JP2005160055A - Ip phone terminal system, router apparatus, call control method, and program - Google Patents

Abstract

<P>PROBLEM TO BE SOLVED: To utilize a VoIP within a private IP network in inexpensive configuration. <P>SOLUTION: A call setting request packet is transmitted to a VoIP router 100 from an adapter 2 to which a private IP address is given (a telephone set 31 is connected). The VoIP router 100 transmits to the adapter 2 a call connection completion packet containing information that connection is completed. Call control information contained in the call setting request packet is then extracted therefrom and a call setting request packet to be transmitted to an uplink side is generated. The VoIP router 100 transmits the generated call setting request packet to an SIP server 7 via a communication network 6. When the VoIP router 100 receives the call connection completion packet, a speech path between the telephone set 31 and an opposite side telephone set is connected, so that speaking becomes possible. <P>COPYRIGHT: (C)2005,JPO&NCIPI

Description

  The present invention relates to an IP telephone terminal system, a router device, a call control method, and a program that enable IP voice communication from within a private IP (Internet Protocol) network.

IP telephone services using VoIP (Voice over Internet Protocol) and SIP (Session Initiation Protocol) are beginning to spread. A communication device that provides an IP telephone service is disclosed in Patent Document 1, for example.
Japanese Patent Laying-Open No. 2003-169163 (page 3-5, FIG. 1)

  In order to effectively use a global IP address with a small number of assignments, routers, gateways, etc. are installed at the boundary between the internal network and the external network, and the internal communication network uses private IP addresses for communication. Often done. At this time, the router or gateway converts the private IP address and the global IP address using a technique called NAT (Network Address Translation).

  On the other hand, in the conventional IP telephone service, the IP telephone service cannot be used unless a global IP address is used. For this reason, it is necessary to receive the global IP address assignment for the desired number of lines, and the cost is increased accordingly.

In order to solve such a problem, a method of using an IP telephone service by sharing one global IP address is also considered. For example,
1) Install a call connection protocol firewall proxy 2) Install a SIP (Session Initiation Protocol) server 3) Install a device that supports UPnP (Universal Plug & Play).

The method 1) is a method in which a proxy is installed in a private network, the proxy detects a packet for an IP telephone transmitted from a device in the private network, and opens a predetermined port toward the external network.
The method 2) is a method of relaying a call using a SIP server in cooperation with a SIP server installed on an external network.
The method 3) uses a mechanism that automatically sets appropriate parameters and starts operation when a device that supports UPnP is installed on the network.

  Although these methods can reduce the number of global IP addresses used, they have the following drawbacks. That is, in the case of 1) and 2), the equipment to be installed is expensive. In the case of 3), the external SIP server, the adapter connected in the private network, etc. must also support UPnP. It is not always available.

  For this reason, there is a problem that it is not realistic to use the IP telephone service from within the private IP network.

  The present invention has been made in view of the above problems, and provides an IP telephone terminal system, a router device, a call control method, and a program that enable IP voice communication from within a private IP network with an inexpensive configuration. With the goal.

An IP telephone terminal system according to a first aspect of the present invention is:
An IP telephone terminal system comprising an adapter for connecting a telephone, and a router device whose uplink side is connected to an IP (Internet Protocol) network and whose downlink side is connected to the adapter via the network,
The router device is
Call accepting means for accepting a call setting request addressed to the self from the adapter;
Destination specifying means for specifying a destination connected to the uplink from the call setting request received by the call receiving means;
Call transmission means for creating and transmitting a second call setting request to the destination specified by the destination specifying means based on the call setting request;
Response reception means for receiving a response to the second call setting request created by the call transmission means;
When the response received by the response receiving means is information indicating that an uplink-side call has been set based on the second call setting request transmitted by itself, between the adapter and the destination Means to set up a call path for
It comprises.

  According to the present invention, voice communication is terminated between the adapter and the router device and between the router device and the external device, and IP voice communication can be performed from within the private IP network.

The destination specifying means is, for example,
Determine whether a call destination is set in the call setting request addressed to the self,
When it is determined that a call destination is set, the destination is the destination,
When it is determined that the call destination is not set, a predetermined destination is set.

The router device
First voice packet receiving means for receiving voice packets from the adapter;
First conversion means for converting the voice packet received by the first voice packet receiving means by a predetermined method;
First voice packet transmission means for transmitting the voice packet converted by the first conversion means toward the uplink;
Second voice packet receiving means for receiving voice packets from the uplink;
Second conversion means for converting the voice packet received by the second voice packet receiving means by a predetermined method;
Second voice packet transmitting means for transmitting the voice packet converted by the second converting means to the adapter;
It is desirable to comprise.

The call setup request preferably includes first method designation information for designating a method for converting a voice packet.
In this case, the router device
Comprising a first extraction means for extracting the first method designation information from the call setting request received by the call reception means;
The first conversion means performs conversion based on the first method designation information extracted by the first extraction means.

