JP2006333291A - Ip telephone system - Google Patents

Abstract

<P>PROBLEM TO BE SOLVED: To provide an IP telephone system which will discard a session under execution of a retransmission process and shifts to call processing using the same session, when a call request is generated during the execution of the retransmission process. <P>SOLUTION: The hand set of an operation part 202 is made into offhook by the user of an IP telephone 101, and a recall request signal is outputted to an IP telephone control part 211 during the execution of retransmission processing of a "BYE" command for 'a period of 32 seconds' by the IP telephone 101, the retransmission processing of the "BYE" command under execution is interrupted. In this time, the IP telephone 101 discards the session used by the retransmission processing, namely, erase the session ID stored in a built-in memory. Then, the IP telephone 101 shifts to recall processing, according to the inputted recall request signal and transmits an "INVITE" command to an SIP server 104, by using the same session ID which has previously been discarded. <P>COPYRIGHT: (C)2007,JPO&INPIT

Description

  The present invention relates to an IP telephone apparatus, and more particularly to an IP telephone apparatus that provides a voice call service.

  2. Description of the Related Art Conventionally, an IP telephone apparatus using SIP (Session Initiation Protocol) is required to operate in the same manner as a general public telephone apparatus (hereinafter referred to as a PSTN telephone apparatus) via an exchange because of the nature of a telephone.

  However, the actual operation of the IP telephone apparatus is performed while maintaining the command transmission / reception relationship of “request-response” with the other IP telephone apparatus, and the operation related to the call of incoming / outgoing call → call → end call is performed. The state is changed while synchronizing with the other party IP telephone apparatus.

  The operation of an IP telephone device using SIP is defined by the “SIP” protocol of RFC3261 (Non-patent Document 1). In this “SIP” protocol, for example, when a user of an IP telephone device goes on-hook at the end of a call, a “BYE” command is transmitted from the IP telephone device to the partner IP phone device, and “200 OK” is transmitted from the partner IP phone device. The call processing is completed when the command is responded.

  Further, in the “SIP” protocol, a retransmission procedure is defined when a failure occurs such that the “200 OK” command returned from the counterpart IP telephone apparatus that transmitted the “BYE” command cannot be received. A specific example of the retransmission procedure at the end of the call will be described with reference to the flowchart shown in FIG. FIG. 4 shows a retransmission procedure at the end of the call executed between the terminals A and B, which are IP telephone devices, via the SIP server.

In FIG. 4, when the user goes on-hook on the terminal A side, a “BYE” command is transmitted from the terminal A to the terminal B via the SIP server. In the terminal B, a “200 OK” command in response to the “BYE” command is transmitted to the SIP server. Then, when the “200 OK” command is transmitted from the SIP server to the terminal A, if the “200 OK” command does not reach the terminal A due to a line failure between the SIP server and the terminal A, the terminal A performs the “200 OK” command. The “BYE” command resending procedure for <32 seconds> is started until it is received.
RFC3261 “Session Initiation Protocol” June, 2002

  However, when the retransmission procedure according to the “SIP” protocol is executed between IP telephone apparatuses as described above, the following problems occur. For example, as shown in FIG. 4, when a user performs an off-hook operation while a retransmission procedure is being executed and a new call processing is to be performed, the retransmission procedure is being executed. There arises a problem that the recall process cannot be executed.

  That is, the user cannot recognize that the IP telephone apparatus is performing the retransmission procedure immediately after on-hook, and there is a possibility that such a situation will occur. In the conventional PSTN telephone apparatus, it is recognized by the user that the re-call processing can be performed immediately after the end of the call, and the situation that the re-call process similar to the PSTN telephone apparatus cannot be realized in the IP telephone apparatus As a result, the basic performance is impaired, and improvement is desired.

  As a proposal for solving the problem of the re-call processing in such an IP telephone apparatus, there is, for example, one using a plurality of sessions supported by the IP telephone apparatus. This means that when a call request is generated during the retransmission procedure as described above using a plurality of sessions supported by the IP telephone apparatus, the re-call processing is performed using the free session. Is. However, even in this case, if there is no available session, it is not possible to perform the re-calling process, which is not a sufficient solution.

