Lan telephone system

BACKGROUND OF THE INVENTION

The This invention relates generally telephone voice transmission systems and particularly to a local area network (LAN) that is capable of to provide telephony services.

In a typical modern office building has each employee a separate phone line. Instead of all the necessary Phone lines from the local to buy phone company, most office have their own private telephone networks established. A private telephone network can calls from or into the building convey, without the scale of the telephone companies PSTN need to claim. So an office building must only buy a smaller number of access or trunks to his private and public to connect telephone network. Such private telephone networks are as crosstalk systems known (PBX).

In addition to a crosstalk plant has a modern office typically has a number of Workstations (PCs) that linked via a local area network (LAN) are. The LAN connects the PCs in the office to share data, printers or other computer peripherals. Although many different LAN configurations are possible, the personal computers are connected via a Connection medium to one or more central distribution (hubs, Switches) connected to a transmission between network nodes enable. A primary Computer or file server typically stores a large amount of data and decides on the data distribution in the network with reference to a data transmission protocol.

Traditional to lead LANs and PBXs from different functions and were as independent Systems developed. For example. distributed a typical Ethernet LAN digital Data with data rates up to 10 Mbits / s at the same time via a common bus all power-target connections. As a result of the realized in the Ethernet LAN "at all" principle several units can in network send data simultaneously. Therefore, collisions can occur, so that the data must be retransmitted. In contrast, builds an extension location separate assigned point-to-point connections on, typically at lower data rates in Kbit / s range work. Furthermore must PBXs that required for connection to the Public Switched Telephone Network Monitoring run and signaling functions and manage internal connections. The implementation of the Real-time telephony functions required real-time event management and NF-distribution is generally a LAN architecture incompatible.

attempts to integrate PBXs and LANs have been unsuccessful, namely partly due to the increased Effort to create a single system, which demands the satisfies both systems. additionally to the high cost are often the functionality and behavior of the application the integrated system compared to separate dedicated Systems affected. equips one example. a PBX with the required additional LANs data capacity from, the cost of it prohibitive without the expected by dedicated LANs performance to provide.

On typical office So now works with two separate and independent networks: a PC-LAN to distribute computer data and a PBX for the Telephony. The hardware infrastructure shall be independent in the networks from one another and separated. Each network requires its own, just assigned to it physical transmission medium such as. coaxial cable, twisted pair cables, etc. Traditionally, the extension technology, the terminals, the control computer and the building wiring applied separately; they are of the two networks together neither used be improved synergistically their performance and behavior.

Of the Term "computer telephony integration (CTI) refers to a system that telephony functions with a computer improves or controls. For this, the PBXs and Computer interfaced with each other coupled to given call information on the computer so that database Aufsuchvorgänge and screen messages to the called terminal are possible. Other implementations use separate servers with new buses out to determine the ability speech processing directory. Recently, CTI developers offer devices that a standard PC added the functions of a private branch exchange allow to realize. The same PC that operates in LAN, can now serve the private branch exchange to realize.

Despite the opportunities offered by the CTI integration still remain artifacts the different development of PBXs and LANs. Although one the PBXs and LANs with standard PCs - even within one and the same device - build can, the two networks remain separate and independent. the LAN operates with its own data transport protocol and physical Connecting medium between the devices on the network. The PBX uses the intrinsic telephony signaling and switching or Switchgear and a separa<?page 3?>tes assigned physical connection medium to transmit voice data.

Since Recently were (Asynchronous Transfer Mode) envisaged ATM networks, to digital data with multimedia voice and video transmissions integrated on a single fast line (pipe) to be transmitted. For this purpose, shares and packaged ATM digital data into small blocks or Cells of a fixed length of 53 bytes and transmits it with high bandwidth of 25 Mbits / s or more. Although ATM networks to Transporting data, voice and video intended were, does the transfer low real-time voice traffic latency over ATM LANs little attention found. So efforts were an ATM voice transmission on high-capacity WANs, university backbone and long-haul or remote networks concentrated.

the ATM Forum has ATM standards LANs developed. A strength ATM is the ability the network, a particular broadcast an appropriate Betriebsgüte- (QoS) class assign. ATM networks can guarantee, that high standards in terms of available bandwidth and minimum delay for such Compounds are respected, a service with predictable quality features need. Thereby, a reliable voice transmission possible in ATM networks. Although the bandwidth requirements for voice transmission from other LAN technologies are easily fulfilled, the ATM technology today provide the predictable operation goodness for bidirectional Real-time transmission is required.

The <patcit><text>EP 0664658</text></patcit> deals with the problem of soft further or handover (handover) of a cell phone conversation in the transition from one cell to the next. Raises the User initiates a call, are its CDMA signals from the nearest added base station, repackaged and an ATM network to the destination address of the call closest Interface in the public Telephone network sent. At the interface to the public switched telephone network the ATM signals to voice signals are converted back, the transferable public PCM telephone network are. Entering the user a new cell and passes the signal strength Request a further or handover (Handover) from the base station of the new cell packs the data corresponding to and sending to the same ATM network to the same Interface to the public Telephone network. The interface can then choose which data they wants to use, and guarantees as a soft and smooth handover. One also monitored at the ATM network connected Call Processor the handover and the route guidance (routing). The ATM network also handles data transmission.

SUMMARY OF THE INVENTION

The present invention, which is defined in claim 1 below, used in its preferred embodiment, CTI technology to create a distributed private branch exchange (PBX) via a local area network (LAN). For simplification and expansion of conventional Telephony, the system uses the power of a workstation via graphical User interfaces (GUIs) and standard interfaces such as Object Linking and Embedding (OLE) from. The LAN telephone system includes in each Workstations a unique multi-port station module both the network data interface and an interface to a standard telephone provides. High-quality voice communications receives one by use of the real-time streaming, wherein the digitized speech for transmission over the asynchronous network or for the local implemented storage on the computer hard drive for later playback directly to cells is.

On another network module (or large systems other network modules) can be connected to the network server. This PSTN module provides an interface from the LAN telephone system trunks or trunk groups, local the Telephone company provides. It combines telephone trunk interfaces with digital signal processing for caller ID, DTMF and Ruffortgangserfassung as well as a real-time voice streaming, to the voice to facilitate the LAN.

The Workstations are connected and the server over a associated ATM switch, which network traffic by conventional ATM protocols sends, as defined in standards of the ATM Forum are. Using the procedure described in this patent unique adapter modules can not the network only conventional ATM traffic, but also the transport of high quality Language program and the conversion of voice information from Analog or digital signals to ATM cells and support back.

A other system component is the telephone node (stroke), the use of the associated not any computers phones allows - for example. of phones in a factory area or in conference rooms. In the preferred embodiment connects this facility the hub via a LAN connection the network and allows the connection of eight or more phones.

the contains system Software that this unique, for voice transmission furnished LAN used to create a distributed PBX, the initiation and completion of telephone calls between connected to client PCs Remote&lt;?page 4?&gt;speakers to phone node and over outside trunks to public Telephone network enables. This PBX differs from known implementations, that a standard ATM LAN the usual Backplane connection between Official and station interfaces replaced and that the voice traffic over the same set of leads as the LAN data runs. conventional PBX signaling trunk / station or station / station is to implemented network messages, the information on real-time Telefonieeregnissen transmitted over the network or commands to network adapters, under which the corresponding signals and behaviors to support normal voice transmission generated or commands to connect voice media streams using standard ATM connection and message protocols will.

The PBX-control software runs on a computer on the network - usually on the server (or servers in large systems) - and has a telephony service provider module on. Telephony applications incl. Voicemail (electronic mailbox) automatic extending ( "auto attendant "), CTI applications, a client telephony assistant as graphical user interface (GUI) a configuration and management GUIs and a Vermittlerkonsol GUI the server and client computer network implemented.

