Patent Document

CLAIM OF PRIORITY 
   This application makes reference to, and claims all benefits accruing under 35 U.S.C. §119 from an application for VOICE OVER INTERNET PROTOCOL SYSTEM ENABLE TO CONTROL GAIN DYNAMICALLY AND METHOD THEREOF earlier filed in the Korean Intellectual Property Office on 17 Feb. 2003 and there duly assigned Serial No. 2003-9912. 
   BACKGROUND OF THE INVENTION 
   1. Field of the Invention 
   The present invention relates to a Voice over Internet Protocol system and, more particularly, to a Voice over Internet Protocol system having a dynamic gain control function and a method for providing a dynamic gain using the system wherein in the process of converting Pulse Code Modulation (PCM) data to a Voice over Internet Protocol packet or vice versa, a dynamic gain value is assigned in accordance with the type of call, and PCM data can be amplified and outputted in accordance with the assigned gain value. 
   2. Description of the Related Art 
   Generally, communication apparatuses connected to telephone lines include a general telephone, a facsimile, and so on. A signal intensity of a communication apparatus connected to a telephone line is changed in accordance with the distance from an exchange. That is, since the telephone line made up of a general copper wire has impedance values changing depending upon its length, a loop current relative to a fixed direct current (DC) of the exchange also changes. 
   Since it may be said that the change of the loop current indicates the change of the length, the change of the signal according to the distance from the exchange is proportional to the change of the loop current. So, a communication equipment user separated far from the exchange hears the other party&#39;s voice as a very low sound or never hears it under certain circumstances. 
   Accordingly, it is necessary to compensate a signal on the attenuating telephone line according to its distance from the exchange. And, it is possible to have the other party&#39;s voice heard well even in communication equipment separated far from the exchange by preparing a gain control circuit in the exchange and controlling a gain of a signal received from a terminal. 
     FIG. 1  is a view showing a representative block construction of an exemplary subscriber terminal gain control circuit of an exchange in the art. 
   Referring to  FIG. 1 , the gain control circuit of the exchange subscriber terminal includes a subscriber connector  110  for connecting the exchange to a subscriber terminal, and an amplification part  120  for amplifying a signal provided from the subscriber connector  110  to a desired level. 
   A subscriber terminal (not shown) to be subscribed and connected to the exchange is connected to the exchange through the subscriber connector  110 . 
   A voice signal applied from the terminal through the subscriber connector  110  is applied to and amplified in the amplification part  120 . Here, an amplifier  126  decides an amplification gain by a ratio of a first resistor  122 , an input resistor, and a second resistor  124 , an output resistor, and the gain can be calculated according to the following expression.
 
amplified gain=amplification gain ratio*20 Log ( R 2/ R 1)  (Expression 1)
 
