SS7 integrated services user part (ISUP) location request messages are encapsulated within SIP. The SIP encapsulated location request message is then routed over the Internet to a destination VoIP positioning center (VPC).

A method and system for providing Voice over Internet Protocol (VoIP) in a wireless communications system using a Packet Data Conversion Protocol (PDCP) with enhanced functionality that allows sub-packets to be processed for transmission over different Radio Bearers (RB) with unequal or different levels of error protection and / or error detection based on perceptual importance of the bits in the sub-packets. The enhanced PDCP being operable to partition a packet into a plurality of sub-packets such that different levels of error protection and / or error detection may be applied to the packet. An overall gain in system capacity can be expected due to transmit power savings provided via unequal error detection and error protection.

A method and apparatus for providing wireless data packet call regulation and call access for wireless Voice over Internet Protocol (VoIP) communications in a wireless network. A first and second time stamp is generated upon the reception and successful transmission of a data packet. A packet delay is determined based upon the first and second time stamps. At least one average delay metric is used to determine whether to provide new call access based on the at least one average delay metric. Another aspect of the present invention is to block a class of users based, in part, on the average sector delay for each user subscriber class. If the average sector delay is greater than that allowed for a class of users, users of that class are denied access. A delay quality of service (QoS) block metric is used to determine if an individual user should be force dropped.

Emergency remote control communications systems and methods provide remote access to, activation of, deactivation of, monitoring of, communications with, and / or otherwise management of a Voice-Over Internet Protocol (VoIP) communications device (and / or another peripheral communications device) in response to an emergency communication originating from the VoIP communications device. An emergency remote control communications software product detects an emergency communications signal, and, in response, generates an emergency remote control communications signal for communication to the VoIP communications device via a communications network.

The present invention is directed to a system and method for sending and receiving Voice-over-Internet-Protocol over a wireless computer network utilizing a hybrid wireless Voice-over-Internet-Protocol telephone. The invention utilizes a phone controller, a wireless handset, a base station and at least one access point. The wireless handset is equipped with both a wireless personal area network transceiver and a wireless local area network transceiver. The base station is equipped with a wireless personal area network transceiver and a network interface card. Voice-over-Internet-Protocol packets are routed to a phone controller which forwards the packets to either an access point for transmission to the wireless handset over the wireless local area network, to a base station for transmission to the wireless handset over the personal area network, or to server for transmission out of the local area network.

An analog telephone adapter (ATA) having a subscriber identity component in the format of a Subscriber Identity Module (SIM) that couples a telephone to a cellular network via Voice over Internet Protocol (VoIP), thereby allowing a cellular service provider to provide an alternative communication service for the home or business. Instead of using the plain old telephone service, a subscriber can plug their analog telephone into the SIM-enabled ATA (SIM-ATA) and receive telephone service from a cellular service provider, eliminating or reducing the need for a traditional Local Exchange Carrier. The SIM-ATA converts signals from analog to digital, and vice-versa. Once the analog telephone signal has been converted to digital, an IP-based protocol (e.g., VoIP) can be used to transmit the telephone call over a digital network. The cellular service provider can then track usage and bill the subscriber accordingly.

Voice Over Internet Protocol (VoIP) emergency calls to an Emergency Response Center (ERC) are handled through a VoIP conference bridge on a VoIP service provider's soft switch. The soft switch works with a VoIP positioning center (VPC) to obtain location information, which is compared against a PSAP database to find an initial best-appropriate PSAP for the location of the emergency caller. The PSAP is issued an Invite message to join the conference, establishing an emergency call. Third parties such as police, ambulance may be issued Invite messages to join the conference. Cold transfers are avoided by Inviting participants to join a single emergency conference rather than passing an emergency call from party to party (e.g., from PSAP to police to ambulance, etc.) The PSAP, other emergency responders, and even the initial VoIP emergency caller may leave and rejoin the VoIP conference without dropping the conference between the others.

