372 results about "VoIP gateway" patented technology

A method and system for secure Voice over Internet Protocol (IP) (VoIP) communications. The method and system provide secure VoIP voice calls, video, Instant Messaging (IM), Short Message Services (SMS), or Peer-to-Peer (P2P) communications while maintaining privacy over the Internet and other communications networks such as the pubic switched telephone network (PSTN) to and from any network device through a virtual private network infrastructure interconnecting private VoIP network devices. The method and system allow a network device to function as an IP private branch exchange (PBX) or a private VoIP gateway and provide and control VoIP voice communications without using other public or private VoIP gateways or VoIP servers or devices on a communications network such as the PSTN or the Internet.

One or more methods and systems of effectively transmitting voice and voice band data from one node to another are presented. In one embodiment, the system comprises an NTP time server generating absolute times to computing devices such as residential voice over internet protocol (VoIP) gateways. The NTP time server generates absolute times in response to NTP time requests made by one or more computing devices such as residential VoIP gateways. In one embodiment, the method comprises determining an adequate rate for requesting absolute times from an NTP server, making periodic requests to the NTP server, obtaining the absolute times from the NTP server, and generating an adjustment parameter for use by a computing device such as a residential VoIP gateway.

Systems and methods for creating a Universal Media Server (UMS) and associated Digital Media Renders (DMRs) are disclosed. This system provides functions such as in-home wired or wireless network for media distribution with PVR functions from a distributed archive while at the same time including functions such as an answering machine, voice recorder, voice over IP gateway, firewall, NAT, DHCP, and security monitoring. A variety of advanced features and functions are also disclosed.

The present invention is a system, apparatus and method for voice over Internet protocol telephone (VoIP) calling using enhanced SS7 signaling packets and may include localized time slot interchanging. A system embodiment of the invention includes originating and terminating VoIP gateway switches in communication with the public switched telephone network (PSTN) and also in communication with an IP-based packet network, such as the Internet, for transmitting packets. The VoIP gateway switches are configured to exchange enhanced SS7 signaling packets over the IP-based packet network for setting up and tearing down VoIP telephone calls. A method of placing a VoIP telephone call in accordance with the present invention includes initiating a telephone call to a destination and connecting the telephone call to an originating VoIP gateway switch using enhanced SS7 signaling packets. The method also includes determining a preferred route from the originating VoIP gateway switch to the destination through an IP-based packet network and through a terminating VoIP gateway switch nearest said destination, and setting up two-way communication through the preferred route using the IP-based packet network using enhanced SS7 signaling packets.

The invention provides a way to fallback to a PSTN call at any time during a VoIP call when Quality of Service in a VoIP network falls below some acceptable level. The PSTN fallback calls can be retrieved “midcall” and rerouted back over the VoIP network. This provides optimal utilization of VoIP without sacrificing the quality of the call connection. Calls are cheaper because PSTN fallback calls are only established temporarily for the amount of time that the QoS problem exists on the VoIP network. Call fallback is conducted in a VoIP gateway by first receiving an incoming call. A Voice over IP (VoIP) call is established for the incoming call over the VoIP network. VoIP packets are encoded from the voice signals in the incoming call and sent over the VoIP network. Quality of service of the VoIP network is monitored during the VoIP call and a fallback call is setup over a PSTN network at any time during the VoIP call when the monitored quality of service of the VoIP network degrades. For a time the voice signals from the incoming call are cross connected to both the output for the fallback call and the output for the VoIP call. When a destination gateway starts receiving the voice signals from the fallback call, the VoIP call is dropped. The quality of service on the VoIP network continues to be monitored during the fallback call. A new VoIP call will be reestablished over the VoIP network during the fallback call when the quality of service of the VoIP network improves. Voice from the incoming call is for a time again cross connected to both the fallback call and the new VoIP call. After the destination gateway starts receiving audio packets again over the new VoIP call, the PSTN fallback call is terminated.

A method and system for communicating information includes evaluation regarding routing of calls performed at a call control point in an ITSP and / or wireless network. A PIC identity associated with another party's preferences is acquired and sent to the call control point. The PIC identity identifies the type of the carrier network (e.g., circuit switched or VoIP) and is used to make further routing decisions. If a circuit-switched carrier is identified, the IP homing leg is terminated at the voice gateway in the recipient's HPLMN and information is sent over to the GMSC where normal, circuit-switched routing procedures. If a VoIP carrier is identified then the called user's roaming number is retrieved from the HLR, and the call is further routed directly over the IP domain towards a voice gateway at the visited network minimizing the number of transcodings. Additional transcoding steps can be avoided if a single encoding is agreed upon according to “tandem free operation” (TFO). Combining inband signaling through the telephony exchanges and out-of-band signaling in the IP network it is possible to achieve, for a mobile subscriber, one encoding and one decoding of a voice.

