WO2006048925A1 - 通信中継方法、通信中継プログラムおよび通信中継装置 - Google Patents
通信中継方法、通信中継プログラムおよび通信中継装置 Download PDFInfo
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- WO2006048925A1 WO2006048925A1 PCT/JP2004/016265 JP2004016265W WO2006048925A1 WO 2006048925 A1 WO2006048925 A1 WO 2006048925A1 JP 2004016265 W JP2004016265 W JP 2004016265W WO 2006048925 A1 WO2006048925 A1 WO 2006048925A1
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- WIPO (PCT)
- Prior art keywords
- connection request
- voice call
- call device
- network
- communication
- Prior art date
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Classifications
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- H—ELECTRICITY
- H04—ELECTRIC COMMUNICATION TECHNIQUE
- H04M—TELEPHONIC COMMUNICATION
- H04M7/00—Arrangements for interconnection between switching centres
- H04M7/12—Arrangements for interconnection between switching centres for working between exchanges having different types of switching equipment, e.g. power-driven and step by step or decimal and non-decimal
- H04M7/1205—Arrangements for interconnection between switching centres for working between exchanges having different types of switching equipment, e.g. power-driven and step by step or decimal and non-decimal where the types of switching equipement comprises PSTN/ISDN equipment and switching equipment of networks other than PSTN/ISDN, e.g. Internet Protocol networks
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- H—ELECTRICITY
- H04—ELECTRIC COMMUNICATION TECHNIQUE
- H04L—TRANSMISSION OF DIGITAL INFORMATION, e.g. TELEGRAPHIC COMMUNICATION
- H04L12/00—Data switching networks
- H04L12/66—Arrangements for connecting between networks having differing types of switching systems, e.g. gateways
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- H—ELECTRICITY
- H04—ELECTRIC COMMUNICATION TECHNIQUE
- H04M—TELEPHONIC COMMUNICATION
- H04M11/00—Telephonic communication systems specially adapted for combination with other electrical systems
- H04M11/007—Telephonic communication systems specially adapted for combination with other electrical systems with remote control systems
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- H—ELECTRICITY
- H04—ELECTRIC COMMUNICATION TECHNIQUE
- H04M—TELEPHONIC COMMUNICATION
- H04M15/00—Arrangements for metering, time-control or time indication ; Metering, charging or billing arrangements for voice wireline or wireless communications, e.g. VoIP
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- H—ELECTRICITY
- H04—ELECTRIC COMMUNICATION TECHNIQUE
- H04M—TELEPHONIC COMMUNICATION
- H04M3/00—Automatic or semi-automatic exchanges
- H04M3/42—Systems providing special services or facilities to subscribers
- H04M3/42195—Arrangements for calling back a calling subscriber
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- H—ELECTRICITY
- H04—ELECTRIC COMMUNICATION TECHNIQUE
- H04W—WIRELESS COMMUNICATION NETWORKS
- H04W4/00—Services specially adapted for wireless communication networks; Facilities therefor
- H04W4/24—Accounting or billing
-
- H—ELECTRICITY
- H04—ELECTRIC COMMUNICATION TECHNIQUE
- H04L—TRANSMISSION OF DIGITAL INFORMATION, e.g. TELEGRAPHIC COMMUNICATION
- H04L65/00—Network arrangements, protocols or services for supporting real-time applications in data packet communication
- H04L65/1066—Session management
- H04L65/1101—Session protocols
- H04L65/1104—Session initiation protocol [SIP]
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- Y—GENERAL TAGGING OF NEW TECHNOLOGICAL DEVELOPMENTS; GENERAL TAGGING OF CROSS-SECTIONAL TECHNOLOGIES SPANNING OVER SEVERAL SECTIONS OF THE IPC; TECHNICAL SUBJECTS COVERED BY FORMER USPC CROSS-REFERENCE ART COLLECTIONS [XRACs] AND DIGESTS
- Y02—TECHNOLOGIES OR APPLICATIONS FOR MITIGATION OR ADAPTATION AGAINST CLIMATE CHANGE
- Y02D—CLIMATE CHANGE MITIGATION TECHNOLOGIES IN INFORMATION AND COMMUNICATION TECHNOLOGIES [ICT], I.E. INFORMATION AND COMMUNICATION TECHNOLOGIES AIMING AT THE REDUCTION OF THEIR OWN ENERGY USE
- Y02D30/00—Reducing energy consumption in communication networks
- Y02D30/70—Reducing energy consumption in communication networks in wireless communication networks
Definitions
- the present invention relates to a communication relay method, a communication relay program, and a communication relay device that relay communication between a first voice call device and a second voice call device.
- PSTN Public Switched Telephone Network
- IP networks such as PCs, PDAs, landline phones, mobile phones, and PHS
- the applicant assigns a phone call to any of a plurality of devices used by the user at the point of time using a transfer destination selection function of a “call transfer server” (specifically, a SIP server).
- a “call transfer server” specifically, a SIP server.
- Propose a "Ubiquitous IP phone system” that can be transferred to the most suitable device with!! Japanese Patent Application No. 20 04-079590.
- the call forwarding destination is determined based on the current location of the user and preset preferences. For example, even if the phone is working on the same phone number, the user is still present.
- the device that is actually called such as an extension telephone for the person's seat, or a mobile phone when going out, will change from time to time.
- the present applicant has proposed a mobile phone type dual terminal equipped with a plurality of wireless communication devices, for example, PHS power WLAN (Japanese Patent Application No. 2004-199333). And on this device,
- the above dual terminal waits with PHS as long as PHS can be used and supplies power to the WLAN as much as possible.
- the terminal is within the PHS range, it turns off the WLAN regardless of whether it is within or outside the WLAN range.
- the above terminals are dual terminals, they are virtually equivalent to PHS when waiting in the PHS area. Therefore, in order to call this from the IP network, it is necessary to first turn on the WLAN and connect it to the IP network, and then to register the necessary items (IP address, etc.) in the SIP server 2000.
