Abstract: Provided is an apparatus and method for providing a voice service without discontinuance in a mobile communication system. The apparatus includes a controller for receiving a system information block of a first network system or a system information block of a second network system from an interface module, and transmitting a network selection change request to the interface module to change a network system mode; and a network selector for, when the system information block of the first network system does not comprise voice switched domain information for location area update, driving a timer according to a selection of a user, changing the network system mode into the second network system, and executing location area update to thereby provide the voice service.

Abstract: A method of providing communication service, includes receiving user identification information and first communication service criteria. The user identification information identifies a user, while the first communication service criteria describe a first communication service requested by the user. The method also includes generating a first communication service order based on the first communication service criteria. The first communication service order includes the user identification information and the communication service criteria; The method further includes selecting premise equipment for the user and generating an equipment order, wherein the equipment order includes the user identification information and identifies the selected premise equipment. Additionally, the method includes transmitting the first communication service order to a first service provider and transmitting the equipment order to an equipment provider.

Abstract: Caller ID information provided coincident to a Voice over IP (VoIP) call is validated by one or more network devices. An originating router detects an offered IP address associated with a calling party node, for example, upon receiving a SIP Invite message initiated by the calling party node. The originating router determines the validity of the offered IP address and hence determines the validity of the Caller ID information based on the validity of the offered IP address. The originating router sends indicia of the validity to a terminating router, connected to a called party node, by inserting a verification bit in the SIP Invite message and sending the transformed SIP Invite message to the terminating router. Alternatively, the terminating router may query the originating router for indicia of validity. Upon receiving the indicia of validity, the terminating router may notify the called party node and so inform the called party whether it can rely on the Caller ID information.

Abstract: A method of network resource release processing is provided, which includes the following steps. After user equipment (UE) using idle mode signaling reduction (ISR) mechanism registers to two 3rd Generation Partnership Project (3GPP) communication networks, when the UE changes from the 3GPP network to a non-3GPP communication network, a serving gateway (serving GW) receives a message sent from a peer endpoint network element (NE), and deletes network resources established by the two 3GPP communication networks for the UE according to the message. A mobility management NE and a serving GW are also provided. Through the method and device of network resource release processing, the resources are released when the UE using the ISR mechanism changes from the 3GPP network to the non-3GPP communication network.

Abstract: Setup of a Voice over Internet Protocol (VoIP) call is initiated and an authentication token is received for the VoIP call that is set up, that indicates that the VoIP call is authorized. The authentication token is inserted into packets for the VoIP call. The packets, including the authentication token therein, are transmitted into an IP network. The authentication token may be placed in an IP version 6 (IPv6) flowID field.

Abstract: A Legacy User Proxy CSCF (“L-CSCF”) provides multimedia services from a SIP based network to legacy access networks and devices. The L-CSCF includes a legacy service control state machine; a P-CSCF state machine and associated protocols; a mapping between a user's legacy network address and a corresponding network entry point; a database and logic for mapping between legacy user addresses and corresponding authentication data; legacy specific QoS control mechanisms; and an interworking function to handle incompatibilities between the legacy and SIP call models. The L-CSCF may be disposed logically between the legacy network and the multimedia network to translate and process communications such that legacy network and multimedia network devices need not require extensive modifications in order to support operation.

Abstract: The systems and methods described herein include, among other things, processes for providing low latency packet transport in ad hoc networks. To this end, the systems, in certain embodiments, employ an architecture for communication nodes in MANETs, in which the delay introduced in “routing” or “relaying” the packet is reduced. In these systems invention, the data plane of the OSI stack at relay nodes is collapsed so that bits entering the node are re-energized “on-the-fly”, with only a brief pause. A transit key in each packet, inserted by the originator of the packet, is used to index into a transit table to determine one of three actions to perform—drop, keep, or retransmit the packet. A control plane determines, for each packet, whether it should be re-energized or not. The control algorithm ensures that only nodes along the path from the source to the destination re-energize (relay) the packet.

Abstract: A method of establishing a voice communication session with a multimedia access device employable in a multimedia communication system. In one embodiment, the method includes initiating a session request from a first endpoint communication device employing an instant messaging client and coupled to a packet based communication network. The method also includes processing the session request including emulating the instant messaging client for a second endpoint communication device coupled to said packet based communication network. The second endpoint communication device is a non-instant messaging based communication device. The method still further includes establishing a voice communication session between the first and second endpoint communication devices in response to the session request.

