JP2009218889A - Sip call termination method - Google Patents

Abstract

<P>PROBLEM TO BE SOLVED: To provide an SIP call termination method which enables call termination even if an SIP-gateway apparatus fails, and/or a connection line is in use and busy. <P>SOLUTION: The transmitting SIP-GW apparatus transmits a transmitting signal to the receiving SIP-GW apparatus (S10), determines whether a reply signal is supplied from the receiving SIP-GW apparatus (S12), determines whether insufficient supply of the reply signal is continued for 10 seconds (S14). When continued for 10 seconds, an IP address in a plurality of receiving SIP-GW apparatuses for storage is extracted (S16). When the reply signal is received within 10 seconds, the transmitting signal is transmitted again (S18), to determine the supply of the reply signal (to S12). After extracting the IP address, the presence of the extracted IP address is determined (S20). The IP address extracted according to the presence of the extracted IP address is set to the IP address of the transmitting signal to be transmitted next, the transmitting signal is transmitted (S22) to determine the supply of the reply signal (to S12). <P>COPYRIGHT: (C)2009,JPO&INPIT

Description

  The present invention relates to a SIP (Session Initiation Protocol) receiving method, and more specifically, a plurality of gateway devices using a SIP interface are installed, a failure occurs, and a line of one gateway device is blocked during use. However, the present invention relates to a SIP bypass incoming method in which a plurality of other SIP-gateway devices are diverted to receive incoming calls.

  Conventionally, telephone lines between headquarters and branch offices have been used by companies such as Nippon Telegraph and Telephone Corporation (NTT) between private branch exchanges (PBXs) installed at the headquarters and branch offices. An analog or digital telephone line provided by a communication carrier, a so-called leased line, has been borrowed and used on a monthly basis. In recent years, with the spread of IP (Internet Protocol), dedicated lines as well as analog and digital telephone lines are shifting to use of IP networks. The advantage of using an IP network is that it is much cheaper than conventional analog and digital telephone lines.

  The conventional PBX does not have a function to connect to an IP leased line. The PBX is connected via a gateway device that converts an analog / digital dedicated line interface to an IP interface when connecting to an IP dedicated line, that is, an IP dedicated line. In recent years, the SIP (Session Initiation Protocol) interface defined in IETF (Internet Engineer Task Force) RFC (Request For Comments) 3261 has been used as a protocol for connecting telephones between GWs connected to IP leased lines. Generally popular.

Conventionally, in a PBX connected by a SIP-gateway device, if the SIP-gateway device of the opposite PBX is out of order, it is possible to receive a detour while there are other SIP-gateway devices that can receive the call. There wasn't. This is because there was no procedure for making an incoming call by detour.
IETF RFC3261 Session Initiation Protocol (SIP), June 2002

  In the case of the conventional technology, in the case of an IP leased line where PBXs are connected via a SIP-gateway device, whether the incoming call can be received by the terminating SIP-gateway device opposite the originating PBX This is not possible until the SIP-gateway device tries to receive a call. For this reason, if the opposite SIP-gateway device on the opposite side is out of order or cannot receive a call while all lines are being used, the opposite SIP-gateway device on the opposite side will fail for the outgoing signal (ini-INVITE) of the SIP procedure. In this case, the SIP-gateway device on the called side cannot return a response signal (100 trying / 180 ringing) to the calling-side SIP-GW to the outgoing signal (ini-INVITE).

  In the conventional technology, if the SIP-gateway device on the receiving side is out of order, even if there is another SIP-gateway device that can receive the call on the receiving side, the SIP-gateway device in failure can be diverted and received. could not. As a result, incoming calls via the failed SIP-gateway device could not be made until the failure was resolved.

  In addition, if all lines connected to one SIP-gateway device are busy and blocked, the SIP-gateway device on the receiving side indicates that the incoming call is not possible (488 not acceptable here or 600 busy everywhere) Is sent back to the SIP-gateway device on the caller side. The SIP-gateway device on the calling side receives a signal indicating that the incoming call cannot be received as a response signal, and transmits a busy tone indicating that the incoming call cannot be received to the originating SIP phone.

