Abstract:
When a server has received an Invite Request for a telephone connected to a public telephone line, the server transfers the Invite Request to a gateway terminal. Upon receiving the Invite Request, the gateway terminal originates a call to the telephone. The server inserts a telephone number of the telephone into one of an area other than an area assigned for a URI of the gateway terminal within a start-line of the Invite Request, an area other than the start-line of the Invite Request, and a message other than the Invite Request. The gateway terminal reads out the telephone number of the telephone from one of the area other than an area assigned for a URI of the gateway terminal within a start-line of the Invite Request, the area other than the start-line of the Invite request and the message other than the Invite request and then originates, in a call origination step, a call to the telephone by using the telephone number that has been read out in a readout step.

Description:
BACKGROUND OF THE INVENTION  
       [0001]     1. Field of the Invention  
         [0002]     The present invention relates to a signaling method for establishing a telephone communication link, and more specifically, to a signaling method for establishing a telephone communication link between a terminal connected to an IP (Internet Protocol) network such as the Internet or intranet and a telephone set connected to a public telephone network.  
         [0003]     2. Description of the Related Art  
         [0004]     An SIP (Session Initiation Protocol), which has been developed for inviting a user to a certain session, is used for establishing, changing, or disconnecting a multimedia session on the Internet.  
         [0005]     Conventionally, two signaling methods have been used for establishing communication between a terminal (“user agent” in terms of the SIP) connected to the Internet and a telephone set connected to a public telephone network.  
         [0006]      FIG. 1  shows a configuration of a system that implements the first signaling method and a sequence of the signaling method.  
         [0007]     Referring to  FIG. 1 , when a terminal  901  makes a call to a telephone  904 , firstly, the terminal  901  sends an Invite Request  911  to an SIP redirect server  902 . The to-line of the header field in the Invite Request  911  describes “0312341234@domin.com” obtained by combining “0312341234”, which is a telephone number of the telephone  904  and “domin.com”, which is a name of a domain that the terminal  901 , SIP redirect server  902  and an SIP gateway server  903  belong to.  
         [0008]     Upon receiving the Invite Request  911 , the SIP redirect server  902  sends back a reply  912  including an address of the SIP gateway server  903  to the terminal  901 .  
         [0009]     Upon receiving the reply  912 , the terminal  901  sends an Invite Request  914  to the SIP gateway server  903 . As is the case with the Invite Request  911 , the start-line of the Invite Request  914  describes URL of the gateway server  903 , and the to-line of the header field describes “0312341234@domin.com”.  
         [0010]     When receiving the Invite Request  914 , the SIP gateway server  903  reads out “0312341234” from “0312341234@domin.com” described in the header field of the Invite Request  914 , connects to a public telephone network, and makes a call to the telephone  904  whose telephone number is “0312341234” ( 915 ).  
         [0011]      FIG. 2  shows a configuration of a system that implements the second signaling method and a sequence of the signaling method.  
         [0012]     Referring to  FIG. 2 , when the terminal  901  makes a call to the telephone  904 , firstly, the terminal  901  sends an Invite Request  921  to an SIP proxy server  905 . The to-line of the header field in the Invite Request  921  describes “0312341234@domin.com” obtained by combining “0312341234”, which is a telephone number of the telephone  904  and “domin.com”, which is a name of a domain that the terminal  901 , SIP proxy server  905  and SIP gateway server  903  belong to.  
         [0013]     Upon receiving the Invite Request  921 , the SIP proxy server  905  sends an Invite Request  922  to the SIP gateway server  903  based on the expectation that the SIP gateway server  903  can handle the destination indicated by the to-line of the header field in the Invite Request  922 . The header field in the Invite Request  922  also describes “0312341234@domin.com”.  
         [0014]     Upon receiving the Invite Request  922 , the SIP gateway server  903  reads out “0312341234” from “0312341234@domin.com” described in the header field of the Invite Request  922 , connects to a public telephone network, and makes a call to the telephone  904  whose telephone number is “0312341234” ( 923 ).  
