Abstract: A method and apparatus for enabling multiple services to be invoked in a specific precedence per call are disclosed. For example, the present method allows a subscriber to interact with a plurality of services provided by a service provider on a single call.

Abstract: Various embodiments of methods and apparatuses for managing authentication key contexts are described herein. In various embodiments, the methods and apparatuses include purging an authentication key context of a supplicant after handing off the supplicant, even the authentication key has not expired.

Abstract: A method of failure detection on an Internet Protocol (IP) forwarding plane, which includes: IP nodes of the two ends obtain Bidirectional Forwarding Detection (BFD) identifier information which is used in the BFD, moreover start a BFD session, encapsulate and transport a BFD packet between the IP nodes running the BFD session and implement the failure detection according to the BFD packet transported between the IP nodes. The method of this invention realizes the failure detection via starting the BFD session and via transporting the BFD packet between the inter-domain IP nodes or the intra-domain IP nodes.

Abstract: The present invention provides a dynamic real-time quality management of packetized communications in a network environment. Packetized communications are monitored by and exchanged between wireless Access Points (APs) and wireless terminals or by quality monitoring modules located within network segments or at network vertices. The processing unit analyzes the packetized communications to identify communication signatures associated with the packetized communications. The processor then uses these signatures to identify network impediments to the exchange of the packetized communications. These impediments may take the form of coding problems in which case an appropriate coding scheme is employed by the programmable COder/DECoder (CODEC) to convert incoming packetized communications to incoming user communications, and outgoing user communications to outgoing packetized communications. These impediments may also take the form of communication problems along and between the various network segments.

Abstract: A method and system for extending Camp-On Capabilities to invitees to an ongoing conference call. In one embodiment, a persistent invitation is extended to a first network station client to join a conference call in response to a determination that the first network station client is unavailable to accept a non-persistent invitation to join the conference call, and the first network station client is added to the conference call when the first network station client accepts the persistent invitation to join the conference call. In another embodiment, a persistent invitation to join an ongoing conference call is presented to an initially unavailable network station client, and the persistent invitation to join the ongoing conference call is accepted in response to user input to a network station client including but not limited to the initially unavailable network station client and another network station client where the user has been authenticated.

Abstract: A method for joining a virtual talk group includes facilitating communications of a virtual talk group. The virtual talk group includes a plurality of endpoints of different communication networks. The method also includes recording the communications of the plurality of endpoints of the virtual talk group. The method also includes adding a first endpoint to the virtual talk group and presenting the first endpoint with the recorded communications of the plurality of endpoints of the virtual talk group.

Abstract: A virtual private communications network (VPCN) and method of using the VPCN. The VPCN includes a communications server and at least one communications trunk connecting the communications server to a public telephone network. A web server and one or more digital telephones may be connected to the communications server. A remotely connected device (such as a PC or WAP) communicating with the communications server acts as a locally connected digital telephone. A remote telephone located in the vicinity of the remotely connected device provides telephonic communications in cooperation with the communications server controlled by the remotely connected device.

Abstract: A computer system having an internet phone with a hands-free function is provided. The computer system includes a computer host and an internet phone. The computer host includes a multimedia player program, an internet communication program, a loudspeaker and a microphone device. The internet phone includes a handset, a telephone microphone and a hands-free switching key. When the hands-free switching program is executed, the audio output task and the audio input task of the multimedia player program are respectively implemented by the loudspeaker and the microphone device of the computer host, and the audio output task and the audio input task of the internet communication program are respectively implemented by the handset and the telephone microphone of the internet phone.

Abstract: In an IP telephone terminal, a function implementation controlling unit is configured so as to be capable of controlling implementation of a function instructed by a file received via IP telephone communications when the communicating unit receives the file. The configuration data storing unit stores configuration data that is used by the function implementation controlling unit to implement the function. The configuration data acquisition controlling unit performs control to acquire configuration data from the configuration data storing unit when a communicating unit receives a read file that is transmitted from an external IP telephone terminal. The response file generation controlling unit performs control to generate a response file including the configuration data. The response file transmission controlling unit controls a communicating unit to transmit the response file to the external IP telephone terminal that has transmitted the read file.