The response preferably includes second method designation information for designating a method for converting a voice packet.
In this case, the router device
Second extraction means for extracting the second method designation information from the call setting request received by the call reception means;
The second conversion means performs conversion based on the second method designation information extracted by the second extraction means.

  According to the present invention, it is possible to deal with various modulation schemes, and the difference in modulation scheme can be absorbed by the router device.

In the IP telephone terminal system, the uplink side can be connected to a public telephone line network.
In this case, the destination specifying means
When a call destination is set in the call setting request addressed to the self, it is determined whether the destination is for the public line network,
If the call destination set in the call setting request addressed to the self is for the public line network, the predetermined destination is the destination for the public line network,
When the call destination set in the call setting request addressed to the self is for an IP network, the predetermined destination is the destination for the IP network,
The router device includes signal transmitting means for generating and transmitting a signal for setting a call when the destination specified by the destination specifying means is for the public line network.

  According to the present invention, an analog telephone line and a VoIP line can be selected and used.

The router device according to the second aspect of the present invention provides:
A router device connected to an uplink and a downlink connectable to a global IP network,
Call accepting means for accepting a call setting request addressed to the self from the downlink;
Destination specifying means for specifying a destination connected to the uplink from the call setting request received by the call receiving means;
Call transmission means for creating and transmitting a second call setting request to the destination specified by the destination specifying means based on the call setting request;
Response reception means for receiving a response to the second call setting request created by the call transmission means;
When the response received by the response receiving means is information indicating that the call on the uplink side is set based on the second call setting request transmitted by itself, the link between the downlink and the destination Means to set up a call path for
It comprises.

The call control method according to the third aspect of the present invention is:
A router device connected to the uplink and the downlink that can be connected to the global IP network,
From the downlink, a call acceptance step for accepting a call setting request addressed to itself;
A destination identification step for identifying a destination connected to the uplink from the call setting request received in the call reception step;
A call transmission step of creating and transmitting a second call setting request to the destination specified in the destination specifying step based on the call setting request;
A response reception step of receiving a response to the second call setting request created in the call transmission step;
When the response received in the response reception step is information indicating that the call on the uplink side is set based on the second call setting request transmitted by itself, between the downlink and the destination Setting up a call path for
Is provided.

The program according to the fourth aspect of the present invention is:
To computers connected to the uplink and downlink that can be connected to the global IP network,
From the downlink, a call acceptance procedure for accepting a call setup request addressed to itself,
A destination identification procedure for identifying a destination connected to the uplink from the call setup request received in the call acceptance procedure;
A call transmission procedure for creating and transmitting a second call setting request to the destination specified in the destination specifying procedure based on the call setting request;
A response reception procedure for receiving a response to the second call setting request created in the call transmission procedure;
When the response received in the response reception procedure is information indicating that the call on the uplink side is set based on the second call setting request transmitted by itself, between the downlink and the destination To set up a call path for
Is executed.

  According to the present invention, it is possible to perform IP voice communication from within a private IP network with an inexpensive configuration.

Embodiments according to the present invention will be described below with reference to the drawings.
FIG. 1 is a block diagram showing a configuration of an IP (Internet Protocol) telephone terminal system 1 according to an embodiment of the present invention.

  As shown in FIG. 1, an IP telephone terminal system 1 according to the present embodiment includes a VoIP (Voice over IP) router 100 and an IP telephone adapter 2b or a VoIP adapter that are interconnected via a local IP network 5. The telephone terminal 31a connected to 2a and the telephone terminal 31b connected to the VoIP router 100 are configured, for example, in the business office I.

  Outside the office I, there is a global (wide area) IP network 6 such as the Internet and a public switched telephone network (PSTN) 9, and the VoIP router 100 is connected to them.

  A SIP (Session Initiation Protocol) server 7 is disposed on the global IP network 6, and a telephone terminal 31c is connected via a VoIP adapter 2c. The global IP network 6 is connected to the PSTN network 9 via the gateway 8. A telephone terminal 31 d is also connected to the PSTN network 9.

  The VoIP adapters 2a and 2c have the same configuration and function. The telephones 31a, 31b, 31c, and 31d have the same functions and configurations.

  The VoIP adapters 2a and 2c connect general telephones 31a and 31c to an IP network (local IP network 5 or global IP network 6).

  Specifically, the VoIP adapters 2a and 2c generate a call setting request packet (connection request packet) or the like for requesting call setting at the start of a call, and transmit the call setting request packet or the like onto the IP network 5 or 6. At this time, the VoIP adapters 2a and 2c include information specifying the PCM (Pulse Coded Modulation) modulation method used during a call and information specifying the RTP (Realtime Transfer Protocol) encoding format in the call setting request packet. A call setting request packet (connection request packet) from the VoIP adapter 2 a is supplied to the SIP server 7 via the local IP network 5, the VoIP router 100, and the global IP network 6. The call setting request packet from the VoIP adapter 2 c is supplied to the SIP server 7 via the global IP network 6. In addition, the VoIP adapter 2 a receives response information for the call setting request from the SIP server 7 via the VoIP router 100. The VoIP adapter 2 c receives a response to the call setting request from the SIP server 7 via the global IP network 6.