- PPPhone system of “International System Research Co., Ltd.” (more specifically, WakeOn Ring technology “PPPush” adopted in the system).
- PPPush WakeOn Ring technology
- the caller places an IP phone call from the calling terminal 2001 via the PSTN, and through its telephone strength SP_S (PSTN ⁇ SIP) gateway 2003 and SIP server 2000, Furthermore, if the call is transferred to the receiving terminal 2002 via the PSTN,
- the present invention has been made in view of the above, and a communication relay method, a communication relay program, and the like capable of relaying communication between a calling terminal and a receiving terminal on a relatively inexpensive route, And it aims at providing a communication relay apparatus.
- the present invention transfers the first connection request from the first voice call device to the second voice call device, and sends the connection request to the second voice call device.
- the connection request is transferred to the first voice call device.
- the present invention transfers the first connection request from the first voice communication device to the second voice communication device through the first network, and from the second voice communication device to the second voice communication device. Net ⁇ When the second connection request is received through the network, the first connection request is transferred to the second voice communication device through the second network.
- connection request for is routed through B.
- the present invention transfers the first connection request received through the predetermined network to the second voice communication device through the network, cancels the connection request, and When the second connection request is received from the voice call device, the connection request is transferred to the first voice call device.
- connection request in which a charge is generated twice due to passing through the same network twice between a calling terminal and a receiving terminal is Canceled before one charge is incurred.
- the present invention receives a connection request from the first voice communication device through the first network, specifies a second network to which the connection request is transferred, and And the second network is not identical, the connection request is transferred to the second voice communication device through the second network.
- connection request that causes a double charge due to passing the same network twice between the calling terminal and the receiving terminal is Discarded before charges are incurred.
- the communication relay method, the communication relay program, and the communication relay device according to the present invention have an effect that the communication between the transmitting terminal and the receiving terminal can be relayed on the relatively cheapest route.
- FIG. 1 is an explanatory diagram showing an outline of a “call back method” realized by the communication relay device according to the first embodiment of the present invention.
- FIG. 2 is an explanatory diagram showing an outline of a “callback method” realized by the communication relay device according to the first embodiment of the present invention.
- FIG. 3 is an explanatory diagram of a hardware configuration of the communication relay device according to the first embodiment of the present invention.
- FIG. 4 is an explanatory diagram of a functional configuration of the communication relay device according to the first embodiment of the present invention.
- FIG. 5 is a flowchart of various request processing procedures in the communication relay device according to the first embodiment of the present invention.
- FIG. 6 is a flowchart showing a procedure of an incoming call process in the incoming call terminal 102 according to the first embodiment of the present invention.
- FIG. 7 is an explanatory diagram showing an outline of the “reconnecting method” realized by the communication relay device according to the second embodiment of the present invention.
- FIG. 8 is an explanatory diagram showing an outline of a “reconnecting method” realized by the communication relay device according to the second embodiment of the present invention.
- FIG. 9 is an explanatory diagram of a functional configuration of the communication relay device according to the second embodiment of the present invention.
- FIG. 10 is a flowchart of a process procedure of various requests in the communication relay device according to the second embodiment of the present invention.
- Fig. 11 is a flowchart showing a procedure of an incoming call process in the incoming call terminal 702 according to the second embodiment of the present invention.
- FIG. 12 is an explanatory diagram showing an outline of the “reconnect + reconnect scheme” realized by the communication relay device according to the third embodiment of the present invention.
- FIG. 13 is an explanatory diagram showing an outline of the “reconnect + reconnect scheme” realized by the communication relay device according to the third embodiment of the present invention.
- FIG. 14 is a flowchart illustrating processing procedures of various requests in the communication relay device according to the third embodiment of the present invention.
- FIG. 15 is a flowchart of the incoming call processing procedure at the incoming call terminal 1202 according to the third embodiment of the present invention.
- FIG. 16 is a flowchart showing the procedure of the calling process in the calling terminal 1201 according to the third embodiment of the present invention.
- FIG. 17 is an explanatory diagram showing an outline of the “recall method” realized by the communication relay device according to the fourth embodiment of the present invention.
- FIG. 18 is an explanatory diagram of a functional configuration of the communication relay device according to the fourth embodiment of the present invention.
- FIG. 19 is a flowchart of processing procedures of various requests in the communication relay device according to Embodiment 4 of the present invention.
- FIG. 20 is an explanatory view showing a problem of the prior art.
- FIG. 21 is an explanatory view showing another problem of the prior art.
- FIG. 1 and FIG. 2 are explanatory diagrams showing an outline of the “callback method” realized by the communication relay device according to the first embodiment of the present invention.
- the calling terminal 101 may be any device capable of VoIP calls using SIP (Session Initia tion Protocol), but here it is a telephone (IP phone) connected to the IP network via a VoIP adapter, for example.
- SIP Session Initia tion Protocol
- IP phone IP phone
- a caller makes a call to a representative number previously assigned to each called party from the calling terminal 101, for example, “050—XXXX—XXXX”, a SIP URI (Uniform Resource Identifier) that means the number, For example, an INVITE request destined for “sip: + 81—050—XXXX—XXXX @ •••” is sent to the SIP server 100 via the IP network ((1) in the figure).
- the URI is referred to as a “virtual URI”.
- the SIP server 100 has the transfer destination selection function described above, and sends an INVITE request to the callee's virtual URI to any one of the callee's actual SIP URIs (at least one). Forward to.
- the above actual SIP URI is referred to as “incoming terminal URI”, and one selected as the forwarding destination is particularly referred to as “selected URI”.
- the correspondence relationship between the virtual URI and the receiving terminal URI and which of the receiving terminal URIs is the currently selected URI are held in the “forwarding destination table” of the SIP server 100.
- the receiving terminal URI to which “*” is added is the selected URI.