Abstract: The present invention provides a technique for providing a forced hold service such as is used for an emergency services call, which is supported at least in part over a packet network. The forced hold service acts to effectively hold a connection for the call with a called party, even when the caller takes an action that would normally end a call, such as going on hook, pressing end, or the like. When the caller takes an action that would normally end the call, the forced hold service allows the caller to automatically reconnect to the emergency services provider over the held connection upon going offhook, pressing send, or the like. Alternatively, the emergency services provider can effectively re-engage the call wherein the caller is reconnected over the held connection upon going offhook, pressing send, or the like.

Abstract: A system, device and method for implementing a special call service are disclosed. The method includes receiving a session establishment request message carrying a service type identifier, determining a service type identified by the service type identifier in the session establishment request message, and implementing a special call service with a service processing logic corresponding to the service type.

Abstract: A mobile wireless communication terminal includes a wireless communication module and a controller. The wireless communication module is configured to communicate with other communication terminals over a wireless interface. The controller is configured to establish a connection with a participant wireless communication terminal via the wireless communication module, to maintain a play list representing song files to be played, to receive a participant song identification from the participant wireless communication terminal, wherein the participant song identification corresponds to a participant song file stored on the participant wireless communication terminal, to add the participant song identification to the play list, and to thereafter execute playback of the song files represented by the play list, including retrieving the participant song file from the participant wireless communication terminal for playback.

Abstract: An Internet compatible dialer pad is used to dial into an Internet server to provide services similar to those found on the Plain Old Telephone System (“POTS”). The dialer pad has an integrated modem set, an extended keypad with alphanumeric entry keys and function keys, display screen and display electronics that renders visual call progress information to the user as well as other communications indicators and related information about the current Internet connection. The dialer uses the Public Switched Telephone System (“PSTN”) and standard LAN/WAN technology to give the user entry into a plurality of Internet calling functions. An Internet database is maintained and permits the dialing party to obtain callee information by entering alphanumeric characters via the dialer. Links from the PSTN to an Internet data base are not restricted to a specific digital data protocol.

Abstract: A method for sharing distributed media resources includes determining at a first call manager that a telephony device controlled by the first call manager requires the use of a media resource device. The method further includes selecting an appropriate media resource device from a media resource group list associated with the telephony device. In addition, the method includes communicating an allocation request to a device process that is associated with the selected media resource device and that is executing at a second call manager controlling the selected media resource device.

Abstract: A method of automatically sharing and transferring information between a Work at Work (WAW) ISDN telephone and a Work at Home (WAH) ISDN telephone. This information includes, calling line identifier delivery from the WAW telephone to the WAH telephone, facilitating subsequent transferring of the call to the WAH telephone if desired. Message waiting status is also shared, facilitating the coordination of message waiting lamps at both the WAW and WAH telephone. Information from one telephone to the other is carried transparently through the ISDN network as user-to-user information, calling party subaddress, D channel X.25 packet message, lower layer compatibility or a combination of these elements. The telephones automatically establish ISDN calls between themselves and swap information pertaining to the status of the companion telephones.

Abstract: To control a conference circuit between multiple terminal devices in a packet-oriented communication network, circuit-oriented conference control signaling is transmitted via a packet-oriented transport protocol between a first terminal device and a communication server of the communication network. Setting up of packet-oriented media channels between the terminal devices and a packet-oriented conference mixing device is initiated by the communication server depending on the conference control signaling.

Abstract: An IP phone system is connected to an IP phone board and used for communicating with an UPnP devices under Universal Plug and Play (UPnP) Protocol. The UPnP device is provided with a device end digital media server and a device end digital media player. The IP phone system comprises an IP phone interface used for receiving AV data transmitted from an IP phone board, an AV converter used for converting the AV data from conforming to a Digital Living Network Alliance (DLNA) type to conforming to an IP phone type and vice versa, a phone end digital media server connected to the AV converter and transmitting the converted AV data to the device end digital media player and a phone end digital media player receiving the AV data transmitted from the device end digital media server, connected to the AV converter and transmitting the converted AV data to the IP phone board.

Abstract: A system is described for managing messages, communications and/or documents comprising: a plurality of task agents, each task agent to perform one or more specified message processing actions, document processing actions, and/or communications processing actions on different types of messages, documents and/or communication channels, respectively; and one or more manager agents to coordinate the actions of the plurality of task agents responsive to a plurality of message and/or document processing rules.