   Here, R 1  and R 2  indicate an input resistor and an output resistor respectively. As shown in expression 1, the amplification gain can be variable by properly controlling the ratio of the first resistor  122  of the input resistor and the second resistor  124  of the output resistor. 
   An analog voice signal amplified and applied from the amplification part  120  is applied to PCM converter  128  to be converted to a digital signal, and the converted digital signal is applied to a controller  130  so as to perform a process needed for exchanging. 
   However, in case of using the resistor  122  and the resistor  124  which have fixed values respectively, receiver sensitivity is transmitted as it is, regardless of the condition of a cable since fixed transmission and receiver gains are maintained despite the change of input state due to an external condition. 
   Since a call gain control circuit in the art has only a unidirectional gain value using fixed elements, it becomes difficult to control the transmission and receiver gains in accordance with characteristic of each line in the real situation that may have various states. 
   In order to solve this problem, a programmable gain control circuit has been developed wherein transmission and reception lines each have a number of resistors which are the same for each line in number and serially connected to each line and on/off switches are connected in parallel to the resistors so that the on/off of the switches are controlled by the controller in accordance with states of the cables and then the values of the resistors to determine the controlled transmission and receiver gains. 
   On the other hand, a Voice over Internet Protocol (referred to as a VoIP, hereinafter) is a communication service of new mode wherein a voice communication is performed not through an existing communication network, a Public Switch Telephone Network (referred to as a PSTN, hereinafter), but through an Internet network. Since the communication method using the Internet network uses a packet-based network different from existing communication methods, a user does not have to pay for charges of domestic/international phone lines separately so that it is possible to perform the voice communication at a lower fare. 
   The VoIP has a faculty of transmitting video information as well as audio information using an H.323 Protocol being an ITU-T (International Telecommunication Union—Telecommunication) standard that provides fundamental principles for voice, video, and data communications over the IP (Internet Protocol) network including the Internet. One of the H.323 entities defined in the H.323 protocol is a gatekeeper. The gatekeeper binds H.323 endpoints present in a packet-based network (i.e., an IP-based network) in one control zone defined as a “Zone”, and then controls/manages the bound H.323 endpoints. 
   A VoIP system using the Internet network as a back-bone has an exemplary construction as shown in  FIG. 2 . 
   Referring to  FIG. 2 , the VoIP system has an Internet  208  used as the back-bone, and the Internet  208  is connected to gateways  206  and  210  and to personal computers  216  and  218  (referred to as PCs, hereinafter). 
   The gateways  206  and  210  are correspondingly connected to PSTNs  204  and  212  that also are connected to telephone terminals  202  and  214  respectively. Terminals such as the phone terminals  202  and  214  and the PCs  216  and  218  are endpoints which are capable of communicating by voices (essential), images (option) and data (option) during a one-to-one communication or a conference. 
   Such terminals can perform a real-time and a bidirectional communication with the gateways  206  and  210  and other terminals. The gateways  206  and  210  are elements that enable terminals (for example, PCs  216  and  218 ) connected to the Internet  208  being a packet-based network and terminals (for example, telephone terminals  202  and  214 ) connected to the PSTNs  204  and  212  or an Integrated Service Digital Network (referred to as an ISDN, hereinafter) to perform the real-time and the bidirectional communication. 
   Briefly, gateways  206  and  210  perform a real-time compression and a protocol transformation of voices and facsimile data inputted from the PSTNs  204  and  212  and send the data to the Internet  208 . 
   The Internet phone (IP-phone) can be classified into 3 types according to the kind of terminal used at both ends, that is, PC to PC, PC to phone and phone to phone. 
   Generally, the technical principle of the Internet phone is made up of a voice encoding and compressing technology, a real-time data transmission technology, a packet recovery technology, a gateway technology, and so on. 
   The voice encoding technology employs a low bit rate, high compression rate and high voice quality encoding technique in order to transmit the voice information without damaging the voice quality. 
   The voice encoding technology includes a PCM, an adaptive prediction coding, a Global System for Mobile communication (referred to as GSM, hereinafter), a Linear Predictive Coding (referred to as LPC, hereinafter), and so on, and the above technologies are now used. The real-time transformation technology includes a Real Time Transport Protocol (referred to as RTP, hereinafter). The RTP receives much recognition in the transmission quality over the Internet and is mainly used since 1995. 
   Also, a gateway embodying technology is to embody a gateway which is a network connection apparatus transforming analog voice information in order to transmit the information from an existing line exchange network to a packet exchange network. 
   A basic function of the gateway is processed in a digital signal processor. The gateway performs a voice compression capability using a compression algorithm, a waiting capability, and a removal capability so that it is possible to transform and transmit PCM voice data to a VoIP packet, and vice versa. 
   Here, in the process that the digital signal processor transforms the PCM data into the VoIP packet by performing the voice compression capability using the compression algorithm, the waiting capability, and the removal capability, the strength of voice may become too low or too high. Therefore, the gateway has a capability of controlling a gain value before compressing the PCM data to the VoIP packet, like the gain controlling circuit in the exchange described above. 
   However, since the gain value in the gateway is already assigned by the operator as a fixed value (of course, the gain value is generally determined by a test according to an environmental characteristic of the gateway), it is difficult to dynamically change the gain value according to the kind of call. 
   SUMMARY OF THE INVENTION 
   Therefore, the present invention has been made in view of the above problem, and it is an object of the present invention to provide a VoIP system having a dynamic gain control capability and a method for providing a dynamic gain using the system wherein in case that the VoIP gateway operates in an interworking state with a key telephone or a private exchange, a gain value can be dynamically assigned according to the type of call. 
   In accordance with an aspect of the present invention, there is provided a Voice over Internet Protocol (VoIP) system having a dynamic gain control function, comprising: a private exchange, on receiving a call setup signal from an extension telephone, for determining the type of telephone using an extension number of the telephone and transmitting a gain value assigned according to the type of determined telephone; and a VoIP gateway for a) receiving the gain value from the private exchange, storing the value, amplifying PCM voice data transmitted from the private exchange according to the stored gain value when a call setup is completed, transforming the amplified PCM voice data into a VoIP packet and transmitting the packet to a receiver, and b) transforming the VoIP packet transmitted from the receiver into PCM voice data, amplifying the transformed PCM voice data according to the stored gain value, and transmitting the amplified data to the private exchange. 