- the INVITE request from the calling terminal 101 is transferred to the receiving terminal 102 uniquely identified by the selected URI after the SIP server 100 changes the destination to the virtual URI power selecting URI.
- the selected URI is a SIP URI that means an identifier (specifically a telephone number) on the PSTN, such as “tel: + 81—070—XXXX—XXXX”
- the above request is first sent to the SP gateway 104 (2 in the figure) is converted into a PSTN call and then transferred to the receiving terminal 102 via the PSTN ((3) in the figure).
- a call charge between the SP gateway 104 and the call receiving terminal 102 is generated even when the call charge is not normally generated, specifically, even when the call receiving terminal 102, which is a dual terminal, is within the WLAN range. End up.
- the selected URI is (a) a SIP URI corresponding to a device that can be connected to a relatively expensive (high communication cost) network and an inexpensive (low communication cost) network.
- SIP URI meaning the former identifier
- SIP URI meaning the PHS number of a dual terminal that can be connected to the PSTN or IP network
- SIP URI meaning the PHS number of a general PHS is (b).
- a SIP URI meaning a PHS number corresponds to (a) if the number is a dual-terminal PHS number, or (b) if it is a general PHS PHS number.
- the type of SIP URI corresponding to (b) is called ⁇ normal ''.
- the caller must pay the call charge that should not be borne by the caller (specifically, the call charge between the S-P gateway 104 and the receiving terminal 102). This is the case when the selected URI is (a), that is, when the type is not “normal”. There In Example 1, if the selected URI type is not “normal”, the CANCEL request is sent immediately after the INVITE request is transferred from the SIP server 100 to the S—P gateway 104 ((2) in the figure). As a result, the call from the SP gateway 104 to the receiving terminal 102 is disconnected before the call charge is generated (specifically, about one call) ((3) in the figure).
- the SIP server 100 also forwards the INVITE request from the calling terminal 101 to the announcement service 105 together with the SP gateway 104 ((4) in the figure), and the announcement service 105 to the calling terminal 101
- the 2XX response (success) and the ACK request from the calling terminal 101 to the announcement service 105 are sequentially transferred.
- the announcement service 105 calls the caller to answer the caller's power, and is instructed to hang up and wait.
- a BYE request is transmitted from the calling terminal 101 to the announcement service 105 ((5) in the figure), the call from the calling terminal 101 is disconnected.
- the called terminal 102 that is so-called “one-off” from the S-P gateway 104 connects to the IP network by the WLA N or connects to the PSTN by the PHS and immediately calls back.
- SIP is used.
- the “transfer source table” of the server 100 the correspondence relationship between the receiving terminal 102 and the calling terminal 101 as the callback destination is held.
- the SIP Sano 100 transfers the INVITE request received from the calling terminal 101 to the SP gateway 104 and the announcement service 105, the virtual URI described in the To header of the request ((1) in the figure) "Receiver Virtual”) and the URI of the calling terminal 101 described in the From header (calling terminal URI; "Caller VoIP”) is registered in the "transfer source table” that holds it.
- the receiving terminal 102 connects to the IP network via the WLAN and calls back to the SP gateway 104
- the INVITE request from the receiving terminal 102 is the destination SP gateway 104. Is transferred to the SIP server 100 (Fig. 1 (6)).
- the INVITE request received by the SIP server 100 includes (a) a normal INVITE request (such as (1) in the figure) and (b) an INVITE request triggered by a past phone call.
- a normal INVITE request such as (1) in the figure
- b an INVITE request triggered by a past phone call.
- the type of I NVITE request corresponding to (a) is called “normal”, and the type of INVITE request corresponding to (b) is called “callback”.
- it can be specified by referring to the transfer destination table and the transfer source table above whether an INVITE request is the above. Specifically, in the forwarding destination table, if the virtual URI associated with the SIP URI described in the From header of the request is also registered in the forwarding source table (b), it is not registered. (A).
- the SIP server 100 specifies the destination of the request as a source terminal URI (corresponding to the virtual URI in the transfer source table).
- “Caller VoIPj” is substituted and transferred to the calling terminal 101 (Fig. 1 (7)).
- 2XX response (success) and ACK request RTP connection is established between the called terminal 102 and one calling terminal 101. (Fig. 1 (8)).
- the SP gateway 104 is the destination.
- An INVITE request is transmitted from the SP gateway 104 to the SIP server 100 (FIG. 2 (7)).
- the SIP server determines whether or not it is a callback in the same manner as described above.
- the destination of the request is transferred to the originating terminal URI retrieved from the transfer source table and transferred (Fig. 2 (8)). ).
- 2XX response uccess
- ACK request RTP connection is established between SP gateway 104 and calling terminal 101 (Fig. 2 (9)).
- the PS gateway 103 of FIG. 1 and FIG. 2 may be omitted (the invention of the first embodiment can be implemented without the PS gateway 103). It is shown for comparison.
- FIG. 3 is an explanatory diagram showing the hardware configuration of the communication relay device (specifically, the SIP server 100 shown in FIG. 1) according to the first embodiment of the present invention.
- CPU 301 controls the entire device.
- the ROM 302 stores a boot program and the like.
- the RAM 303 is used as a work area for the CP 301.
- the HDD 304 controls the read Z write of data to the HD 305 in accordance with the control of the CPU 301.
- HD305 stores data written according to the control of HDD304 To do.
- the FDD 306 controls data read Z write to the FD 307 according to the control of the CPU 301.
- the FD 307 stores data written according to the control of the FDD 306.
- the FD307 is an example of a detachable recording medium. Instead of the FD307, a CD-ROM (CD-R, CD-RW), MO, DVD (Digital Versatile Disk), memory force, etc. may be used. .
- the network IZF 308 is connected to a network such as LANZWAN, and controls transmission and reception of data between the network and the inside of the apparatus.
- the bus 300 connects the above-described units.
- this device may have consoles such as a display, a mouse, and a keyboard.