BRIEF DESCRIPTION OF THE DRAWINGS

<figref idrefs="S41">1</figref> shows diagrammatically an embodiment of the telephone network according to the invention;

<figref idrefs="S42">2</figref> shows diagrammatically the multiport PSTN module of the telephone network of <figref idrefs="S41">1</figref>;

<figref idrefs="S43">3</figref> shows diagrammatically the multiport station module of the telephone network of <figref idrefs="S41">1</figref>;

<figref idrefs="S44">4</figref> shows the telephony node with eight telephones as a block diagram;

<figref idrefs="S45">5</figref> shows a real-time voice streaming in the telephone network <figref idrefs="S41">1</figref>;

<figref idrefs="S46">6</figref> shows a schematic view of the control logic of the multi-port PSTN module of the <figref idrefs="S42">2</figref>;

<figref idrefs="S47">7</figref> shows the server software architecture of the telephone network of the <figref idrefs="S41">1</figref>;

<figref idrefs="S48">8th</figref> shows the client software architecture of the telephone network of the <figref idrefs="S41">1</figref>; and

<figref idrefs="S49">9</figref> shows the Bedienerkonsol GUI.

DETAILED DESCRIPTION OF THE DRAWINGS

ATM LAN telephone network

What the drawings are concerned, now is the <figref idrefs="S41">1</figref> a LAN telephone network or a distributed PBX <figref>10</figref> according to the present invention. A network connection Server <figref>12</figref> provides an interface between the public Telephone network (PSTN) <figref>16</figref> or a wide area network (WAN) and the ATM network switch <figref>14</figref> ready. Although the present embodiment is shown with an ATM network, it is appreciated that they can also be implemented with other types of networks -. for example, Ethernet or a Cells-in-frames-Ethernet. The telephony network server<figref>12</figref> is so equipped, he said ATM LAN telephony network <figref>10</figref> telephony functions a PBX provides, as more fully described below. The ATM network switch<figref>14</figref> transmitted Network messages by sending of ATM cells, the computer data, the network monitoring, signaling or signaling and multiple media streams between the telephony network server <figref>12</figref> and the client machines <figref>18</figref> transfer, on the telephone <figref>11</figref> are connected. The media streams may multimedia images, Video or NF- or voice telephony traffic include.

Of the Telephony Network Server <figref>12</figref> and the client computer <figref>18</figref> are preferably general purpose computer such as IBM-compatible workstation (PCs). It may also be Sun workstations, DEC Alpha machines or other server and workstation PCs or workstation computer act.

On typical network server includes an N processor Intel Pentium class or fast or a fast RISC processor such as the Alpha processor Fa. Digital Equipment Corporation (DEC) on. The server works typically with 64 MB or more RAM memory and at least GBytes of hard disk space. Additional stores are typically Floppy or tape-Drives and a CD-ROM drive. a keyboard and a can on the server Mouse for control purposes as well as a graphics monitor for tracking the software design be connected. The server typically uses faster Disk technology such as Fast-Wide SCSI-2 and can interchangeable with redundant, "hot" be equipped power supplies and other hardware innovations.

Of the Telephony Network Server <figref>12</figref> contains the PBX software <figref>85</figref>. monitoring the, treated signaling and the development of telephone connections. These monitoring software the status of all client telephone <figref>11</figref> the network in real time and respond in time to telephony events that a &lt;?page 5?&gt;usual Telephone service results. These include conventional signaling tones as the official and the busy signal, the call sign, etc. and the production and the completion of media streams between Phones on the LAN. The PBX software<figref>85</figref> uses the multi-port modules, the ATM LAN and PCs to implement standard PBX features such as initiating and completing telephone calls either in the network or to external trunks <figref>17</figref>, The ability to calls switch to the hold state to pass calls to park, and resume, conference call, and the caller ID. Telephony Applications such as voice mail and auto clear through to application software implemented with the PBX as a provider of telephony services.

As the <figref idrefs="S41">1</figref> shows, the switch <figref>14</figref> over a Standard UTP 3 cabling with the telephony network server <figref>12</figref> and the client machines, <figref>18</figref> connected. The connection between the Switch <figref>14</figref> and every client computer <figref>18</figref> leads both the LAN computer data, telephony monitoring and News and the various media streams. The requirements for the connector, Lines and the wirings used in each case as defined in the ATM Forum specifications.

Of the Network switch <figref>14</figref> is preferably an ATM-25 network switch, the data of 25 Mbits / s transfers. It contains a optional ATM-155 interface for connection to faster ATM backbone networks. On suitable ATM network switch supports four to 24 or more ports. Several ATM network switches can be stacked to enhance the port capacity. Selected ports can Ethernet ports for LAN-based Printers or other peripheral hardware Old record. ATM network switches are currently provided by a number of producers -. eg Fa ATM Ltd.. - Available. An example for one for the present invention suitable ATM network switch is the model Virata Fa. ATM Ltd., a 24-port ATM switch.

Of the Client PC <figref>18</figref> is preferably a general-purpose computer as a IBM-compatible standard PC preferably with an Intel 486, Pentium or faster processor. The client uses at least 8 MB and preferably 16 Mbytes of general purpose RAM, and typically operates with 500 MB or more hard disk capacity. Additional stores are typically inter alia, floppy or tape removable drives and a CD-ROM drive. To the client are a keyboard and a mouse for control purposes connected and a graphics monitor for tracking the software processes.

Multiport PSTN module

As the <figref idrefs="S42">2</figref> shows, the telephony network server <figref>12</figref> With the multi-port PSTN module <figref>20</figref> Equipped whose Interconnection and software a trunk interface <figref>22</figref> and an ATM network interface <figref>24</figref> implement and a RAM buffer memory <figref>27</figref> with control logic <figref>26</figref> having, to media stream the Office interface <figref>22</figref> and the ATM network switch <figref>14</figref> convert. The trunk interface <figref>22</figref> handles the connection to the trunks <figref>17</figref> the PSTN <figref>16</figref>, The ATM network interface<figref>24</figref> offers conventional Software and interconnection so that the telephony network server <figref>12</figref> on the ATM-LAN <figref>20</figref> can access. The buffer memory<figref>27</figref> and the control logic <figref>26</figref> implement an effective handover of media streams between the trunk interface <figref>22</figref>, The telephony network server <figref>12</figref>. the digital signal processor (DSP) <figref>23</figref> and ATM network interface <figref>24</figref>. as described in detail below described.

The Trunk Interface <figref>22</figref> implements the conventional monitoring the transfer telephony trunks and Zeichengabe- or signaling protocols, which for connection to the external trunks <figref>17</figref> of PSTN <figref>16</figref> required are. The trunks<figref>17</figref> perform various telephony-like signals as broadcast monitoring and -zeichengabe and NF-voice signals, fax or modem data for the conventional Telephone service (POTS). additionally can the trunks <figref>17</figref> other transmission formats such as T1, ISDN or fiber-based services transmit telephony or multimedia data images, video, text or AF signals. In the preferred embodiment provides the Office interface <figref>22</figref> Access to 16 or more POTS trunks.

A ATM network interface <figref>24</figref> preferably comprises conventional Circuitry for connection to an ATM line as specified in the Standards UNI 3.1 ATM Forum and the LAN emulation specification (LANE) on.

On Example of a suitable ATM interface is from the Fa. ATM, Ltd. (ATML) Cambridge, England, available with sales office in Sunnyvale, CA, USA. includes The ATML interface the ATM interface electronics and a modern RISC processor (ARM, "advanced RISC machine ").