   In accordance with another aspect of the present invention, there is provided a Voice over Internet Protocol (VoIP) system having a dynamic gain control function, comprising: a private exchange, on receiving a call setup signal from an extension telephone, for determining the type of receiver using a telephone number of the receiver and transmitting a gain value assigned according to the type of determined receiver; and a VoIP gateway for a) receiving the gain value from the private exchange, storing the value, amplifying PCM voice data transmitted from the private exchange according to the gain value stored when a call setup is completed, transforming the amplified PCM voice data into a VoIP packet and transmitting the packet to the receiver, and b) transforming the VoIP packet transmitted from the receiver into PCM voice data, amplifying the transformed PCM voice data according to the stored gain value, and transmitting the amplified data to the private exchange. 
   In accordance with another aspect of the present invention, there is provided a Voice over Internet Protocol (VoIP) system having a dynamic gain control function, comprising: a private exchange, on receiving a call setup signal from an extension telephone, for determining the type of telephone using an extension number, determining the type of receiver using a telephone number of the receiver, and transmitting a gain value assigned according to the type of determined receiver and the receiver; and a VoIP gateway for a) receiving the gain value from the private exchange, storing the value, amplifying PCM voice data transmitted from the private exchange according to the stored gain value when a call setup is completed, transforming the amplified PCM voice data into a VoIP packet and transmitting the packet to the receiver, and b) transforming the VoIP packet transmitted from the receiver into the PCM voice data, amplifying the transformed PCM voice data according to the stored gain value, and transmitting the amplified data to the private exchange. 
   Preferably, the system in accordance with the present invention further comprises a gatekeeper for, on receiving the call setup signal whose receiver is the extension telephone of the private exchange, determining the type of telephone using the telephone number of the receiver and transmitting the gain value assigned according to the type of determined telephone to the VoIP gateway, and wherein the VoIP gateway a) stores the gain value transmitted from the gatekeeper, transforms the VoIP packet transmitted from the sender to the PCM voice data when the call setup is completed, amplifies the transformed PCM voice data and transmits the data to the private exchange, and b) amplifies the PCM voice data transmitted from the private exchange according to the stored gain value, transforms the amplified data into the VoIP packet and transmits the data to the sender. 
   Preferably, the system in accordance with the present invention further comprises a gatekeeper for, on receiving the call setup signal whose receiver is the extension telephone of the private exchange, determining the type of sender using an IP address of the sender, and transmitting the gain value assigned according to the determined sender to the VoIP gateway, and wherein the VoIP gateway a) stores the gain value transmitted from the gatekeeper, transforms the VoIP packet  11  transmitted from the sender to the PCM voice data when the call setup is completed, amplifies the transformed PCM voice data and transmits the data to the private exchange, and b) amplifies the PCM voice data transmitted from the private exchange according to the stored gain value, transforms the amplified data into the VoIP packet and transmits the data to the sender. 
   Preferably, the system in accordance with the present invention further comprises a gatekeeper for, on receiving the call setup signal whose receiver is the extension telephone of the private exchange, determining the type of telephone of the receiver using the telephone number of the receiver, determining the type of sender using the IP address of the sender, and transmitting the gain value assigned according to the type of telephone of the determined sender and receiver to the VoIP gateway, and wherein the VoIP gateway a) stores the gain value transmitted from the gatekeeper, transforms the VoIP packet transmitted from the sender to the PCM voice data when the call setup is completed, amplifies the transformed PCM voice data and transmits the data to the private exchange, and b) amplifies the PCM voice data transmitted from the private exchange according to the stored gain value, transforms the amplified data into the VoIP packet and transmits the data to the sender. 
   Preferably, in the system in accordance with the present invention, the VoIP gateway includes: a Public Switched Telephone Network (PSTN) connector for providing an interface with the private exchange; an Internet network connector for providing a connection with an Internet network; a media processor for a) amplifying the PCM voice data transmitted from the private exchange through the PSTN connector according to the established gain value when the call setup is completed, transforming the amplified data into the VoIP packet and transmitting the packet to the receiver through the Internet network connector, and b) transforming the VoIP packet transmitted from the receiver through the Internet network connector into the PCM voice data, amplifying the transformed PCM voice data according to the gain value, and transmitting the amplified data to the private exchange through the PSTN connector; and a main controller for receiving the gain value from the private exchange through the PSTN connector, storing the gain value, providing the media processor with the gain value when the call setup is completed, and amplifying the PCM voice data according to the stored gain value. 
   Preferably, in the system in accordance with the present invention, the media processor includes: a memory for storing the gain value transmitted from the main controller; a digital signal processor for a) amplifying the PCM voice data transmitted from the private exchange through the PSTN connector according to the gain value when the call setup is completed, transforming the amplified data into the VoIP packet and transmitting the packet to the receiver through the Internet network connector, and b) transforming the VoIP packet transmitted from the receiver through the Internet network connector into the PCM voice data, amplifying the transformed PCM voice data according to the gain value and transmitting the data to the PSTN connector; and a controller for receiving the gain value from the main controller, storing the gain value in the memory, providing the digital signal processor with the gain value when the call setup is completed, and amplifying the PCM voice data according to the gain value. 
   Preferably, in the course that the private exchange transmits the gain value to the VoIP gateway, the private exchange transmits the gain value using a call setup message. 
   In accordance with another aspect of the present invention, there is provided a Voice over Internet Protocol (VoIP) system having a dynamic gain control function, comprising: a gatekeeper for, on receiving a call setup signal whose receiver is an extension telephone of a private exchange, determining the type of sender using a telephone number of the sender and transmitting a gain value assigned according to the type of determined sender; and a VoIP gateway for a) storing the gain value transmitted from the gatekeeper, transforming a VoIP packet transmitted from the sender into PCM voice data when a call setup is completed, and amplifying the transformed PCM voice data and transmitting the data to the private exchange, and b) amplifying the PCM voice data transmitted from the private exchange according to the stored gain value, transforming the data into the VoIP packet and transmitting the packet to the sender. 