         [0015]     The following documents can be taken as conventional art documents related to the present invention. RFC 3261 written standards (http://www.ietf.org/rfc/rfc3261.txt) relates to an SIP; RFC 2327 written standards (http://www.ietf.org/rfc/rfc2327.txt?number=2327) relates to an SDP (Session Description Protocol) concerning contents described in the body of an SIP message; RFC 3550 written standards (http://rfc3550.x42.com/) relates to an RTCP (Real time Control Protocol) for controlling transmission of an RTP (Real Time Packet); and “How to Add MSN Messenger Services for PC-to-Phone Functionality to Cisco Packet Voice Networks (http://www.cisco.com/warp/public/cc/techno/tyvdve/sip/prodlit/mpcph_wp.htm)” relates to the abovementioned two signaling methods.  
         [0016]     To implement the aforementioned two signaling methods, the SIP gateway server must be used. The SIP gateway server is a device that has an SIP server function and gateway function. Since the SIP server function part is expensive, the SIP gateway server is an expensive product to use.  
       SUMMARY OF THE INVENTION  
       [0017]     An object of the present invention is to provide a signaling method that allows a communication connection between a terminal connected to an IP network and a telephone set connected to a public telephone network to be established without the use of an SIP gateway server, and a server and gateway terminal for use in the method.  
         [0018]     According to a first aspect of the present invention, there is provided a signaling method comprising: a transfer step in which, when having received an Invite Request for a telephone connected to a public telephone network, a server transfers the Invite Request to a gateway terminal; and a call origination step in which, when having received at least the Invite Request, the gateway terminal originates a call to the telephone.  
         [0019]     The signaling method according to the first aspect of the present invention may further comprise: a readout step in which the server reads out a telephone number of the telephone from the received Invite Request; an insertion step in which the server inserts a URI of the gateway terminal into a start-line of the received Invite Request; and an insertion step in which the server inserts the telephone number of the telephone into one of an area other than an area assigned for the URI of the gateway terminal within the start-line of the Invite Request to be sent to the gateway terminal, an area other than the start-line of the Invite Request to be sent to the gateway terminal, and a message to be sent to the gateway terminal other than the Invite Request sent to the gateway terminal.  
         [0020]     The signaling method according to the first aspect of the present invention may further comprise a readout step in which the gateway terminal reads out the telephone number of the telephone from one of the start-line of the Invite Request received from the server, the area other than the start-line, and the message received from the server other than the Invite Request from the server, wherein the gateway terminal originates, in the call origination step, a call to the telephone by using the telephone number that has been read out in the readout step.  
         [0021]     The signaling method according to the first aspect of the present invention may further comprise an insertion step in which the server inserts the telephone number of the telephone into the start-line of the header in the Invite Request to be sent to the gateway terminal.  
         [0022]     The signaling method according to the first aspect of the present invention may further comprise a readout step in which the gateway terminal reads out the telephone number of the telephone from the start-line of the header in the Invite Request received from the server, wherein the gateway terminal originates, in the call origination step, a call to the telephone by using the telephone number that has been read out in the readout step.  
         [0023]     The signaling method according to the first aspect of the present invention may further comprise an insertion step in which the server inserts the telephone number of the telephone into a header field of the Invite Request to be sent to the gateway terminal.  
         [0024]     The signaling method according to the first aspect of the present invention may further comprise a readout step in which the gateway terminal reads out the telephone number of the telephone from the header field of the Invite Request received from the server, wherein the gateway terminal originates, in the call origination step, a call to the telephone by using the telephone number that has been read out in the readout step.  
         [0025]     The signaling method according to the first aspect of the present invention may further comprise an insertion step in which the server inserts the telephone number of the telephone into a body of the Invite Request to be sent to the gateway terminal.  