Of the ARM control processor <figref>28</figref> is to monitor the transmission and Reception of ATM cells between the ATM network switch <figref>14</figref> and the telephony network server <figref>12</figref> programmed. Desgl. is the ARM control processor<figref>28</figref> in capable network messages between the ATM network switch <figref>14</figref> and the telephony network server <figref>12</figref> and to transmit the media content the messages to the Office interface <figref>22</figref> to send. In the preferred implementation ar&lt;?page 6?&gt;processed the network using the TCP / IP protocol. The network messages contain computer data, monitoring Telephony shipments, signaling and various media streams. Of the control processor <figref>28</figref> oriented computer data containing network messages via the PC bus <figref>29</figref> the multi-port PSTN module <figref>20</figref> directly from ATM network switch <figref>14</figref> to the telephony network server <figref>12</figref>,

Of the control processor <figref>28</figref> manages the real-time event handling telephone, as far as it concerns the telephony trunk interfaces. He managed the efficient use of the resources of the DSP <figref>23</figref> for detecting the caller ID, DTMF, and the Ruffortgangsverfolgung other conventional Forms of signaling on trunks. In the preferred embodiment is the DSP <figref>23</figref> a TMS320C50 the Fa. Texas Instruments etc., on the digital standard software algorithms for the digital telephony signal processing Fa. HotHaus Technologies, Richmond, British Columbia, Kanda to run. The control processor<figref>28</figref> also manages the production telephony tones to select and other purposes, and controls the connection state, impedance matching and echo cancellation the e inzelnen exchange connections<figref>22</figref> at the Multiport PSTN module <figref>20</figref>,

furthermore manages the control processor <figref>28</figref> the diversion of media streams from incoming trunks <figref>17</figref> on client computers <figref>18</figref> about the ATM network or directly to and from the server's hard drive for storage and later playback, so realize that to voice mail and the exchange automatic leave. This blank media streams directly to a to a standard telephone line <figref>17</figref> connected external caller or the network for the purpose of voice playback on client Telephone <figref>11</figref> in the Send network.

furthermore manages the control processor <figref>28</figref> intrusion more Media streams to the DSP <figref>23</figref>. so that they do to the conference call between a plurality of the System - either LAN or PSTN lines <figref>17</figref> - Connected callers bring together leave.

all these telephony features are ultimately of the PBX software controlled that using a socket-based programming interface to a standard protocol such as TCP / IP to the control processor <figref>28</figref> communicates. Messages from the control processor <figref>28</figref> over the network sent to the PBX software <figref>85</figref> in server <figref>12</figref> from Real-time telephony events to the connected trunks <figref>17</figref> to inform, and received coming from the PBX to the call monitoring to implement. Some of these messages control the establishment and the degradation of media streams for the Voice transmission.

PBX trunk control messages are from the host processor in the server <figref>12</figref> on the PC bus <figref>29</figref> directly to control processor <figref>28</figref> where the multi-port PSTN module. In contrast, be network news, the media streams of digital representations comprise of real-time voice content, between the trunk interface <figref>22</figref> and the ATM network switch <figref>14</figref> using the DSP <figref>23</figref>, of Buffer RAM <figref>27</figref> and the control logic <figref>26</figref> transfer. The buffer RAM <figref>27</figref> and the control logic <figref>26</figref> to implement a first-in-first-out (FIFO) for transmitting -Datenpufferung digital representations of voice signals between the NF-ATM network and the synchronous exchange interface. The operation of buffering in implementing a real-time voice streaming is hereinafter described in detail.

As the <figref idrefs="S41">1</figref> represents, is a primary function of the network server <figref>12</figref>. an interface between the PSTN <figref>16</figref> incoming Trunks and the ATM network switch <figref>14</figref> represent. For example. packed the telephony network server <figref>12</figref> the various synchronous telephony signals, the trunks, the <figref>17</figref> lead to an asynchronous standard cell format of 53 bytes of fixed length, which the ATM interface <figref>24</figref> to the ATM network switch <figref>14</figref> passes.

Multiport station module

As the <figref idrefs="S43">3</figref> shows, the client PC <figref>18</figref> with a Multiport station module <figref>30</figref> with ATM network interface <figref>34</figref>. a conventional Phone Interface <figref>32</figref>, A DSP <figref>33</figref> and the control logic <figref>36</figref> equipped. This hardware can desired Sound signals by programming the appropriate algorithms in the digital signal processing <figref>33</figref> generate - example. by official and call signs. Furthermore, the multi-port station module <figref>30</figref> in a position, to detect - tone symbols from the connected telephone - for example DTMF dial tones. and decode. The multi-port station module<figref>30</figref> contains a small Switching power supply, the voltages to simulate the exchange battery and supplies of Rufleitungszuständen.

The ATM network interface <figref>34</figref> allows the client PC <figref>18</figref> via conventional Circuitry and software access to the ATM LAN. The telephone line interface<figref>32</figref> puts digitized voice and audio signals to analog signals to and provides a conventional POTS interface (with exchange battery and ringing voltages) to a Standard telephone set of type 2500 prepared with a standard RJ-11 connector is connected.

The control logic <figref>36</figref> facilitates data transfer between the ATM network interface <figref>34</figref>. the client PC <figref>18</figref>, The DSP <figref>33</figref> and the telephone line&lt;?page 7?&gt;interface <figref>32</figref>, The ATM network interface <figref>34</figref> preferably comprises conventional electronic circuitry for connection to the ATM line based on the specification of the ATM Forum UNI 3.1 on. Similar to Multiport PSTN module the telephone network server <figref>12</figref> is a suitable ATM interface of Fa. ATML available. The ATM network interface <figref>34</figref> includes the necessary ATM interface circuitry and preferably a RISC processor (ARM) control processor as <figref>38</figref>. is programmed to the transmission and reception of ATM cells between the ATM network switch <figref>14</figref> and the client PC <figref>18</figref> to be monitored.

Of the control processor <figref>38</figref> is also capable of news from ATM network switch <figref>14</figref> at the client-PS <figref>18</figref> or the telephone line interface <figref>32</figref> to judge. For example. be network news, the computer data and / or monitoring and signaling signals from or to the telephone line from the ATM network switch <figref>14</figref> included on the PC bus <figref>29</figref> on the client PC <figref>18</figref> directed. In contrast, network news with media stream the net <figref>10</figref> and the telephone line interface <figref>32</figref> on the DSP <figref>33</figref> and the RAM memory of the control processor <figref>38</figref> passed, wherein a FIFO buffering is implemented, as described in more detail below. Since the Client PC <figref>18</figref> must cope with a few media streams can be the FIFO buffering the control processor with internal software using existing storage space do.

Of the control processor <figref>38</figref> managed telephony events differ the telephone station interface in real time. It controls calls on the telephone <figref>11</figref> and manages the efficient use of resources of the DSP <figref>33</figref> for the detection of DTMF dialed digits from the connected telephone <figref>11</figref> and generation standard signaling tones as the Amtszeichen-, busy and ringback tone. In the preferred Implementation is in the used digital signal processor a TMS320C50 the Fa. Texas Instruments or similar Processor chip and for the digital signal processing software algorithms standardized Fa. Hot-House Technologies, Richmond, British Columbia, Canada, used.

Of the Control processor also handles the rerouting of media streams from telephone <figref>11</figref> to other client PCs <figref>18</figref> the ATM network or to the PSTN module <figref>20</figref> purpose of following the line bundle <figref>17</figref> and driving media streams directly to and from the hard disk of the client PC or directly from of or the server hard disk on the network, respectively for storage and later back playing. also manages the control processor <figref>38</figref> intrusion of multiple media streams to the DSP <figref>33</figref>. so that they are in the sense of a conference call with several Merge callers / participants leave.

all these features will eventually from the PBX software in the server <figref>12</figref> controlled using the a socket-based programming interface to a standard protocol such as TCP / IP to communicate with the control processor. Messages on the Sent network to the PBX of real-time telephony events regarding of the telephone 11- eg removed handset or choose - to inform.. In reaction then releases from the PBX are received to the call monitoring to implement. Some of these messages control the development and Degradation of media connections voice transmission.

