Abstract: In one embodiment, a scheme is disclosed for managing call routing in a network environment including a circuit-switched (CS) network and an IP multimedia subsystem (IMS) network. When a call is originated by a user equipment (UE) device in the CS network, call information associated with the call is provided to a call continuity control function (CCCF) network node disposed in the IMS network. At the CCCF node, a pool of E.164 numbers are maintained as IP multimedia routing numbers (IMRNs) which are mapped to or otherwise associated with called party numbers. The CCCF node dynamically allocates a select IMRN with respect to a called party number received from the UE device and returns it to the UE device. The dynamically allocated IMRN is then utilized for routing the call towards the called party.

Abstract: Methods and apparatus to provide extended voice over Internet protocol (VoIP) services are disclosed. An example residential gateway comprises a radio frequency (RF) detector to detect a presence of a cellular communication device, a service controller to enable a VoIP communication device when the cellular communication device is present, wherein the VoIP communication device and the cellular communication device are different devices.

Abstract: When a transmission request for new data is issued under a situation where a plurality of data are transmitted to a specific service area from a plurality of service areas, a feeder link station searches an available frequency to be allocated to the new data in a down-link frequency band of the specific service area, and when any of data communications is completed, allocates a down-link frequency band used in the data communication as a down-link frequency band for the new data. A satellite repeater arranges the new data on the down-link frequency band right after the data communication is completed following a relay control by the feeder link station.

Abstract: Disclosed are systems and methods that provide centralized or nodal inmate management and telephone call processing capabilities to controlled environment facilities. An inmate management and call processing system serves a plurality of facilities and includes an inmate information database. The database is shared across the facilities and contains inmate records that may be accessed and modified by each facility as the inmate is transferred among those facilities. Each inmate record may include, among other information, contact information of third parties whom may be notified of the inmate's arrest and/or subsequent transfers of the inmate to different facilities. The inmate management and telephone call processing system may use third party contact information to establish accounts used to charge calls or transactions made by an inmate while residing at the facility.

Abstract: A communication apparatus provides services to a communication device. The communication apparatus comprises a private base station, such as a Home Node B, for communicating with a communication device authorized to use the private base station and a gateway, such as a Home Node B gateway, communicatively coupled to the private base station for providing access to an IP Multimedia Subsystem, IMS, network and at least one other communication network, such as a Circuit Switched, CS, network. The private base station is arranged to select a route for providing a service to the communication device through the private base station and gateway, the selected route being one of a route between the communication device and the IMS network and a route between the communication device and the at least one other communication network. The private base station is arranged to select a route based on the service to be provided.

Abstract: Communication networks and methods are disclosed that provide call delivery to a dual mode wireless device served by a wireless data network (e.g., WiFi/IMS network) and a wireless voice network (e.g., cellular network). A call control function in the wireless data network is adapted to receive a call to the dual mode wireless device. The call control function transmits a ping message to the dual mode wireless device over the wireless data network to determine if the device is available over the wireless data network for the call. If the call control function receives a response to the ping message from the device, then the call control function attempts to set up the call over the wireless data network. If the call control function does not receive a response to the ping message, then the call control function may attempt to set up the call over the wireless voice network.

Abstract: A method which allows standard telephone users to audio conference with video conferencing participants over IP networks in a private secure environment. A dial-out is performed from one or more conference client terminals bridging audio between the Internet and the PSTN networks. The process uses a mixed mode hybrid network architecture for call set-up, initialization and teardown including the method to mix audio at the desktop terminal instead of in a general purpose server as in the prior art. The method conferences video and audio between multiple clients and include audio from a standard telephone network within the conference. A virtual private network connects all of the IP clients together including the voice over IP server used to transcode the proprietary audio into the H.323 standard for transport into the telephony network.

Abstract: When an IP terminal on the Internet side transmits a name resolution request, which requests to resolve the FQDN of a public server connected to a router, to a SIP/DNS server in which SIP and DNS cooperate to manage the status of the data line of the router and when the data communication line status of the router is a disconnect status, a data communication line connection request instruction is transmitted to the router. The router connects the data communication line and notifies the SIP/DNS server about the result. The SIP/DNS server transmits the IP address of the router to the IP terminal as a response.