   In accordance with another aspect of the present invention, there is provided a Voice over Internet Protocol (VoIP) system having a dynamic gain control function, comprising: a gatekeeper for, on receiving a call setup signal whose receiver is an extension telephone of a private exchange, determining the type of sender using an IP address of the sender and transmitting a gain value assigned according to the determined sender; and a VoIP gateway for a) storing the gain value transmitted from the gatekeeper, transforming a VoIP packet transmitted from the sender into PCM voice data when a call setup is completed, amplifying the transformed PCM voice data and transmitting the data to the private exchange, and b) amplifying the PCM voice data transmitted from the private exchange according to the stored gain value, transforming the data into the VoIP packet, and transmitting the packet to the sender. 
   In accordance with another aspect of the present invention, there is provided a Voice over Internet Protocol (VoIP) system having a dynamic gain control function, comprising: a gatekeeper for, on receiving a call setup signal whose receiver is an extension telephone of a private exchange, determining the type of telephone of a receiver using a telephone number of the receiver, determining the type of sender using an IP address of the sender, and transmitting a gain value assigned according to the type of telephone of the determined receiver and the determined sender; and a VoIP gateway for a) storing the gain value transmitted from the gatekeeper, transforming a VoIP packet transmitted from the sender into the PCM voice data when a call setup is completed, amplifying the transformed PCM voice data, and transmitting the data to the private exchange, and b) amplifying the PCM voice data transmitted from the private exchange according to the stored gain value, transforming the data into the VoIP packet, and transmitting the packet to the sender. 
   Preferably, in the course that the gatekeeper transmits the gain value to the VoIP gateway, the gatekeeper transmits the gain value using a call setup message. 
   In accordance with another aspect of the present invention, there is provided a method for providing a gain dynamically in a Voice over Internet Protocol (VoIP) system, comprising the steps of: when receiving a call setup signal from an extension telephone, determining the type of telephone by a private exchange using an extension number of the extension telephone; transmitting a gain value assigned according to the type of telephone determined in the step a) to a VoIP gateway by the private exchange; and by the VoIP gateway, storing the gain value transmitted from the private exchange and amplifying PCM voice data according to the gain value when a call setup is completed. 
   In accordance with another aspect of the present invention, there is provided a method for providing a gain dynamically in a Voice over Internet Protocol (VoIP) system, comprising the steps of: a) when receiving a call setup signal from an extension telephone, determining the type of receiver using a number of the receiver by a private exchange; b) transmitting a gain value assigned according to the type of receiver determined in the step a) to a VoIP gateway by the private exchange; and c) by the VoIP gateway, storing the gain value transmitted from the private exchange and amplifying PCM voice data according to the gain value when a call setup is completed. 
   In accordance with yet another aspect of the present invention, there is provided a method for providing a gain dynamically in a Voice over Internet Protocol (VoIP) system, comprising the steps of: a) when receiving a call setup signal from an extension telephone, by a private exchange, identifying the type of telephone of a sender using the extension number of the extension telephone and determining the type of telephone of a receiver using the number of the receiver; b) transmitting a gain value assigned according to the type of telephone of the sender and the type of telephone of the receiver determined in the step a) to a VoIP gateway by the private exchange; and c) by the VoIP gateway, storing the gain value transmitted from the private exchange and amplifying PCM voice data according to the gain value when a call setup is completed. 
   Preferably, the step c) includes the steps of: c-1) storing the gain value transmitted from the private exchange; c-2) amplifying the PCM voice data transmitted from the private exchange according to the gain value, transforming the data into a VoIP packet and transmitting the packet to the receiver, after the call setup is completed; and c-3) transforming the VoIP packet transmitted from the receiver into the PCM voice data, amplifying the transformed PCM voice data according to the gain value and transmitting the data to the private exchange. 
   Preferably, the method in accordance with the present invention further comprises the steps of: d) on receiving the call setup signal whose receiver is the extension telephone of the private exchange, determining the type of telephone using the telephone number of the receiver by the gatekeeper; transmitting the gain value assigned according to the determined type of telephone to the VoIP gateway by the gatekeeper; and by the VoIP gateway, storing the gain value transmitted from the gatekeeper and amplifying the PCM voice data according to the stored gain value when the call setup is completed. 
   Preferably, the method in accordance with the present invention further comprises the steps of: d) on receiving the call setup signal whose receiver is the extension telephone of the private exchange, determining the type of sender using an IP address of the sender by the gatekeeper; transmitting the gain value assigned according to the type of sender determined in the step d) to the VoIP gateway by the gatekeeper; and by the VoIP gateway, storing the gain value transmitted from the gatekeeper and amplifying the PCM voice data according to the stored gain value when the call setup is completed. 
   Preferably, the method of the present invention further comprises the steps of: d) on receiving the call setup signal whose receiver is the extension telephone of the private exchange, determining the type of receiver using the telephone number of the receiver and determining the type of sender using the IP address of the sender, by the gatekeeper; e) transmitting the gain value assigned according to the type of telephone of the receiver and the type of sender determined in the step d) to the VoIP gateway by the gatekeeper; and by the VoIP gateway, storing the gain value transmitted from the gatekeeper and amplifying the PCM voice data according to the stored gain value when the call setup is completed. 
   Preferably, the method of the present invention further comprises the steps of: d) on receiving the call setup signal whose receiver is the extension telephone of the private exchange, determining the type of telephone using the telephone number of the receiver by the gatekeeper; e) transmitting the gain value assigned according to the type of telephone determined in the step d) to the VoIP gateway by the gatekeeper; and f) by the VoIP gateway, storing the gain value transmitted from the gatekeeper and amplifying the PCM voice data according to the stored gain value when the call setup is completed. 
   Preferably, the method of the present invention further comprises the steps of: d) on receiving the call setup signal whose receiver is the extension telephone of the private exchange, determining the type of sender using the IP address of the sender by the gatekeeper; transmitting the gain value assigned according to the type of sender determined in the step d) to the VoIP gateway by the gatekeeper; and storing the gain value transmitted from the gatekeeper and amplifying the PCM voice data according to the stored gain value when the call setup is completed by the VoIP gateway. 
   Preferably, the method of the present invention further comprising the steps of: d) on receiving the call setup signal whose receiver is the extension telephone of the private exchange, determining the type of telephone of the receiver using the telephone number of the receiver, and determining the type of sender using the IP address of the sender by the gatekeeper; transmitting the gain value assigned according to the type of telephone of the receiver and the type of sender determined in the step d) to the VoIP gateway by the gatekeeper; and storing the gain value transmitted from the gatekeeper and amplifying the PCM voice data according to the stored gain value when the call setup is completed, by the VoIP gateway. 