Phone node (hub)

The <figref idrefs="S44">4</figref> shows the telephone node (hub) <figref>15</figref> as a block diagram. This means allows it, not the PCs assigned telephone to the network <figref>10</figref> to join. In the preferred implementation includes the establishment of an interface <figref>42</figref> for eight telephones. The operation largely corresponds to the above-described PSTN module <figref>20</figref>, except that instead of several trunk interface multiple telephone interfaces to be used. For example. to lead the ATM interface <figref>44</figref>, The RAM buffer memory <figref>47</figref> and the control processor <figref>48</figref> similar Functions as the ATM interface <figref>34</figref>, The RAM buffer memory <figref>37</figref> and the control processor <figref>38</figref>That in the description of the station module <figref>30</figref> discussed are. In the preferred implementation is a switching power supply<figref>45</figref> use, which can serve eight telephones with Amtsbatterie- and ringing voltages. The Operation of real-time voice streaming largely corresponds to that of the PSTN module <figref>20</figref>, Also multiple voice lines operated.

Voice streaming and steering

The <figref idrefs="S45">5</figref> shows voice streaming and -Lenkungsfunktionen that a multiport module <figref>50</figref> as the PSTN module <figref>20</figref>, The station module <figref>30</figref> and the telephone node (<figref>15</figref>) Performs. For example. be the telephony connection<figref>55</figref> analog or digital Voice signals from a telephone (in the case of a station module <figref>30</figref> or the telephone node <figref>15</figref>) Or from a trunk line (in the case the PSTN module <figref>20</figref>) over a A line interface circuit <figref>52</figref> transfer. After the line interface<figref>52</figref> puts a CODEC <figref>51</figref> (Eg. Texas Instruments TCM29C13 or National Semiconductor TP 3054) the analog voice signal to a standard synchronous digital form to (eg. PCM). For example. becomes for 64 Kbit PCM every 125 microseconds, a new 8-bit sample generated in synchronism; Thus one obtains 8000 samples&lt;?page 8?&gt;per Second. it is appreciated that the CODEC<figref>51</figref> not used , when the connection to digital lines or devices takes place. The CODEC<figref>51</figref> is the synchronous digital data on the DSP <figref>53</figref>Where the telephony signal acquisition and generation and the line management take place.

The synchronous data then go to the function block <figref>56</figref> and an associated module control processor <figref>58</figref>, The synchronous the implement to asynchronous data and the latter as a media stream on enter one of the ports. done The synchronous / asynchronous conversion in function block <figref>58</figref> under FIFO data buffering. The function block<figref>56</figref> and the module control processor <figref>58</figref> accept data bytes from the synchronous data stream in a FIFO memory buffer until enough data for too shipping network data packet present. In an ATM network are eg. 32-48 bytes of data for an ATM cell and a more data cell stored to temporal uncertainty (jitter) catch that in the transmission always occur in asynchronous LAN. The number of transmitted per cell bytes depends Kom promissen regard to the required network latency and the Synchronization process from which are chosen by an expert. Has the desired Number of accumulated data bytes, a packet of the asynchronous data at the network interface <figref>54</figref> given and asynchronously in the LAN sent to a distal module.

The above synchronous / asynchronous conversion can be from any of the <figref idrefs="S42">2</figref>. <figref idrefs="S43">3</figref> and <figref idrefs="S44">4</figref> described Multiport modules <figref>15</figref>. <figref>20</figref>. <figref>30</figref> Performing FIFO buffering of the respective control logic <figref>26</figref>. <figref>36</figref>. <figref>46</figref> among Management by the corresponding control processor <figref>28</figref>. <figref>38</figref>. <figref>48</figref>, Multiport modules that need a larger line capacity (Eg. The PSTN module <figref>20</figref> the <figref idrefs="S42">2</figref>use) additional fast RAM buffer memory <figref>20</figref>To which the control processor <figref>28</figref> and the DSP <figref>23</figref> Have access. Multiport modules with lower Throughput requirements (eg. The multi-port station module <figref>30</figref> the <figref idrefs="S43">3</figref>) use only the RAM memory of the control processor <figref>38</figref>, A more detailed Discussion of the operation of these circuits and related software follows in the description of <figref idrefs="S46">6</figref>,

Around to enable two-way communication, implement the Funtionsblock <figref>56</figref> and the module control processor <figref>58</figref> a Return path on the asynchronous data streams from the LAN port <figref>57</figref> for telephony port <figref>55</figref> Posted can be as follows. Asynchronous data flows from the LAN port <figref>57</figref> be of the network interface <figref>54</figref> adopted and from control processor <figref>58</figref> and function block <figref>56</figref> out the asynchronously implemented for synchronous form. The asynchronous / synchronous data conversion takes place inversely to the above-described synchronous / asynchronous conversion. For example. be in an ATM network asynchronous data cells from the functional block<figref>56</figref> and module Steuerprozuessor <figref>58</figref> receive and FIFO buffering converted to synchronous data cells. The synchronous data will therefore be reissued, the form in which it over the line interface <figref>52</figref> to a connected telephone <figref>11</figref> (as the station module <figref>30</figref>), A line <figref>17</figref> (like in PSTN module <figref>20</figref>) Or a digital interface such as a transmitted T1 line leave. After that Syn can be chrondaten example. Byte-wise on the DSP<figref>53</figref> at the (possibly present) CODEC <figref>51</figref> pass on.

The unidirectional temporal total delay (latency) for implementation and the transfer in Network and by two multiport modules <figref>50</figref> , in the case an ATM network are typically less than 20 ms, which demands the for a quality voice transmission fulfilled. The synchronization in the network and the two multi-port modules is achieved either by using the 8 kHz broadcast capability of ATM networks or by sending timing and sequence information with all or selected Network data packets or cells, extracting this information in the other module and exploit this information to resynchronize under PLL using conventional techniques.

Of the Module control processor <figref>58</figref> may asynchronous data streams and a PC port <figref>59</figref> to steer. On the PC port<figref>59</figref> can the asynchronous data, a host computer as the PC network server <figref>12</figref> or a PC client <figref>18</figref> order processing desired access. For example. can the data from the PC port <figref>59</figref> overwritten in larger buffers and periodically transmitted to the system disk or further processed. The stored data can be later for playback via either the control processor <figref>58</figref>, The FIFO buffer, the DSP <figref>53</figref>. the (possibly present) CODEC <figref>51</figref> and the line interface <figref>52</figref> read back or for storage, playback or processing directly from the control processor <figref>58</figref> about the network interface <figref>54</figref> passed to another network receiver module or the LAN.

The Voice-streaming circuits and software lead to: <ul><li>1. providing an interface to conventional Telephone operation as analog or digital telephones or usual analog or digital trunks;</li><li>2. reacting language from analog to digital form, possibly vice versa; </li><li>3. reacting of voice data from the synchronous (eg. PCM) in the asynchronous format (such as ATM cells) and back;</li><li>4. Initiate speech from the line interface Stel&lt;?page 9?&gt;le to the network and back;</li><li>5. conducting digital language of the line interface or from the LAN to the local hard disk to store and later reading; and</li><li>6. directing digital language from the LAN to the local hard drive for Save and later Reading from the network or from the line interface connector.</li></ul>

The above Leitbarkeit of speech signals is the basis for sending voice in ATM LAN <figref>10</figref> in the preferred embodiment and also Applications such as voice mail and automatic switching that both locally or in the network, storing and re-reading of require voice data.

For example. the voicemail feature uses the functionality described above as follows out. Achieved an external caller the system via the telephony port<figref>55</figref>. will be digitized speech signal, if necessary, and from the synchronous PCM implemented in asynchronous form. Then, the data are expressed as stream over the Port <figref>59</figref> performed on the processor of the PC host, the typically in large Buffer of at least several KB of data writes. For better System efficiency will typically compresses data and periodically stored in a file on the hard disk. A system user is then able to access the stored voice data, by the system from the PSTN via the telephony port <figref>55</figref> or the LAN port <figref>57</figref> responsive. In the latter case, the Data as a stream over the power in the station module <figref>30</figref> played the user be to connected directly to a telephone <figref>11</figref> reproduced to become; alternatively can the file the voice data compressed containing up until the provide storage of the client PC, and then locally on the bus <figref>29</figref> of Client PC to the station module <figref>30</figref> over played and finally on the connected telephone <figref>11</figref> to be reproduced.