- FIG. 4 is an explanatory diagram showing a functional configuration of the communication relay device according to the first embodiment of the present invention (specifically, the SIP server 100 shown in FIG. 1), and FIG. It is a flowchart which shows the process sequence of various requests. The function of each part in Fig. 4 will be explained in sequence in the flowchart of Fig. 5.
- step S501 When the proxy 402 of the SIP server 100 receives the NVITE request (step S501: Yes), the request is output to the transfer destination specifying unit 403 and the INVITE type specifying unit 403a of the transfer destination specifying unit 403 Specify the request type. If the type is “normal” (step S502: Yes), the type of the selected URI associated with the virtual URI of the destination of the above request is specified. Is identified.
- the transfer destination specifying unit 403 is the proxy 402.
- the INVITE request, the selected URI, and the SIP URI of the announcement service 105 input from are output to the request generation unit 404.
- Step S504 and these are sequentially transferred by the proxy 402 (Step S504).
- P505 A CANCEL request destined for the selected URI is transferred after the SP gateway 104 calls the receiving terminal 102 in response to an INVITE request destined for the URI.
- the transfer destination specifying unit 403 instructs the transfer source table update unit 405 to register the pair of the virtual URI of the INVITE request input to the proxy 402 and the calling terminal URI in the transfer source table 401 ( Step S506).
- step S503 when the type of the selected URI is “normal” (step S503: Yes), the forwarding destination specifying unit 403 sends the INVITE request input from the proxy 402 and the selected URI to the request generating unit 404.
- the request generation unit 404 generates an INVITE request with the URI as the destination by replacing the destination of the request with the URI (step S507).
- the request generation unit 404 outputs the above request to the proxy 402, and after the request is transferred (step S508), the normal processing after the INVITE request transfer such as the response transfer or the ACK request transfer is performed. Perform (Step S509).
- step S502 if the type of INVITE request input from proxy 402 is "callback" (step S502: No), that is, the request is triggered by another INVITE request previously transferred in steps S501-S506. If the request is received, the transfer destination specifying unit 403 outputs the request and the originating terminal URI registered in the transfer source table 401 when transferring the other request to the request generation unit 404. Then, the request generation unit 404 generates an INVITE request with the URI as the destination by replacing the destination of the request with the URI (step S510), and the request is transferred to the calling terminal 101 by the proxy 402 (step S510). S 511).
- the transfer destination specifying unit 403 instructs the transfer source table update unit 405 to check the virtual URI and the originating terminal URI registered in the transfer source table 401 when transferring another INVITE request that has been triggered. Delete the pair (step S512). Thereafter, the proxy 402 performs normal processing after transferring the INVITE request, such as transferring a response or transferring an ACK request (step S509).
- step S501 No
- step S5 server 100 performs other processing corresponding to the received request (step S513).
- FIG. 6 is a flowchart showing the procedure of the incoming call process in the incoming call terminal 102 according to the first embodiment of the present invention.
- Step S601 If there is an incoming call from the S—P gateway 104 while waiting by PHS (Step S601: Yes, Step S602: Yes. Note that the incoming call from the S—P gateway 104 can be identified by the telephone number). 102 waits for call disconnection without outputting a ring tone, turns on the WLAN that is turned off (step S603), determines whether it is outside the zone Z of the WLAN (step S604), and based on this result, the WLA Select one of NZPHS as the communication method. In addition to the above judgment results, you may select one of the communication methods based on the communication cost, call quality, preset preferences, etc.
- step S605 If the WLAN is selected (step S605: Yes), the receiving terminal 102 connects to the IP network via the WLAN, and sends an INVITE request to the S-P gateway 104 (step S606).
- SIP call (step S607) is started.
- the WLAN that was turned on at the incoming call is turned off again (step S608).
- a REGISTER request is sent to the SIP server 100 before the INVITE request.
- step S605 when PHS is selected (step S605: No), the receiving terminal 102 turns off the WLAN that has been turned on again (step S609).
- step S610 connect to the PSTN with PHS and place a call to the telephone number of S—P gateway 104 (step S610), so that a normal PHS call (step S611. To become a call).
- a force other than the SP gateway 104 is applied to the telephone strength S, a ring tone is output (Step S602: No, Step S612), and after the called party answers the phone, it becomes a normal PHS call (Step S602). S611).
- FIG. 1 and FIG. 2, and FIG. 5 and FIG. 6 will be described in association with each other.
- the SIP server 100 since the call forwarding destination from the calling terminal 101 is not “normal”, the SIP server 100 The gateway 104 is instructed to call the incoming terminal 102 with a so-called “one-off” method that does not involve the power of call charges, and the outgoing terminal 101 is connected to the announcement service 105 (FIG. 5, steps S501 to S506).
- the receiving terminal 102 connected to the IP network via the WLAN and returned an INVITE request.
- steps S601-S606 or as a result of terminating terminal 102 connecting to PSTN via PHS and calling S-P gateway 104 (Fig. 6, steps S601-S605, S609-S610), S-P gateway
- the SIP server 100 identifies that the request is a callback and transfers the destination after changing the destination to the calling terminal 101 (step S501 in FIG. 5). — S502, S510— S512).
- the SIP server 100 transfers the response to the receiving terminal 102 or the SP gateway 104 and receives an ACK request from the receiving terminal 102 or the SP gateway 104. Then, the request is transferred to the calling terminal 101 (step S509 in FIG. 5).
- the PSTN is used only (3).
- the communication path is the same as the dotted line arrow in Fig. 20 (however, the direction of the arrow is reversed due to the callback), and compared to the conventional solid line arrow, the call charge between S-P gateway 2004 and receiving terminal 2002 Can be reduced.
- Example 1 in addition to making the caller feel uncomfortable (because the phone call must be hung up), the caller and the caller can actually talk. It takes time to become. Therefore, the caller power is also called back as in the first embodiment. As in the second embodiment described below, if the caller is connected in the conventional manner, the communication path is relatively expensive. Try to reconnect the call from the calling party to the called party via the relatively inexpensive communication path B.