  If the VoIP adapter 2a knows the global IP address of the VoIP adapter 2c to which the telephone 31c of the call destination is connected, the VoIP adapter 2a may communicate directly without going through the SIP server 7. For example, when the telephone terminal 31a notifies a telephone number registered in advance as a telephone call destination, the VoIP adapter 2a reads out the global IP address of the telephone call destination and requests call setting to that address. Send a control packet.

  The VoIP adapters 2a and 2c convert analog audio signals transmitted from the telephones 31a and 31c into digital signals by a predetermined method (in this embodiment, RTP (Real-time Transfer Protocol)) during a call, The digital signal is encoded to generate a voice packet, which is transmitted over the IP network 5 or 6. The VoIP adapters 2a and 2c decode the voice packet supplied from the IP network 5 or 6 and perform D / A conversion to reproduce an analog voice signal and supply it to the telephones 31a and 31c.

  Further, when the telephone 31a is on-hooked (when the call is disconnected), the VoIP adapters 2a and 2c generate a call end notification packet (connection end packet) for requesting the end of the call, and the like on the IP network 5 or 6 Send to. End notification packets from the VoIP adapters 2 a and 2 b are supplied to the SIP server 7 through the local IP network 5, the VoIP router 100, and the global IP network 6. Further, the end notification packet from the VoIP adapter 2 c is supplied to the SIP server 7 via the global IP network 6. Further, the adapters 2 a and 2 b receive response information for notifying the end of the call from the SIP server 7 via the VoIP router 100. Further, the VoIP adapter 2 c receives a call end notification from the SIP server 7 via the global IP network 6.

  The telephone terminals 31a, 31b, 31c, and 31d are composed of general telephone terminals and have a telephone function unit for executing a voice call. For example, when a speaker picks up the telephone 31 to start a call and dials a telephone number, a signal indicating the telephone number is transmitted to the adapter 2. When the speaker has a conversation, an analog voice signal corresponding to the conversation is transmitted to the adapter 2 as a sound packet.

  The local IP network 5 is a communication network such as a LAN (Local Area Network) based on a TCP / IP (Transmission Control Protocol / Internet Protocol) communication protocol. A private address (local IP address) that can be identified only within the communication network 5 is assigned to the VoIP router 100, the adapter 2, and the like connected to the communication network 5.

  The global IP network 6 is a communication network such as the Internet based on the TCP / IP communication protocol, for example. A global address is given to the VoIP router 100 together with the above private address. More specifically, a private address is assigned to the “LAN port” side, and a global address is assigned to the “WAN port”.

  The SIP (Session Initiation Protocol) server 7 is constituted by an information processing device such as a server or a personal computer (PC) disposed in an Internet service provider, and performs processing such as bandwidth control, call control, billing, authentication, and the like. Do.

  The gateway 8 includes an information processing device such as a dedicated processing device, a server, or a personal computer, and connects the global IP network 6 and the PSTN network 9.

  The telephone 31a in the IP telephone terminal system 1 and the external telephone 31c or 31d are i) an IP telephone network, ii) an IP telephone network and a public switched telephone network (PSTN), iii) a PSTN network, A call can be made via any of the above.

  Specifically, the telephone 31a and the telephone 31c can communicate with each other via i) an IP telephone network (local IP network 5, VoIP router 100, global IP network 6). The telephone 31a and the telephone 31d are connected to the IP telephone network (local IP network 5, VoIP router 100, global IP network 6), gateway 8, and public switched telephone network (PSTN) 9 of ii). Or iii) PSTN network (local IP network 5, VoIP router 100, PSTN network 9).

  In the case of i), the SIP server 7, the VoIP adapter 2c, or the telephone 31c with the built-in VoIP adapter 2c is an uplink device, and in the case of ii), the gateway 8 or the SIP server 7 is up. In the case of iii), a device (for example, a telephone exchange) in the PSTN network 9 becomes an uplink device. The apparatus on the downlink side is the VoIP adapter 2a.

  In the case of i) and ii), the VoIP router 100, the VoIP adapter 2, the SIP server 7 and the like transmit and receive a packet called a SIP (Session Initiation Protocol) packet for setting a call. The SIP packet according to the present embodiment includes a “call ID” for identifying a call, a “telephone number” indicating the other party's telephone number, a “source IP address” indicating the caller, and a “transmission destination” indicating the destination. IP address "and the like.