In conjunction with a data communication network (53) carrying multiple telephony signals and allowing for connection of telephone sets (17), a system and method in which two external feeders (55a, 55b) connect to the data network (53) at two distinct points via two distinct devices. The data network can be based on dedicated wiring or can use existing in premises medium such as telephone, powerlines or CATV wiring. In the latter case, the wiring can still carry the original service for which it was installed. The external telephone connections can be based on the traditional PSTN, CATV network, cellular telephone network or any other telephone service provider network, using specific adapter for any medium used. In the case of connection to a POTS telephone signal, VoIP gateway (or any other converter) is required.

The VoIP gateway 50a in accordance with the present invention serves to send a connection request to either the IP network 2 or the PSTN 1 on the basis of the information stored in the VOIP gateway 50a and the destination telephone number contained in the connection request as sent from the telephone terminal 70a. Also, even if a connection request is decided to be sent to the IP network 2, the VOIP gateway 50a sends the connection request to the PSTN 1 when the IP network state judgment unit 56 judges that the communication through the IP network 2 is impossible. Furthermore, if a destination telephone number includes predetermined identification information, the VOIP gateway 50a can forcibly send the connection request to the PSTN 1.

An integrated voice gateway system for use within a company which can route a voice telephone call between parties at two different locations over an IP network or over the PST NETWORK. The system can route a voice telephone call from a first location within the system to a second location within the system via the IP network, and then from the second location to a third location via the PST NETWORK. The integrated voice gateway system includes a gateway server which serves as an intranet / Internet telephony gateway. The gateway server routes intra-company voice or facsimile (fax) calls, over the company's intranet or the public Internet. The gateway server provides an alternate voice network to the PST NETWORK for a company. This alternate network is provided at a much lower cost. The gateway server is a combination of hardware and software components which reside on a PC server platform. The gateway server is coupled to a customer premise telephone system, i.e. a PBX via a T1 or E1 trunk for larger systems, or an analog trunk for smaller systems. The gateway server is coupled to the company's intranet via industry standard connections. The gateway servers in a multi-site company are coupled together via the company's intranet or wide area network (WAN) into a gateway network. The gateway server uses PBX call status links to provide many unique and useful features which are otherwise unavailable. The gateway server uses T1 inband ANI, PRI, QSIG or industry standard CTI applications programming interfaces (API) and works with any PBX which supports any of these call status links. The gateway server is equipped with a database of user and gateway objects and attributes, and provides many unique features including caller's name based on caller phone number, address translation, gateway network routing information, user authentication, etc. This database can be integrated with industry standard enterprise directory services systems including any directory which supports the Lightweight Directory Access Protocol (X.500) (LDAP) interface.

Systems and method for providing voice-activated presence information are disclosed. According to one embodiment, the system includes a voice gateway in communication with a presence server. The presence server is for determining presence information of an individual. The voice gateway is for voicing the presence information to a caller after receiving the presence information from the presence server. The voice gateway may also place an outgoing communication to the individual. The presence server may determine whether the individual is present and available on a communication network.

A communication-status notification apparatus enabling a subscriber to observe various kinds of communication status in a network easily via the subscriber's own terminal in a communication system. The apparatus includes a request analysis section for discriminating whether or not voice data received by gateway equipment from a subscriber terminal contains a request on monitoring / controlling or notifying of a communication status in the network and for analyzing the content of the request when contains, a communication-status monitor / control section for monitoring / controlling the communication status responsive to the content of the request analyzed by the request analysis section based on a processing status of the voice data in the gateway equipment, and a communication-status notification section for notifying the subscriber terminal of the communication status monitored / controlled by the communication-status monitor / control section via the gateway equipment responsive to the content of the request analyzed by the request analysis section. The apparatus is useful when applied to VoIP gateway equipment or the like used for a VoIP communication system.

A method is provided that includes receiving a request from a communication device to establish a communication session with a mobile station, the mobile station being operable to roam between a private and a public network. The mobile station is signaled via a cellular data network that a call is being initiated for the mobile station. Signaling information may be exchanged with a voice gateway such that one or more voice circuits are established. A signaling pathway may be established between an Internet protocol private branch exchange (IP PBX) and the mobile station via the cellular data network. The establishment of the signaling pathway is substantially concurrent with the establishment of one or more of the voice circuits. One or more features associated with a private network are delivered to the mobile station during the communication session as an end user moves between the public and private networks.

A method of adaptively routing voice packets in an IP network receiving a request for a new call at a voice gateway, determining a destination gateway for the new call, determining an availability of at least one route for routing the new call across the IP network to the destination gateway, and deciding whether to admit the new call into the IP network based on whether a route is available over which voice packets for the new call can be routed. A call admission decision is made based on whether a direct, single-path route or a two-path route is available, where each path is an LSP tunnel whose availability is determined using a token bucket technique. If no route is available, the voice gateway does not admit the call into the IP network.