- FIG. 7 and FIG. 8 are explanatory diagrams showing an outline of the “reconnection method” realized by the communication relay device according to the second embodiment of the present invention.
- the SIP server 700 searches the forwarding destination table for the selected URI associated with the virtual URI, and determines the destination of the request. Replace with the selected URI and transfer. If this selected URI is a SIP URI that means the PHS number of a dual terminal, the above request is forwarded to the SP gateway 704 ((2) in the figure) and converted to a PSTN call, via the PSTN. Is transferred to the receiving terminal 702 ((3) in the figure). The steps so far are the same as in the callback method of the first embodiment.
- the terminating terminal 702 called from the S—P gateway 704 selects one of the WLAN ZPHS as a communication means, and when selecting the WLAN, transmits a REGISTER request to the SIP server 700 as shown in FIG. 7 ( (4) in the figure).
- the REGISTER request received by the SIP server 700 includes (a) a normal REGISTER request and (b) a REGISTER (such as (4) in the figure) triggered by a past phone call.
- a REGISTER request is the above or not can be specified by referring to the transfer destination table and transfer source table described above.
- the virtual URI associated with the SIP URI (the SIP URI where the IP address is registered) described in the To header and From header of the above request is stored in the forwarding source table. Is also registered (b), otherwise (a).
- the type of REGISTER request corresponding to (a) is called “normal”.
- the SIP server 700 sends a CANCEL request to the S-P gateway 704 ((5) in the figure).
- the call to the receiving terminal 702 is disconnected ((6) in the figure).
- the destination of the INVITE request received from the calling terminal 701 is transferred to the SIP URI in which the receiving terminal 702 has registered the IP address ((4) in the figure) (Fig. Medium (7)).
- the SP gateway 704 Upon receiving the above signal, the SP gateway 704 transmits a 2XX response to the SIP server 700 ((5) in the figure), and the SIP server 700 further forwards the response to the calling terminal 701 (in the figure). (6)). After an ACK request is transferred, an RTP connection is established between the calling terminal 701—S—P gateway 704 ((7) in the figure).
- FIG. 9 is an explanatory diagram showing the functional configuration of the device
- FIG. 10 is a flowchart showing the processing procedure of various requests in the device. The function of each part in FIG. 9 will be described in sequence in the flowchart of FIG.
- step S1001 When the proxy 902 power NVITE request of the SIP server 700 is received (step S1001: Yes), the request is output to the transfer destination specifying unit 903 at any time, and the transfer destination type specifying unit 903a of the transfer destination specifying unit 903 With reference to the forwarding destination table 900, the type of the selected URI associated with the virtual URI of the destination of the request is specified.
- step S1002 If the selected URI type is "normal” (step S1002: No), the transfer source table update unit 905 is instructed to set the combination of the virtual URI of the request and the calling terminal URI. It is registered in the transfer source table 901 (step S 1003). If the selected URI type is “normal” (step S 1002: Yes), the process of step S 1003 is omitted.
- the request and the selected URI are output to the request generation unit 904, and the request generation unit 904 replaces the destination of the request with the URI, so that the URI is the destination.
- An INVITE request is generated (step S1004).
- the proxy 902 performs normal processing after transferring the INVITE request, such as transferring a response or transferring an ACK request (step S1006).
- the registrar 906 of the SIP server 700 receives the REGISTER request (step S1007: Yes)
- the R of the registrar 906 follows the normal SIP registration process (step S1008).
- the EGISTER type specifying unit 906a specifies the type of the request.
- step S 1009: Yes If the type is “normal” (step S 1009: Yes), the process directly returns to step S 100 1. On the other hand, if the type is not “normal” (step S 1009: No), that is, if the above request is triggered by the previous INVITE request transferred in steps S1001 to S1006, the registrar 906 Outputs the INVITE request, the selected URI that is the transfer destination of the INVITE request, and the originating terminal URI registered in the transfer source table 901 when the INVITE request is transferred to the request generation unit 904.
- step S1010 are generated (step S1010), and these are sequentially transferred by the proxy 902 (step S1011).
- the registrar 906 instructs the transfer source table update unit 905 to delete the pair of the virtual URI and the transmission terminal URI registered in the transfer source table 901 when the triggered IN VITE request is transferred.
- the proxy 902 performs normal processing after transferring the INVITE request, such as transferring a response or transferring an ACK request (step S1006). If a request other than an INVITE request or a REGISTER request is received (step S1001: No, step S1007: No), the SIP server 700 performs other processing corresponding to the received request (step S1013).
- FIG. 11 is a flowchart showing the procedure of the incoming call process in the incoming call terminal 702 according to the second embodiment of the present invention. If there is an incoming call from the S—P gateway 704 while waiting by PHS (step S1101: Yes, step SI 102: Yes), the receiving terminal 702 turns OFF the WL AN that is OFF (step S 1103). ) Determine whether the zone is outside the Z zone of WL AN (step S1104), and select one of the WLANZPHS as a communication method based on the result. In addition to the above judgment results, communication means may be selected based on communication costs, call quality, and pre-set preferences.
- step S1105 If the WLAN is selected (step S1105: Yes), the receiving terminal 702 is the WLAN. To connect to the IP network and send a REGISTER request to the SIP server 700 (step S 110). On the other hand, when PHS is selected (Step S1105: No), the WLAN that is turned on is set to OF F (Step S1107) and a ring tone is output (Step S1108). After the called party answers the call, it is normal. (Step S 1109).
- step S 1101 Yes, step S1102: No
- the receiving terminal 702 outputs a ring tone (step S1108), and the callee After answering the call, it becomes a normal PHS call (step S 1109).
- step S1111 When an INVITE request is received by the WLAN that was turned ON in step S 1103 (step S1101: No, step SI 110: Yes), the receiving terminal 702 outputs a ring tone (step S1111), and the incoming call is received. After the person answers the call, it becomes a normal SIP call (step S1112). At the end of the call, turn off the WLAN that was turned on when the call was received (step S1113)
- FIG. 7 and FIG. 8, and FIG. 10 and FIG. 11 are explained in correspondence with each other.