  When a call is made via the IP telephone network of i), the SIP server 7, the adapter 2, or the telephone 31 incorporating the adapter 2 serves as an uplink side device, and the IP telephone network and public telephone line of ii) When making a call through the network, the gateway 8 or the SIP server 7 becomes an apparatus on the uplink side, and when making a call through the PSTN network 9 of iii), a device in the PSTN network 9 (for example, a telephone exchange) ) Becomes the equipment on the uplink side. The device on the downlink side is the adapter 2.

  The VoIP router 100 connects the local IP network 5 and the global IP network 6 and relays a voice call from the adapter 2a to another adapter 2c. At this time, the VoIP router 100 terminates this voice call (VoIP adapter 2a- (local IP network 5) -VoIP router 100- (global IP network 6) -VoIP adapter 2c), and two voice communications, that is, ( Voice communication of the VoIP adapter 2a-local IP network 5-VoIP router 100) and voice communication of (VoIP router 100-global IP network 6-VoIP adapter 2c).

  The VoIP router 100 includes a “LAN port” for connecting to an internal network, a “TEL port” for directly connecting a telephone, a “PSTN port” for connecting to a PSTN network, and a “LAN port” for connecting to an external network. WAN port ". As shown in the figure, “LAN port” is connected to the local IP network 5, “TEL port” is connected to the telephone 31b, “PSTN port” is connected to the PSTN network 9, and “WAN port” is connected to the global IP network 6. Has been.

  The configuration of the VoIP router 100 is shown in FIG. As illustrated, the VoIP router 100 includes a central processing unit 101, a crosspoint control unit 102, a PCM (Pulse Code Modulation) codec 103, and communication control units 105, 106, 107, and 108.

  The central processing unit 101 includes, for example, a CPU (Central Processing Unit) and a storage device (for example, a RAM (Random Access Memory) and a ROM (Read Only Memory)). The central processing unit 101 controls each unit of the VoIP router 100 by reading and executing an operation program stored in the RAM or ROM.

  The central processing unit 101 processes information (SIP packets) transmitted from the adapter 2 and the uplink side device, sets up and controls a call. Then, information (SIP packet) indicating a call setting result or the like is generated and transmitted to the adapter 2 and the device on the uplink side.

  The central processing unit 101 determines whether the call goes through the communication network 6 or the PSTN network 9 according to the telephone number of the other party included in the URI included in the SIP packet. In other words, if the telephone number starts with “050”, it is an IP telephone, so that it is via the communication network 6, and if the telephone number is other than that, it is via the PSTN network 9.

  The central processing unit 101 manages calls. The central processing unit 101 generates a “call management table” for managing calls in the predetermined storage device. FIG. 3 shows an example of information described in the “call management table”.

As shown in the figure, in the “call management table”, records having “call ID”, which is information for identifying a call, as a primary key are created, and each record has “source IP address”, “transmission” Information indicating “destination IP address”, “internal PCM format information”, “internal RTP format information”, “external PCM format information”, “external RTP format information”, “internal codec information”, “external codec information”, etc. To be recorded.
“Source IP address” and “Destination IP address” store IP addresses accessible by the VoIP router 100.
The “internal PCM format information” and the “internal RTP format information” store information on the conversion format used when the VoIP router 100 transmits and receives IP telephone packets for the downlink. As described above, the same information is used in this embodiment.
“External PCM format information” and “external RTP format information” store information on the conversion format used when the VoIP router 100 transmits and receives IP telephone packets for the uplink.
“Internal codec information”, information indicating which PCM codec 103 the path between the VoIP router 100 and the VoIP adapter 2a passes through is stored.
“External codec information”, information indicating which PCM codec 103 the path between the VoIP router 100 and the uplink side device passes through is stored.

  When the VoIP router 100 receives a call control request from the VoIP adapter 2a, the central processing unit 101 creates a record having “call ID” included in the call control request as a primary key in the “call management table”. Information related to “source IP address” and “destination IP address” is retrieved from information included in the call control request and stored. Then, the PCM codec 103 to be used is determined, and information regarding the determined PCM codec 103 is stored in “internal codec information”.

  When the VoIP router 100 receives information indicating a call setting result or the like from the uplink side device, the central processing unit 101 stores a record including the “call ID” included in the information in the “call management table” ”And information related to“ external PCM format information ”and“ external RTP format information ”is searched from the information included in the information and stored. Then, the PCM codec 103 to be used is determined, and information regarding the determined PCM codec 103 is stored in “external codec information”.

  Returning to FIG. 2, the crosspoint control unit 102 includes a switch, a predetermined storage device (for example, a RAM), and the like, and according to an instruction from the central processing unit 101, the communication control units 105 and 106 and the communication control unit 107, A call path with 108 is set and released.

  The crosspoint control unit 102 creates a “exchange status management table” in the storage device in order to manage the currently set call. FIG. 4 shows an example of information described in the “exchange status management table”.