Abstract: Session control for emergency services sessions (E-sessions) originating in a circuit-switched subsystem (CS) or a multimedia subsystem (MS) is provided by the visited MS and is anchored at a domain transfer function (DTF), which is a service provided by a visited MS that is currently serving the user. As such, all call signaling for the E-session is passed through the DTF. The DTF enables efficient and effective subsystem transfers of the E-session between the CS and MS, while maintaining the E-session between the user element and the PSAP. The DTF may also provide to the PSAP location information bearing on the location of the user element when the E-session is established as well as provide updated location information when transfers between the CS and MS occur.

Abstract: A network node is provided. The network node includes a component configured to use a value in a Session Initiation Protocol message. The value indicates a supported transport addressing scheme and is used to determine whether to route a related communication session through a transport addressing scheme translation component.

Abstract: Apparatus for establishing a packet-based connection between a video display equipment (such as a set-top box) having an analog modem and a remote computer. The translation apparatus receives a connection initiation request from the video display apparatus and responsive to the connection initiation request establishes a modem connection with the video display apparatus. The translation apparatus further receives communication data from the video display apparatus via the modem connection and transmits the communication data to the remote computing apparatus via a packet-based network. The translation apparatus can also receive communication data from the remote computer and relay the same to the video display apparatus. The connection can be established upon powering up of the video display apparatus, upon loading of an Interactive TV application or upon other pre-determined event. The connection can be maintained “always-on” or substantially “always-on”.

Abstract: A method, application server, and Serving Call/Session Control Function (S-CSCF) for allocating a SIP Application Server to a subscriber in an IP Multimedia Subsystem. A Home Subscriber Server (HSS) identifies for the subscriber, a set of provisioned initial filter criteria, which contains at least one generic SIP Application Server identity. The HSS sends the filter criteria to an S-CSCF allocated to the subscriber. The S-CSCF resolves the generic SIP Application Server identity into a plurality of application server addresses, with one of the addresses being allocated to the subscriber for use in provisioning a service to the subscriber. The S-CSCF caches the address allocated to the subscriber for subsequent use.

Abstract: Systems and methods are disclosed that provide femto-based wireless coverage in a communication network. This can involve providing an interworking function that communications with a femto base station or femto cell to provide connectivity to the core network. The interworking function can provide service and mobility management where a femto cell (such as a home node B (HNB)) is served concurrently by an IMS core and a legacy core. The interworking function can also provide service through a femto cell to a variety of mobile nodes such as legacy devices and IMS capable devices. The interworking function also provides the ability for handoffs to occur between the core networks and between a femto cell and a macro cell.

Abstract: A method of initiating outbound calls through a telephony interface. The method accepts an inbound call from a user, authenticates the inbound call against a predetermined user identity, accesses an address book associated with the user identity, receives an spoken request from the user, selects a telephone record in the address book associated with the spoken request, initiates an outbound call to the telephone record; and connects the inbound call with the outbound call responsive to a connection established on the outbound call. In an alternative embodiment the method accepts an inbound call, authenticates the inbound call against a predetermined user identity, receives a dual-tone frequency-modulated input representing a desired outbound call target, initiates an outbound call to the outbound call target, and connects the inbound call with the outbound call responsive to a connection established on the outbound call. A corresponding computer system for each embodiment is also provided.

Abstract: A method and apparatus is provided for communication between a first terminal capable of using both a Circuit Switched (CS) call and an Internet Protocol Multimedia Subsystem (IMS) session, and a second terminal capable of using the IMS session, in a communication system supporting a Combined CS call and IMS session (CSI service). If the second terminal sends a Session Initiation Protocol (SIP) request (INVITE) message to originate a voice service or a multimedia service including the voice service with the first terminal, a CSI Application Server (AS) of an IMS domain managing the first terminal separates a voice service-related component included in the request message from a multimedia service-related component, and generates and sends first and second request messages to the first terminal.

Abstract: The present invention is related to methods and systems for reducing or canceling echoes during fax communications. In an embodiment, during a fax call, a first fax signal from a first fax device intended for a second fax device is detected. Based at least in part on detecting the first fax signal from the first fax device or on the termination of the first fax signal, an echo audio packet is muted, wherein the echo audio packet is at least partly the result of the first fax signal being transmitted over a network, the network including at least a data network.

Abstract: A method of configuring a port on a network device includes receiving a frame of information, determining that the frame includes maximum transmission unit (MTU) information, and configuring a prioritized receive queue on the port with an MTU size based on the MTU information. The MTU size is different than another MTU size of another prioritized receive queue on the port. A network interface includes a port, a de-multiplexer coupled to an output or the port, and prioritized receive queues that each have an MTTU size. A first prioritized receive queue has an MTU size different than a second prioritized receive queue.