Abstract: A method of network resource release processing is provided, which includes the following steps. After user equipment (UE) using idle mode signaling reduction (ISR) mechanism registers to two 3rd Generation Partnership Project (3GPP) communication networks, when the UE changes from the 3GPP network to a non-3GPP communication network, a serving gateway (Serving GW) receives a message sent from a peer endpoint network element (NE), and deletes network resources established by the two 3GPP communication networks for the UE according to the message. A mobility management NE and a Serving GW are also provided. Through the method and device of network resource release processing, the resources are released when the UE using the ISR mechanism changes from the 3GPP network to the non-3GPP communication network.

Abstract: A Session Initiation Protocol (SIP) message adapted for use by a multi-media services provider system to form a multi-media communication path between at least a calling communication device adapted to operate using a first protocol (e.g. SIP) and at least a destination communication device adapted to operate using a second protocol, such as Integrated Services Digital Network User Part (ISUP). The SIP message includes a header region having a number of header fields, a first body region having Session Description Protocol (SDP) information related to the calling communication device and a second body region having ISUP related addressing information associated with the destination communication device.

Abstract: A system for providing third-party call control in a telecommunications environment is provided. The system comprises, a call-control mechanism for providing service logic and routing intelligence, a control application for providing service-logic description and command instruction for implementing third-party controlled call connections, a call-switching mechanism for providing an abstract state of switching matrix and for commutation of external and internal call legs and a commutation application for making and breaking call connections according to commands sent from the control application.

Abstract: A server-based architecture for mobile conferencing is presented. Session-based mobile conferencing is established by transmitting necessary server network address information through page-mode based messaging services to establish connections among the various mobile devices with the server.

Abstract: A communication system capable of improving telephone communication service by effectively integrating communications with IP phones and with mobile phones. A connection controller controls a connection interface with a mobile phone and obtains the phone number of the mobile phone being connected thereto. A connect information transmitter transmits connect information including identification information and the phone number. A connect information manager stores and manages the received connect information. When a calling IP phone sends a request to call the mobile phone, a communication controller determines based on the connect information whether the mobile phone is connected to a receiving IP phone. If yes, the communication controller makes a call to the receiving IP phone to establish communication between the calling and receiving IP phones.

Abstract: A mechanism is provided for transmitting data from a first processor of a data processing system to a second processor of the data processing system. In one or more switches, a set of virtual channels is created, the one or more switches comprising, for each processor, a corresponding switch in the one or more switches. The data is transmitted from the first processor to the second processor through a path comprising a subset of processors of a set of processors in the data processing system. In each processor of the subset of processors, the data is stored in a virtual channel of a corresponding switch before transmitting the data to a next processor. The virtual channel of the corresponding switch in which the data is stored corresponds to a position of the processor in the path through which the data is transmitted.

Abstract: A component of the wireless network detects when a party who is sending an electronic message desires to have the sender's ID information blocked. If a determination is made that the sender's ID information is to be blocked, the network component removes the ID information from the communication and transmits the message to the intended recipient without the sender's ID information being included in the transmission.

Abstract: A wireless location system may include geolocation of a wireless device connected to Voice-over-Internet-Protocol (VoIP) adapter. The VoIP adapter may include a wireless transceiver or a wireless location determining receiver that facilitates the location of a wireless device connected to the VoIP adapter. The wireless transceiver or the location determining receiver may provide location information to an emergency dispatcher.

Abstract: A system and method for processing a plurality of requests for multi-media services received at a call control element (CCE) defined on the system from a calling communication device. The system includes a Network Routing Element, a Service Broker (SB), at least a primary media severs (MS) and at least a secondary MS, a plurality of application servers (ASs) and a plurality of border elements, all of which are coupled to the CCE. The SB is adapted to receive a plurality of requests including parameters for requesting multi-media services, via the CCE, and to selectively redirect the requests to one or more ASs for providing feature processing for the requests. The ASs can instruct either the primary MS or secondary MS, via the CCE, to form a preliminary interactive communication path with the calling communication device for collecting caller-entered data, which can be validated prior to providing the feature processing.