- the SIP server 700 first receives a call from the calling terminal 701 to the receiving terminal 702 via the PSTN. However, if there is a REGISTER request from the receiving terminal 702 (FIG. 11, step S1101—S 1106), it is canceled and then transferred to the receiving terminal 702 via the IP network (see figure S1001—S1005). 10 steps S1007—S1012). When there is a 2X X response from the receiving terminal 702, the response is transferred to the calling terminal 701, and when there is an ACK request from the calling terminal 701, the request is transferred to the receiving terminal 702 (FIG. 10 step S10 06). ).
- the S-P gateway 704 force ⁇ SIP Sano 7 00 Since the 2XX response is returned, the SIP server 700 transfers the response to the calling terminal 701 and, when receiving an ACK request from the calling terminal 701, transfers the request to the SP gateway 704 (step S1006 in FIG. 10).
- step S1006 in FIG. 10 is performed when the SIP server 700 receives a 3XX-6XX response from the S—P gateway 704, on the condition that there is no REGISTER request from the receiving terminal 702 within a predetermined time. It may be modified to forward to. In other words, even if a connection failure is notified from the SP gateway 704, it is not immediately notified to the transmitting terminal 701, but the registration from the receiving terminal 702 is waited for the predetermined time.
- the telephone made by the SP gateway 704 in (3) is disconnected before the incoming terminal 702 responds. Call charges are not incurred throughout 10).
- the communication path is the same as the dotted line arrow in FIG. 20, and the call charge between the SP gateway 2004 and the receiving terminal 2002 can be reduced compared to the solid line arrow in the prior art.
- FIG. 8 is the same as the solid line arrow of FIG. 20, that is, in the above-mentioned ubiquitous IP telephone system, it is the same as the case where a call from an IP terminal to a representative number is transferred to a non-IP terminal.
- S-P gateway 2004—calling terminal 2002 is charged to the caller.
- the REGISTER request transmitted from the receiving terminal 702 to the SIP server 700 is essentially a request for SIP registration.
- this is a kind of connection request, that is, a transmitting terminal. It is regarded as a request from the incoming terminal 702 that it should be connected to the 701 via the IP network.
- the SIP server 700 can be said to be redirecting the telephone once transferred via the PSTN via the IP network. it can.
- the above connection request must be a REGISTER request from the receiving terminal 702, not necessarily some kind of notification (not necessarily an incoming call) indicating that the receiving terminal 702 is ready to receive calls on the WLAN. It may not be a notification from the terminal 702 itself, etc.).
- FIG. 12 and FIG. 13 are explanatory diagrams showing an outline of the “reconnect + reconnect scheme” realized by the communication relay device according to the third embodiment of the present invention.
- Example 2 described above is a case where a call is made from an IP terminal (for example, an IP phone), but Example 3 is a non-IP terminal, specifically, This is the case when making a power call such as a conventional phone or dual terminals when outside the WLAN area.
- Example 3 is a non-IP terminal, specifically, This is the case when making a power call such as a conventional phone or dual terminals when outside the WLAN area.
- the calling terminal 2001 makes a call via the PSTN and the called terminal 2002 also receives a call via the PSTN, the call charge between the calling terminal 2001—PS gateway 2003 and the S—P Call charges between gateway 2004 and receiving terminal 2002 are doubled.
- the calling terminal 1201 makes a call via the PSTN ((1) in the figure), and the PS gateway 1203 force and the SIP Sano 1200 are passed through.
- the PS gateway 1204 power is further transmitted from the SIP server 1200 immediately after the INVITE request is transferred when the call is made via the PSTN ((4) in the figure).
- the call from the S—P gateway 1204 to the receiving terminal 1202 is disconnected before the call charge is incurred (specifically, about one call). ((4) in the figure).
- the receiving terminal 1202 transmits a REGISTER request to the SIP server 1200 as shown in FIG. 12 ((5) in the figure). If the above request type is not normal, the SIP server 1200 will register the destination of the INVITE request received from the PS gateway 1203 ((2) in the figure) and the destination terminal 1202 will register the IP address. ((5) in the figure) Transfer to the SIP URI that was changed ((6) in the figure).
- the 2XX response is received from the receiving terminal 1202 ((7) in the figure), the response is transferred to the PS gateway 1203 ((8) in the figure), and further, after the ACK request is transferred, An RTP connection is established between P—S gateway 1203—receiving terminal 1202 ((9) in the figure). After that, a response signal is returned from the P-S gateway 1203 to the calling terminal 1201 via the PSTN ((10) in the figure).
- the receiving terminal 1202 receives a call from the SP gateway 1204 as shown in FIG. 13 ((4) in the figure). Wait for the phone to come up (do nothing from you). S—P gateway 1204 Because the phone with 1204 power is cut off by one time, it is not enough to answer with PHS, and neither power nor WL AN can be used Z Because it is not used, the incoming terminal 1202 is PHS with S—P gateway 12 04 There is no choice but to call back S-P gateway 1204 to SIP server 1 If the INVITE request is transferred to the PS gateway 1203 via 200 and the calling terminal 1201 is called from the PS gateway 1203 via the PSTN, the above-mentioned double charging problem may occur.
- the calling terminal 1201 instead of the receiving terminal 1202 doing nothing, the calling terminal 1201 automatically makes a call again.
- the calling terminal 1201 according to the third embodiment does not connect the call to the called party's representative number within the predetermined time (the response signal is not returned from the PS gateway 1203 within the predetermined time), Select another caller's phone number and try again.
- the other telephone number may be stored in advance in the calling terminal 1201, or may be acquired in real time from the SIP server 1200 or the like.