Abstract: A method and apparatus for implementing a presence based universal camp-on feature in packet-based telephony systems are disclosed. A first packet-based device is subscribed to a line status of a second packet-based device. A first notification is received into the first packet-based device from the second packet-based device when the line status of the second packet-based device is in a busy state. A second notification is received into the first packet-based device from the second packet-based device when the line status of the second packet-based device is in an idle state. A user of the first packet-based device is prompted with an option to establish a communication session with the second packet-based device upon receiving the second notification into the first packet-based device. A communication session is established between the first packet-based device and the second packet-based device in response to receiving an accept-call command into the first packet-based device.

Abstract: In a method, system and apparatus for providing an endpoint-to-endpoint transcoding free connection, a customer premises equipment (CPE) at a first or second endpoint encodes and decodes voice data received from and sent to a respective one of first and second media gateways according to an initial codec type. The first and second media gateways perform a Transcoding Free Operation (TFO) negotiation to determine if there is a common codec type among the first and second media gateways and first and second endpoints. If such a common codec type is determined, the first and second media gateways switch from encoding voice data according to the initial codec type to the common codec type and send an indication of the switch to the respective one of the first and second endpoints.

Abstract: In one embodiment the invention is a method. The method includes receiving data from a first party into a multiplexing device or a plurality of multiplexing devices. The method also includes receiving from a second party into a first jitter buffer, processing the data from the second party, and sending output data from the first jitter buffer to a multiplexing device or a plurality of multiplexing device. The method also includes receiving data from a third party into a second jitter buffer, processing the data from the third party, and sending output data from the second jitter buffer to a multiplexing device or a plurality of multiplexing devices. The method further includes sending data from one or a plurality of the multiplexing devices to a first output device. The method further includes sending data from one or a plurality of the multiplexing devices to a second output device. Moreover, data is further sent from one or a plurality of the multiplexing devices to a third output device.

Abstract: A VoIP telephone system which enables a user to receive and play audio files, such as music, over a telephone handset. The telephone system uses a media terminal adaptor (MTA) to provide voice communications to a telephone base station which communicate with a plurality of telephone handsets. The MTA stores a plurality of audio files which may be streamed to the handsets via the base station to be played back to the users. The audio files may be retrieved from a PC connected to the MTA or downloaded from a server through the IP network.

Abstract: Methods and apparatus for customizing the audio parameters of a networked voice communications device (NVCD) are presented. One method includes receiving a settings file, determining a group parameter, extracting at least one audio control parameter from the settings file based upon the group parameter, deriving audio processing parameters based upon a value selected from the at least one audio control parameter, and controlling the audio characteristics of the networked voice communications device using the audio processing parameters. An apparatus having audio characteristics configured by a settings file is also presented, which includes a processor, and memory which is functionally coupled to the processor, where the memory stores a group parameter and executable instructions, and the executable instructions cause the processor to control the audio characteristics of the networked voice communications device based upon the group parameter.

Abstract: In addition to a conference server 100, a record server 200 is provided. The conference data of a real-time conference is stored in a storage section 214 of the record server 200 (S2). A conference terminal 300a performs connection processing to the conference (S1) and then transmits a catch-up playback request to the record server 200 (S3). The record server 200 determines whether or not catch-up playback is enabled (S4) and then distributes the past conference data (S6), and notifies the conference terminal 300a that the catch-up playback end point has come (S7-1). Accordingly, the connection of the conference terminal 300a is switched to the conference server 100 (S9) and the conference terminal 300a is connected to the real-time conference (S10).

Abstract: In one embodiment, the invention provides a method comprising loading a set of transport components into a memory; initializing each transport component of the set of transport components to listen on a particular conferencing interface using a listen string to associate with a network interface, each transport component of the set of transport components listening to a different conferencing interface, the listen string encapsulating parameters to one or more calls to listen; receiving an incoming call signal on the network interface having an incoming conferencing interface; processing the incoming call signal to detect the incoming conferencing interface; and launching an application based on the incoming conferencing interface. Other embodiments of the invention are also described.

Abstract: Aspects of the present invention are generally directed to providing convenient access to contextual information related to one or more voicemail messages. In this regard, a user interface that presents contextual information related to a user's voicemail messages is implemented on an IP phone. The user interface is configured to receive contextual information from a remote device that manages the user's voicemail box. Based on contextual information displayed, input to navigate between the one or more voicemail messages may be received. For example, a user may selectively identify voicemail messages represented on the user interface that will be played. Moreover, a user may generate commands from the user interface for navigating between the voicemail messages.

