TL;DR: In this paper, a call management computer intercepts incoming calls and controls the handling of such calls according to instructions received from the users' workstations, which are accessed via the digital data network.
Abstract: A Call Management System provides for management of calls directly by system users at their workstation computers via a digital data network such as a digital networks not controlled via the user's telephone instruments as in prior systems. A call management computer intercepts incoming calls and controls the handling of such calls according to instructions received from the users' workstations, which are accessed via the digital data network. Trunk circuits are monitored and controlled using digital signal processors to proactively identify the called party, the calling party and the call type (voice, Fax, data) and control and to monitor all calls. Each different type of call is managed differently and automatically through direct user workstation controls and/or user-generated rules to provide special treatment for designated callers. Multiple calls to user at the same time may be handled with no busy signals to callers. Only one number is needed for a user to receive voice Fax and data calls. Fax and data transmissions are automatically received.
TL;DR: In this article, the authors propose an auto-switching mechanism between two peer PBXs over either a public switched telephone network (PSTN) or a data network based on cost considerations for handling each such call and called directory numbers, monitoring quality of service (QoS) then provided through the data network and switching such calls back and forth between the PSTN and data network, as needed, in response to dynamic changes in the QoS.
Abstract: Apparatus (200), and accompanying methods for use therein, for a telephony gateway intended for use, e.g., paired use, at opposite ends of a data network connection, in conjunction with at each end, e.g., a private branch exchange (PBX) (14, 44) for automatically routing telephone calls, e.g., voice, data and facsimile, between two peer PBXs over either a public switched telephone network (PSTN) (20) or a data network (30), based on, among other aspects, cost considerations for handling each such call and called directory numbers, monitoring quality of service (QoS) then provided through the data network and switching ('auto-switching') such calls back and forth between the PSTN and the data network, as needed, in response to dynamic changes in the QoS such that the call is carried over a connection then providing a sufficient QoS. To support auto-switching, the apparatus embeds, using call independent signaling, certain call-specific information, as non-standard data, within various conventional H.323 messages that transit between the paired gateways. Furthermore, for added local redundancy, this apparatus utilizes peered border elements (430, 430') within an H.323 administrative domain.
TL;DR: 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 is described in this article.
Abstract: 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.
TL;DR: In this paper, a method and system for secure Voice over Internet Protocol (IP) (VoIP) communications is proposed. But it is not suitable for VoIP voice calls, video, Instant Messaging (IM), Short Message Services (SMS), or Peer-to-Peer (P2P) communications while maintaining privacy over the Internet.
Abstract: 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.
TL;DR: In this article, a telephone system supports communication with user devices over both a cellular radio network as well as over an Internet Protocol (IP) network, and enables roaming and active call handoff between cellular and IP domains.
Abstract: A telephone system supports communication with user devices over both a cellular radio network as well as over an Internet Protocol (IP) network, and enables roaming and active call handoff between cellular and IP domains. Components of the system interact with conventional cellular telephone systems, for example, by emulating behavior of control components, providing proxy services for conventional components, transporting cellular telephone control communication over IP connections, or by simulating cellular operating characteristics of user devices operating in an IP domain.