         [0026]     The signaling method according to the first aspect of the present invention may further comprise a readout step in which the gateway terminal reads out the telephone number of the telephone from the body of the Invite Request received from the server, wherein the gateway terminal originates, in the call origination step, a call to the telephone by using the telephone number that has been read out in the readout step.  
         [0027]     The signaling method according to the first aspect of the present invention may further comprise an insertion step in which the server inserts the telephone number of the telephone into an acknowledge request to be sent to the gateway terminal.  
         [0028]     The signaling method according to the first aspect of the present invention may further comprise a readout step in which the gateway terminal reads out the telephone number of the telephone from the acknowledge request received from the server, wherein the gateway terminal originates, in the call origination step, a call to the telephone by using the telephone number that has been read out in the readout step. The signaling method according to the first aspect of the present invention may further comprise an insertion step in which the server inserts the telephone number of the telephone into a Real Time Control Protocol packet to be sent to the gateway terminal.  
         [0029]     The signaling method according to the first aspect of the present invention may further comprise a readout step in which the gateway terminal reads out the telephone number of the telephone from the Real Time Control Protocol packet received from the server, wherein the gateway terminal originates, in the call origination step, a call to the telephone by using the telephone number that has been read out in the readout step.  
         [0030]     According to a second aspect of the present invention, there is provided a server comprising: receiving means for receiving an Invite Request for a telephone connected to a public telephone network; and a transfer means for transferring the Invite Request to a gateway terminal when the server has received the Invite Request.  
         [0031]     The server according to the second aspect of the present invention may further comprise a readout means for reading out a telephone number of the telephone from the received Invite Request; a first insertion means for inserting a URI of the gateway terminal into a start-line of the received Invite Request; a second insertion means for inserting the telephone number of the telephone into one of an area other than an area assigned for the URL of the gateway terminal within the start-line of the Invite Request to be sent to the gateway terminal, an area other than the start-line of the Invite Request to be sent to the gateway terminal, and a message to be sent to the gateway terminal other than the Invite Request sent to the gateway terminal.  
         [0032]     In the server according to the second aspect of the present invention, the second insertion means may insert the telephone number of the telephone into the start-line of the header in the Invite Request to be sent to the gateway terminal.  
         [0033]     In the server according to the second aspect of the present invention, the second insertion means may insert the telephone number of the telephone into a header field of the Invite Request to be sent to the gateway terminal.  
         [0034]     In the server according to the second aspect of the present invention, the second insertion means may insert the telephone number of the telephone into a body of the Invite Request to be sent to the gateway terminal.  
         [0035]     In the server according to the second aspect of the present invention, the second insertion means may insert the telephone number of the telephone into an acknowledge request to be sent to the gateway terminal.  
         [0036]     In the server according to the second aspect of the present invention, the second insertion means may insert the telephone number of the telephone into a Real Time Control Protocol packet to be sent to the gateway terminal.  
         [0037]     According to a third aspect of the present invention, there is provided a gateway terminal comprising: receiving means for receiving at least an Invite Request for a telephone connected to a public telephone network; and a call origination means for originating a call to the telephone based on an Invite request when the gateway terminal has received at least the Invite Request for the telephone.  
         [0038]     The gateway terminal according to the third aspect of the present invention may further comprise a readout means for reading out a telephone number of the telephone from one of an area other than an area assigned for the URI of the gateway terminal within the start-line of the Invite Request received from the server, the area other than the start-line of the Invite Request to be sent to the gateway terminal, and the message received from the server other than the Invite Request received from the server, wherein the call origination means originates a call to the telephone by using the telephone number read out by the read out means.  
         [0039]     In the gateway terminal according to the third aspect of the present invention, the readout means may read out the telephone number of the telephone from the start-line of the header in the Invite Request received from the server.  
         [0040]     In the gateway terminal according to the third aspect of the present invention, the readout means may read out the telephone number of the telephone from a header field of the Invite Request received from the server.  