  SUMMARY OF THE INVENTION The present invention has been made in view of the above problems, and an object of the present invention is to provide a SIP incoming method that enables incoming calls even when a SIP-gateway device fails and / or a connected line is busy.

  In order to solve the above-described problem, the present invention provides a caller and an incoming call via a dedicated line that communicates between a plurality of gateway devices based on IP (Internet Protocol) to the callee side facing the gateway device on the caller side. A communication terminal device is connected to each of the gateway devices on the side, and a method for receiving an incoming call based on SIP (Session Initiation Protocol) for communication with each of the communication terminal devices, the method includes: A first step of transmitting a transmission signal to the device; a second step of determining whether or not a response signal is supplied from the gateway device on the incoming side to the gateway device on the outgoing side in the gateway device on the outgoing side; A third step of determining whether or not the lack of supply has continued for a first predetermined time, and a caller side when continuing for a first predetermined time A fourth step of extracting IP addresses in a plurality of gateway devices on the receiving side stored in the gateway device, and retransmitting the outgoing signal when the response signal is received within a first predetermined time, The fifth step returns to step 2, the sixth step after the fourth step to determine the presence or absence of the extracted IP address, and the extracted IP address is transmitted next depending on the presence of the extracted IP address A seventh step of setting the IP address of the transmission signal to be transmitted, transmitting the transmission signal, and returning to the second step.

  According to the SIP incoming method according to the present invention, a call origination signal is transmitted from a caller-side gateway device to a callee-side gateway device, and a response signal is transmitted from the callee-side gateway device to the caller-side gateway device. It is determined whether or not the response signal has been supplied, and it is determined whether or not the supply of the response signal has continued for a first predetermined time. When the response signal continues for the first predetermined time, the originating gateway device stores it. After extracting the IP address in the plurality of gateway devices on the receiving side and receiving the response signal within the first predetermined time, retransmit the outgoing signal, return to the determination of supply of the response signal, and after extracting the IP address Determine the presence or absence of the extracted IP address, set the extracted IP address as the IP address of the outgoing signal to be sent next, according to the presence of the extracted IP address, this outgoing signal Even if the gateway device facing the calling side breaks down by returning to the determination of supply of the response signal, it is possible to quickly bypass the other gateway device that can receive incoming calls and communicate with the communication terminal device. Become.

  Next, an embodiment of a SIP incoming method according to the present invention will be described in detail with reference to the accompanying drawings. Referring to FIG. 1, in the embodiment of the SIP terminating method according to the present invention, in a private branch exchange system, a calling signal is transmitted from a calling-side gateway device to a called-side gateway device (step S10). It is determined whether or not a response signal is supplied from the gateway device on the side to the gateway device on the transmission side (step S12), and it is determined whether or not the supply of the response signal has continued for a first predetermined time (step S14). ), When continuing for the first predetermined time, the IP addresses of the plurality of gateway devices on the incoming side stored in the gateway device on the outgoing side are extracted (step S16), and a response signal is received within the first predetermined time. When the message is received, the outgoing signal is retransmitted (step S18), the process returns to the determination of supplying the response signal, and after extracting the IP address, the presence or absence of the extracted IP address Determine (step S20), set the extracted IP address according to the presence of the extracted IP address as the IP address of the outgoing signal to be transmitted next, send this outgoing signal (step S22), and supply the response signal By returning to this determination, even if the gateway device facing the transmission side breaks down, it is possible to quickly bypass the other gateway device that can receive incoming calls and communicate with the communication terminal device.

  In this embodiment, the SIP incoming method of the present invention is applied to the SIP-gateway device 10. The illustration and description of parts not directly related to the present invention are omitted. In the following description, the signal is indicated by the reference number of the connecting line in which it appears.