   
     BRIEF DESCRIPTION OF THE DRAWINGS 
     A more complete appreciation of the present invention, and many of the attendant advantages thereof, will become readily apparent as the same becomes better understood by reference to the following detailed description when considered in conjunction with the accompanying drawings in which like reference symbols indicate the same or similar components, wherein: 
       FIG. 1  is a view showing a representative block construction of an exemplary subscriber terminal gain control circuit of an exchange in the art; 
       FIG. 2  is a view showing an exemplary construction of a VoIP system using a general Internet network as a back-bone; 
       FIG. 3  is a view showing a VoIP system having a dynamic gain control capability in accordance with a preferred embodiment of the present invention; 
       FIG. 4  is a view showing an inner block diagram of a VoIP gateway; 
       FIG. 5  is a view showing a detailed block construction of an IP network connector and a media processor shown in  FIG. 4 ; 
       FIG. 6  is a view showing an operation flow chart of a method for providing a dynamic gain in a sender using a VoIP system in accordance with an embodiment of the present invention; and 
       FIG. 7  is a view showing an operation flow chart of a method for providing a dynamic gain in a receiver using a VoIP system in accordance with another embodiment of the present invention. 
   

   DETAILED DESCRIPTION OF THE PREFERRED EMBODIMENTS 
   Now, a VoIP system having a dynamic gain control capability and a method for providing a dynamic gain using the system in accordance with a preferred embodiment of the present invention will be described in detail with reference to the annexed drawings. 
     FIG. 3  is a view showing a VoIP system having a dynamic gain control capability in accordance with a preferred embodiment of the present invention. 
   Referring to  FIG. 3 , a VoIP system having a dynamic gain control capability in accordance with a preferred embodiment of the present invention includes a key telephone/private exchange  330 , a VoIP gateway  340 , and a gatekeeper  350 . 
   The key telephone/private exchange  330  is a telephone exchange system which enables a predetermined number of external telephone lines used in enterprise fields such as a public office, a company, a factory and a hotel to be shared with all members, and phone calls between internal users connected to extension lines to be connected automatically. 
   A main object of the key telephone/private exchange  330  is to reduce an expenditure occurring in case that all members of a public office, a company, a factory and a hotel have their own general telephone lines. 
   The key telephone/private exchange  330  is owned not by a telephone company but privately owned by a public office, a company, a factory and a hotel and is managed on their own responsibility. The key telephone/private exchange  330  employed an analog mode originally but the trend of it is recently changing to a digital mode. 
   The key telephone/private exchange  330  has an analog telephone  310  and a digital telephone  320  which are connected to it, and it provides exchange connections between one extension telephone  310  and another extension telephone  320  or between the extension telephones  310  and  320  and external telephone lines (telephone lines of telephone office via). 
   Here, the analog telephone  310  is a telephone used in a general home and is connected to the key telephone/private exchange  330  through an analog interface. The digital telephone  320  is a special telephone which is manufactured in order to make full use of the key telephone/private exchange, and is connected to key telephone/private exchange  330  through a digital interface. 
   The key telephone/private exchange  330  receives, from a manager, information on whether the connected telephone is the analog telephone  310  or the digital telephone  320 , makes a database by correlating the information with the extension numbers and stores the database in it. When a call setup request is made from the connected telephones  310  and  320 , the key telephone/private exchange  330  looks up the extension number and determines whether the telephone currently requesting the call setup is the analog telephone  310  or the digital telephone  320 . 
   The information on the kind of telephone correlated with the extension number is transmitted to the gatekeeper  350  coupled tightly with the key telephone/private exchange  330  and is managed in it. 
   The key telephone/private exchange  330  manages a gain table shown in Table 1, below, and this gain table is used to determine an amplification ratio when the PCM voice data are amplified on the VoIP gateway  340  in accordance with the kind of sender telephones  310  and  320 . 
   When the key telephone/private exchange  330  senses a call setup request from an extension line, it determines whether the sender telephone is an analog telephone  310  or a digital telephone  320  by identifying the extension telephone number. And, the key telephone/private exchange  330  looks up an IP address translation table transmitted from the gatekeeper  350  so as to be a database and managed by it, and determines whether the receiver (called) telephone  360  is an IP telephone  360  or a VoIP gateway  400 . Here, using the called telephone number, the key telephone/private exchange finds the IP address corresponding to it and determines whether the receiver is an IP telephone  360  or a VoIP gateway  400 . On the basis of the determination result, the key telephone/private exchange  330  looks up the gain table of Table 1, identifies the gain value and transmits the identified gain value to the VoIP gateway  340 . 
   The key telephone/private exchange  330  transmits the gain table information of Table 1 to the gatekeeper  350  so as to be stored in a database and managed by it. 
   