  As shown in the figure, in the “exchange status management table”, a record having a “call ID” as a primary key, which is information for identifying a call, is created, and each record has “internal codec information” and “external code”, respectively. Information indicating “codec information” and the like is recorded.

  When the VoIP router 100 receives a call control request from the VoIP adapter 2a, the crosspoint control unit 102 creates a record with the “call ID” included in the call control request as a primary key in the “exchange status management table”. Hold until the call is terminated. “Internal codec information” and “external codec information” store information indicating the PCM codec 103 determined by a predetermined method.

  Returning to FIG. 2, the PCM codec 103 is composed of an analog / digital converter, and the like. The audio data is sampled, quantized and PCM-modulated into digital data (packets). PCM modulation schemes include simple PCM, differential PCM, and adaptive PCM. In this embodiment, the modulation scheme can be specified for each call. For example, audio data and information specifying a modulation method are received from the communication control unit 105, and the audio data is transferred to the crosspoint control unit 102 as PCM-encoded data (packets) by the specified modulation method. The PCM codec 103 also performs inverse conversion.

The communication control units 105, 106, 107, and 108 are configured by a communication device such as a predetermined network interface card (NIC), an input / output buffer, and the like, and communicate with other devices.
The communication control unit 105 is directly connected to the telephone 31b via the TEL port.
The communication control unit 106 is connected to the adapter 2 via the communication network 5 via the LAN port.
The communication control unit 107 is connected to the PSTN network 9 via a PSTN port.
The communication control unit 108 is connected to the SIP server 7, the gateway 8, and the external VoIP adapter 2c via the WAN port and the global IP network 6.

  The input / output buffers included in the communication control units 105, 106, 107, 108 are input / output buffers for audio data or audio packets processed by the connected PCM codec 103. In response to an instruction from the central processing unit 101, the communication control units 105, 106, 107, and 108 transmit audio data or audio packets to an input / output buffer directed to an appropriate PCM codec 103.

Hereinafter, the operation of the IP telephone terminal system 1 according to the present embodiment will be described.
In the following description, a telephone call is made from the telephone 31a connected to the local IP network 5 (via the VoIP adapter 2a) to the telephone 31c connected to the global IP network 6 outside the office I via the IP telephone network. Shall.

  After a call request is issued from the VoIP adapter 2a, the VoIP router 100, and the uplink side device (any one of the VoIP adapter 2c, the SIP server 7, and the gateway 8), the call is set The operation during will be described.

  Note that when the other party of the telephone 31 connected to the IP telephone network is known in advance (that is, when the sender's VoIP adapter 2a or VoIP router 100 knows the global IP address of the other party) The VoIP adapter 2c on the receiving side becomes an apparatus on the uplink side. Otherwise, the SIP server 7 or the gateway 8 is on the uplink side depending on where the partner telephone 31 is installed on the network. Become a device.

  After a call between the two telephones 31 is set up, voice packets are transmitted and received between these telephones 31. Then, when the call is finished and one of the telephones 31 is in the on-hook state, the call is finished in the same manner as the following operation.

  FIG. 5 is a chart of signals transmitted and received between devices when a call is established.

  As shown in the figure, the caller of the business entity I picks up the telephone 31a and dials the telephone number. Then, as shown in FIG. 5, the telephone number reaches the connected VoIP adapter 2a in the form of a voice signal (step S1).

  The VoIP adapter 2a issues an “INVITE” command of the SIP packet, and requests the VoIP router 100 to set a call (step S2). The “INVITE” command includes the telephone number and IP address of the telephone 31 that is the other party. The transmission source is the VoIP adapter 2 and the transmission destination is the VoIP router 100. This “INVITE” command is transmitted to the communication control unit 106 of the VoIP router 100.

The communication control unit 106 provides the received “INVITE” command to the central processing unit 101 (step S2).
The central processing unit 101 controls the communication control unit 106 to display a message indicating that processing is in progress (100 Trying), a message indicating that a call is being made (180 Ringing), and a message indicating that a call has been set with the other party (200 OK). It transmits to the VoIP adapter 2a in succession (steps S3 to S5).

  The VoIP adapter 2a that has received the message indicating that the call has been set with the other party transmits a delivery confirmation (ACK) command to the communication control unit 106 (step S6). Through the processing so far, the call between the adapter 2 and the VoIP router 100 is terminated.

  On the other hand, the central processing unit 101 determines the PCM codec 103 between the communication control unit 106 and the crosspoint control unit 102 used in the call. Based on the received “INVITE” command, a record for managing a new call is created in the “call management table”. First, information such as “call ID”, “source IP address”, “destination IP address”, “internal codec information”, and the like of this record is determined.

  The central processing unit 101 then instructs the crosspoint control unit 102 to set a call. Upon receiving the instruction, the crosspoint control unit 102 creates a record for managing the call in the “exchange status management table”.