Abstract: A method of managing input/output/access options and caller greetings within a SmartPhone (or similar) environment. The method is based on grouping Caller IDs (CIDs) of like community, activity, achievement, interests or functionality. These groups are assigned a Group ID code (GID) conferring specific rights to callers through GID-profiles and menu and scripting options that are capable of drawing upon a prescribed suite of variables linked to files, resident applications and peripherals. Rights are further augmented if the CID is associated with multiple GIDs or conferred individual rights. Unlinked CIDs are further scanned for matching prefix-based (area code) GIDs, thus presenting the possibility of individual (rather than governmental) control of Do Not Call (DNC) functionality.

Abstract: The present invention relates to a Universal Serial Bus (USB) storage device having IP telephone capable of originating/receiving a call and providing on-line content. A USB connector (61) and USB memory (63) are installed in a housing (62). A unique phone number belonging to the USB storage device and a content program, used to automatically connect to the server of an ITSP, are stored in the USB memory (63). The USB storage device includes call origination/reception means enabling Internet telephony, and alarm means for sounding an alarm to recognize an incoming call. When inserted into a USB port of a computer, the USB storage device is automatically connected to the server, and a telephone number is input on a monitor, thus enabling call origination and call reception.

Abstract: A system and method is provided to allow users in a remote location, such as a construction site, to both send and receive telephone calls, radio transmissions, and data without directly connecting to the plain old telephone system (POTS). The wireless telephone and data system is a rugged, portable, integrated system comprising an external gateway, WiFi network host, and Internet protocol-enabled telephone unit. When connected to an AC power source in the remote location, the system connects a variety of wireless devices to telephone, radio phone, cellular, and Internet access to facilitate communication of telephone calls, radio calls, and data transfer both internally and with the outside world.

Abstract: A method for performing radio bearer mapping in an user equipment of a wireless communications system includes setting a plurality of multiplexing options of a frequency division duplex radio bearer, called FDD RB hereinafter, when the user equipment operates in a Cell Dedicated Channel state, each of the plurality of multiplexing options corresponding to a multiplexing mode, determining whether an RB of the user equipment has one of the plurality of multiplexing options, and using a multiplexing mode corresponding to a multiplexing option of the plurality of multiplexing options for performing RB mapping when the RB has the multiplexing option.

Abstract: At least one exemplary embodiment of the present invention includes a method comprising the activities of detecting, via a public switched telephone network, a caller's dialing of a telephone number; and after said detecting activity, and prior to call setup, providing to the caller a voicemail status notification. At least one exemplary embodiment of the present invention includes a method comprising the activities of placing a telephone call via a public switched telephone network, and receiving, via the public switched telephone network, a voicemail status notification after placing the telephone call and prior to call setup. It is emphasized that this abstract is provided to comply with the rules requiring an abstract that will allow a searcher or other reader to quickly ascertain the subject matter of the technical disclosure. This abstract is submitted with the understanding that it will not be used to interpret or limit the scope.

Abstract: Method for maintaining a virtual presence of a first remote telephone user in a PBX system having a frame relay network connection between two endpoint routers while permitting the first remote user to make local calls includes first signaling a PBX to represent the remote telephone as being off hook; routing a telephone call placed at the remote telephone in accordance with a defined protocol; and when the routed telephone call is terminated, second signaling the PBX to restore the on-hook status of the remote telephone. Preferably, the first and second signaling are performed in-band, in accordance with an FRF.11 or VToA AAL2 voice over packet protocol. The telephone call-routing may be to a public switched telephone network (PSTN) local to the remote telephone or it may be to another remote telephone user at the same site within the PBX system as the first remote telephone user.

Abstract: To provide a PTT system that includes a plurality of PTT-capable mobile phones each including a display unit, and a server which mediates a PTT call between mobile phones out of the plurality of mobile phones. The server includes a storage unit, a retrieval unit, and a sending unit. The storage unit stores images of users of the plurality of mobile phones and phone identifiers of the plurality of mobile phones in correspondence with each other. The retrieval unit retrieves, from the storage unit, an image corresponding to a phone identifier of a mobile phone involved in the PTT call. The sending unit sends the retrieved image to at least one other mobile phone involved in the PTT call. A display unit in the at least one other mobile phone displays the image sent from the server.

Abstract: A distributed intelligence conferencing system is disclosed, having a plurality of conferencing nodes to connect groups of participants to a conference. Each of the conferencing nodes provides for the connection of one or more participants to the conference. Each node includes a DSP for distributed signal processing, eliminating the need for a central processor.