  The SIP-gateway (GW) device 10 has a gateway function for comprehensively communicating voice and / or data between an analog telephone line and / or a digital telephone line and an IP (Internet Protocol) network. The SIP-GW apparatus 10 includes a line signal / packet conversion unit 12, a packet switch 14, and a control unit 16, as shown in FIG.

  In addition, when communicating with the counterpart SIP-GW device, the SIP-GW device 10 controls call connection using SIP (Session Initiation Protocol) and transmits a voice signal using VoIP (Voice over Internet Protocol). In addition, when transmitting a FAX signal, it transmits using FoIP (FAX over Internet Protocol), and when transmitting information, that is, data, it transmits using TCP (Transmission Control Protocol) or UDP (User Datagram Protocol). Have

  The line signal / packet conversion unit 12 divides the signal 18 output from, for example, the PBX installed on the transmission side into blocks of a predetermined length, that is, packets according to the control signal 20 from the control unit 16, and further connects Each packet has a function of assigning a destination to each packet based on the previous telephone number, assigning a transmission order, storing a predetermined amount of packets, processing the packets, and outputting the processed packets 22 to the packet switch 14. Further, the line signal / packet conversion unit 12 receives and accumulates the packet 22 output from the packet switch 14 in response to the control signal 20 from the control unit 16, and stores the telephone number of the connection destination stored in the packet 20 , Extract the destination of each packet and the transmission order given to each packet, assign a telephone number to an analog telephone line and / or digital telephone line, convert it to the signal format of each line, and convert the converted signal 18 Has a function to output to PBX.

  The line signal / packet conversion unit 12 further includes a subscriber loop interface circuit (SLIC: Subscriber-Loop-Interface-Circuit and a telephone line interface function, supply of call current to the installed PBX, transmission of a ringing signal, It has functions including a subscriber interface function such as end-of-call monitoring and a service function corresponding to the subscriber type.

  The packet switch 14 receives the packet 22 output from the line signal / packet conversion unit 12, and the input packet 22 is selected as the packet 26 in accordance with the control signal 24 from the control unit 16 and is selected by the control unit 16 It has the function of controlling the route toward the destination and outputting it to the IP network. As a result, since the packet switch 14 transmits the packet 22 output from the line signal / packet conversion unit 12 to the PBX installed at the other party, one of the plurality of SIP-GW apparatuses installed at the other party of the communication Send the route to the destination. The packet switch 14 has a function of inputting the supplied packet 26 and outputting the input packet 26 to the line signal / packet conversion unit 12.

  The control unit 16 has a function of controlling the entire components of the SIP-GW apparatus 10. This control is performed by a CPU (Central Processing Unit) 28. The control unit 16 has a SIP-GW bypass list 30 that stores an IP address indicating a destination. This list 30 is a database. The reading of the SIP-GW bypass list 30 in the control unit 16 will be described in detail later.

  Next, FIG. 3 shows a configuration example of a private branch exchange system between the head office and the branch office to which the SIP-gateway device 10 is applied. As shown in FIG. 3, the private branch exchange system 32 includes a head office side component 34 and a branch office side component 36, and the components 34 and 36 communicate via an IP network 38.

  Here, the IP network 38 is not limited to the Internet, and may be a local area network (LAN), a wide area network (WAN), and a virtual private network (VPN). Good.

  The head office side component 34 includes a SIP phone 40, a PBX 42 and a SIP-GW device 10, and the branch office side component 36 includes a SIP-GW device 44, 46 and 48, a PBX 50 and a SIP phone 52. SIP phones 40 and 52 may be analog or digital telephones.

  The SIP-GW apparatus 10 is connected to the PBX 42 via an analog telephone line and / or a digital telephone line, and exchanges a line signal 18. The PBX 42 is connected to the SIP phone 40 via an analog telephone line and / or a digital telephone line, that is, an extension line 54.