Abstract: An IP (Internet Protocol) session for a mobile node is carried out through the use of a virtual care-of address. A foreign agent sends an agent advertisement to the mobile node to allow the mobile node to choose from a list of IP addresses of the foreign agent. To perform virtual care-of address services, the foreign agent ties the virtual care-of address to a mobile node so that an intelligent and dynamic selection of tunnels to be used for the IP session can occur. Therefore, traffic for an IP session is not limited to transmission over the single particular tunnel that corresponds to an IP address initially selected by the mobile node. Rather, the virtual care-of address shifts the tunneling decision from the mobile node to the foreign agent. Supporting multiple tunnels between home agent and foreign agent allows resilience, redundancy, and service-level differentiation to mobile node traffic without involving the mobile node in the process.

Abstract: Providing an interface device interfacing between an access network and a back-bone network comprises: providing a management device, receiving data readings from a transceiver device indicative of the current or recent line rate of a DSL connection of the access network, and providing the interface device to restrict data flowing to the DSL connection to a new maximum rate selected in accordance with the received data readings from the transceiver device indicative of the current or recent line rate of the DSL connection. When the new maximum rate is greater than the existing maximum rate for the DSL connection at which the interface device is currently provided, the interface device is provisioned (or re-provisioned) only after a delay period. The delay period depends upon the increase amount of in the new maximum rate compared to the existing maximum rate so that the greater the increase, the shorter the delay period.

Abstract: A system and a method for releasing quality of service (QoS) resources in a mobile communication network enable QoS resources allocated to a mobile terminal to be released, if, after the mobile terminal is allocated with resources for QoS through an Evolution-Data Optimized (EVDO) Revision A (rA) network to receive data serves, the mobile terminal escapes from the EVDO rA network without first performing a resource release request to the EVDO rA network. The system and method may be implemented in the mobile terminal, the access network, the text message server, or any combination thereof, for releasing the allocated QoS resources without requiring the mobile terminal to re-connect to the EVDO rA network.

Abstract: A system for determining a requirement in a converged communication system is provided. The system includes a network analyzer 108 that is operable to: (a) determine a packet-switched port usage for at least one of (i) a set of packet-switched stations, (ii) a set of packet-switched trunks, and (iii) a set of packet-switched communications, and a circuit-switched port usage for at least one of (iv) a set of circuit-switched stations, (v) a set of circuit-switched trunks, and (vi) a set of circuit-switched communications in at least a portion of an enterprise network; and (b) based on the determined packet-switched and circuit-switched port usages, determine a functional component requirement to comply with a selected communication specification.

Abstract: A method, a session managing node and an arrangement for providing a network-service address of at least one network-service of an IMS network (401) of a first operator to a third party service provider (403), wherein the third party service provider has no business agreement with the first operator. A first application (404) of a mobile station is communicating with a second application (405) via a first network (406), resulting in the set-up of an IMS-session, between the mobile station and the third party service provider. A network-service address of at least one network-service, provided by the IMS network of the first operator, is inserted into a SIP message of the IMS-session in a session managing node (600) of the IMS network of the first operator and delivered to the third party service provider, where it is forwarded to the second application. If required, the second application may access the one or more network-services via the first network, using the retrieved network-address.

Abstract: A clearinghouse is used to determine caller location for VoIP calls. The clearinghouse maintains (i) a first correlation between a first IP address and a first ISP and (ii) a second correlation between a second IP address and a second ISP. The clearinghouse receives the first IP address from a VoIP service provider (VSP). The first IP address is a source address of a request from a VoIP telephone to set up a call, received by the VSP from the first ISP. The clearinghouse references the first correlation and queries the first ISP with the first IP address. The clearinghouse then receives from the first ISP a location associated with the first IP address. The clearinghouse transmits this location to an entity such as the VSP or, for a 911 call, perhaps a PSAP. The clearinghouse may select the PSAP based at least in part on the location.

Abstract: A hybrid switch module configured on a common support structure providing networks the ability to automatically reconfigure point-to-point links, per software controls, to optimize network topology by combining a network switching device with a physical layer switch using switching logic. The hybrid switch module is an integrated unit that can operate at a physical layer, routing signals between dedicated ports, and operate as a packet switch for non-dedicated ports.

Abstract: A multi-core die is provided that allows packets to be communicated across the die using resources of a packet switched network and a circuit switched network.