Abstract: An enhanced IP telephone taking advantage of the many call-related functions provided by existing PBX systems. The enhanced IP telephone includes a thin computer telephony integration (CTI) client transmitting to a CTI server a command associated with a particular call-related function provided by a PBX unit. The CTI server translates the command to a PBX command, and transmits the PBX command to the PBX unit for providing the particular call-related function to the enhanced IP telephone. Instead of the thin CTI client, the enhanced IP telephone may include an enhanced IP signaling protocol stack for accessing the PBX functionality using its signaling protocol. A CTI translator translates the enhanced signaling command to a CTI command for transmitting to the CTI server.

Abstract: A portable telephone having music playback functions has a first storage for storing music data of plural musical tunes and a second storage (namely, a cache memory) for storing at least one musical tune to be reproduced. A musical tune that may be likely to be reproduced is selected from among plural musical tunes in accordance with prescribed rules, so that the music data of the selected musical tune are precedently transferred from the first storage to the second storage to enable smooth reproduction in response to an incoming call whose telephone number specifies the selected musical tune.

Abstract: Processing of multimedia calls in a packet-based network builds on existing call model processing by maintaining the use of the basic call state models to control the overall two-party call and the use of connection views to associate multiple two-party calls and provide access to advanced supplementary services. Through the use of a multimedia view processor, the system separates multimedia connection handling from call handling. The basic call models are extended to support new messages and enhanced messages for creating, modifying, and deleting multimedia view objects and communicating modifications to the media streams between call state model processors and protocol specific processors.

Abstract: A data communications device includes a supervisory circuit, a power supply, and a power circuit. The power circuit includes a data communications port, a power supply connection coupled to the power supply, and a power controller coupled to the data communications port and the power supply connection. The power controller is configured to provide a power signal from the power supply connection to the data communications port in response to communication with the supervisory circuit. Upon loss of communication with the supervisory circuit, the power controller is configured to selectively continue to provide the power signal from the power supply connection to the data communications port when a local parameter has a first value, and discontinue providing the power signal from the power supply connection to the data communications port when the local parameter has a second value.

Abstract: A method and apparatus for digital subscriber line (xDSL) communications between one or more digital signal processors (DSPs) and analog front ends (AFEs) each coupled to corresponding subscriber line(s). The apparatus transports channels of data between subscribers and the DSP(s). The apparatus includes a bus for the transport of digital data, a DSP AFE interfaces. The DSP interfaces couples the DSP to the bus. The DSP interface accepts downstream channels of digital data from the DSP and transmits packets each associated with a corresponding one of the downstream channels to the bus. Each of the packets identifies a targeted AFEs coupled to a selected one of the subscriber lines. The AFE interfaces each couple an associated one of the AFEs to the bus. Each of the AFE interfaces transmits selected packets to a selected one of the subscriber lines for the transport to the subscriber.

Abstract: A method and apparatus for using SONET technology to efficiently carry both data and voice traffic by Add/Drop Multiplexing of both isochronous traffic, such as voice and video, and non-isochronous traffic, such as data, in a SONET/SDH fiber-optic ring topology. A portion of a SONET/SDH frame structure (STS-1, STM-1 etc.) is shared by a number of nodes connected to an optical SONET/SDH ring (either directly or sitting behind a SONET/ADM). The inherent column structure of the SONET/SDH frame is used define dynamic channels (flexible bandwidth channels) between the nodes. Each dynamic channel can be assigned a priority and optionally a minimum bandwidth. The channel to column mapping is varied dynamically for data traffic by a centralized bandwidth manager which serves as an arbiter for requests for bandwidth.

Abstract: An enhanced IP telephone taking advantage of the many call-related functions provided by existing PBX systems. The enhanced IP telephone includes a thin computer telephony integration (CTI) client transmitting to a CTI server a command associated with a particular call-related function provided by a PBX unit. The CTI server translates the command to a PBX command, and transmits the PBX command to the PBX unit for providing the particular call-related function to the enhanced IP telephone. Instead of the thin CTI client, the enhanced IP telephone may include an enhanced IP signaling protocol stack for accessing the PBX functionality using its signaling protocol. A CTI translator translates the enhanced signaling command to a CTI command for transmitting to the CTI server.