Abstract: A method for exchanging media description information between user agents using a session initiation protocol (SIP) comprises the steps of: providing information on an OS and sound module of a first user agent so that the first user agent generates, in a media description field of a session description protocol (SDP), an SIP message including the information on the OS and sound module of the first user agent, and sends it to a second user agent; and setting a processing format for voice data of the second user agent so that the second user agent determines the voice data format supportable by the first user agent from the information on the OS and sound module of the first user agent, compares it with the voice data format supportable by the second user agent, and sets a format commonly supportable by the first and second user agents as the voice data format of the second user agent.

Abstract: A method for branching data in a mobile communication terminal to perform data communication between a mobile station and a network which have media access control sublayers. In a data sending mode, each of the media access control sublayers of the mobile station and network attaches logical channel types based on traffic characteristic information and a radio bearer status to a media access control header contained in data to be sent. Then, each of the media access control sublayers branches the data to be sent, to transport channels corresponding to the attached logical channel types. In a data receiving mode, each of the media access control sublayers determines logical channels corresponding to logical channel types of a media access control header contained in received data. Then, each of the media access control sublayers branches the received data to the determined logical channels.

Abstract: Provided are a real time reproduction method of a file being received according to a non real time transfer protocol and a video apparatus thereof. The video apparatus includes a communication module which is connected to an external device to communicate with the external device; a buffer which stores data received through the communication module; a decoder which decodes input data; a controller which stores data constituting a file received from the external device in the buffer via the communication module according to a non real time transfer protocol, and inputs a part of the data stored in the buffer to the decoder before all of data constituting the file is completely received; and an output unit which reproduces data decoded at the decoder in a real time.

Abstract: Some embodiments provide a method of registering a user equipment (UE) in a communication system that includes a licensed wireless communication system and a generic access network (GAN) that has a generic access network controller (GANC). The method sends a register request message from the UE to the GANC that indicates a GAN mode capability of A/Gb only for the UE. When the GANC has a GAN mode capability of A/Gb, the GANC registers the UE with the GAN. When the GANC has a GAN mode capability of Iu only, the GANC rejects the register request message. When the GANC has a GAN mode capability of both A/Gb and Iu, the GANC registers the UE based on a set of GANC mode selection rules that the GANC applies for registering UEs with the GAN.

Abstract: A method of establishing video-conferences between audio communication equipments (ACE1, ACE2) comprises a) using a user audio communication equipment (ACE1) to initiate a first call with another user audio communication equipment (ACE2), b) routing the first call to an exchange node (SSPN) of the telephone network (TN) in order it transmits a message, comprising the telephone numbers of the user audio communication equipments (ACE1, ACE2), to a service node (SN) connected to the telephone (TN) and Internet (IN) networks, c) authorizing the exchange node (SSPN) to establish the first call when the telephone numbers are stored in a memory (MY) of the service node (SN), in order to set up an audio communication between the audio communication equipments (ACE1, ACE2), d) determining, if required, the addresses of the user video communication equipments (VCE1, VCE2) in order to establish a video communication therebetween, in parallel with the audio communication.

Abstract: An IP telephone apparatus is connected to an ENUM server and a Web server via an IP network. The ENUM server has a database that stores an IP telephone URI as terminal information of the IP telephone apparatus and a Web server URI corresponding to the IP telephone apparatus. The Web server stores image data captured by a camera. The IP telephone apparatus transmits to the ENUM server a query on terminal information of a destination IP telephone apparatus; receives from the ENUM server the terminal information of the destination IP telephone apparatus; acquires from the received terminal information a Web server URI corresponding to the destination IP telephone apparatus and performs visual communication with the Web server based on the Web server URI; and displays captured image data received from the Web server in the visual communication.

Abstract: A method for accessing content through multiple devices, e.g. Telephone (1) and/or a data device (4) such as a Personal Digital Assistant (PDA) or Web Enabled Mobile Phone, is discussed where a user can initiate interaction with data through any device (mode), and then switch to another mode of interaction without starting over. The discussed method also allows a user to return to the same point in an application where a call session was lost or intentionally disengaged before completion. The user can regain access to the original call session via any telecommunications (voice or non-voice) device. Access can be accomplished whether the device is the same or different from that used for the original session.