  The central processing unit 101 rewrites the “source address” included in the “INVITE” command received by the communication control unit 106 with its own global IP address, and sets the “destination address” to the VoIP that connects the telephone 31c of the communication partner. The “INVITE” command in which the information specifying the PCM modulation method and the information specifying the RTP encoding format are deleted is transmitted to the uplink side device (step S7).

  When the device on the uplink side receives the “INVITE” command, it transmits a message indicating that processing is in progress to the communication control unit 108 of the VoIP router 100.

  More specifically, when the VoIP adapter 2c is directly called, the uplink side device is the VoIP adapter 2c. When the uplink side device receives the “INVITE” command, the VoIP adapter 2c is processing. Is transmitted to the communication control unit 108 of the VoIP router 100 (global IP address of the VoIP adapter 2c).

  On the other hand, if the IP address of the communication destination is not known, the “INVITE” command is transmitted to the SIP server 7, and the SIP server 7 specifies the communication destination VoIP adapter 2c and transfers the “INVITE” command. When the communication destination is on the PSTN network 9 via the gateway 8, the processing is via the gateway 8.

  Next, the apparatus on the uplink side instructs the adapter 2 to ring the bell of the other party's telephone 31 (step S8), and performs communication control of a message indicating that processing is in progress (100 Trying) and a message indicating that it is ringing (180 Ringing). It transmits to the part 108 (step S9, S10). When the other party's telephone 31 enters an off-hook state (step S11), a message (200OK) indicating that a call has been set with the other party is transmitted to the communication control unit 108 (step S12).

  When the communication control unit 108 receives a message (200 OK) indicating that a call has been set with the other party, the VoIP adapter 2c that connects the other party's telephone 31 to the ACK command (may be incorporated in the gateway 8). Send to. With the processing so far, the call between the VoIP router 100 and the VoIP adapter 2 that connects the telephone 31 on the other side is terminated.

  The communication control unit 108 transfers to the central processing unit 101 a message indicating that a call has been set with the other party. When the central processing unit 101 receives this message, it determines the PCM codec 103 between the communication control unit 108 and the crosspoint control unit 102 used in the call. Then, the information stored in the “call management table” is supplemented. That is, the “external PCM format information” and the “external RTP format information” that have not been determined are complemented.

  Further, the central processing unit 101 transmits an instruction to connect the two PCM codecs 103 used in the call to the crosspoint control unit 102. The cross point control unit 102 connects between designated PCM codecs 103. This makes it possible to start a call between the two telephones 31.

  It should be noted that after the call between the VoIP adapter 2a and the VoIP router 100 is terminated (step S13), the call between the VoIP router 100 and the uplink side device is terminated and the call is actually possible. In the meantime, the VoIP router 100 transmits a voice packet indicating the pending status to the adapter 2 (step S14).

  Then, the VoIP adapter 2a transmits an acknowledgment signal (ACK) to the VoIP router 100 (step S15), and further, the VoIP router 100 transmits an acknowledgment signal (ACK) to the uplink side device (step S16).

  Even when a call request is transmitted from outside the communication network 5, the IP telephone terminal system 1 operates in the same manner as in the above-described operation example.

  Thereafter, direct voice packet transmission / reception is performed between the transmission-side VoIP adapter 2a and the counterpart-side VoIP adapter 2 via the VoIP router 100. At this time, the voice packet is converted by the VoIP router 100 in accordance with the registration in the call management table and sent to the other party.

  For example, as shown in FIG. 6A, the VoIP adapter 2a includes, as header information, transmission destination IP address = global IP address of the call destination VoIP adapter 2c, transmission source IP address = own local IP address, and payload. The voice packet including the voice data based on the PMC format and the RTP format based on its own setting is sent to the VoIP router 100.

For example, a voice packet from the VoIP adapter 2 a is received by the communication control unit 106, and header information is provided to the central processing unit 101.
The central processing unit 101 refers to the call management table, identifies the call ID from the IP address of the transmission source, and determines the payload of the identified call ID based on the information registered in the call management table and the exchange status management table. A PMC codec 103 for decoding and a PCM codec 103 for re-encoding the decoded payload in accordance with the request format of the destination, and further, cross-point control based on the destination address A data transfer route is designated in the section 102.

  As a result, the voice packet from the VoIP adapter 2 a received by the communication control unit 106 is decoded by the PCM codec 103 corresponding to the PCM format adopted by the VoIP adapter 2 a, and is transmitted via the crosspoint control unit 102. The destination setting VoIP adapter 2 c is encoded again by the PCM codec 103 corresponding to the PCM format, and is supplied to the communication control unit 108. The communication control unit 108 generates header information including the global IP address of the transmission destination provided from the central processing unit 101 and the global IP address of its own (WAN port) in the recoded payload. The voice packet shown in FIG. 6B is generated by adding to the payload. The communication control unit 108 transmits the generated voice packet to the global IP network 6.