         [0041]     In the gateway terminal according to the third aspect of the present invention, the readout means may read out the telephone number of the telephone from a body of the Invite Request received from the server.  
         [0042]     In the gateway terminal according to the third aspect of the present invention, the readout means may read out the telephone number of the telephone from an acknowledge request received from the server.  
         [0043]     In the gateway terminal according to the third aspect of the present invention, the readout means may read out the telephone number of the telephone from a Real Time Control Protocol packet received from the server.  
         [0044]     The present invention can eliminate the use of the SIP server having a gateway function, thereby allowing a communication link between the terminal connected to an IP network and telephone connected to a public telephone network to be established at low cost. 
     
    
     BRIEF DESCRIPTION OF THE DRAWINGS  
       [0045]      FIG. 1  is a view showing a configuration of a system that implements a first conventional signaling method and a sequence of the signaling method;  
         [0046]      FIG. 2  is a view showing a configuration of a system that implements a second conventional signaling method and a sequence of the signaling method;  
         [0047]      FIG. 3  is a view showing a configuration of a system that implements a signaling method according to an embodiment of the present invention and a sequence of the signaling method;  
         [0048]      FIG. 4  is a table showing a data structure of an Invite Request that a terminal according to the embodiment of the present invention sends to an SIP server;  
         [0049]      FIG. 5  is a table showing a first example of the start-line and header field of the Invite Request that an SIP server according to a first example of the present invention sends to a gateway terminal;  
         [0050]      FIG. 6  is a table showing a second example of the start-line and header field of the Invite Request that an SIP server according to the first example of the present invention sends to a gateway terminal;  
         [0051]      FIG. 7  is a table showing a third example of the start-line and header field of the Invite Request that an SIP server according to the first example of the present invention sends to a gateway terminal;  
         [0052]      FIG. 8  is a table showing a fourth example of the start-line and header field of the Invite Request that an SIP server according to the first example of the present invention sends to a gateway terminal;  
         [0053]      FIG. 9  is a table showing a fifth example of the start-line and header field of the Invite Request that an SIP server according to the first example of the present invention sends to a gateway terminal;  
         [0054]      FIG. 10  is a table showing a sixth example of the start-line and header field of the Invite Request that an SIP server according to the first example of the present invention sends to a gateway terminal;  
         [0055]      FIG. 11  is a table showing a seventh example of the start-line and header field of the Invite Request that an SIP server according to the first example of the present invention sends to a gateway terminal;  
         [0056]      FIG. 12  is a table showing an eighth example of the start-line and header field of the Invite Request that an SIP server according to the first example of the present invention sends to a gateway terminal;  
         [0057]      FIG. 13  is a table showing a ninth example of the start-line and header field of the Invite Request that an SIP server according to the first example of the present invention sends to a gateway terminal;  
         [0058]      FIG. 14  is a table showing a tenth example of the start-line and header field of the Invite Request that an SIP server according to the first example of the present invention sends to a gateway terminal;  
         [0059]      FIG. 15  is a table showing an eleventh example of the start-line and header field of the Invite Request that an SIP server according to the first example of the present invention sends to a gateway terminal;  
         [0060]      FIG. 16  is a table showing a twelfth example of the start-line and header field of the Invite Request that an SIP server according to the first example of the present invention sends to a gateway terminal;  
         [0061]      FIG. 17  is a table showing a thirteenth example of the start-line and header field of the Invite Request that an SIP server according to the first example of the present invention sends to a gateway terminal;  
         [0062]      FIG. 18  is a table showing a fourteenth example of the start-line and header field of the Invite Request that an SIP server according to the first example of the present invention sends to a gateway terminal;  
         [0063]      FIG. 19  is a table showing a fifteenth example of the start-line and header field of the Invite Request that an SIP server according to the first example of the present invention sends to a gateway terminal;  
         [0064]      FIG. 20  is a table showing a sixteenth example of the start-line and header field of the Invite Request that an SIP server according to the first example of the present invention sends to a gateway terminal;  
         [0065]      FIG. 21  is a table showing a seventeenth example of the start-line and header field of the Invite Request that an SIP server according to the first example of the present invention sends to a gateway terminal;  
         [0066]      FIG. 22  is a table showing an eighteenth example of the start-line and header field of the Invite Request that an SIP server according to the first example of the present invention sends to a gateway terminal;  
         [0067]      FIG. 23  is a table showing a nineteenth example of the start-line and header field of the Invite Request that an SIP server according to the first example of the present invention sends to a gateway terminal;  
         [0068]      FIG. 24  is a view showing a configuration of a system that implements a signaling method according to the first example of the present invention and a sequence of the signaling method;  
         [0069]      FIG. 25  is a view showing a configuration of a system that implements a signaling method according to a second example of the present invention and a sequence of the signaling method; and  
         [0070]      FIG. 26  is a view showing a configuration of a system that implements another signaling method according to the second example of the present invention and a sequence of the signaling method. 