Abstract: A method for branching data in a mobile communication terminal to perform data communication between a mobile station and a network which have media access control sublayers. In a data sending mode, each of the media access control sublayers of the mobile station and network attaches logical channel types based on traffic characteristic information and a radio bearer status to a media access control header contained in data to be sent. Then, each of the media access control sublayers branches the data to be sent, to transport channels corresponding to the attached logical channel types. In a data receiving mode, each of the media access control sublayers determines logical channels corresponding to logical channel types of a media access control header contained in received data. Then, each of the media access control sublayers branches the received data to the determined logical channels.

Abstract: Embodiments of the present invention provide systems and methods for recording media communications an exemplary method is comprised of the following steps: receiving instructions at a gateway to record an incoming call; receiving the incoming call from a calling communication device; determining whether to record the incoming call based on the received instructions; routing the incoming call to a called communication device in a customer center; responsive to the determination to record the incoming call, duplicating the media communications via the gateway; transmitting the duplicated media communications associated with the incoming call via the gateway; and recording the duplicated media communications.

Abstract: In one embodiment, a scheme is disclosed for managing call continuity in a network environment including a circuit-switched (CS) network and an IP multimedia subsystem (IMS) network wherein a SIP Invite message having a Domain Transfer-URI contained in the Request-URI that is operable to trigger a return of a routable number is utilized. Responsive to the SIP Invite message from a UE device, a network node provides a SIP response message (e.g., SIP 380 (Alternative Service) Response) which includes one or more radio access technologies and an alternative domain available to the UE device for continuing a call from one domain to another domain.

Abstract: Methods, systems, and computer program products for determining a serving home subscriber server (HSS) in a communications network are described. One method includes obtaining a subscriber identifier from a query message. An exceptions-based data structure contained in a database is accessed to locate a database entry associated with the subscriber identifier. Similarly, a range-based data structure contained in the database is accessed to locate the database entry associated with the subscriber identifier if the exceptions-based data structure does not contain the database entry. The method also includes acquiring serving HSS data corresponding to the located entry from either the exceptions-based data structure or the range-based data structure.

Abstract: According to one embodiment, a telephone exchange apparatus includes a plurality of processors whose connect communication lines differing in communication protocol from one another, a memory which stores a plurality of items of protocol unique information to process communication signals corresponding to the plurality of processors, a communication controller which is connected to the plurality of processors via control buses with smaller transmission capacities than those of the plurality of communication lines and reads the corresponding-protocol unique information from the memory based on notified protocol identification information and makes the corresponding processor execute signal processing by the protocol unique information, and a call controller which notifies protocol identification information corresponding to a communication line to be added to the communication controller, when a setting request for a call is issued.

Abstract: A serverless telephony system (STS) establishes and maintains a voice, data and video exchange between telephony devices using Internet Protocol (IP). The STS in a preferred embodiment comprises software running on a telephony device with a standard packet network connection over a local area network (LAN) or a wide area network (WAN), or a combination, which may include the Internet network. The new system enables one telephony device to establish a connection with another telephony device without intervention by any third device such as a switch or a server. In one embodiment an IP-PBX (Internet Protocol-Private Branch Exchange) application is provided having no server presented in the network for establishing a call between any two IP phones on a Local Area Network (LAN), providing substantially all services typically expected from a standard PBX, and yet functions in compliance with one or more of a H.323, a MGCP, a SIP protocol or any one of other standard protocols.

Abstract: A voice communication system configured over an IP phone system, with which users could establish voice calls between analog telephone devices and IP phones through an IP phone system without changing their conventional usage method is provided. To allow analog telephone devices to be able to login to the IP phone system, the communication mediation device is required to replace an analog telephone device by be an IP phone user in the IP phone system, and to process voice communications by using analog-digital signal conversion. Using the pre-established registration information, thus allowing the IP phone upon only determining the destination telephone number, it is then capable of automatic converting the telephone number to the corresponding IP phone account so as to proceed with a call through the IP phone system.

Abstract: Systems and techniques are disclosed relating to wireless communications. The systems and techniques involve wireless communications wherein a packet data session may be established between a wireless communications device and a packet data serving node to support a network connection to a packet-switched network. The wireless communications device may be configured to receive a notification of an incoming call from a circuit-switched network even when the network connection is active.