     
       
             
             
             
             
             
           
         
             
               TABLE 1 
             
             
                 
             
             
               Receiver/ 
               Analog 
               Digital 
                 
               External VoIP 
             
             
               sender 
               telephone 
               telephone 
               IP telephone 
               gateway 
             
             
                 
             
           
           
             
               Analog 
               X 
               X 
               9 db 
               10 db 
             
             
               telephone 
             
             
               Digital 
               X 
               X 
               8 db 
               13 db 
             
             
               telephone 
             
             
               IP telephone 
                9 db 
                8 db 
               X 
               X 
             
             
               External VoIP 
               10 db 
               13 db 
               X 
               X 
             
             
               gateway 
             
             
                 
             
           
        
       
     
   
   As described above, the key telephone/private exchange  330  is installed in a building or a predetermined place and employs an external telephone line ‘endowed a telephone number’ from the telephone station. And, the key telephone/private exchange  330  enables the extension lines endowed their own numbers to communicate freely using the analog telephone  310  or the digital telephone  320  and provides various functions including switchover, holding, broadcasting, conference, and so on so that users can perform their businesses with ease and efficiency. The key telephone/private exchange  330  also identifies gain values and informs the VoIP gateway  340  of the gain values. So, the VoIP gateway  340  can look up the gain values when it compresses PCM voice data. 
   The key telephone/private exchange  330  can be connected to the VoIP gateway  340  using a digital line such as an E1/T1  370  and an analog line such as a loop line. 
   The key telephone/private exchange  330  can be connected to another VoIP gateway  400  outside through a PSTN (public switch telephone network)  401  using an E1/T1  390  or a loop line as an external telephone line trunk. 
   The VoIP gateway  340  is adapted to connect the PSTN to the VoIP network. Generally, in order to embody a VoIP voice communication, it is needed to employ protocols for controlling is call, for example, a protocol SS7 (Signaling System 7) for controlling call of the PSTN, an H.323 protocol for Internet, an SIP (Session Initiation Protocol) and so on. 
   The VoIP gateway  340  is needed to control an inter-transformation between a call control protocol used for an Internet and a PSTN when both networks are interworked and the media. 
   Generally, the VoIP gateway  340  can be classified according to its service. For example, the VoIP gateway  340  includes a built-in type gateway which is mounted on a Key Telephone System (KTS) or a Private Branch exchange (PBX)  330  as a card form, a server type gateway which is mounted on a platform such as a window Network Terminal (NT), a stand-alone type gateway which is independently constructed from other terminals, and so on. 
   The stand-alone type gateway can be classified into a TANDEM (Trunk and ENM (ear &amp; mouth)) function and a stand-alone function. The TANDEM function stand-alone gateway is a stand-alone gateway for supporting an interworking between heterogeneous lines. 
   The TANDEM function stand-alone gateway is connected to a private exchange and/or a key telephone system  330  through an internal T1/E1 interface, a loop start trunk interface and an SLC (Subscriber Line Circuit) interface. 
   The stand-alone type gateway of stand-alone function is connected to a plurality of telephone terminals directly. Accordingly, in connection with the present invention, the VoIP gateway  340  may be the built-in type gateway and the server type gateway which is mounted on a platform such as a window NT (window Network Terminal), and a TANDEM type gateway among the stand-alone type gateways which are independently constructed from other terminals. 
   The VoIP gateway  340  is connected to gatekeeper  350  through an Internet (IP) network  380 . 
   Main functions of the VoIP gateway  340  are to compress the PCM voice data received from the key telephone/private exchange  330 , transform the data into VoIP packets and transmit them to over the Internet network  380 , or transform the VoIP packet received from the Internet network  380  into PCM data and transfer the data to the key telephone/private exchange  330 . 
   Here, the VoIP gateway  340  stores a gain value transmitted from the key telephone/private exchange  330 . And then, in case that a call setup is completed and there exists a voice data exchange, when transforming PCM voice data into a VoIP packet, the VoIP gateway  340  amplifies and transforms the PCM voice data according to the stored gain value. And, in case of transforming the VoIP packet into the PCM voice data and outputting them, the key telephone/private exchange  330  amplifies the transformed PCM voice data and outputs them. 
   Of course, in case that a sender is not the key telephone/private exchange  330  but an IP telephone  360  or an external VoIP gateway  400 , the VoIP gateway  340  stores a gain value transmitted from the gatekeeper  350 , and then in case that the call setup is completed and there exists the voice data exchange, when transforming the PCM voice data into the VoIP packet, amplifies and outputs the PCM data according to the stored gain value. And, in case of transforming the VoIP packet into the PCM voice data and outputting the data, the VoIP gateway  340  amplifies the transformed PCM voice data and outputs them. 
   The gatekeeper  350  is one of H.323 Entity which is defined in the H.323 protocol being a multimedia communication standard of ITU-T, which is an apparatus for controlling, managing and integrating H.323 end points (gateway, terminal, MCU, and so on) existing in a packet-based network after making them one control area defined as a Zone. 
   Main functions of the gatekeeper  350  include an address translation function for translating the Alias name or a destination name into a network (IP) address name, a bandwidth control function of a call authentication (RAS) function for properly distributing a protocol related with the gatekeeper  350  and a bandwidth being a limited resource of a network to each end point in the H.323 of Registration/Admission/Status (RAS) and checking if they reaches to a limit values and then performing a blocking, a call control function for connecting/releasing call between one end point and another end point, and additional maintenance functions such as billing, statistics, and so on. 
   Such a gatekeeper  350  is connected to the IP telephone  360  through the VoIP gateways  340  and  400  through the IP network  380 . 
   The gatekeeper  350  manages the IP address translation table used for mapping the Internet telephone number and the IP address in order to perform the address translation function, which transmits IP address translation table information to the key telephone/private exchange  330  so that the key telephone/private exchange  330  can make the IP address translation table its database to be stored and managed. 
   When the gatekeeper  350  receives a call setup request for sharing the VoIP gateway  340  from the IP telephone  360  or the external VoIP gateway  400 , it analyses a sender IP address and determines whether the sender is the IP telephone  360  or the external VoIP gateway  400 . The gatekeeper  350  identifies information on the type of telephone related the extension number which is transmitted from the key telephone/private exchange  330  (here, called extension telephone number is used) and determines whether the called extension telephone is the analog telephone  310  or the digital telephone  320 . 
   Then, the gatekeeper  350  obtains the gain value by looking up the gain table (Table 1) transmitted from the key telephone/private exchange  330 , and transmits the obtained gain value to the VoIP gateway  340 . 