     
    
     DETAILED DESCRIPTION OF THE PREFERRED EMBODIMENTS  
       [0071]     A preferred embodiment of the present invention will be described below in detail with reference to the accompanying drawings.  
         [0072]      FIG. 3  is a view showing a configuration of a system that implements a signaling method according to an embodiment of the present invention and a sequence of the signaling method.  
         [0073]     Referring to  FIG. 3 , when a terminal  101  makes a call to a telephone  104 , firstly, the terminal  101  sends an Invite Request  111  to an SIP server  102 . The start-line of the Invite Request  111  describes “0312341234@domin.com” obtained by combining “0312341234”, which is a telephone number of the telephone  104  and “domin.com”, which is a name of a domain that the terminal  101 , SIP server  102  and a gateway terminal  103  belong to.  
         [0074]     The gateway terminal is a user agent in terms of the SIP to which a gateway function is added.  
         [0075]     Upon receiving the Invite Request  111 , the SIP server  102  sends an Invite Request  112  to the gateway terminal  103 . The start-line of the Invite Request  112  describes “gateway-terminal@domin.com”, which is SIP URL of the gateway terminal  103 .  
         [0076]     The SIP server  102  deletes the telephone number of the telephone  104  from the start-line of the Invite Request  112  and inserts URI of the gateway terminal  103 . Alternatively, the SIP server  102  inserts the telephone number of the telephone  104  into the start-line, header field, or body of the Invite Request  112 .  
         [0077]     For example, the start-line describes a telephone number as follows: “SIP; (URI of gateway terminal) USER=Phone+(telephone number)”, or “SIP; (URI of gateway terminal) tag (telephone number)”.  
         [0078]     The above descriptions are merely an example and are not meant to limit the description method of a telephone number in the start-line.  
         [0079]     For example, the body describes a telephone number as follows with URI of the to-line: “SIP; (URI of gateway terminal) USER=Phone+(telephone number)”, or “SIP; (URI of gateway terminal) tag (telephone number)”.  
         [0080]     “To URI” in the body may be described without change from the original description as follows: “TO; (telephone number)@(domain) USER=Phone”.  
         [0081]     The above descriptions are merely an example and are not meant to limit the description method of a telephone number in header field.  
         [0082]     Further, the SIP server  102  may insert the telephone number of the telephone  104  into a massage that the SIP server  102  sends to the gateway terminal  103  after the Invite Request  112 . That is, the SIP server  102  may insert the telephone number of the telephone  104  into, for example, an ACK request or a RTCP packet.  
         [0083]     Upon receiving the Invite Request  112 , the gateway terminal  103  reads out the telephone number of the telephone  104  from the header field or body of the Invite Request  112 . Alternatively, the gateway terminal  103  may read out the telephone number of the telephone  104  from a message that the gateway terminal  103  receives from the SIP server  102  after the Invite Request  112 . That is, the gateway terminal  103  may read out the telephone number of the telephone  104  that has been inserted into, for example, the ACK request or the RTCP packet. The gateway terminal  103  then connects to a public telephone network and makes a call to the telephone  104  whose telephone number is “0312341234” ( 113 ).  