Abstract: A method, system, and device for converting session control signaling are provided. The method includes at least one of the following steps: a first adaptation unit converts a session control signaling message of a circuit switched (CS) session leg into a session control signaling message of an IP multimedia subsystem (IMS) session leg; or the first adaptation unit converts a session control signaling message of an IMS session leg into a session control signaling message of a CS session leg. In embodiments of the present invention, an adaptation unit intercepts a CS signaling message after an inter-mobile switching center (MSC) handover, and converts CS signaling messages to IMS signaling messages and vice versa, so that the IMS domain may continue to control a session after the inter-MSC handover.

Abstract: A method for improving a random access procedure for a mobile device of a wireless communication system is disclosed. The method includes initiating the random access procedure, starting a time alignment timer of the mobile device when receiving a time alignment message transmitted by a base station of the wireless communication system, and controlling the random access procedure according to expiry of the time alignment timer, wherein the time alignment timer of the mobile device is used for determining a synchronization state between the mobile device and the base station and the time alignment message is utilized by the base station to update a timing advance for the mobile device.

Abstract: A device receives an emergency call invite generated by a user device associated with an Internet protocol (IP) multimedia subsystem (IMS) media gateway, and forwards the emergency call invite to a location resource function/route determination function (LRF/RDF) device for determining public safety answering point (PSAP) routing information. The device receives, from the LRF/RDF device, the PSAP routing information based on the emergency call invite, and routes the emergency call invite to a particular PSAP based on the PSAP routing information. If the emergency call invite is routed to a legacy PSAP, the emergency call is routed to the legacy PSAP via legacy mobile switching center (MSC) direct trunks. If the emergency call invite is routed to an IP PSAP, the emergency call is routed to the IP PSAP via an IP connection.

Abstract: An IP telephony system allows users of the IP telephony system to register extension telephony devices with the IP telephony system. An extension telephony device is one that is provided with service by a separate telephony service provider. Once an extension telephony device is registered, a user can obtain communications services from the IP telephony system using the extension telephony device. A extension telephony device may be tied to a user's main telephony services account with the IP telephony system such that when the user obtains communications services from the IP telephony system using an extension telephony device, the user will be billed for those communications services through the user's main account.

Abstract: According to one aspect of the teachings presented herein, a “smart” phone or other wireless communication device includes a “modem” processor configured to support cellular communication network signaling and an “applications” processor configured to host call control and/or user applications, such as by providing a desired Operating System (OS) for hosting such applications. In at least one embodiment of a wireless communication device contemplated herein, the modem processor implements a cellular network signaling controller, a Generic Access Network (GAN) manager, and a privately routable, first Internet Protocol (IP) stack. Further, the applications processor implements a publicly routable, second IP stack that implements routing, e.g., a Network Address Translation (NAT) routing, for routing GAN traffic to and from the first IP stack on a local IP link bridging the first and second IP stacks.

Abstract: A system, terminating node, method, and computer program product enable an originating node in a public network to communicate with a terminating node in a private network by sending a non-IP-based message from the originating node to the terminating node, thereby causing the terminating node to initiate IP-based communication with the originating node. Because a client behind a NAT and/or firewall in a private network will generally be able to connect and establish a TCP/IP session to a server in the public network, this improves the ability of the originating node to communicate with the terminating node whenever desired. Once the IP-based communication session is established, the two nodes can exchange data in both directions until the connection between them is closed by either node. As such, unmodified client and server TCP/IP applications are able to traverse a NAT/firewall, without the need for configuration of an intermediary node.

Abstract: A system including a first network configured to receive IP device data from an IP device and to provide one or more IP addresses of the IP device based on the IP device data; a second network comprising: a second network location database configured to store physical location information, and a second network location server configured to receive the one or more IP addresses of the IP device from the first network, and query the second network location database to determine physical location information of the IP device based on the one or more IP addresses.

Abstract: The present invention relates to a method and system for monitoring and evaluating multimedia telephonic and computer-communication interactions. The multimedia quality monitoring system may include an event manager, which may receive a notification associated with a VoIP communication between an agent and a customer and recording servers, which record data associated with the VoIP communication.

Abstract: User Equipment (UE) receiving non-voice packet switched service in an Evolved Universal Mobile Telecommunications System Terrestial Radio Access Network (eUTRAN) cell undergoes handover to a circuit switched GSM/Edge Radio Access Network (GERAN)/UTRAN cell in response to a requirement for voice service. The voice service is established in the circuit switched domain of a GERAN/UTRAN cell to ensure service continuity regardless of UE mobility that would otherwise not be possible if the voice service was established in the eUTRAN cell. A Gs-like interface between a Mobile Switching Center (MSC) of the CS Core Network and the Mobility Management Entity (MME) of the Evolved Packet Core provides an interface for page requests and location update procedures to be performed.