  Furthermore, another example in which a retransmission procedure occurs is shown in the flowchart of FIG. The parts indicated by (1) to (3) in FIG. 5 are other examples in which the retransmission procedure occurs. (1) shows a case where the “200 OK” command transmitted from the terminal B that has received the “BYE” command does not reach the SIP server due to a line failure between the terminal B and the SIP server. (2) shows a case where the “BYE” command transmitted from the terminal A does not reach the terminal B due to a line failure between the terminal B and the SIP server when the command is transmitted to the terminal B via the SIP server. Yes. (3) shows a case where when the “BYE” command transmitted from the terminal A is transmitted to the SIP server, it does not reach the SIP server due to a line failure between the terminal A and the SIP server. Even when these re-transmission procedures occur, the problem of the above-mentioned re-call processing occurs.

  The present invention has been made in view of such a point, and when a call request is generated during the execution of a retransmission procedure, the session that is executing the retransmission procedure is discarded and the call processing is performed using the same session. It is an object of the present invention to provide an IP telephone apparatus and a method for making a call to the IP telephone apparatus.

  The present invention relates to an IP telephone apparatus connected via an IP network, and a call is made between an operating means for inputting an incoming / outgoing call instruction, an end of call instruction, and the like, and a partner IP telephone apparatus based on SIP as a protocol. Control means for controlling a connection procedure, and when the call instruction is input from the operation means during execution of the retransmission procedure for the counterpart IP telephone apparatus based on the SIP, the control means A configuration is adopted in which the procedure is terminated and the process proceeds to call processing using the same session.

  According to the present invention, when a call request is generated during the retransmission procedure, the session that is executing the retransmission procedure is discarded, and the same session is used to shift to the calling process. It is possible to provide an IP telephone apparatus that realizes a re-call function equivalent to the above.

  An IP telephone apparatus according to a first aspect of the present invention is an IP telephone apparatus connected via an IP network, and is based on an operating means for inputting an incoming / outgoing call instruction, an end of call instruction, and the like, and SIP as a protocol. Control means for controlling a call connection procedure with the counterpart IP telephone apparatus, and the control means originates a call from the operation means while executing a retransmission procedure for the counterpart IP telephone apparatus based on the SIP. When an instruction is input, a configuration is adopted in which the retransmission procedure is terminated and the process proceeds to call processing using the same session.

  According to this configuration, when a call request is generated during the retransmission procedure, the PSTN telephone device terminates the session that is executing the retransmission procedure and shifts to the call processing using the same session. It is possible to provide an IP telephone apparatus that realizes a re-call function equivalent to the above.

  An IP telephone apparatus according to a second aspect of the present invention is the IP telephone apparatus according to the first aspect, wherein the control means is configured to receive a call termination instruction from the operation means and then to call the other party based on the SIP. When a call instruction is input from the operation means during the re-transmission procedure related to the call termination with the IP telephone apparatus, the re-transmission procedure is terminated and the process proceeds to call processing using the same session. Take the configuration.

  According to this configuration, when a call request is generated during the retransmission procedure after the end of the call, the session that is executing the retransmission procedure is terminated, and the same session is used to move to the calling process. Thus, it is possible to provide an IP telephone apparatus that realizes a re-calling function equivalent to that of the PSTN telephone apparatus without giving the user a sense of incongruity.

  An IP telephone apparatus according to a third aspect of the present invention is the IP telephone apparatus according to the second aspect, wherein the control means stores session information during a call connection with the counterpart IP telephone apparatus, and performs the retransmission procedure. When the session being executed is discarded and the retransmission procedure is terminated, a configuration for controlling the erase timing of the stored session information is adopted.

  According to this configuration, it is possible to realize a recall function equivalent to that of the PSTN telephone apparatus simply by adding a program.

  A calling method for an IP telephone apparatus according to a fourth aspect of the present invention is a calling method for an IP telephone apparatus connected via an IP network, and is based on SIP as a protocol. When a call instruction is input during a re-transmission procedure related to a call termination with a partner IP telephone apparatus based on the SIP after the call connection, the re-transmission procedure is terminated. Use the same session and move to call processing.

  According to this method, when a call request is generated during the retransmission procedure, the session in which the retransmission procedure is being executed is terminated, and the PSTN telephone device is used to shift to the call processing using the same session. It is possible to provide a calling method for an IP telephone apparatus that realizes a re-calling function equivalent to the above.

  Hereinafter, an embodiment of the present invention will be described in detail with reference to the drawings.

  FIG. 1 is a diagram showing an overall configuration of an IP telephone system according to the present embodiment. In FIG. 1, an IP telephone system 100 includes a plurality of IP telephone apparatuses (hereinafter referred to as IP telephones) 101, 102, and 103 and an SIP server 104 connected to each other via an IP network 105. The IP telephone system may be configured by connecting three or more IP telephone devices without being limited to the configuration shown in FIG.