- the hardware configuration of the communication relay device according to the third embodiment of the present invention (specifically, the SIP server 1200 shown in FIGS. 12 and 13) is the same as that of the first embodiment shown in FIG. Since there is, explanation is omitted.
- the functional configuration of the above apparatus is almost the same as that of the second embodiment shown in FIG. 9, and the differences will be sequentially described in the flowchart of FIG.
- FIG. 14 is a flowchart showing a processing procedure for various requests in the apparatus.
- Example 2 shown in Fig. 10 The difference from Example 2 shown in Fig. 10 is that the CANCEL request generation and transfer processing is in the processing loop when receiving a REGISTER request in Fig. 10 (steps S 1010 ⁇ S 1 011). In Fig. 14, it is the point (steps S 1404 ⁇ S 1405) in the processing loop when receiving the INVITE request that triggered the request. (If a request other than an INVITE request or a REGISTER request is received, SIP The server 1200 performs other processing corresponding to the received request (Step S1401: No, Step SI409: No, Step S1415)).
- the request generation unit 904 for the IN VITE request received by the SIP server 1200, the type of the selected URI specified by the transfer destination type specification unit 903a. Is not “normal”, an INVITE request and a CANCEL request with the URI as the destination are generated, and these are sequentially transferred by the proxy 902 (steps S 14 01 to S 1405). If the selected URI type is “Normal”, an INVITE request is generated with the URI as the destination, and after this is forwarded by the proxy 902, the proxy 902 forwards the response, ACK request, etc. Then, normal processing is performed after the INVITE request is transferred (steps S1401 to S1402, S1406 to S1408).
- the request generation unit 904 for the REGI STER request received by the SIP server 1200, if the type power S specified by the REGISTER type specification unit 906a is not "normal", the IP address An INVITE request whose destination is the registered SIP URI is generated and transferred to the receiving terminal 1202 by the proxy 902 (steps S1409 to S1414).
- FIG. 15 is a flowchart showing the procedure of the incoming call process in the incoming call terminal 1202 according to the third embodiment of the present invention.
- the processing in steps S 1501 to S 1513 in the figure is the same as the processing in steps S 1101 to S 1113 shown in FIG.
- step S1505 No, step S1507
- the call of the S-P gateway 1 204 is disconnected! Because PHS calls are not possible, step S 1501 [Return!
- FIG. 16 is a flowchart showing the procedure of the calling process in the calling terminal 1201 according to the third embodiment of the present invention.
- a calling instruction is issued for a specific telephone number, for example, a representative number of the called party (step S1601: Yes)
- the calling terminal 1201 calls the number via the PSTN (step S1602).
- the above call is converted to an I NVITE request by the PS gateway 1203, transferred to the SIP server 1200, and when a 2XX response is returned from the SIP server 1200, a response signal is sent from the PS gateway 1203 to the calling terminal 1201. Is returned (step S 1603: Yes), and then a normal voice call is made (step S 1604).
- Step S1603: No, Step S1605: No the calling terminal 1201 waits for the above signal for a predetermined time from the call (Step S1603: No, Step S1605: No), and responds from the PS gateway 1203 even if the time has elapsed. If there is no phone number (Step S1603: No, Step S1605: Yes), if there is another phone number (Step S1606: Yes), it is still possible to make a call to the above called party. Above (step S1607), the corresponding number is called (step S1602). If all the telephone numbers of the called party are called and there is still no response (step S 1606: No), the process ends at that point.
- FIG. 12 and FIG. 13 are related to FIG. 14 and FIG. 16.
- the SIP server 1200 does not wait for a REGISTER request from the receiving terminal 1202, and If the incoming terminal 1202 receives a REGISTER request ( Figure 15 steps S 1501-S 1506), the IP address is sent to the registered SIP URI. The request is transferred (steps S1409 to S1414 in Fig. 14).
- the SIP server 1200 force NVITE request is canceled (FIG. 14 Steps S1401 to S1405).
- Terminal 1201 selects the other telephone number of the called party, in this case the PHS number of the dual terminal, and makes a call again (steps S 1601 to S 1607 in FIG. 16).
- FIG. 17 is an explanatory diagram illustrating an outline of the “recall method” realized by the communication relay device according to the fourth embodiment of the present invention.
- Example 4 causes double charging.
- the INVITE request (specifically (2) in the figure) that is transferred from the P-S gateway 1703 to the S-P gateway 1704 via the SIP server 1700 is discarded by the SIP server 1700. Do not forward to that destination.
- the calling terminal 1701 selects another telephone number of the called party, for example, a PHS number of a dual terminal.
- the call is corrected 4 times ((3) in the figure) and a response signal is returned from the receiving terminal 1702 ((4) in the figure), it becomes a normal PHS call.
- FIG. 18 is an explanatory diagram showing the functional configuration of the device, and FIG. The function of each part in FIG. 18 will be described in sequence in the flowchart of FIG.
- the proxy 1801 of the SIP server 1700 receives the INVITE request (step S19 01: Yes)
- the request is output to the transfer destination specifying unit 1802 at any time, and the transfer availability determining unit 1802a of the transfer destination specifying unit 1802
- the transfer destination table 1800 is referenced to determine whether the request can be transferred (step S 1902).
- a SIP URI indicating a telephone number is described in the To header and From header.
- step S1902 If transfer is not possible (step S1902: No), the process returns to step S1901 to wait for reception of another request, while if transfer is possible (step S 1902: Yes), the request generation unit 1803 By changing the destination of the request to the selected URI in the transfer destination table 1800, an INVITE request for the URI is generated (step 1903). After this is transferred by the proxy 1801 (step S1904), the proxy 1801 performs normal processing after the INVITE request transfer, such as a response transfer or an ACK request transfer (step S 1905).
- step S1901 If a request other than an INVITE request is received (step S1901: No)
- the SIP server 1700 performs other processing corresponding to the received request (step S1 906).
- the receiving terminal 1702 of the fourth embodiment may be any device capable of receiving a call on the PSTN, and the processing in the device is the same as that of the conventional technology.