Abstract: A system for establishing a universal or overlapping wireless broadband network composed of a plurality of Access Points (AP). These APs may be coupled to existing copper transmission lines, and may work alone or in groups to provide wireless network access. Different types of APs may be utilized to perform various communication-functions, and these nodes may obtain power and/or communication resources from the existing copper transmission lines. Further, peripheral devices may be coupled to nodes to provide ancillary functionality to the system.

Abstract: A system for establishing a universal or overlapping wireless broadband network composed of a plurality of Access Points (AP). These APs, or nodes, may be coupled to existing copper transmission lines, and may work alone or in groups to provide wireless network access. Different types of APs may be utilized to perform various communication-functions, and these nodes may obtain power and/or communication resources from the existing copper transmission lines, and communication resources from either the existing copper transmission lines or from fiber optic transmission lines. Further, peripheral devices may be coupled to nodes to provide ancillary functionality to the system.

Abstract: A communication connection between a calling communications terminal and a further communications terminal, the connection setup being initiated through the exchange of an invite message and a number of acknowledgment messages between the calling communications terminal, the further communications terminal and a higher-ranking communication-management module, the connection modalities relevant for the calling communications terminal being agreed in a first connection initiation sequence between the higher-ranking communication-management module and an application-management module allocated to the calling communications terminal, and a second connection initiation sequence being provided for agreeing the connection modalities relevant for the further communications terminal between the higher-ranking communication-management module and an application-management module allocated to the further communications terminal.

Abstract: A method for managing services for a network service provider includes the steps of receiving a Feature Request Message from a switching node associated with a mobile station within a non-circuit switched network at a registration node. The Feature Request Message includes a code for activating a feature within the registration node enabling connections to the non-circuit switched network. A service request is received for providing a network service to the mobile station at the registration node. The service request is routed to the switching node associated with the non-circuit switched network responsive to the feature within the registration node enabling connection to the non-circuit switched network being activated.

Abstract: A user equipment (UE) implementing a message recovery method is disclosed, supporting messages of both the circuit switched (CS) domain and the packet switched (PS) domain. The UE roams in a wireless telecommunications system and performs a call setup procedure. When the UE receives an NAS message comprising a domain identifier and a protocol discriminator, the UE determines whether the domain identifier and the protocol discriminator are of the same domain. If the domain identifier and the protocol discriminator are of the same domain, the UE proceeds with processes corresponding to the NAS message. If the domain identifier and the protocol discriminator are of different domains, the NAS message is determined to be an error message, and the UE modifies the domain identifier of the NAS message to generate a recovered NAS message, and determines whether the recovered NAS message is compatible to the call setup procedure.

Abstract: A smart detection and recovery method is implemented in a terminal for detecting statuses of a host. The method comprises steps of: receiving a datagram; determining if the datagram complies with a predetermined format? If “YES” then proceed; determining if a MAC address contained in the datagram is the same with a currently used MAC address pre-stored in the terminal? If “YES” then proceed; determining if an IP address contained in the datagram is the same with a currently used IP address pre-stored in the terminal? If “NO” then re-connect or guide the terminal to the IP address contained in the datagram, if “YES” then handle the datagram by normal operations. Therefore, changes of the IP address of the host can be detected as early as possible, such that the terminal can be re-connected or guided to the host and services of the host can recovered as soon as possible.

Abstract: The present invention proposes a solution for providing IMS services and in particular mid-call services to users having circuit-switched controlled terminals and being not adapted to provide IMS services to the users. In particular, it is proposed to introduce a new node type called Mobile Access Gateway Control Function (MAGCF). This new node combines the logical functionality of a cellular switching center and the logical functionality of packet-based logic. The invention discusses a concept of the MAGCF handling mid-calls, which comprises identification of the received mid-call request, generating in accordance to the identified mid-call a corresponding message, tracking the status of the performed mid-calls.

Abstract: A voice communication system for achieving a public switch telephone network (PSTN) communication and an internet communication is provided. The system comprises a host, a control device and a telephone. The invention features advantages of both the PSTN and the internet, and further covers the natural shortcomings of each other.