The present invention provides a method and an apparatus for controlling service quality of data communications in a wireless network in which quality of service control for voice over internet protocol packets is provided on a downlink shared channel. A method is provided for a wireless communications between at least a first and a second service user. The method comprises prioritizing transmission of voice over internet protocol packets for the first and second service users over a shared downlink channel based on a first end-to-end delay that the first service user experiences relative to a second end-to-end delay that the second service user experiences. For example, such a priority adaptation may provide quality of service control for voice over internet protocol over a high-speed downlink packet access channel based on an end-to-end delay that a mobile-to-mobile user may be currently experiencing relative to a mobile-to-land user. In this way, a scheduler at a base station of a communications system may maximize use of voice over internet protocol capacity while maintaining an acceptable level of end-to-end delay for different users of mobile stations.

Methods and systems enable the injection of sound effects, for example professional voices, music, special effects, etc. into electronic communications, for example mobile telephone conversations, land line telephone conversations, voice over Internet protocol conversations, etc. In various disclosed embodiments of the invention, the sound injection occurs through a simplified process, each sound effect being classified both by category and specific sound. In different embodiments, a sound effect is selected through the selection first of a category followed by a specific sound, or by the direct selection of a specific sound from within a category.

The Federal Communications Commission Order 05-116, E911 Requirements for IP-Enabled Service Providers, (FCC Order) requires interconnected Voice over Internet Protocol (VoIP) service providers to deliver all emergency service calls to a VoIP service user's local emergency service operator as a standard feature of the service. Under the FCC Order, VoIP service providers must supply emergency service if the VoIP service user accesses the VoIP service over broadband and if the VoIP service provider interconnects to and from the Public Switched Telephone Network (PSTN). Embodiments of the present invention guarantee compliance with the FCC Order by routing a call as either a VoIP call over the Internet or as a conventional call over the PSTN, verifying that at least one of the VoIP or PSTN telephony connections supports emergency service and routing calls over that connection or, if there is no connection supplying emergency service, restricting calling over the VoIP connection.

A method and apparatus for surveillance of voice over Internet protocol (VoIP) communications is provided. A server (112) associated with a network provider receives information to identify communications associated with a party under surveillance (200). The server uses the information to direct communications associated with the party under surveillance to a surveillance server (130) (202). The communications are analyzed to identify any initial VoIP data (204). If initial VoIP data is detected, identifiers for subsequent related VoIP communications are determined from the initial VoIP data (206). The identifiers for subsequent VoIP communications are used to identify subsequent VoIP packets. Those subsequent VoIP packets are then stored and analyzed as needed by an agency conducting surveillance.

A method, apparatus, system, network element, and software product are used to provide a traffic signal, such as a voice over internet protocol (VoIP) signal, via high speed packet access. The traffic signal is prepared and provided via a high speed transmission, and a control signal is gated to produce a discontinuous transmission. This reduces interference, and therefore increases traffic capacity.

A system is provided for verifying an identity of a calling party. The system includes a storage that stores voice sample information in association with information that identifies an individual. The system also includes a receiver that receives, from a communications platform called by the calling party, a request to authenticate the identity of the calling party as the individual, using a digital audio sample packetized according to voice over internet protocol and obtained from the calling party during the call. Additionally, the system includes a processor that compares the digital audio sample to the voice sample information to authenticate the identity of the calling party as the individual.

In a Voice over Internet Protocol (VoIP) video communication method and apparatus, first and second VoIP-based terminals store video event indexes and images mapped thereto with respect to state information of a user and terminal, and establish a session using session establishment information defining the video event. The first terminal then transmits a data packet including the video event index, and the second terminal receives the data packet and outputs an image mapped to the video event index of the images stored in advance. Thus, by transmitting and receiving a predefined index alone without transmitting user and terminal states as a real time data-type image, the VoIP video communication method and apparatus prevent unnecessary waste of bandwidth in video communication.