A VoIP routing method and system and a program storage device, readable by a machine, tangibly embodying a program of instructions executable by the machine to perform the VoIP routing method includes: receiving information as to whether a failure has occurred in VoIP gateways of VoIP system; and establishing a routing path by selecting a VoIP gateway where a failure has not occurred to bypass a VoIP gateway where a failure has occurred in accordance with the received information.

A method and system for creating a virtual roaming solution for a MSISDN using a softphone over an IP network. The system involves (i) implementation of a novel virtual mobile network (VMN) comprising virtual visitor location register (vVLR), virtual home location register (vHLR) and virtual multiple switching centre (vMSC) on an IP server responsible for managing IP call traffic administration, and (ii) implementation of a novel mobile to internet gateway (MIG) comprising an VoIP gateway for diverting call traffic from the mobile network to the IP network, and an IP server with vMSC functionality to translate routing information from the VMN to GSM network so as to appear to the GSM network as a traditional mobile operator. The system dynamically registers the subscriber to the IP network, and provides valid routing information to the MSC (Mobile Switching Centre) or public telephone switch to route the call over to the NGN (next generation network) operator in the IP space.

The present invention relates to a system and method for wireless telecommunication in a packet-based network comprising a Software Radio Port (SRP) which functions as a radio base station and a VoIP gateway to interconnect the wireless network with the VoIP packet network. Together with a Network Server Platform (NSP) and VoIP call-server, the SRP combines mobile call processing signaling with the VoIP call signaling to establish calls between the mobile and VoIP device or between mobiles. The SRP establishes the voice path to the mobile station over the air and the RTP media path to a party over a packet network for a call. These two paths are interconnected at the SRP so that an end-to-end voice path is established.

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.

Embodiments connect a call in which at least one party is a VoIP call party and monitoring resulting VoIP signals for unauthorized call activity, such as three-way call activity. The monitoring may include monitoring the call for suspend and / or resume events to detect the unauthorized call activity, the suspend and resume events may be generated by a telephone system and passed into a VoIP system associated with the VoIP call party. The monitoring may be carried out by an agent disposed between a VoIP gateway and the VoIP call party or by the VoIP gateway itself.

A Bluetooth terminal, relay, and system for determining a network configuration automatically and for transferring to an optimal waiting state. A Bluetooth terminal is provided with a profile functioning as a headset for communicating a terminal on the partner side via a voice gateway terminal, includes an input key for accepting an input operation including a transmitting operation, a receiving operation, and a call termination operation. A waiting state selecting member is provided for selecting a waiting state to be transferred upon call termination from a first waiting state in which an SCO link is disconnected and an ACL link is maintained, a second waiting state in which the SCO link and the ACL link are both disconnected. The waiting state selecting member selects the first waiting state after communication without input key operation being terminated, and selects the second waiting state after communication with input key operation being terminated.

A voice gateway (18) in a telecommunications network (1) includes a plurality of telephony port modules (102). Each telephony port module (102) receives telephony voice signals from a public switched telephony network (13). Each telephony port module (102) includes one or more digital signal processors (110) that perform one or more processing functions on the telephony voice signals. A particular telephony port module (102) may receive a telephony voice signal and use its associated digital signal processor (110) to process the received telephony voice signal or transfer the received telephony voice signal for processing to any digital signal processor (110) on any telephony port module (102). Telephony signals may also be transferred for processing to digital signal processors (110) on another voice gateway (18) in a voice gateway system.

An IP converged system includes a VoIP ALG module and a policer module. The VoIP ALG module acquires dynamically changing RTP IP / port information of a packet by parsing a VoIP SIP message, and transmits the RTP IP / port information to the policer module. The policer module sets IP / port, which provides a real-time data service, by referring to the information from the VoIP ALG module, and discriminatively sets a packet processing condition for a non-real-time data service and a packet processing condition for the real-time data service. The VoIP ALG module and the policer module share RTP IP / port information, dynamically determined by the negotiation between VoIP gateways or VoIP terminals, in call setup / release, so that the policer can discriminately drop or mark VoIP packets by referring to the RTP IP / port information.

A network system for enabling voice interaction between communications-center applications and human agents remote from the center has a primary server connected to the network the server controlling at least one routing point used by the center, a secondary server connected to the network the secondary server for generating and serving voice extensible markup language, a voice gateway associated with the secondary server, the gateway for executing voice extensible markup language and recognizing speech input, and a software platform based in the primary server and distributed in part as a server application to the secondary server, the software suite functioning as a data transformation interface between the center applications and the gateway. In a preferred use agents and applications communicate bi-directionally using VXML.