         [0084]     Examples of the present invention will next be described.  
       EXAMPLE 1  
       [0085]     In the Example 1, the telephone number of the telephone is inserted into the header field of an Invite Request.  
         [0086]     An Invite Request  111  that the terminal  101  sends to the SIP server  102  is as shown in  FIG. 4 . Referring to  FIG. 4 , the Invite Request includes a start-line, header field, empty line and body. The start-line describes “0312341234@domin.com” obtained by combining “0312341234”, which is a telephone number of the telephone and “domin.com”, which is a name of a domain that the terminal  101 , SIP server  102  and gateway terminal  103  belong to.  
         [0087]     Upon receiving the Invite Request  111 , the SIP server  102  sends an Invite Request  112  to the gateway terminal  103 . The start-line and header field of the Invite Request  112  are, for example, as shown in  FIG. 5 . Note that the Invite Request  112  and Invite Request  111  have the same body. Referring to  FIG. 5 , the start-line of the Invite Request  112  describes “gateway-terminal@domin.com”, which is SIP URL of the gateway terminal  103 . The telephone number of the telephone  104  is inserted into an item “Via” included in the header field.  
         [0088]     Instead of inserting the telephone number of the telephone  104  into “Via” as shown in  FIG. 5 , the SIP server  102  may insert the telephone number of the telephone  104  into “Via” in the manner as shown in  FIG. 6  or  FIG. 7 . Further, the SIP server  102  may insert the telephone number of the telephone  104  into “From” as shown in  FIG. 8 . Further, the SIP server  102  may insert the telephone number of the telephone  104  into “From” as a second element of “tag” as shown in  FIG. 9 . Further the SIP server  102  may insert the telephone number of the telephone  104  into “From” as an element of “tag” in the manner as shown in  FIG. 10 . In  FIG. 10 , character string “tel” is inserted before the telephone number “0312341234” for clarification of the tag. Further, the SIP server  102  may insert the telephone number of the telephone  104  into “To” as shown in  FIG. 11 . Further, the SIP server  102  may insert the telephone number of the telephone  104  into “To” as a second element of “tag” as shown in  FIG. 12 . Further the SIP server  102  may insert the telephone number of the telephone  104  into “To” as an element of “tag” in the manner as shown in  FIG. 13 . In  FIG. 13 , character string “tel” is inserted before the telephone number “0312341234” for clarification of the tag. Further, the SIP server  102  may insert the telephone number of the telephone  104  into “Call-ID” as an element of “tag” as shown in FIG.  14  or  15 . Further, the SIP server  102  may insert the telephone number of the telephone  104  into “Cseq” as shown in  FIG. 16  or  FIG. 17 . In  FIG. 17 , character string “tel” is inserted before the telephone number “0312341234” for clarification of the tag. Further, the SIP server  102  may insert the telephone number of the telephone  104  into “Contact” as shown in  FIG. 18, 19  or  20 . In  FIG. 20 , character string “tel” is inserted before the telephone number “0312341234” for clarification of the tag. Further, the SIP server  102  may insert the telephone number of the telephone  104  into “Content-type” as shown in  FIG. 21, 22  or  23 . In  FIG. 22 , character string “tel” is inserted before the telephone number “0312341234” for clarification of the tag.  
         [0089]     Upon receiving the Invite Request  112 , the gateway terminal  103  reads out the telephone number of the telephone  104  that has been inserted into any of the items in the header field of the Invite Request  112 . Thereafter, the gateway terminal  103  connects to a public telephone network and makes a call to a telephone  104 .  
         [0090]      FIG. 24  is a sequence diagram showing a message sending procedure according to the Example 1. Referring to  FIG. 24 , the same procedure as the embodiment is performed until the gateway terminal  103  has made a call to the telephone  104 .  