Of the Users can also leave a message for someone else, by voice data via the LAN port <figref>57</figref> to control processor <figref>58</figref> and then over the PC port <figref>59</figref> be conducted to and stored compressed to be, as described above.

In an alternative implementation is not the voice data file on the server hard disk <figref>12</figref>of, but on the Client PCs <figref>18</figref> stored. Playback is then via the LAN and eventually on Line interface <figref>52</figref> or locally via the PC bus to the connected station module <figref>30</figref>,

Multiport PSTN module - control logic

The <figref idrefs="S46">6</figref> shows a block diagram of the control logic for the PSTN module <figref>20</figref> the <figref idrefs="S42">2</figref>, In the preferred embodiment The operation of the transmission and reception of voice media streams as follows.

all <figref>5</figref> ms receives the control processor <figref>28</figref> (<figref idrefs="S42">2</figref>) from the network <figref>10</figref> an ATM cell with 40 bytes of useful component. These 40 bytes are a unique Base Address each channel to the buffer memory <figref>27</figref> transfer. to display the actual handover address adds the address logic <figref>69</figref> the buffer RAM (<figref idrefs="S46">6</figref>) an offset to the base address, which it from the control processor via its interface <figref>66</figref> receive has.

Asynchronous receives to the above, the control processor <figref>27</figref> (<figref idrefs="S42">2</figref>) from the clock generator <figref>64</figref> about the interface <figref>66</figref> the control processor every 5 ms an interrupt, the reports that the buffer memory 40 bytes of data to send over the maintains network ready. The control processor <figref>28</figref> presented for each channel a unique Base address to which the address logic <figref>69</figref> of the buffer RAM alternately the offset 0 or 40 added before the address to the Data buffer is placed.

On or more 8-bit CODEC shift registers <figref>62</figref> are chained, to the PCM transmit bus to dine. This bus supplies in turn a corresponding number of CODECs, corresponding to the number of the shift registers. In the preferred embodiment, find 16 shift registers and 16 CODECs use. The entry for CODEC shift register chain <figref>62</figref> via the PCM receive bus. The CODECs are programmed certain time slots in PCM bus to address so that the end a frame period the shift register which the transmit data for CODEC n contained, now contains the CODEC from the n received data. Every shift register <figref>62</figref> is assigned to a holding register; at the End of a block, the contents of the holding register and the shift register replaced and DSP interface <figref>60</figref> a DSP interrupt from the clock generator <figref>64</figref> to the DSP <figref>23</figref> adopted. At the Operation of this interrupt reads the DSP <figref>23</figref> each holding register in the CODEC shift registers <figref>62</figref> out and writes the data in the receive buffer RAM <figref>27</figref> each Channel. He then describes each register with data from the associated transmit buffer RAM<figref>27</figref>, While the handover the data between the holding registers <figref>62</figref> and the buffer RAM <figref>27</figref> can the DSP <figref>23</figref> can manipulate them.

Around the synchronized exchange of tax information between the control processor <figref>28</figref> and the DSP <figref>23</figref> to facilitate, for each channel a pair of mailboxes <figref>63a</figref>. <figref>63b</figref> provided. Each &lt;?page 10?&gt;Write to the mailbox <figref>63b</figref> by control processor <figref>28</figref> causes an interrupt to the DSP <figref>23</figref> via the DSP interface <figref>60</figref>, To the CPU load <figref>28</figref> reduce, triggers a write operation the DSP <figref>23</figref> in the mailbox <figref>63a</figref> no interrupt the control processor from; instead, examines the control processor<figref>28</figref> the mailbox <figref>63a</figref> every 5 ms while the clock generator <figref>64</figref> induced cell-available interrupts.

the Interface Module <figref>68</figref> is the trunk group of control processor <figref>28</figref> about the control processor interface <figref>66</figref> (Together with the address decoding <figref>65</figref> the control processor) is used to individual trunks <figref>17</figref> to prove. Process in reverse , the control processor <figref>28</figref> Ringing voltages on the trunks <figref>17</figref> capture.

The DSP address decode logic serves individual CODEC shift registers <figref>62</figref> and a specific mailbox <figref>63a</figref> or <figref>63b</figref> select.

The CODEC programming from the control processor <figref>28</figref> done by the control processor interface <figref>66</figref> together with the control processor address decode <figref>65</figref> under use the CODEC control register interface <figref>67</figref>,

The Control logic for the multi-port station module <figref>30</figref> operates similarly to the for the above PSTN module <figref>20</figref>Except that the functions of the Buffer address logic <figref>69</figref> (<figref idrefs="S46">6</figref>) And the Buffer RAM <figref>27</figref> (<figref idrefs="S42">2</figref>) Now in the control processor, and the associated running RAM will. The additional Processing overhead is tolerable, since only a few channels each are simultaneously active. This achieves a lower cost Implementation with fewer components. The other difference is the replacement of the trunk interface module <figref>68</figref> (<figref idrefs="S46">6</figref>) against a station interface module, which control processor the <figref>38</figref> (<figref idrefs="S43">3</figref>) allows on the line from the station module <figref>30</figref> to telephone <figref>11</figref> the hook condition (Approved handset) the concern of the battery voltage and the concern of a ringing voltage capture.

Not shown is the control logic <figref>46</figref> the telephony node, which differs from the server control logic <figref>26</figref> only through the exchange the trunk interface module with a multi-station Telephony Interface Module having a functionality similar to that the above-discussed single station interface module <figref>78</figref> differs.

Server-Software

The <figref idrefs="S47">7</figref> shows the architecture of the server software. Generally, the software is with conventional C ++ - compilers and other software development tools for operating systems such as Microsoft Windows NT, Windows 95 and UNIX developed. In the preferred embodiment, is in Server Microsoft Windows NT and Windows client computer 95 or Windows NT employed.

A Key component running on the server, the PBX control software <figref>85</figref>That all telephony resources the system manages - incl. the telephone <figref>11</figref> and the connections to the PSTN <figref>16</figref> (<figref idrefs="S41">1</figref>). The PBX software <figref>85</figref> controls both the local trunk connections the server to the trunks <figref>17</figref> and other trunk group to distributed payphones at the remote server or through the ATM LAN Client computers.

The Communication between the PBX software <figref>85</figref> and the local PSTN module <figref>20</figref> (<figref idrefs="S42">2</figref>) via the Rufsteuerschnittstelle <figref>87</figref> the PBX network, a socket-based Programming interface to send and receive messages directly from the Serverbus <figref>29</figref> (<figref idrefs="S42">2</figref>) To / from the control processor in module <figref>20</figref> and also to communicate with remote telephony modules via a similar socket-based mechanism allows, which according to a standard protocol such as TCP / IP messages over the ATM network shipped. The same interface can also be used are to news the internet to control remote telephony resources also be sent to the places connected Internet. The slower Reactions of the Internet are more intelligent client software considered.

Of the PBX operation is controlled by data in a configuration database <figref>82</figref> are stored. This software allows system administrators, functions as the allocation of telephone connections, trunk connections, user options as the number of rings before the voicemail switch, Call forwarding, caller blocking and designate to vermitteltelndn manually terminals control. Access to this database via GUIs, facilitate the setting up and managing the PBX.