Abstract: A method for selecting radio access system in a multi access system handling packet data services where a terminal can communicate with at least two different types of access systems. According to the invention the method comprises the steps of:—determining (S23) packet characteristics of each packet to be transmitted;—selecting (S31) one of the available accesses for the transmission of each packet, the selection being based on the determined packet characteristic for this specific packet; and—transmitting (S37) the packet through the selected access.

Abstract: A method for branching data in a mobile communication terminal to perform data communication between a mobile station and a network which have media access control sublayers. In a data sending mode, each of the media access control sublayers of the mobile station and network attaches logical channel types based on traffic characteristic information and a radio bearer status to a media access control header contained in data to be sent. Then, each of the media access control sublayers branches the data to be sent, to transport channels corresponding to the attached logical channel types. In a data receiving mode, each of the media access control sublayers determines logical channels corresponding to logical channel types of a media access control header contained in received data. Then, each of the media access control sublayers branches the received data to the determined logical channels.

Abstract: A session set-up for communicating data via a data network is disclosed. First, a session establishment between a communication apparatus and a serving entity is requested for a service from the serving entity supporting the service. Then, when a requirement of a user connection for the service is determined, an association with a data network for the service is activated via a core network.

Abstract: An apparatus. The apparatus includes a network interface card in communication with a telephone, a functional circuit in communication with the network interface card, and an Ethernet port in communication with the functional circuit. The functional circuit is for constructing an Ethernet frame.

Abstract: A system and method are provided with a computing system, capable of binding SMS, voice and other media transactions into a continuous, stateful messaging process that maintains awareness of the communications event over time delays and media transitions. Communications events between two or more communicators over one communications medium are bridged by the system to a second communications medium, according to the needs of the communicators. Various applications of the system permit selective or automated bridging of the communications event based on predetermined events and timeline formulae.

Abstract: In an embodiment, a network router includes plural interfaces and a chain walker which, upon receipt of a first protocol packet in a first interface, processes the packet by walking the packet through a decapsulation chain associated with the interface and the first protocol and an encapsulation chain associated with an identified second interface and identified second protocol. The walker passes the packet to elements of the decapsulation chain to produce a decapsulated packet. The walker passes the decapsulated packet to elements of an encapsulation chain to produce an encapsulated packet. Each chain includes at least one packet-processing element. An element receives the packet from the walker as processed by a previous element, the packet being further processed as appropriate. The walker, upon receipt of the first protocol packet, passes the packet to elements of a demultiplexer chain associated with the first interface to produce a demultiplexed packet.

Abstract: A personal Interactive Voice Response (IVR) system includes a plurality of IVR servers communicating personal IVR service node information in a network and a plurality of personal IVR service node clients coupled to the plurality of IVR servers communicating the personal IVR service node information. The plurality of personal IVR service node clients have a client-server relationship with the plurality of IVR servers. The personal IVR system further includes facilities for managing transmission of the personal IVR service node information using a Voice over Internet Protocol (VoIP) technology.

Abstract: Tandem-free vocoder operations (TFO) between non-compatible communication systems may be enabled through hardware modifications at communication elements within each system. In one aspect, each infrastructure entity in System 1 comprises an intra-system TFO Frame Generator G1, an intra-system TFO Frame Extractor E1, and a TFO Frame Extractor E2 of System 2, which is non-compatible to System 1. Each infrastructure entity in System 2 comprises an intra-system TFO Frame Generator G2, an intra-system TFO Frame Extractor E2, and a TFO Frame Extractor E1 of System 1.

Abstract: A signalling gateway for transmitting messages between a first network and a second network, where the first network comprises an SS7 network and the second network comprises an IP network, the second network comprising a plurality of application servers, the signalling gateway being operable to store SS7 characteristic information corresponding to the application servers, the signalling gateway being operable to receive a message via the first network, the message comprising destination information and origin information identifying a calling entity, read the destination information and identify an application server in accordance with the destination information, transmit the message to the application server, the message comprising the destination information and origin information, receive a response from the application server, the response comprising a destination identifier corresponding to the origin information and an origin identifier corresponding to the destination information, substitute the SS7

Abstract: A telecommunications terminal hosts a conference mixer adapted to enable an at least audio conference between a first conference peer and at least two further conference peers. The conference mixer includes for each of the at least two further conference peers, a respective first data buffer configured to buffer portions of at least an audio data stream received from the respective conference peer; a first audio data stream portions mixer fed by the first data buffers and configured to: a) get audio data stream portions buffered in the first data buffers; b) mix the audio data stream portions from the first data buffers to produce a first mixed audio data portion; and c) feed the first mixed audio data portion to a rendering device of the telecommunications terminal, wherein the first audio data stream portions mixer is configured to perform operations a), b) and c) upon receipt of a notification from the rendering device indicating that the rendering device is ready to render a new mixed audio data portion.