Abstract: A telephone system for undetected provision of an auxiliary service during a telephone conversation between at least two participants over a communications network. The system includes a handset, having an input mechanism operable by a requesting participant to submit a request for an auxiliary service to be provided during the conversation, and a service processor, adapted to receive the request over the network and to provide the requested auxiliary service responsive to the request, such that at least one of the other participants receives substantially no indication of the request.

Abstract: A method is disclosed, where the RX unit of an IP telephone waits for a first silence indication packet to be received. When the first silence indication packet is received, the RX begins attenuating incoming packets. The rate of attenuation may occur at a rate responsive to an indication provided by the TX unit. The RX unit also mixes comfort noise with the attenuated incoming packets. The RX unit the waits for a second silence indication packet to be received. When the second silence indication is received the RX unit mixes in comfort noise at a level responsive to an indication provided by the TX unit. The RX units then waits for voice packets to be received from the TX unit. When normal voice activity resumes, the RX unit stops the generation of comfort noise and begins playing voice packets again.

Abstract: The subject invention is directed to a method and system for removing music or other unwanted noise from a conference call particularly when the music or other unwanted noise is not directly under the control of the telecommunication equipment the user is attached to and using. The method and system detects the unwanted sound, deactivates the sound source or disconnects from the line carrying the sound, monitors the disconnected line for voice energy, and reconnects the line when voice energy is detected. This capability provides the telecommunications equipment user with control over unwanted music sources being provided by equipment located either in the private or public networks.

Abstract: A method to transfer data from a digital data modulator device to a digital data demodulator device over a subscriber line. The digital data modulator device has a detection circuit for detecting incoming idle ATM cells, and a cascade coupled null data cell generator, a trellis code modulator, a data encoder/mapper, a constellation points rotation circuit and a constellation points shifting circuit for shifting constellation points to lower power points. The digital data demodulator device has a detection circuit for detecting received null data, and a cascade coupled de-shifting circuit for shifting constellation points with the lowest power back to points corresponding to null data cells, a de-rotation circuit for de-rotating the constellation points, a data decoder/de-mapper and a trellis code de-modulator. Transmission power of idle data is minimized, and the random rotation of low power constellation points simplifies the training and tracking of the frequency equalizer in the demodulator.

Abstract: A system operating in the intelligent peripheral servers (IP) of an intelligent network, allowing a telephone user to access the service applications (SLPs) of the peripheral server it is connected to or other service applications (SLPs) located on a remote server (SCP) of the intelligent network. The system comprises a programming interface to develop service applications operating in the intelligent peripheral and a handler modifying the standardized communication protocol between the intelligent peripheral server and the remote servers. The service applications located in the servers (IP and SCP) are developed using the programming interface, the handler being adapted to each intelligent network protocol such as ITU/ETSI or BELLCORE or WIN standards used in direct mode, through a direct network communication between the remote and intelligent peripheral servers or a communication through the central exchange (SSP) of the switched telephone network to which the telephone user is connected.

Abstract: A process, a control system (CS), a terminal (TER), an exchange, an incoupling device, a decoupling device, and a combination device for providing functions to a telecommunications terminal (TER) which is connected to an exchange (EX) via a subscriber line (SL). Therein a connection (VL1, VL2) of the subscriber line is carried out via a control system (CS) on the network side while a decoupling device (DECP) decouples the connection from the subscriber line and an incoupling device (INCP) incouples the connection once again into the subscriber line. The control system provides the functions to the terminal by manipulation of data which are transmitted on the connection.

Abstract: A multi-line telephony system and method deliver multiple concurrent telephone conversations to a customer premise via a single traditional twisted-pair communications connection, and support telephone services via a traditional dial-up connection. Improved packet-based communications are performed using the multi-line telephony system and method with better transmission characteristics, such as transmission speed, over traditional twisted-pair wire connections to provide H.

Abstract: A method of operating a telecommunications system with a switching facility and with subscriber channels over which digital data are transmissible in the form of data packets each including a destination address, and such a telecommunications system are characterized in that the data packets of two or more subscriber channels to be sent to the Internet are combined onto a single concentrating channel.

Abstract: Method and apparatus for sending and displaying short messages from data network users which are linked to a telephonic network. The telephonic network comprises telephones, each with a display for visually displaying short messages to subscribers of the telephonic network. The apparatus holds a dialogue between the data network users and a software agent embedded in the telephonic network. The dialogue provides the telephonic network the content, the sender ID and the address information of each short message, stores the content and address information in a database, and displays the content of the messages on the telephone display of each destined subscriber.