   Here, the IP telephone  360  is also called an Internet telephone which enables users to perform a voice communication in the VoIP net. 
   Now, an operation of the VoIP system in accordance with an embodiment of the present invention having the construction described above will be explained. 
   When a user makes a phone call using the analog telephone  310  or the digital telephone  320  and a call passing the VoIP gateway  340  is generated, the key telephone/private exchange  330  identifies the extension number of the telephone making a phone call and determines if the sender telephone is the analog telephone  310  or the digital telephone  320 . 
   Then, the key telephone/private exchange  330  searches for a called telephone number by looking up the IP address translation table transmitted from the gatekeeper  350  and determines if the receiver is the IP telephone  360  or the external VoIP gateway  400 . 
   After then, the key telephone/private exchange  330  identifies a gain value by looking up the gain table (Table 1) and transmits the gain value together with a call setup signal. Of course, the key telephone/private exchange  330  may transmit the call setup signal to the VoIP gateway  340  first, and then transmit the gain value using a special message. 
   As an example, when a user makes a phone call to the IP telephone  360  using the analog telephone  310 , the key telephone/private exchange  330  transmits a gain value of 9 db to the VoIP gateway  340 . 
   As an example, also, when a user makes a phone call to the external VoIP gateway  400  using the digital telephone  320 , the key telephone/private exchange  330  transmits a gain value of 13 db to the VoIP gateway  340 . 
   The VoIP gateway  340  which has received a call setup signal including a gain value from the key telephone/private exchange  330  extracts the gain value included in the call setup signal transmitted and stores it in a memory, and transmits the call setup message to the IP telephone  360  or the external VoIP gateway  400 . Of course, in case of receiving the gain value from the key telephone/private exchange  330  through a special message, it is possible to extract the gain value from the message transmitted and store it in a memory. 
   When the VoIP gateway  340  tries to compress and transmit the PCM voice data using a codec in case that a call setup is normally made to the IP telephone  360  or the external VoIP gateway  400 , it amplifies the PCM voice data according to the amplification ratio determined in response to the stored gain value before compressing them and then compresses and transmits the amplified PCM data using the codec. 
   Also, the VoIP gateway  340  transforms the VoIP packet transmitted from the called IP telephone  360  or external VoIP gateway  400  into the PCM voice data and then, according to the amplification ratio determined in response to the stored gain value, amplifies and transmits the PCM voice data to the key telephone/private exchange  330 . 
   In case that an external call is received, for example, when the VoIP gateway  340  receives a call setup signal transmitted from the IP telephone  360  or the VoIP gateway  400 , the call setup signal passes through the gatekeeper  350 . Here, the gatekeeper  350  identifies the gain value and transmits it to the VoIP gateway  340 . 
   At first, when the gatekeeper  350  receives a signal for requesting a call setup with the telephones  310  and  320  connected to the key telephone/private exchange  330  from the IP telephone  360  or the external VoIP gateway  400 , it identifies a sender IP address and determines whether the sender is the IP telephone  360  or the VoIP gateway  400 . 
   The gatekeeper  350  identifies an IP address included in the call setup signal and determines whether the sender requesting the call setup is the IP telephone  360  or the VoIP gateway  400 . 
   The gatekeeper  350  identifies the called telephone number and determines whether the receiver is the analog telephone  310  or the digital telephone  320  using information on the type of telephone related with the extension telephone number received from the key telephone/private exchange  330 . 
   The gatekeeper  350  obtains a gain value by looking up the gain table (Table 1) transmitted from the key telephone/private exchange  330 , and transmits the obtained gain value to the VoIP gateway  340 . 
   Then, in case that a call setup is normally made to the analog telephone  310  or the digital telephone  320 , the VoIP gateway  340  transforms the VoIP packet into PCM voice data and then, according to the gain value, amplifies and outputs the transformed PCM voice data. 
   Also, when the VoIP gateway  340  tries to compress and transmit the PCM voice data using the codec, it amplifies the PCM voice data according to the amplification ratio determined in response to the stored gain value before compressing the PCM voice data, and then compresses and transmits the amplified PCM data using the codec. 
   On the other hand, though the gain value of this embodiment is determined in consideration of the sender and the receiver both, it may be possible to determine the gain value by merely referring to the type of telephones  310  and  320  connected to the key telephone/private exchange  330 . 
     FIG. 4  is a view showing an inner block diagram of a VoIP gateway. 
   Referring to  FIG. 4 , the VoIP gateway includes a subscriber line connector  402 , a switch  403  for connecting between subscribers who are connected through lines, a media processor  405  for compressing and decompressing common voice, a PSTN connector  407  for connecting to the PSTN and receiving an E1/T1 digital line of a key telephone/private exchange  408 , an IP network connector  406  for connecting to the IP network  409 , and a main controller  404 . 
   In the VoIP gateway described above, when the PSTN connector  407  interfaced with the key telephone/private exchange  408  requires the main controller  404  to make a call setup with the IP network  409 , the main controller  404  checks the state of the IP network  409  and then performs the call setup to the IP network  409  through the IP network connector  406 . 
   The media processor  405  compresses the PCM voice data inputted from the key telephone/private exchange  408  through the PSTN connector  407 , and then transmits the data to the IP network connector  406 . And, the media processor  405  also decompresses packet data inputted from the IP connector  406  and relays the call to the key telephone/private exchange  408  through the PSTN connector  407 . 
   Here, the main controller  404  extracts and stores a gain value included in a call setup message received from the key telephone/private exchange  408  (a special message may be used) and then provides the media processor  405  with the stored gain value when the call setup is completed. 
   Then, the media processor  405  amplifies the PCM voice data inputted according to the gain value, and compresses and transmits the amplified PCM voice data as VoIP packets. 
   Also, the media processor  405  transforms the VoIP packet into the PCM voice data according to the gain value, and then amplifies and outputs the transformed PCM voice data. 
   On the other hand, when the main controller  404  received an external call setup signal through the IP network connector  406  (of course, the main controller may receives the gain value from the gatekeeper using a special message), it extracts the gain value included in the call setup signal and stores the value, and provides the media processor  405  with the gain value stored after the call setup is completed. 
   Then, the media processor  405  transforms the VoIP packet into the PCM voice data according to the gain value, and then amplifies and outputs the transformed PCM voice data. 
   Also, the media processor  405  amplifies the PCM voice data inputted according to the gain value, and then compresses and transmits the amplified PCM voice data as VoIP packets. 
     FIG. 5  is a view showing a detailed block construction of the IP network connector and the media processor shown in  FIG. 4  and connected to PSTN connector  407  and key telephone/private exchange  408 . 