  The IP phone 101 (102, 103) has a function capable of performing a voice call with another IP phone via the IP network 105.

  In response to a call connection request from the IP telephones 101, 102, 103 connected to the IP network 105, the SIP server 104 is connected to the called IP telephones 101, 102, 103 based on the SIP protocol. In addition to executing control, transmission / reception of voice data to / from the IP telephone connected to the call is controlled.

  FIG. 2 is a block diagram showing a configuration of IP phone 101 of the present embodiment. IP phones 102 and 103 have the same configuration as IP phone 101.

  In FIG. 2, an IP phone 101 includes an IP phone control unit 211, a SIP memory 212, a VoIP memory 213, and a network I / F unit 214, a numeric keypad, a function key, a handset, a display unit, etc. (not shown). The operation unit 202 is mainly configured.

  The IP telephone control unit 211 controls calling / incoming call processing, call termination processing, retransmission processing, and the like with the counterpart IP phone based on the SIP program stored in the SIP memory 212. Further, the IP telephone control unit 211 controls transmission / reception of voice packets with the other party IP telephone to which the call is connected based on the VoIP program stored in the VoIP memory 213. Furthermore, the IP telephone control unit 211 executes calling / incoming call processing and call termination processing in response to an operation signal input from the operation unit 202, and displays display signals such as calling number and called number. It outputs to 202 and displays on a display part. The IP telephone control unit 211 has a built-in memory (not shown) for storing a session ID in use when a call connection is established with a partner IP telephone.

  The SIP memory 212 stores a SIP program for executing calling / incoming call processing, call termination processing, retransmission processing, and the like based on the SIP protocol. The VoIP memory 213 stores a VoIP program for executing transmission / reception processing of voice packets via the IP network 105.

  The network I / F unit 214 has an interface function for connecting to the IP network 105, and executes transmission / reception of commands and voice packets based on SIP under the control of the IP telephone control unit 201.

  The operation unit 202 includes ten keys, function keys, a handset, a display unit, and the like (not shown). When the user operates a numeric keypad, function key, or handset at the time of outgoing or incoming call, the operation unit 202 transmits an operation signal to the IP telephone control unit 211, and a display signal input from the IP telephone control unit 211 To display the calling number, the called number, and the like on the display unit.

  Next, in the IP telephone system 100 of FIG. 1, the user of the IP telephone 101 made a call request during the execution of the retransmission procedure that occurred when the call between the IP telephone 101 and the IP telephone 102 ended (end call). The operation in this case will be described with reference to the flowchart shown in FIG. It is assumed that IP phone control unit 211 of IP phone 101 stores a session ID being used by the telephone call in a built-in memory (not shown).

  In FIG. 3, IP phone 101 and IP phone 102 make a call via SIP server 104 (step S301). When the user of IP phone 101 performs an on-hook operation of handset of operation unit 202 when the call ends, the on-hook signal is displayed. The data is output from the operation unit 202 to the IP telephone control unit 211. When the on-hook signal is input, IP telephone control unit 211 shifts to a call-ending operation and sends a “BYE” command from network I / F unit 214 to SIP server 104 (step S302).

  The SIP server 104 sends the “BYE” command received from the IP phone 101 to the IP phone 102 (step S303). When the IP telephone 102 receives the “BYE” command from the SIP server 104, the IP telephone control unit 211 recognizes that the call is over, and sends a “200 OK” command to the SIP server 104 as a response command (step). S304). Then, the SIP server 104 sends the “200 OK” command received from the IP phone 102 to the IP phone 101 (step S305).

  At this time, it is assumed that a line failure or the like has occurred between the SIP server 104 and the IP phone 101 and the “200 OK” command sent from the SIP server 104 does not reach the IP phone 101.

  When the IP phone control unit 211 of the IP phone 101 does not receive the “200 OK” command that is a response to the “BYE” command, the IP phone control unit 211 proceeds to the retransmission procedure defined in the SIP, and “200 OK” that is the response to the “BYE” command. In order to receive the “command”, the timer for <32 seconds> is started and retransmission processing of the “BYE” command is started (step S306).

  In this case, the “BYE” command transmitted from the IP phone 101 via the SIP server 104 reaches the IP phone 102 (steps S306 and S307), but a line failure occurs between the SIP server 104 and the IP phone 101. Therefore, the “200 OK” command transmitted from the IP phone 102 via the SIP server 104 does not reach the IP phone 101 (steps S308 and S309).