  Conversely, the voice packet from the VoIP adapter 2c has a global IP address assigned to the communication control unit 108 as the destination IP address as shown in FIG. 6C, and is received by the communication control unit 108. Is done. The header information of the voice packet is also supplied to the central processing unit 101. The central processing unit 101 identifies the call ID from the transmission source address, etc., identifies the PCM codec used for decoding and recoding from the information set in the identified call ID, and sends it to the crosspoint control unit 102 A data transmission route is instructed, and a local IP address of a transmission destination is designated to the communication control unit 106. As a result, the voice packet from the VoIP adapter 2 c received by the communication control unit 108 is decoded by the PCM codec 103 corresponding to the PCM format adopted by the VoIP adapter 2 c, and is transmitted via the crosspoint control unit 102. The destination setting VoIP adapter 2 a is encoded again by the PCM codec 103 corresponding to the PCM format, and is supplied to the communication control unit 106. The communication control unit 108 generates header information including the global IP address of the VoIP adapter 2c of the transmission source and the local IP address of the transmission destination provided from the central processing unit 101 in the recoded payload. Is added to the payload to generate the voice packet shown in FIG. The communication control unit 106 transmits the generated voice packet to the local IP network 5.

  When one of the telephones 31 in a call is on-hooked, the call is disconnected by the same processing as described above, and information regarding the call is deleted from the call management table and the exchange status management table.

  More specifically, when one of the telephones 31 in a call is on-hooked, the VoIP adapter outputs a BYE request indicating the end of the call. This is supplied to the SIP server 7 and transferred to the VoIP router 100 when the on-hook side is the VoIP adapter 2c. The VoIP router 100 terminates this, sends an acknowledgment signal (200OK) to the VoIP adapter 2c via the SIP server 7, and generates a BYE packet addressed to the VoIP adapter 2a (the destination address is the local IP address). ) Send to the VoIP adapter 2a. When the on-hook side is the VoIP adapter 2a, the BYE request is supplied to the VoIP router 100. The VoIP router 100 terminates this, generates a BYE packet of the VoIP adapter 2c via the SIP server 7 (the source address is the global IP address of the WAN port), and transmits it to the VoIP adapter 2c via SIP7. . Also, an acknowledgment signal (200 OK) is sent to the VoIP adapter 2a.

  Furthermore, the central processing unit 101 deletes information regarding the specified call ID from the call management table and the exchange status management table.

  When communication with the telephone on the PSTN network 9 is designated from the telephone 31a or 31b, the VoIP router 100 starts from the telephone number (for example, prefix 0 * is 03, 06, etc.) from the PSTN network 9 The communication control unit 107 is controlled to perform processing with the other party telephone, and voice communication between the communication control unit 105 or 106 and the communication control unit 107 is enabled. Also at this time, the communication is terminated between the VoIP adapter 2a and the VoIP router 100, between the VoIP router 100, and between the PSTN lines 9.

  As described above, according to this embodiment, without performing routing using IP, between the VoIP adapter 2a and the VoIP router 100, and between the VoIP router 100 and a partner device (a partner VoIP adapter 2c, SIP) Since communication is terminated between the server 7 and the gateway 8), NAT can be exceeded, and IP voice communication between the local (private) network and the outside becomes possible. Further, since communication is terminated between the VoIP adapter 2a and the VoIP router 100, between the VoIP router 100, and between the PSTN networks 9, the PSTN network connected to the VoIP router 100 in which the NAT function is set. 9 can be connected. .

The processing at the time of establishment of a call is not limited to the example shown in FIG. 5, and various modifications and applications are possible. Modified examples are shown in FIGS.
In the example of FIG. 7, when the caller dials the telephone number, the VoIP adapter 2a issues an “INVITE” command of the SIP packet and requests the VoIP router 100 to set the call (step S101).

  The VoIP router 100 transmits a message indicating that processing is in progress (100 Trying) and a message indicating that it is being called (180 Ringing) to the VoIP adapter 2a (steps S102 and S103).

  On the other hand, the central processing unit 101 creates a record for managing a new call in the “call management table”, registers necessary items, and further manages the call in the “exchange status management table”. Create a record. Then, the “source address” included in the “INVITE” command received by the communication control unit 106 is rewritten to its own global IP address, and the “destination address” is set to the IP of the VoIP adapter 2c that connects the telephone 31c of the other party. An “INVITE” command in which the information specifying the PCM modulation method and the information specifying the RTP encoding format are deleted is transmitted to the uplink side device (step S104).

  When the uplink side apparatus receives the “INVITE” command, the uplink side apparatus instructs the adapter 2 to ring the bell of the telephone 31 of the other party, a message indicating that the process is in progress (100 Trying), and a ringing state. A message (180 Ringing) is transmitted to the communication control unit 108 (steps S105 and S106). When the other party's telephone 31 enters an off-hook state, a message (200 OK) indicating that a call has been set with the other party is transmitted to the VoIP router 100 (step S107).