  The branch-side component 36 consists of SIP-GW devices 44, 46 and 48 each connected to an IP network 38, and each SIP-GW device 44, 46 and 48 connected to a PBX 50 with an analog telephone line and / or a digital telephone line. And supply line signals 56, 58 and 60 to the PBX 50, respectively. The PBX 50 is connected to the SIP phone 52 via an analog telephone line and / or a digital telephone line, that is, an extension 62.

  Although not shown, each of the SIP-GW apparatuses 44, 46 and 48 includes a line signal / packet conversion unit 12, a packet switch 14, and a control unit 16. The control unit 16 of each of the SIP-GW apparatuses 44, 46, and 48 does not have the SIP-GW detour list 30, but may be provided to each of the SIP-GW apparatuses 44, 46, and 48.

  In the present embodiment, in the private branch exchange system 32 between the head office and the branch office, a plurality of SIP-GW apparatuses 44, 46 and 48 are installed in the PBX 50 of the incoming call destination. In this configuration, even if one of the SIP-GW devices 44, 46, and 48 fails, if there is one other SIP-GW device that can function normally and can be received, it bypasses the failed SIP-GW device. FIG. 1 shows a procedure for enabling an incoming call to an incoming SIP-GW apparatus.

  First, a transmission signal is transmitted from the head office side SIP phone 40 to the branch office side SIP phone 52 via the SIP-GW apparatus 10 by extension (step S10: ini-INVITE).

  Next, the SIP-GW apparatus 10 determines whether or not a response signal (100trying / 180rining) to the outgoing signal has been received (step S12). When the response signal is received (YES), the process proceeds to the normal partner call process via the connector A. If no response signal is received (NO), the process proceeds to timeout determination (to step S14).

  Next, the SIP-GW apparatus 10 determines whether or not a situation in which a response signal cannot be received has timed out (step S14). The predetermined time when the response signal cannot be received is 10 seconds. This time is determined by the detour timer of the SIP-GW device 10 for timeout. If not timed out (NO), the process proceeds to retransmission of the outgoing signal (to step S16). If timed out (YES), the process proceeds to IP address extraction (to step S18).

  The SIP-GW apparatus 10 retransmits the outgoing signal (step S16). After the retransmission, the process proceeds to reception determination of the response signal (go to step S12).

  Next, the SIP-GW apparatus 10 extracts an IP address from the list data of the branch office side SIP-GW apparatus according to the timeout (step S18). The control unit 16 sequentially extracts the IP addresses in the SIP-GW bypass list 30. The control unit 16 may select an IP address so that the extracted IP address is not extracted twice.

  As a result of extracting the IP address, it is determined whether there is an IP address corresponding to another SIP-GW device (step S20). When the IP address is extracted (YES), the process proceeds to the outgoing call process (to step S22). If the IP address has not been extracted (NO), the process proceeds to the retransmission timeout determination process (to step S24).

  In the transmission process, the destination of the IP packet, the request URI (Uniform Resource Identifiers) of the transmission signal (ini-INVITE), and the IP address part of the to header are set as the extracted IP address from the extracted IP address. The SIP-GW apparatus 10 transmits a transmission signal to the branch office-side SIP-GW apparatus (step S22). After the transmission signal is transmitted, the process proceeds to reception determination of the response signal (to step S12).

  The retransmission timeout determination process determines whether it is within 32 seconds described in RFC3261 (step S24). If the specified time is exceeded (YES), the outgoing call processing is shifted to the end, and is ended. If it is within the specified time (NO), the process proceeds to retransmission of the transmission signal (to step S16).

  When operated in this way, even if the opposite SIP-GW device is out of order, if there is another SIP-GW device that can receive incoming calls, the SIP-GW device can quickly bypass the other SIP-GW device that can receive incoming calls. It becomes possible to make a call with the phone 52.