The Telephone service provider software <figref>84</figref> is connected to the PBX software <figref>85</figref> coupled. The software <figref>84</figref> contains Functionalities, as TAPI, the 2.0 ( "Telephony Applications Programming Interface ", programming for Tele fonieanwendungen) of Fa. Microsoft (Redmond, Washington, USA) and the TSAPI ( "Telephony Server Applications Programming Interface ", programming interface for telephony server applications) Fa. NOVELL (Orem, Utah, USA) offer. The service provider allows the software-based control by telephone tasks such as assembly and disassembly of conversations handoff in the hold state, the Rufweitergabe, parking and the Like&lt;?page 11?&gt;resumption of talks, the build-up of conference calls and call monitoring. It provides a programming interface, the Telephony control the by applications such as voice mail, automatic switching and the GUI for placement officers simplified.

Client software

The <figref idrefs="S48">8th</figref> shows the architecture of the client software. The control processor software<figref>95</figref> in the Client NIC <figref>30</figref> (<figref idrefs="S43">3</figref>) Uses a socket-based Programming for sending and receiving messages to / from the PBX software in the server using a protocol such as TCP / IP over the ATM-LAN <figref>10</figref> (<figref idrefs="S41">1</figref>). The client PC applications using a software module <figref>93</figref> for the provision , Telephony services to the telephony service provider <figref>84</figref> in the server <figref>12</figref> (<figref idrefs="S41">1</figref>) Access to PBX telephony services via the LAN <figref>10</figref> to obtain. These services are applications as the operator console GUI <figref>90</figref>, The setup and configurable GUI <figref>92</figref> and a telephone wizard GUI <figref>91</figref> in taken to claim that the all the telephony usage by the operator simplify and enhance client PCs.

Vermittlungskonsol- and telephone assistant GUIs

The <figref idrefs="S49">9</figref> shows the Vermittlungskonsol GUI that the conventional key telephone, to the operators normally apply to incoming calls to control and forward replaced by a pop-up window, that facilitates call management directly on the computer screen. In the following description, the term GUI is used to the Phone Assistant GUI or to designate the Vermittlungskonsol GUI. These two applications have a common way of working; they have the same He apparition form and feel is equal to. The Vermittlungskonsol GUI provides all the functionality of the telephone assistant GUI and other functions such as. the possibility of controlling the Status of all PBX lines through the mediation of the person in Monitor view and the identification of any other on the PBX active Operator consoles.

The GUI allows a user to more than one call at a given telephone line manage using of Call options like Reply, hooking, manufacturing and canceling the hold state, parking, accommodating, staff and Blind Transfer.

be talking the call section ( "Call View ") <figref>100</figref> as Picture symbols (icons) <figref>104</figref> graphically displayed so that the Users using conventional Windows mechanisms and mouse actuations having a conversation can interact. A conversation can be choose; then can be menu options and the panels a Toolbar (Toolbar) it apply. Appears eg. An incoming Call in the conversation view <figref>100</figref>need the user only to the call symbol <figref>104</figref> double click to answer him. If a call is accepted, the user can on an extension number (set) in the "hot List" <figref>101</figref> double, turn on him. Other Rufsteueroptionen are protected by the Menu <figref>110</figref> and the toolbar buttons always available. to Display of each straight through questions call serve colors; eg be. the currently selected Call red and all other gray shown. For different call or serve this week states different icons. An incoming, just ringing Call appears eg. As a flashing phone icon, a by asked Call the phone off hook.

For the selected call is the GUI in a status window <figref>111</figref> additional Information from, namely, the call state, the call duration, the calling apparatus and the name and extension number of the other call participant. Where possible, is the name of a caller in addition appears to his number.

The the user activities to be undertaken are optimized for convenience and efficiency. For example. let yourself with a single drag-and-drop a call from another on a Entry in the list view <figref>101</figref> to hand over.

To the Generating calls can a number input with the keyboard in the data entry field <figref>102</figref>. a telephone keypad <figref>103</figref> in the GUI or the Tastatenfeld of the telephone <figref>11</figref> even use. In the preferred embodiment and allows for the use of Microsoft Windows 95 or Windows NT, a Number also from the "phone book" of the Microsoft program Select Exchange.

The for the Users most important numbers are in a "hot List" <figref>101</figref> stored, which is always visible in the GUI and enables the user to by double-clicking the mouse a call to one of the numbers depose. The "Hot List" can be set up by the user and contains Phones, the other by connectors or numbers in the public may be telephone network. In the preferred embodiment, which works with Microsoft Windows 95 or Windows NT, can the Add user numbers from the Exchange phonebook for "Hot List". The Number of Entries the list is unlimited.

The can list Also for use a speed dial. A single click on a Lis&lt;?page 12?&gt;teneintrag selects the GUI to the number. The list also serves to simplify the Call or call control. Dragging a Rufsymbols from the call section on a shunt in the "Hot List" directs the user switching through a on this apparatus.

Does the "Hot List" a large number of entries, allows a text search mechanism for the user to locate an entry quickly, by means of the data input field <figref>102</figref>, For example. can a live agent so configure the list that they all In addition to connections contains at the PBX. If there is the name of a person to be called, are the live agent the first few letters of the name; the GUI will search the associated List entry, roll it into view and marks it for the live agent. Listing can be customized with additional Expand numbers. For example. leaves there is an entry "company" with a mouse click expand so that all Departments are displayed, which lends itself to the individual In addition to connections be expanded. With this hierarchy, the number of all times keep visible numbers minimal.

For incoming Calls, the user can instruct the GUI to handle the call. With a single press a button <figref>105</figref> answered the GUI this, plays a announcement stored and places the call on hold or transfer to voicemail. The user can perform these operations while talking about a another call continues. These single-field call handling simplifies operation when the User has to work more than one call.

The GUI is designed to run in the background. receives the User a reputation, it appears on top of each other currently running application. Will users eventually a call settle, brings a double-click on an icon on the desktop (desktop) the GUI in the foreground.

The GUI is the direct use of the handset for call control again. If, for example. The user the handset from, to make a call, the GUI displays the icon of the new call as <figref>104</figref> on the screen. Both call control methods can be applied together on the same call. For example. can the user lift the handset, This action places the dial tone, and then a number of means Choose GUI.

On Call Log <figref>106</figref> is guided, the all incoming to the user line and displays outgoing calls. For example. is still logged an incoming, but not answered call, so that the user can view the caller's identification later.

The contains Vermittlungskonsol GUI a monitoring field ( "Monitoring view") <figref>107</figref>, the the status of all the PBX each ongoing talks indicates. This view is performed in real time and are state changes the calls immediately. In the "hot List "for each monitored call to the associated Participants also entries an activity icon <figref>108</figref> output.

For each talk , the switching person the caller, the called party, the call state (For example through., Hold state, call in progress), the call duration (Optionally provided) and the holding period (if a call on hold was seen connected). Each participant of a conversation is in the "Hot List" <figref>101</figref> With an activity icon <figref>108</figref> marked. The mediation may select an entry of "hot list" and only monitor the call in a separate window.

The GUI provides other on the desktop running applications a Telephone service provided. In the preferred embodiment, the Microsoft Windows 95 or Windows NT is running, supports the GUI is a drag-and-drop function, in an automatic Selection is made when the user drop a phone number in the GUI window leaves. The GUI also acts as an OLE Automation Server, the other automation client - such as Microsoft Word, Excel and Access -allows, settle out of GUI calls. This OLE Automation Interface allows the client not only the call origination but also a Full Ring and call control.

mains operation

We now examine how the network works to typical telephony operations to realize. First was investigated as a call on LAN is discontinued. This discussion focuses on the Use of the telephone to make and receive calls. Phone calls can settle well and received using the intermediary and telephone assistant GUIs under the control of the User of the client computer. The operation is the same as described above, but with the dialing and answering calls can initiate directly on the computer with application software, the the PBX services directly through the local phone service provider <figref>84</figref> accesses (<figref idrefs="S47">7</figref>). This gives the user additional Information- and control options, as in the preceding section on GUI features and functions described.