- the calling terminal 1701 of the fourth embodiment may be any device that can make a call on the PSTN, but, as with the calling terminal 1201 of the third embodiment, a telephone call for a predetermined time or more is performed according to the procedure of FIG. It is assumed that it has a function to call back to another number when it is not connected.
- Example 4 INVITE requests that cause double billing when transferred are limited to SIP server 1700, and calling terminal 1701 will make another call over another route. Double charge does not occur. More generally speaking, if there is a relatively expensive communication path A and an inexpensive communication path B, if the telephone power S is applied via A first, the power on the path (for example, Since the SIP server 1700) discards it as shown in Example 4 and makes a call via B again, the communication path when the call is finally connected is always the cheapest.
- calling terminal 101Z701 is an IP terminal
- calling terminal 1201 Z 1701 is a non-IP terminal.
- the INVITE request received by the SIP server includes (a) an INVITE request sent by the calling terminal itself, and (b) an INVITE request transferred by the PS gateway after converting the call from the calling terminal.
- a SIP server having both functions of the SIP server 700 of the second embodiment and the SIP server 1200 of the third embodiment is prepared, and when (a) is received, the procedure of FIG. When (b) is received, it is processed according to the procedure shown in Fig. 14.
- S of Example 1 Prepare a SIP server that has the functions of both the IP server 100 and the SIP server 1700 of the fourth embodiment. When (a) is received, the procedure of FIG. 5 is performed. When (b) is received, the procedure of FIG. 19 is performed. Make sure to do it.
- the communication relay method, the communication relay program, and the communication relay device that are effective in the present invention are useful for selecting an optimum route when there are a plurality of communication routes between the calling terminal and the receiving terminal.
- it is suitable for a case where a relatively inexpensive communication path becomes practically unusable for some reason (for example, power saving).
Abstract
Description
Claims
Priority Applications (4)
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JP2006542195A JP4627760B2 (ja) | 2004-11-02 | 2004-11-02 | 通信中継方法、通信中継プログラムおよび通信中継装置 |
PCT/JP2004/016265 WO2006048925A1 (ja) | 2004-11-02 | 2004-11-02 | 通信中継方法、通信中継プログラムおよび通信中継装置 |
EP04799468A EP1809010A4 (en) | 2004-11-02 | 2004-11-02 | COMMUNICATION RELAY METHOD, COMMUNICATION RELAY PROGRAM, AND COMMUNICATION RELAY APPARATUS |
US11/797,363 US20080212764A1 (en) | 2004-11-02 | 2007-05-02 | Communication relay apparatus, communication relay method, and computer product |
Applications Claiming Priority (1)
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PCT/JP2004/016265 WO2006048925A1 (ja) | 2004-11-02 | 2004-11-02 | 通信中継方法、通信中継プログラムおよび通信中継装置 |
Related Child Applications (1)
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US11/797,363 Continuation US20080212764A1 (en) | 2004-11-02 | 2007-05-02 | Communication relay apparatus, communication relay method, and computer product |
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WO2006048925A1 true WO2006048925A1 (ja) | 2006-05-11 |
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ID=36318945
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PCT/JP2004/016265 WO2006048925A1 (ja) | 2004-11-02 | 2004-11-02 | 通信中継方法、通信中継プログラムおよび通信中継装置 |
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US (1) | US20080212764A1 (ja) |
EP (1) | EP1809010A4 (ja) |
JP (1) | JP4627760B2 (ja) |
WO (1) | WO2006048925A1 (ja) |
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JP2008109191A (ja) * | 2006-10-23 | 2008-05-08 | Nec Infrontia Corp | 携帯電話端末および電話制御方法 |
JP2008219733A (ja) * | 2007-03-07 | 2008-09-18 | Softbank Mobile Corp | 移動無線電話による通信と無線lanアクセスポイントによる通信の選択方法及びシステム |
JP2020150296A (ja) * | 2019-03-11 | 2020-09-17 | 三菱自動車工業株式会社 | 車両の無線通信装置 |
JP2020198636A (ja) * | 2015-09-25 | 2020-12-10 | Line株式会社 | 効率的な呼処理のためのシステムおよび方法 |
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US7212615B2 (en) * | 2002-05-31 | 2007-05-01 | Scott Wolmuth | Criteria based marketing for telephone directory assistance |
US9367846B2 (en) * | 2004-11-29 | 2016-06-14 | Jingle Networks, Inc. | Telephone search supported by advertising based on past history of requests |
US9363228B2 (en) * | 2009-12-15 | 2016-06-07 | Qualcomm Innovation Center, Inc. | Apparatus and method of peer-to-peer communication |
KR101909982B1 (ko) * | 2011-12-22 | 2018-10-23 | 삼성전자 주식회사 | VoIP 게이트웨이 장치, 이의 제어방법 및 이를 포함하는 시스템 |
US9762628B2 (en) | 2013-02-19 | 2017-09-12 | Avaya Inc. | Implementation of the semi-attended transfer in SIP for IP-multimedia subsystem environments |
US9467570B2 (en) * | 2013-11-20 | 2016-10-11 | Avaya Inc. | Call transfer with network spanning back-to-back user agents |
KR102231859B1 (ko) * | 2014-09-01 | 2021-03-26 | 삼성전자주식회사 | 무선 통신 시스템에서 단말이 서비스 연결을 유지하는 장치 및 방법 |
US11233831B2 (en) * | 2019-07-31 | 2022-01-25 | Centurylink Intellectual Property Llc | In-line, in-call AI virtual assistant for teleconferencing |
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Also Published As
Publication number | Publication date |
---|---|
EP1809010A1 (en) | 2007-07-18 |
JP4627760B2 (ja) | 2011-02-09 |
JPWO2006048925A1 (ja) | 2008-05-22 |
EP1809010A4 (en) | 2010-01-27 |
US20080212764A1 (en) | 2008-09-04 |
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