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

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.

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

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.

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 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.

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.

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, 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.

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 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.

A method, for example, is described in the present application. Such a method can include detecting or establishing a circuit-switched call. The method can also include establishing a voice-over-internet-protocol standby call in parallel to the circuit-switched call. The method can additionally include activating the voice-over-internet-protocol standby call to become an active voice-over-internet protocol call when a handover situation is detected. The method can further include releasing resources of the circuit-switched call when the voice-over-internet-protocol standby call becomes the active voice-over-internet protocol call.

A voice over Internet protocol (VoIP) gateway includes a foreign exchange office (FXO), a foreign exchange station (FXS), and a VoIP processor. A controller of the VoIP gateway sets the VoIP gateway to either a TANDEM (trunk and ENM (ear and mouth)) mode or a standalone mode. In the TANDEM mode, the VoIP gateway transmits an incoming call from the VoIP processor to the FXO and an outgoing call from the FXS to the VoIP processor. In the standalone mode, the VoIP gateway transmits the incoming call from the VoIP processor to the FXS and the outgoing call from the FXS to the VoIP.

An example of a solution provided here comprises providing a measurement process including: (a) transmitting a test stream over a transmission path; and (b) measuring a quality-of-service indicator for a real-time communication application based on the transmitting; utilizing the measurement process, in continuously sampling a plurality of transmission paths in the real-time communication application's production environment; collecting data from the measurement process; comparing measured values to a threshold value; outputting a representation of compliance or non-compliance with the threshold value; and outputting a trend report based on the data; whereby the real-time communication application may be managed with reference to the threshold value. Such a solution may be selected for a Voice-over-Internet-Protocol application, a video conference application, or a speech-recognition application, to give some non-exclusive examples. One such example comprises measuring a speech-quality indicator for a Voice-over-Internet-Protocol application.

A system for interfacing different types of network communications. In one embodiment, the system includes one or more gateways capable of converting messages from messages that are adapted for a first type of network and/or destination device to messages that are adapted for a second type of network and/or destination device. The second type of network and/or destination device includes a first Voice Over Internet Protocol (VOIP) communications device. A message-analysis module is adapted to employ an address associated with the message to selectively forward the message to one or more of the one or more gateways and/or to a destination device. In a more specific embodiment, the first type of network includes a packet-switched network in communication with the first VOIP phone. The destination device includes a mobile phone in communication with a cellular network. In the specific embodiment, the first type of destination device includes a user option to selectively transition a text-based communications session to a voice-based communications session.

The present invention relates to a healthcare consultation system that includes a knowledge database, a decision support server, an interactive help system and a connection, wherein the connection comprises a remote assistance unit. The present invention also relates to a wearable computer capable of two-way audio/visual conferencing to provide 24/7 real-time healthcare consultation. The present invention further relates to a method for providing training and help services to a healthcare provider, a healthcare manufacturer or a healthcare user by providing a resource center that comprises a knowledge database, a decision support server and an interactive help system, building the knowledge database by compiling information, analyzing the information in the knowledge database to develop the decision support server, establishing a connection between the interactive help system and the healthcare provider, healthcare manufacturer and healthcare user, accessing the decision support server, and delivering education, training and consultation services via the interactive help system, wherein the connection comprises a remote assistance unit. According to a preferred embodiment, the remote assistance unit comprises a heads-up display, a mini personal computer, a wireless local area network, a headset, a camera, voice over internet protocol technology, a microphone, and/or glasses or head gear.