Prefer the Users can make a call, he takes first the listener of the telephone <figref>11</figref> (<figref idrefs="S41">1</figref>) From. the Event from the control processor <figref>38</figref> the multi-port station module (<figref idrefs="S43">3</figref>) about the Telefo&lt;?page 13?&gt;never-interface circuit <figref>32</figref> and the control logic <figref>36</figref> recognized and a message will be the LAN <figref>10</figref> to the PBX software <figref>85</figref> (<figref idrefs="S47">7</figref>) sent to the server. The PBX checks its configuration database <figref>82</figref> and has possibly the client NIC <figref>30</figref> With an over the LAN <figref>10</figref> sent to message, the dial tone from the DSP <figref>33</figref> about the telephony interface <figref>32</figref> on the telephone <figref>11</figref> to . send Chooses of users on the telephone <figref>11</figref> desired by pressing the buttons Extension connection, go standard DTMF tones on the station interface <figref>32</figref> the station module <figref>30</figref> and are from DSP <figref>33</figref> detected. They are from the control processor<figref>38</figref> read and messages are the LAN and the ATM interface <figref>34</figref> to the PBX software <figref>85</figref> in the server <figref>12</figref> sent. The PBX software compares the choice with a stored table valid side port numbers and sends, if the desired Number is found, a message to the destination client PC <figref>18</figref> With the statement at the station module <figref>30</figref>, about the Phone Interface <figref>32</figref> the connected telephone <figref>11</figref> to call. If the user of the target remote speaker from the listener, this event is the target control processor <figref>38</figref> detected, of over the LAN sends an appropriate message to the PBX software. The PBX software<figref>85</figref> sends a message to the control processor <figref>38</figref> the calling and the multi-port station module and it instructs, via the LAN establish a bidirectional media stream, so that a voice transmission possible is.

At this point run voice analog signals from the microphone in the telephone <figref>11</figref> by the telephone station interface <figref>32</figref>That a standard CODEC contains. Data samples are from the control logic <figref>36</figref> the CODEC to the DSP <figref>33</figref> and control processor <figref>38</figref> given. There are voice data in the local memory of the control processor <figref>38</figref> collected, until enough for an ATM cell plus additional present data to compensate for network jitter. Then the data for a cell to the ATM interface <figref>34</figref> is given and an ATM cell via the LAN <figref>10</figref> sent to the target station module where the cell data of the ATM interface <figref>34</figref> to the memory of the control processor <figref>38</figref> given will. The data is byte-wise at the same rate of 8000 samples per second to the DSP <figref>33</figref> and the sent target CODEC and there analogized to connected to the receiver in the target telephone <figref>11</figref> sent back to become.

On reverse transmission path is also established so that bi-directional transmission is possible. Sets of a conversation participants on, this will be from the local interface <figref>32</figref> the station module <figref>30</figref> and the control logic <figref>36</figref> detected, the event from the local control processor <figref>38</figref> recognized and a message about the LAN <figref>10</figref> to the PBX software <figref>85</figref> in server <figref>12</figref> sent, in turn, both multi-port modules instructs the connection to break up.

Calls to external lines run similarly. If the user wishes to settle a reputation, he picks up the receiver on the telephone <figref>11</figref> from. The event is from the control processor of the client NIC via its Telephony interface circuit <figref>32</figref>, The control logic <figref>36</figref> and the control processor <figref>38</figref> recognized and a message about LAN <figref>10</figref> to the PBX software <figref>85</figref> sent to the server. The PBX has typi cally the client NIC with a message on the LAN, the DSP <figref>33</figref> about the telephony interface <figref>32</figref> the dial tone to the telephone <figref>11</figref> to . send Pressing of users on the telephone <figref>11</figref> the digit <figref>9</figref>, becomes a standard DTMF string on the Station Interface <figref>32</figref> the station module <figref>30</figref> cleverly and the DSP <figref>33</figref> detected. This code is from the control processor<figref>38</figref> read and a message about the LAN to the PBX software <figref>95</figref> in server <figref>12</figref> sent. numeral <figref>9</figref> typically serves to display a Außenrufs. At the Receiving the same checks the PBX software <figref>85</figref> the state of PSTN module <figref>20</figref> and takes over the first available "suspended" line, which it will, by on the local PC bus <figref>29</figref> an appropriate notification to control processor <figref>28</figref> sends. The PBX software<figref>85</figref> sends then a message to both the control processors <figref>28</figref>. <figref>38</figref> of triggering Multi-port station module and the multi-port PSTN module and comprises Build it to a bidirectional media stream. The message at the station module <figref>30</figref> is then treated with a standard protocol such as TCP / IP over the LAN <figref>10</figref> and the message to the PSTN module directly on the PC bus of the server to control processor <figref>28</figref> the multi-port PSTN module sent. The user waits for the external dial tone and then dials the desired Connection.

At this point run analog DMTF dialing signals from the client telephone <figref>11</figref> about the Telephone station interface <figref>32</figref>That a standard CODEC contains. This circuit digitizes the analog speech signal. are data samples by the control hardware to the DSP <figref>33</figref> given. There will be the voice signals in the RAM of the control processor <figref>38</figref> collected, until enough of them for an ATM cell are present, plus of data to compensate for temporal uncertainty (jitter) the case of cell transfer always occur in asynchronous LAN. The data of a cell to control processor <figref>38</figref> and then to the ATM interface and finally the cell via the LAN to the PSTN module <figref>20</figref> given. There, the cell is from control processor <figref>28</figref> and the control logic <figref>26</figref> out the ATM interface <figref>24</figref> in the buffer RAM <figref>27</figref> written. This circuit is more complex than in the client so that the PSTN module <figref>20</figref> sixteen or more two-way voice channels loss without substantial performance can efficiently manage. The&lt;?page 14?&gt;Control Hardware <figref>26</figref> takes over the data from the buffer RAM <figref>27</figref> and they are byte-wise to the target CODEC in the trunk interface <figref>22</figref>To which they back analogized and the connecting cable <figref>17</figref> the connected Remote speaker is.

Also A reverse transmission path is constructed so that a bi-directional transmission is possible. The client telephone behaves then like a normal telephone, the right external to the line connected. If a caller hangs up, the station module detected<figref>30</figref> the Event from the local control processor <figref>38</figref> is recognized, and a message goes the LAN <figref>10</figref> to the PBX software <figref>85</figref> in server <figref>12</figref>. in turn, the modules in the client <figref>30</figref> and the web <figref>20</figref> instructs To end the connection and the external conduit for other users Ready to keep.

Incoming Calls are similar treated. The call is from the PSTN module<figref>20</figref> with the line interface <figref>22</figref> and the control logic <figref>26</figref> detected; the control processor<figref>28</figref> reports the event to the PBX software <figref>85</figref>, When instructed by the PBX software, causes the control processor <figref>28</figref> a "pick up" the Office interface <figref>22</figref>, Incoming signals are in a CODEC in the trunk interface <figref>22</figref> of PSTN module digitized, as discussed above, and passed to the DSP, performs the call and DTMF ID. Typically the incumbent Telephone line management of application software for switching automatic, the appropriate voice message plays, instructing the user, enter the desired extension number with the DTMF keypad of the telephone. This notice is taken digitized voice messages, which are typically stored on the server's hard drive. Of the DSP <figref>23</figref> catches the DTMF digits, and outputs the decoded information to control processor <figref>28</figref> of PSTN module <figref>20</figref> Next, the PBX the <figref>85</figref> by means of a message on the bus <figref>29</figref> to the main processor of the notifies the server that the PBX software <figref>85</figref> performs. Has been a valid shunt detected, the PBX each station module <figref>30</figref> at, the connected telephone <figref>11</figref> to call. If a Response, this event is from the control processor <figref>38</figref> of station module <figref>30</figref> detected and a message to the PBX software <figref>85</figref> in the server <figref>12</figref> ball, which then the PSTN module <figref>20</figref> and the station module <figref>30</figref> instructs establish bidirectional media streams, so that a voice transmission possible is.