  Then, while IP phone 101 is executing retransmission processing of the “BYE” command for <32 seconds> (step S310), the handset of operation unit 202 is off-hooked by the user of IP phone 101, and the re-call request signal is IP phone control. When output to unit 211, IP telephone control unit 211 suspends the resending process of the “BYE” command being executed. At this time, the IP telephone control unit 211 discards the session in use by the retransmission process, that is, erases the session ID stored in the built-in memory. In this case, the IP telephone control unit 211 issues a session ID deletion command to the internal memory after the transmission process of the “BYE” command is completed.

  Then, the IP telephone control unit 211 shifts to the recall processing in response to the recall request signal input from the operation unit 202, and sends the “INVITE” command to the SIP server 104 using the same session that was previously discarded. It is sent out (step S311).

  IP phone 101 suspends the retransmission process for <32 seconds> without waiting for the reception of the response “200 OK” command from IP phone 102, but the “200 OK” command sent from IP phone 102 does not respond to SIP. It reaches the server 104. For this reason, IP phone 102 determines that the response transmission to the “BYE” command has been successful and can send a busy tone, so there is no particular problem even if IP phone 101 interrupts the retransmission procedure.

  As described above, when a re-call request is generated during execution of the retransmission process, the IP telephone according to the present embodiment immediately discards the session in which the re-transmission process is being executed, and shifts to the re-call process using the same session. Therefore, it is possible to realize a re-calling function equivalent to that of a PSTN telephone without giving the user a sense of incongruity.

  Further, even when the other retransmission procedures shown in (1) to (3) of FIG. 5 occur, it is possible to apply the process of shifting to the re-call process using the same session and discarding the session. . In these cases as well, there is no trouble caused by interruption of the retransmission procedure in the other party IP phone.

  Therefore, in the IP telephone device according to the present embodiment, even when the user hooks the handset by the end of the call and the retransmission procedure is being executed, the re-call can be made immediately after the end of the call. It is possible to provide an IP telephone device that realizes a recall function.

  Further, in the IP telephone apparatus according to the present embodiment, the session discarded in the retransmission process is used in the recall process, so that the session resource can be effectively used in the IP telephone apparatus that can use a plurality of sessions. it can. For this reason, even in a low-cost IP telephone apparatus that uses a small number of sessions, it is possible to realize a recall function equivalent to that of a PSTN telephone apparatus by simply adding a program.

  The IP telephone apparatus according to the present invention provides an IP telephone apparatus that realizes a re-call function equivalent to that of a PSTN telephone apparatus because a re-call can be made immediately after the end of a call even when a retransmission procedure is being executed. Useful in that it can.

The figure which shows the whole structure of the IP telephone system which concerns on one embodiment of this invention Block diagram showing a configuration of an IP telephone apparatus according to the present embodiment Flow chart showing an example of a retransmission procedure and a re-calling operation executed between IP telephones in the IP telephone system according to the present embodiment Flow chart showing a retransmission procedure at the time of call termination between conventional IP telephone devices Flow chart showing another retransmission procedure at the time of call termination between conventional IP telephone devices

Explanation of symbols

DESCRIPTION OF SYMBOLS 100 IP telephone system 101,102,103 IP telephone apparatus 104 SIP server 105 IP network 201 IP telephone part 202 Operation part 211 IP telephone control part 212 SIP memory 213 VoIP memory 214 Network I / F part

Claims (4)

An IP telephone device connected via an IP network,
An operation means for inputting an incoming / outgoing call instruction or an end of call instruction,
Control means for controlling a call connection procedure with a partner IP telephone device based on SIP as a protocol,
The control means terminates the retransmission procedure and makes a call using the same session when a call instruction is input from the operation means during execution of the retransmission procedure for the counterpart IP telephone apparatus based on the SIP. IP telephone device to be transferred to processing.   The control means receives a call instruction from the operation means during execution of a retransmission procedure related to the call end with the partner IP telephone device based on the SIP after the call end instruction is input from the operation means. 2. The IP telephone apparatus according to claim 1, wherein, when entered, the retransmission procedure is terminated and the process proceeds to call processing using the same session.   The control means stores session information during call connection with the other IP telephone apparatus, and controls the erasure timing of the stored session information when the retransmission procedure is terminated and the retransmission procedure is terminated. The IP telephone apparatus according to claim 2. An IP telephone device calling method connected via an IP network,
Based on SIP protocol, call connection with the partner IP phone device,
If a call instruction is input during a re-transmission procedure related to a call termination with a partner IP telephone device based on the SIP after call connection, the re-transmission procedure is terminated and the same session is used to make a call. Calling method of IP telephone apparatus to shift to call processing.

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