  A specific sequential in the private branch exchange system 32 between the head office and the branch office is shown in FIG. In the sequential example of this embodiment, PBXs 42 and 50 are omitted. When the head office calls the branch office by dialing from the extension, the SIP-GW devices 44 and 46 on the branch office side are out of order. Upon receiving the dial transmission, the SIP-GW apparatus 10 transmits a transmission signal 54 to the SIP-GW apparatus 44 (time T10: ini-INVITE 54). Since the SIP-GW device 44 is out of order, the response signal (100 trying / 180 ringing) cannot be returned to the SIP-GW device 10.

  Originally, in the SIP procedure, the head office-side SIP-GW apparatus 10 continues to retransmit the transmission signal (ini-INVITE) for 32 seconds until a response signal is returned. However, in this embodiment, the retransmission time is set to 10 seconds. This retransmission time can be changed by data or the like. If there is no response from the SIP-GW device 44 for 10 seconds, the SIP-GW device 10 extracts the IP address of the SIP-GW device 46 from the SIP-GW bypass list 30 of the control unit 16, and then transmits the IP packet And set the information in the outgoing signal (ini-INVITE), that is, the IP address of the request URI, to header as the IP address of the SIP-GW device 46, and send it to the SIP-GW device 46 (time) T12: ini-INVTE 56).

  Since the SIP-GW device 46 is also malfunctioning, it cannot respond. The SIP-GW device 10 extracts the IP address of the SIP-GW device 48 from the SIP-GW bypass list 30, sets it as the destination of the next IP packet to be transmitted, and sets the outgoing signal 58 in which the next IP address is set. Make a call (time T14: ini-INVTE 58). SIP-GW # 2 can receive calls. As a result, the private branch exchange system 32 between the head office and the branch office can set the destination to the SIP-GW apparatus 48 that can receive a call.

  The SIP-GW device 48 calls the SIP phone 52 corresponding to the branch office extension in response to the outgoing signal 58. Then, the SIP-GW apparatus 48 returns a response signal 60 to the transmission signal 58 (time T16: 100 trying / 180 ringing 60).

  When the SIP phone 52 on the branch office side responds to the SIP-GW device 48, the SIP-GW device 48 returns a response signal 62 to the SIP-GW device 10 (time T18: 200 OK 62). The SIP-GW apparatus 10 receives the return and transmits a response signal 64 to the SIP-GW apparatus 48 (time T20: ACK 64). From then on, it will be an extension call between the head office and the branch office.

  Next, FIG. 5 shows another embodiment of the SIP bypass incoming call method to which the SIP incoming call method according to the present invention is applied. As for the detouring procedure in the present embodiment, description of the procedure common to the previous embodiment is omitted. In this embodiment, the branch-side SIP-GW apparatus fails and another SIP-GW apparatus is in use because the line is in use.

  In this embodiment, a line blockage determination process is provided between the time-up determination process (step S14) and the IP address extraction (step S26). In the line blocking determination process, it is determined whether or not the line of the SIP phone 52 on the receiving side connected to any of the SIP-GW apparatuses 44, 46, and 48 is blocked. If the line blocking signal (488/600) is received after the time-out (YES), the process proceeds to IP address extraction (to step S16). If the line blocking signal (488/600) is not received (NO), the process proceeds to the normal partner call process via the connector A.

  When operated in this way, even if the opposite SIP-GW device is out of order, if there is another SIP-GW device that can receive incoming calls without waiting for the line to become free, it can receive incoming calls quickly. It is possible to bypass the device and make a call.

  FIG. 6 shows a specific sequential in the private branch exchange system 32 between the head office and the branch office. The SIP-GW apparatus 10 transmits a transmission signal 56 to the SIP-GW apparatus 46 at time T12. Since the SIP-GW apparatus 46 is in the line blocking state, the line blocking signal 64 is returned to the SIP-GW apparatus 10 at time T22 (488/600 64). The SIP-GW apparatus 10 receives the line blocking state of the SIP-GW apparatus 46, extracts the IP address of the SIP-GW apparatus 48, and transmits an outgoing signal 58 to the SIP-GW apparatus 48 (ini-INVITE 58) . Thereafter, the private branch exchange system 32 takes the same procedure as in the previous embodiment and performs a call.