         [0091]     When a communication link between the gateway terminal  103  and telephone  104  ( 114 ) has established, the gateway terminal  103  sends back an OK reply  115  to the SIP server  102 . Upon receiving the OK reply  115 , the SIP server  102  sends back an OK reply  116  to the terminal  101 . The terminal  101  that has received the OK reply  116  then sends an acknowledge request  117  to the SIP server  102 . Upon receiving the acknowledge request  117 , the SIP server  102  sends an acknowledge request  118  to the gateway terminal  103 .  
       EXAMPLE 2  
       [0092]     In the Example 2, the telephone number of the telephone is inserted into the body of the Invite Request.  
         [0093]     The invite Request  111  that the terminal  101  sends to the SIP server  102  is as shown in  FIG. 4 . Referring to  FIG. 4 , the Invite Request includes a start-line, header field, empty line, and body. The start-line describes “0312341234@domin.com” obtained by combining “0312341234”, which is a telephone number of the telephone and “domin.com”, which is a name of a domain that the terminal  101 , SIP server  102  and gateway terminal  103  belong to.  
         [0094]     Upon receiving the Invite Request  111 , the SIP server sends an Invite Request  112  to the gateway terminal  103 . As is the case with the Example 1, the start-line of the Invite Request  112  describes as follows: 
        “INVITE SIP: gateway-terminal@domin.com SIP/2.0”       
 
         [0096]     Like the Invite Request  111 , the header field of the Invite Request  112  is as shown in  FIG. 4 .  
         [0097]     The body of the Invite Request  112  has a configuration according to the present invention. The telephone number can be described in any of the items (any of v, b, . . . , i in  FIG. 4 ) in the body of the Invite Request. Here, it is assumed that “p” that is originally used for representing a telephone number for obtaining session information is used in order to distinguish the description of a telephone number according to the present invention from that of an original item value. In the original case, a telephone number is described using the “p” as: 
        P=0334311831 
 
 whereas, a telephone number is described in the case of the present invention as: 
    p=&lt;sip:0312341234@domin.com;user=phone&gt;.        
 
         [0100]     The gateway terminal  103  has been designed according to the present invention and can distinguish between the original item value and telephone number according to the present invention even in the case where they are described in the same item by determining the format.  
         [0101]     Upon receiving the Invite Request  112 , the gateway terminal  103  reads out the telephone number of the telephone  104  that has been inserted into any of the items in the body of the Invite Request  112 . Thereafter, the gateway terminal  103  connects to a public telephone network and makes a call to the telephone  104 .  
       EXAMPLE 3  
       [0102]     In the Example 3, the telephone number of the telephone is inserted into the header field of the acknowledge request.  
         [0103]      FIG. 25  is a sequence diagram showing a message sending procedure according to the Example 3.  
         [0104]     Referring to  FIG. 25 , the terminal  101  sends an Invite Request  131  to the SIP server  102 . The start-line of the Invite Request  131  describes “0312341234@domin.com”. Upon receiving the Invite Request  131 , the SIP server  102  sends an Invite Request  132  to the gateway terminal  103 . The start-line of the Invite Request  132  describes “gateway-terminal@domin.com”. Unlike the case of the Examples 1 and 2, the telephone number of the telephone  104  is not inserted into the header field and body of the Invite Request  112 . Upon receiving the Invite Request  131 , the SIP server  102  further sends back a TRYING reply  133  to the terminal  101 . The gateway terminal  103  that has received the Invite Request  132  connects to a public telephone network ( 134 ) and sends back a ringing reply  135  to the SIP server  102 . Upon receiving the ringing reply  135 , the SIP server  102  sends back a ringing reply  136  to the terminal  101 . Thereafter, gateway terminal  103  sends back an OK reply  137  to the SIP server  102 . Upon receiving the OK reply  137 , the SIP server  102  sends back an OK reply  138  to the terminal  101 . Upon receiving the OK reply  138 , the terminal  101  sends an acknowledge request  139  to the SIP server  102 . Upon receiving the acknowledge request  139 , the SIP server  102  sends an acknowledge request  140  to the gateway terminal  103 . The telephone number of the telephone  104 , that is, “0312341234” is inserted into the acknowledge request  140 . Upon receiving the acknowledge request  140 , the gateway terminal  103  reads out the telephone number of the telephone  104  from the acknowledge request  140  and makes a call to the telephone  104  ( 141 ). Thereafter, a communication link between the gateway terminal  103  and telephone  104  is established ( 142 ).  