Abstract: A cellular communication system comprises a core network, a first access network not supporting circuit switched services and a second access network supporting circuit switched services. A user equipment detects that an application using circuit switched communication is initiated and accordingly initiates a reselection from the first to the second access network. The user equipment transmits a location update message to a circuit switch mobile switching center of the core network which in response notifies a mobility management entity for the first access network that the access network reselection has been initiated. In response to the notification the mobility management entity initiates a change of a routing association from being with the first access network to being with the second access network. The invention may e.g. allow improved circuit switched fallback for a 3rd Generation Partnership Project cellular communication system comprising an access network supporting only packet switched services.

Abstract: Methods and systems are provided for using a PDSN in a wireless communication network that includes multiple IP addresses and multiple corresponding physical interfaces. Through the use of multiple IP addresses and interfaces, the throughput of the PDSN may be substantially increased. Additionally, the multiple IP addresses and interfaces may be used to provide redundancy in order to protect against software or hardware failures. According to the methods and systems of the invention, moreover, the risk of internal hard handoffs resulting from the use of a PDSN having multiple IP addresses and interfaces is eliminated or at least substantially reduced.

Abstract: An approach provides interdomain traversal to support packetized voice transmissions. A request is received and specifies a directory number for establishing a communication session from a first endpoint to a second endpoint. The first endpoint is behind a first network address translator of a first domain, and the second endpoint is within a second domain. A service provider network is accessed to determine a network address for communicating with the second endpoint based on the directory number, to determine existence of a second network address translator within the second domain, and to establish, if the network address can be determined, a media path between the first endpoint and the second endpoint based on the network address to support the communication session. An encrypted session is established with a proxy server according to a cryptographic protocol to support the media path. The proxy server resides within the second domain.

Abstract: A method for implementing a fallback procedure from a packet switched (PS) radio access technology (RAT) to a circuit switched (CS) RAT is disclosed. A paging message is received at a wireless transmit/receive unit (WTRU) via the PS RAT, the paging message including a WTRU identifier and a domain indicator, wherein the domain indicator indicates whether the paging message is related to a service on the CS RAT. If the WTRU identifier in the paging message matches a stored identifier of the receiving WTRU, then a fallback procedure to the CS RAT is performed. The WTRU responds to the paging message using a CS RAT-based identifier assigned to the WTRU.

Abstract: A method for communication in a communication system includes predefining a plurality of service delivery modes having different signaling and media path configurations in circuit-switched and packet-switched networks. A request to place a call is accepted in the circuit-switched network, and a call service to be provided to the call by a service platform in the packet-switched network is identified responsively to the request. A service delivery mode is selected from the plurality responsively to an attribute of the call indicated by the request. Signaling and media paths are established for the call via the circuit-switched and packet-switched networks in accordance with the signaling and media path configurations of the selected service delivery mode. The call service is provided from the service platform to the call using the established signaling and media paths.

Abstract: Systems and methods are described for implementing an application-level routing protocol for multiparty audio-video conferencing. In one implementation, application-level per-stream routing techniques separately control audio data and video data between conference members hosted on a network. Different audio application-level multicast (ALM) trees are generated by each member, are dynamically updated according to shortest-path-first selection of data delivery paths, and are used to send audio data to the other members of the videoconference. Likewise, different video ALM trees are generated by each member, are dynamically updated according to broadest-path-first selection of data delivery paths, and are used to send video data to the other members of the videoconference. Separate audio and video ALM trees for each member can utilize IP multicast in segments of the network in which IP multicast is enabled.

Abstract: Radiocommunication device comprising a starter software module suitable to allow a simplified radio link of the radiocommunication device with a local radiocommunication network and allow a downloading, by this simplified radio link, of an update of a use software module that is contained in the radiocommunication device and that is suitable to enable a normal radio link to be set up.

Abstract: The invention herein is fallback function communications device comprised of a microprocessor, utilized to perform a predetermined operation and processing of a predetermined input signal and then output a corresponding signal, as well as a minimum of one relay circuit having an amplifier circuit and a relay. The terminal at the first side of the relay is connected to the telephone line tip/ring terminal, while the terminal at the second side of the relay is connected to the Public Services Telephone Network (PSTN) tip/ring terminal of the microprocessor. An off-hook detection circuit is connected to the PSTN tip/ring terminal and also connected to the ring detection terminal of the said microprocessor. A dummy load circuit is connected to the tip terminal and the ring terminal of the PSTN. As such, in the event of Voice Over Internet Protocol (VOIP) system failure, the present invention is capable of automatically switching to the PSTN and thereby maintain telephone system operation.