  In this way, not only a failure but also an error response that cannot be received other than the line blockage (488/600) can be handled in the same way, the failure of the opposite SIP-GW device, and the line Even in the case of a blockage, if there is another SIP-GW device that can receive a call without waiting for the failure to be resolved or the line to be released, it is possible to make a call bypassing the SIP-GW device that can quickly receive the call.

  Note that the present embodiment is not limited to the procedure for confirming the status in the order of failure and line blockage in a plurality of SIP-GW devices on the receiving side, but it is also possible to check the status in the order of line blockage and failure. In the latter situation, it is preferable to confirm the blockage by checking the blockage of the line, the failure, that is, the presence / absence of a response signal and the timeout procedure. Needless to say, the confirmation of the detour may be a combination of the status check in the order of failure and line blocking and the status check in the order of failure and line blocking.

It is a flowchart describing the procedure of the SIP alternative incoming call method in the private branch exchange system of the embodiment to which the SIP incoming call method according to the present invention is applied. It is a block diagram which shows the schematic structure of the SIP-GW apparatus in the private branch exchange system to which the SIP alternative call incoming method of FIG. 1 is applied. FIG. 2 is a block diagram showing a schematic configuration of a private branch exchange system to which the SIP bypass incoming call method of FIG. 1 is applied. 2 is a sequential chart describing an operation procedure in a private branch exchange system to which the SIP bypass incoming call method of FIG. 1 is applied. It is a flowchart describing the procedure of the SIP alternative incoming call method in the private branch exchange system of another embodiment to which the SIP incoming call method according to the present invention is applied. 6 is a sequential chart describing an operation procedure in a private branch exchange system to which the SIP bypass incoming call method of FIG. 5 is applied.

Explanation of symbols

10 SIP-GW equipment
12 Line signal / packet converter
14 Packet switch
16 Control unit
28 CPU
30 SIP-GW bypass list

Claims (4)

A communication terminal device is connected to each of the gateway device on the caller side and the callee side via a dedicated line for communication based on IP (Internet Protocol) between a plurality of gateway devices on the callee side facing the gateway device on the caller side In the method of terminating communication based on SIP (Session Initiation Protocol) for communication with each of the communication terminal devices, the method includes:
A first step of transmitting a transmission signal from the originating gateway device to the terminating gateway device;
A second step of determining whether or not a response signal is supplied from the terminating gateway device to the originating gateway device at the originating gateway device;
A third step of determining whether the lack of supply of the response signal has continued for a first predetermined time;
A fourth step of extracting IP addresses in the plurality of gateway apparatuses on the receiving side stored in the gateway apparatus on the calling side when continuing for a first predetermined time;
A fifth step of retransmitting the outgoing signal and returning to the second step when the response signal is received within a first predetermined time; and
After the fourth step, a sixth step of determining the presence or absence of the extracted IP address;
A seventh step of setting the extracted IP address to an IP address of a transmission signal to be transmitted next according to the presence of the extracted IP address, transmitting the transmission signal, and returning to the second step. SIP incoming method characterized by that. The method of claim 1, wherein the method comprises an eighth step of determining whether or not the second predetermined time has continued in response to the lack of the extracted IP address in the sixth step. Including
A SIP terminating method characterized in that the process is terminated when the judgment of the eighth step is true, and proceeds to the fifth step when the judgment is false.   The SIP receiving method according to claim 2, wherein the second predetermined time is a time specified in RFC (Request For Comments) 3261.   The method according to claim 1, wherein, in the method, whether or not the line of the communication terminal device on the incoming side connected to the gateway device on the incoming side is blocked during the third and fourth steps. A SIP terminating method comprising a ninth step of determining whether or not.

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