         [0105]     When the link establishment has failed, the gateway terminal  103  sends a request error, server error, or global error to the terminal  101 .  
         [0106]      FIG. 26  is a sequence diagram showing another message sending procedure according to the Example 3.  
         [0107]     Referring to  FIG. 26 , the terminal  101  sends an Invite Request  151  to the SIP server  102 . The start-line of the Invite Request  151  describes “0312341234@domin.com”. Upon receiving the Invite Request  151 , the SIP server  102  sends an Invite Request  152  to the gateway terminal  103 . The start-line of the Invite Request  152  describes “gateway-terminal@domin.com”. Unlike the case of Examples 1 and 2, the telephone number of the telephone  104  is not inserted into the header field and body of the Invite Request  152 . Subsequently, the SIP server  102  sends an acknowledge request  153  to the gateway terminal  103 . The telephone number of the telephone  104 , that is, “0312341234” is inserted into the acknowledge request  153 . The SIP server  102  then sends a TRYING reply  154  to the terminal  101 . The gateway terminal  103  that has received the acknowledge request  153  reads out the telephone number from the acknowledge request  153 , then connects to a public telephone network and makes a call to the telephone  104  ( 155 ). Subsequently, the gateway terminal  103  sends back a ringing reply  156  to the SIP server  102 . Upon receiving the ringing reply  156 , the SIP server  102  sends back a ringing reply  157  to the terminal  101 . When a communication link between the gateway terminal  103  and telephone  104  has been established ( 158 ), the gateway terminal  103  sends back an OK reply  159  to the SIP server  102 . Upon receiving the OK reply  159 , the SIP server  102  sends back an OK reply  160  to the terminal  101 . Upon receiving the OK reply  160 , the terminal  101  sends an acknowledge request  161  to the SIP server  102 . Upon receiving the acknowledge request  161 , the SIP server  102  sends an acknowledge request  162  to the gateway terminal  103 .  
         [0108]     When the link establishment has failed, the gateway terminal  103  does not send back the OK reply  159  to the SIP server  102 .  
         [0109]     The manner of inserting the telephone number of the telephone into the header field of the acknowledge request  140  or  153  is the same as the manner of inserting the telephone number into the header field of the Invite Request in the Example 1. Therefore, the header field of the acknowledge request  140  or  153  has the structure as shown in FIGS.  5  to  23 .  
       EXAMPLE 4  
       [0110]     In the Example 4, the telephone number of the telephone  104  is inserted into a predetermined region of the profile-specific extensions in a Receiver Report RTCP Packet described in chapter of 6.2.4 of the RFC3550 written standards.  
         [0111]     The SIP server  102  may send the telephone number of the telephone  104  to the gateway terminal  103  by using a message other than a message for signaling.  
       EXAMPLE 5  
       [0112]     In the Example 5, the SIP server  102  and gateway terminal  103  are built in the same information equipment.  
         [0113]     The present invention can be utilized for establishing a communication connection between a terminal connected to an IP network and a telephone connected to a public telephone network.  
         [0114]     Further, the present invention can be used for the communication connection to the dedicated line for an enterprise telephone system or a VoIP carrier service. Further, the communication medium is not limited to a telephone and the present invention can also be applied to other media, such as a video conferencing stream or e-mail.