Voice Over Internet Protocol (VOIP) devices are invited to a conference bridge based on a pre-defined criteria for the conference bridge (e.g., passengers on a plane. The conference invite messages are transmitted using Internet Protocol, which they may or may not accept. The invited VoIP users may be determined based on their physical proximity to the initial caller and other pre-determined characteristics. A VoIP soft switch includes conference bridges that eliminate the conventional requirement that they dial the phone number of another specific VoIP communications device. Instead, location information relating to the initial VoIP user is passed to the VoIP conference bridge, either from the user's VoIP communication device or from their respective location server. Location and other information is then compared by the VoIP soft switch against other VoIP devices, to find potential VoIP conference participants.

An apparatus that includes a coupling, a signal level measurement circuit, a communication circuit and a processing circuit. The coupling is configured to connect to and received broadband RF signals from a coaxial cable termination of a communication network. The signal level measurement circuit is operably coupled to the coupling, and is operable to generate signal level measurements regarding a first set of the broadband RF signals. The communication circuit operable to obtain voice over internet protocol data packets from a second set of broadband RF signals. The processing circuit is operably connected to the communication circuit, and is operably connected to receive signal level measurements from the signal level measurement circuit. The processing circuit is operable to obtain diagnostic data relating to the second set of broadband RF signals. The processing circuit is further operable to cause communication of information representative of the signal level measurements and the diagnostic data in human-perceptible form.

First and second test devices communicatively coupled to a hybrid fiber / coax (HFC) plant are operative for transmitting data packets of a simulated Voice over Internet Protocol (VoIP) telephone call therebetween without the use of a physical telephone at either test device. Each test device is operative for analyzing data packets received or transmitted thereby and for transmitting its analysis of the received or transmitted data packets to a call management system and a test controller.

A deployable Voice over Internet Protocol (VoIP) system for providing VoIP communication to communication devices via a data network is provided. Wireless telephonic devices are used for wireless communication with an extended range antenna coupled with a base station. VoIP adapters, coupled with the base station, provide telephonic lines upon obtaining network connectivity. A power supply provides power to the base station and the VoIP adapters with the base station, VoIP adapters and power supply being fixedly housed in a portable case.

An apparatus and method are provided for estimating the grade of service (52) and offered traffic (51) for voice over internet protocol calls at a gateway (2) bridging calls between a public switched telephone network (3) and an internet protocol network (4), the gateway (2) having a dial-control management information base. The method comprises the steps of periodically polling the dial-control management information base for dial peer traffic statistics (44), storing the polled data, estimating the carried traffic using the polled data (501), estimating the grade of service (52) by utilizing the Erlang-B formula in an inverse manner (502), operating on the estimated carried traffic obtained in the first estimating step (501), and estimating the offered traffic (51) using the estimated values for the carried traffic and the grade of service (52) obtained in the previous estimation steps (503). In a second embodiment of the invention, a system (FIG. 4) utilizing the method, continuously monitors the grade of service (52) and offered traffic (51) at gateways (2) in an internet protocol telecom network supporting voice over internet protocol. An enhancement of the system further comprises a world wide web interface (46) for generating monitoring reports.

A method and system for providing Voice over Internet Protocol (VoIP) telephony products to purchasers is disclosed herein. Purchasers of the VoIP products are provided VoIP telephony calling capability via an ITSP VoIP system. The method and system include providing an online tool to the purchasers where the online tool enables online purchasing of each of the VoIP telephony products. The method and system further include detecting an order for a VoIP product(s) by a purchaser via the online tool where the order includes data for provisioning the VoIP product in the ITSP network, forwarding the order to the ITSP network and receiving an order confirmation from the ITSP network. The order confirmation indicates successful purchasing and provisioning of the VoIP product. The purchaser may be a reseller who resells the VoIP products to an end customer. Alternatively, the purchaser may be the end customer.