Sets one of the participants on, this is the control processor of the respective Multiport module - either right in the station module <figref>30</figref> or indirectly in the PSTN module <figref>20</figref> - Example. detected on the outside line at the new appearance of the dial tone, by means of algorithms in DSP Ruffortgangsermittlung <figref>23</figref>,

The Architecture of the ATM LAN telephony system offers several advantages: <ul><li>1. Users with both LAN and telephone connections do not require duplicate building wiring.</li><li>2. The system is cost-effective, there existing computing power exploiting synergy in LAN.</li><li>3. The software integration between computer applications and the telephony call processing is more effective, since both on the same network with the same computers work. The transfer between the two requires only a software message and -Zeichengabe with standard protocols without limiting the functionality of interfaces between each other incompatible system blocks.</li><li>4. The multi-port modules <figref>20</figref>. <figref>30</figref> support both the telephony call and voice processing. Thus, any Telephony station or line interface speech applications, Voicemail support - unlike to existing systems in which it is customary to at first the PBX to separate from the voice processing function, and this often separate systems, each with only a limited number of voice access ports. These Warranties is avoided with the architecture described herein.</li><li>5. The wide operating range of modules <figref>20</figref>. <figref>30</figref> allows add of features such as voicemail as software applications.</li><li>6. The system is on user-friendly GUIs easy to use.</li><li>7. In the LAN is carried out for both voice and data features simplified maintenance and administration.</li><li>8. The system can easily be expanded and its capacity to not Way limited.</li><li>9. Large Campus networks can be created by individual workgroups interconnected, each having a LAN PBX system with ATM connections contain between ATM switches.</li></ul>

It it should be noted that both the first as well as the general her following detailed Description as illustrative only and are exemplary and the invention as claimed is explaining should. For example. are other network technologies can be used as ATM, the a QoS support provide for real-time voice transmission. An alternative topology is ATM only for power transmission language parallel one to an existing Ethernet LAN. In a such a configuration, a telephony hub is a directly to the communication server via Ethernet interface for data connection to the existing Ethernet LAN connect. An ATM switch is to be operated only when an increasing number of Telephone necessary.

&lt;?page 15?&gt;

furthermore can digital telephones are used by a suitable digital interface - for example. a standard USB interface - used. For the trunk port would at Instead of analog digital line interfaces such as T1 and SONET usable. Numerous other modifications and variations are also possible. Accordingly, the foregoing detailed description is to be illustrative, not to be construed as limiting. The present invention incl. All equivalents is exclusive of the following claims defined.

LEGEND TO FIGURE LABEL

<figref idrefs="S41">1</figref> - LAN Phone System<dl><dt>client computers</dt><dd>Client computer</dd><dt>Internet / intranet access</dt><dd>Internet / intranet access</dd><dt>Network server</dt><dd>network Server</dd><dt>PSTN</dt><dd>public Telephone network</dd><dt>Switch</dt><dd>node (Switch)</dd><dt>Telephone stroke</dt><dd>Telephone node (Hub)</dd><dt>Telephones</dt><dd>telephone</dd></dl>

<figref idrefs="S42">2</figref> - Multi-port PSTN module<dl><dt>16</dt><dd>public Telephone network</dd><dt>22</dt><dd>Multiline Trunk Interface</dd><dt>23</dt><dd>digital signal processor</dd><dt>24</dt><dd>ATM interface</dd><dt>26</dt><dd>control logic</dd><dt>27</dt><dd>Buffer RAM</dd><dt>28</dt><dd>control processor</dd><dt>LAN server</dt><dd>LAN Server</dd></dl>

<figref idrefs="S43">3</figref> - Multi-port station module<dl><dt>11</dt><dd>telephone</dd><dt>18</dt><dd>Host pc</dd><dt>32</dt><dd>Phone Station Interface</dd><dt>33</dt><dd>digital signal processor</dd><dt>34</dt><dd>ATM interface</dd><dt>35</dt><dd>Switching Power Supply</dd><dt>36</dt><dd>control logic</dd><dt>38</dt><dd>control processor</dd></dl>

<figref idrefs="S44">4</figref> - Phone node (hub)<dl><dt>11</dt><dd>telephone</dd><dt>42</dt><dd>Phone Station Interface</dd><dt>43</dt><dd>digital signal processor</dd><dt>44</dt><dd>ATM interface</dd><dt>45</dt><dd>Switching Power Supply</dd><dt>46</dt><dd>control logic</dd><dt>47</dt><dd>Buffer RAM</dd><dt>48</dt><dd>control processor</dd></dl>

<figref idrefs="S45">5</figref> - Voice streaming<dl><dt>12, 18</dt><dd>client or server computer</dd><dt>Disk storage</dt><dd>disk</dd><dt>Voice Mail Storage ...</dt><dd>Voicemail storage and select</dd><dt>Voice / audio ...</dt><dd>Voice / NF storage and Reading, user interface, Server-PBX software</dd><dt>51</dt><dd>CODEC</dd><dt>Analog / digital ...</dt><dd>Analog / digital conversion of voice signals, possibly even reversed</dd><dt>52</dt><dd>Line interface (s)</dd><dt>53</dt><dd>Digital signal processing</dd><dt>Tone generation ...</dt><dd>tone, Detection of Caller ID, DTMF, Ruffortgang, echo cancellation and conference call</dd><dt>54</dt><dd>Network interface</dd><dt>55</dt><dd>Telephony port</dd><dt>56</dt><dd>Synchronous / asynchronous conversion</dd><dt>Conversion ...</dt><dd>implementation Language packages and vice versa, recovering the synchronization</dd><dt>57</dt><dd>LAN port</dd><dt>58</dt><dd>Module control processor</dd><dt>Cell or package ...</dt><dd>cell or packet handling, network interface, local PC interface, local telephony control</dd><dt>59</dt><dd>PC Connection</dd></dl>

<figref idrefs="S46">6</figref> - Multi-port PSTN module - control logic<dl><dt>60</dt><dd>DSP interface</dd><dt>61</dt><dd>DSP address decoder</dd><dt>62</dt><dd>CODEC shift registers</dd><dt>63a</dt><dd>DSP mailbox</dd><dt>63b</dt><dd>Control processor mailbox</dd><dt>64</dt><dd>clock generator</dd><dt>65</dt><dd>Control processor - address decoder</dd><dt>66</dt><dd>Control processor interface</dd><dt>67</dt><dd>CODEC control register interface</dd><dt>68</dt><dd>Trunk Interface Module</dd><dt>69</dt><dd>Address buffer RAM</dd></dl>

<figref idrefs="S47">7</figref> - Server Software Architecture<dl><dt>10</dt><dd>LAN</dd><dt>16</dt><dd>public Telephone network</dd><dt>19</dt><dd>Internet</dd><dt>80</dt><dd>Voicemail, mediation automatic</dd><dt>81</dt><dd>Vermittlerkonsol GUI</dd><dt>82</dt><dd>Konfiugurations database and GUI</dd><dt>83</dt><dd>CTI software</dd><dt>84</dt><dd>Telephony service providers</dd><dt>85</dt><dd>PBX-control software</dd><dt>86</dt><dd>Internet Access</dd><dt>87</dt><dd>PBX network - Rufsteuerschnittstelle</dd><dt>88</dt><dd>DSP and control processor software</dd></dl>

<figref idrefs="S48">8th</figref> - Client software architecture<dl><dt>10</dt><dd>LAN</dd><dt>90</dt><dd>Vermittlerkonsol GUI</dd><dt>91</dt><dd>Phone Assistant GUI</dd><dt>92</dt><dd>Set-up and configuration GUI</dd><dt>93</dt><dd>provider for telephony services</dd><dt>94</dt><dd>voicemail</dd><dt>95</dt><dd>DSP and control processor software</dd></dl>

<figref idrefs="S49">9</figref> - Vermittlerkonsol- / telephone assistant GUI