Abstract: A method for reducing unnecessary retransmissions due to transmission delays in a wireless network environment, comprising: measuring a packet inter-arrival time with respect to a received data packet; transmitting an acknowledgement packet corresponding to the data packet received at the packet inter-arrival time in excess of a first threshold time when the measured packet inter-arrival time exceeds the first threshold time; and suspending transmission of an acknowledgement packet corresponding to a new data packet when a data packet received after having transmitted the acknowledgement packet is the new data packet that has not previously been received.

Abstract: A system, method and node for overload control in a network. The method includes the steps of receiving an incoming offer by a node, filtering the offer by an in-throttle to determine if the offer is permitted to be processed by the node prior to processing by the node, processing a permitted offer, and sending the processed offer to a designated target. After processing the permitted offer by the node, the offer may be further filtered prior to sending the offer to the designated target.

Abstract: A method of securely initializing subscriber and security data in a mobile routing system when the subscribers are also subscribers of a radio communication network. The method comprises, within the mobile routing system, authenticating subscribers to the mobile routing system using an authentication procedure defined for the radio communication network, collecting subscriber information from relevant nodes of the radio network, and agreeing upon keys by which further communications between the subscribers and the mobile routing system can take place, and using the subscriber information and keys in the provision of mobility services to subscriber mobile nodes and correspondent nodes.

Abstract: A packet and a method for analyzing network latency are disclosed. The disclosed systems and methods measure the latency between nodes in a network and do so while using less bandwidth and processing than traditional methods by using a packet to traverse a network and collect timestamps at various nodes so that the delay in transit time between nodes can be calculated when the packet returns to a server.

Abstract: A facsimile transmission device implements silence suppression based on a state of a facsimile call and detected facsimile signals. A decision process implemented in the facsimile device can determine when silence suppression should be enabled. The facsimile device may be an IP media gateway connected between a PSTN and an IP network, or may be an IP endpoint. The determination of when to enable silence suppression can be based on the state of sending or receiving facsimile transmissions including facsimile operations and/or signaling. The decisions can be based on the type of facsimile transmission including G3 type or SG3 type facsimile transmissions. By enabling silence suppression during facsimile transmissions, significant reductions in bandwidth requirements can be achieved for the facsimile transmission while maintaining facsimile transmission quality.

Abstract: An improved data communication technique may be employed with modems through a packet network. The disclosed technique facilitates a virtual end-to-end connection between two modems such that the two modems can effectively behave as if directly connected to each other, unaware of any modifications to the data being transferred or to the protocols configured within the communication connection. Preferably, a data communication system for carrying out the communication technique demodulates data coming from a first modem, transports the demodulated data in packets between two gateways, and then remodulates the data before delivering to a second modem at the other end.

Abstract: The invention is directed to providing NDP mediation over an IP interworking VLL. ICMPv6 Neighbor Discovery and Inverse Neighbor Discovery messages are intercepted by provider edge (PE) devices to learn both IPv6 interface addresses and link-layer addresses of customer edge (CE) devices before forwarding the messages to the CE devices, and in some cases after altering the messages to make them more meaningful to the receiving CE devices. In this manner, the CE device interface addresses do not need to be communicated over the LDP control link, as in IPv4 ARP Mediation.

Abstract: A subscription method and device based upon the session initiation protocol are disclosed, such that a function of permanent subscription and temporary subscription is achieved, and a time at which a notification is sent can be customized. In the present invention, a mechanism of permanent subscription and temporary subscription is introduced. Upon reception of a SUBSCRIBE message, a notifier sets a temporary subscription flag or a permanent subscription flag for a subscriber according to an identifier carried in the message or service characteristic corresponding to a subscribed event. In the case of the permanent subscription flag, a NOTIFY message from the notifier is continuously sent until the subscriber or the system revokes subscription, and in the case of the temporary subscription flag, the notifier no longer sends a NOTIFY message upon finishing of a subscribed service.