   Referring to  FIG. 5 , the IP network connector  406  includes a central processing unit (CPU)  511 , a memory  512  and a Compact Peripheral Component Interconnect (cPCI) bridge  513 . And, the media processor  405  includes a digital signal processor (DSP)  521 , a central processing unit (CPU)  522 , a memory  523 , and a cPCI bridge  524 . The cPCI bridges  513  and  524  are constituents used to match CPU busses. 
   Here, since a normal media processor  405  has a lower degree of integration of a channel than the IP network connector  406 , it is common that one IP network connector  406  is matched with a number of media processors  405 . At that time, there occurs a task that a number of hardware PBA (Printed Board Assembly) should be matched with a common bus for voice traffic in a backplane in order that the IP network connector  406  transmits and receives the voice traffic to and from the media processor  405 . 
   It is common to use a cPCI bus in the aspect of a bus band and operating with a current level of technology, and it is necessary to use the CPUs  511  and  522  for operating the bus in case of using the cPCI bus. Here, the CPUs  511  and  522  should be used for transferring voice traffic and also for operating the bus. 
   The CPU  511  of the IP network connector  406  is used to process the IP protocol and to operate the cPCI bus, and the cPCI bridges  513  is used to match the CPU bus with cPCI bus. 
   The DSP  521  of the media processor  405  is an essential constituent for embodying vocoding function, and the CPU  522  of the media processor  405  is an essential constituent for controlling the cPCI bridge  524 . 
   The memories  512  and  523  are essential constituents which act as buffers for transmitting and receiving data processed in the CPUs  511  and  522  and the DSP  521 . 
   The cPCI bus is made up of a master and a number of targets wherein an IP network connector  406  acts as the master and a number of media processors act as the targets, in the conventional art. 
   In connection with the present invention, the CPU  522  of the media processor  405  receives a gain value from the main controller  404  of the VoIP gateway and stores it in the memory  523 , and in case of compressing the PCM data inputted from a PSTN connector  407  into the VoIP packet, controls the DSP  521  so that the DSP  521  amplifies the PCM data according to the gain value and then compresses the amplified data. 
   When the CPU  522  of the media processor  405  transforms the VoIP packet inputted from the IP network connector  406  ( FIG. 4 ) into the PCM data, it controls the DSP  521  so that the DSP  521  amplifies the PCM data according to the gain value stored and outputs the data. 
     FIG. 6  is a view showing an operation flow chart of a method for providing a dynamic gain in a sender using a VoIP system in accordance with an embodiment of the present invention. 
   Referring to  FIG. 6 , when a telephone user makes a phone call which passes through a VoIP gateway using an analog telephone or a digital telephone, the analog telephone or the digital telephone transmit a call setup message including a called telephone number to a key telephone/private exchange (S 110 ). 
   Then, the key telephone/private exchange identifies an extension number of a sender and determines whether the type of telephone of the sender is an analog telephone or a digital telephone. 
   Then, the key telephone/private exchange looks up an IP address translation table transmitted from a gatekeeper and determines whether the receiver is an IP telephone or an external VoIP gateway. 
   Then, according to the determination, the key telephone/private exchange obtains a gain value by looking up a gain table and then transmits the gain value together with the call setup message to the VoIP gateway (S 112 ). The obtained gain value corresponds to either the extension telephone making the call, the terminal (end-point) receiving the call (i.e., the IP telephone or the external VoIP gateway) or both the extension telephone making the call and the terminal (end-point) receiving the call. 
   Then, the VoIP gateway extracts the gain value from the call setup message, stores the value (S 113 ), and transmits the call setup message to the gatekeeper (S 114 ). The call setup message is then transmitted to the called IP telephone or external VoIP gateway (S 116 ). 
   Then, when the gatekeeper and the VoIP gateway receive a call response message from the IP telephone or the external VoIP gateway (S 118  and S 120 ), the received call response message is transmitted to the key telephone/private exchange and the telephone (S 122  and S 124 ). 
   When the VoIP gateway tries to compress and transmit the PCM voice data using the codec after the call setup has been completed and the call setup has been normally made to the IP telephone or the external VoIP gateway, the VoIP gateway amplifies the PCM voice data according to the amplification ratio determined in response to the stored gain value and then compresses and transmits the amplified PCM data using the codec. 
   Also, the VoIP gateway transforms the VoIP packet received from the IP telephone or the external VoIP gateway into the PCM voice data, amplifies the PCM voice data according to the amplification ratio determined in response to the stored gain value and transmits the amplified PCM voice data to the key telephone/private exchange. 
     FIG. 7  is a view showing an operation flow chart of a method for providing a dynamic gain in a receiver using a VoIP system in accordance with another embodiment of the present invention. 
   Referring to  FIG. 7 , when a gatekeeper receives a call setup message form an IP telephone or an external VoIP gateway (S 210 ), it identifies an IP address of a sender and determines whether the sender is an IP telephone or an external VoIP gateway by looking up an IP address translation table. 
   Then, the gatekeeper determines whether a receiver is an analog telephone or a digital telephone using information on the kind of telephone for an extension telephone number transmitted from the key telephone/private exchange. And then, the gatekeeper obtains a gain value by looking up the gain table and transmits the gain value obtained together with the call setup message to the VoIP gateway (S 212 ). 
   Then, the VoIP gateway extracts the gain value from the call setup message, stores the gain value (S 213 ) and transmits the call setup message to the key telephone/private exchange (S 214 ). The call setup message is then transmitted to the analog telephone or the digital telephone (S 216 ). 
   Then, when the VoIP gateway receives a call response message from the analog telephone or the digital telephone (S 218  and S 220 ), the received call response message is transmitted to the IP telephone or the external VoIP gateway so that the call setup is made (S 222  and S 224 ). 
   Then, as the call setup is completed, the VoIP gateway transforms the VoIP packet transmitted from the IP telephone or the external VoIP gateway into the PCM voice data, amplifies the PCM voice data according to the amplification ratio determined in response to the stored gain value, and then transmits the data to the key telephone/private exchange. 
   Also, when the VoIP gateway tries to compress the PCM voice data using the codec and transmit the PCM voice data to the IP telephone or the external VoIP gateway, it amplifies the PCM voice data according to the amplification ratio determined in response to the stored gain value, compresses the amplified PCM data using the codec, and then transmits them as VoIP packets. 
   Even though the present invention explains the case that the key telephone/private exchange, the VoIP gateway and the gatekeeper are close coupled and share information in the database, the same method will be applied to the case that the constituents are not coupled closely. 
   Although the preferred embodiments of the present invention have been disclosed for illustrative purposes, those skilled in the art appreciate that various modifications, additions and substitutions are possible, without departing from the scope and spirit of the invention as disclosed in the accompanying claims. 
   EFFECT 
   In accordance with the present invention, when connecting a call to communication equipment (an IP phone or a VoIP gateway) connected to an external IP network through another VoIP gateway in a key telephone/private exchange, a speech quality can be enhanced by enabling the call to have proper gain values according to the type of terminals to be connected to the call.

Technology Category: 5