  When the VoIP router 100 receives the message (200 OK) indicating that the call is set with the other party, the VoIP router 100 transmits an ACK command to the other party apparatus (step S108). With the processing so far, the call between the VoIP router 100 and the adapter 2 connecting the telephone 31 on the other side is terminated.

In addition, the VoIP router 100 transmits a message (200 OK) indicating that the call is set with the other party to the VoIP adapter 2a (step S109).
The VoIP adapter 2a transmits an approval signal (ACK) to the VoIP router 100 (step S110).
With the process so far, the call between the VoIP router 2a and the VoIP router 100 is terminated.

  Thereafter, direct voice packet transmission / reception according to RTP is performed via the VoIP router 100 between the VoIP adapter 2a on the transmission side and the VoIP adapter 2 on the other side.

  In the example of FIG. 8, when the caller dials the telephone number, the VoIP adapter 2a issues an “INVITE” command of the SIP packet and requests the VoIP router 100 to set the call (step S201).

  On the other hand, the central processing unit 101 creates a record for managing a new call in the “call management table”, registers necessary items, and further manages the call in the “exchange status management table”. Create a record. Then, the “source address” included in the “INVITE” command received by the communication control unit 106 is rewritten with its own global IP address, and the “destination address” is the IP address of the adapter 2c to which the telephone 31c of the other party is connected. In addition, an “INVITE” command in which the information specifying the PCM modulation method and the information specifying the RTP encoding format are deleted is transmitted to the uplink side device (step S202).

When the uplink side apparatus receives the “INVITE” command, the uplink side apparatus issues an instruction to ring the bell of the other party's telephone 31 to the VoIP adapter 2a, and displays a message (100 Trying) indicating that the process is in progress. (Step S203).
The VoIP router 100 transmits a message (100 Trying) indicating that processing is in progress to the VoIP adapter 2a (step S204).

The apparatus on the uplink side instructs the adapter 2 to ring the bell of the other party's telephone 31 and transmits a message (180 Ringing) indicating that it is ringing to the VoIP router 100 (step S205).
The VoIP router 100 transmits a message (180 Ringing) indicating that the call is in progress to the VoIP adapter 2c (step S206).
When the partner telephone 31 enters the off-hook state, the partner device transmits a message (200 OK) indicating that the call with the partner is set to the VoIP router 100 (step S207). The VoIP router 100 transmits to the VoIP adapter 2a a message (200OK) indicating that a call has been set with the other party (step S208).

  When the VoIP adapter 2a receives the message (200OK) indicating that the call is set with the partner, the VoIP adapter 2a transmits an ACK command to the partner device (step S209). With the processing so far, the call between the VoIP adapter 2a and the VoIP router 100 is terminated.

  In addition, the VoIP router 100 transmits an approval signal (ACK) to the counterpart device (step S210). With the processing so far, the call between the VoIP router 100 and the partner apparatus is terminated.

  Thereafter, direct voice packet transmission / reception according to RTP is performed via the VoIP router 100 between the VoIP adapter 2a on the transmission side and the VoIP adapter 2c on the other side.

  Next, the internal processing of the VoIP router 2a when establishing a call between the local IP network 5 and the global IP network 6 will be described in more detail with reference to FIG.

  First, upon receiving the “INVITE” command, the communication control unit 106 generates a new record in the call management table shown in FIG. 3, registers a new call ID, a source IP address (local IP address), and a destination IP address (if known), internal PCM format information (PCM format information of VoIP adapter 2a), internal RTP format information, (RTP format information of VoIP adapter 2a), internal codec information (communication with VoIP adapter 2a) (Information for specifying a codec to be used) is registered (step S301).

  Subsequently, telephone number information 0 * 0123456 to be transferred is acquired from the call connection information in the received “INVITE” command and transferred to the call control relay process (step S302).

In the call control relay processing, the relay processing to the IP telephone is recognized from the relay processing prefix “0 *”, and the number “0123456” is transferred to the WAN side call control storing the relay route.
For example, if the relay processing prefix is “03”, the connection to the PSTN network 9 is established. On the other hand, for example, if “050”, the relay processing to the IP phone is recognized, and the number “0123456” is transferred to the WAN side call control storing the relay route (step S303).

In WAN side call control, the number is transmitted. That is, a SIP packet is generated and an INVITE command is transmitted from the communication control unit 108.
When the other party answers (when 200OK is received), the PCM format information and RTP format information of the other party are registered in the call management table (step S304). If it is not completed, the IP address (global IP address) of the destination VoIP adapter 2c is registered (step S304). This completes the call management table for the current call.

  In the call control relay process, after the connection is completed, various information such as internal codec information and external codec information is acquired from the call management table and used for transmission / reception of voice packets (step S305).

  Subsequently, the PCM codec 103 to be used is specified on the LAN side and the WAN side based on the call management information, and registered in the exchange status management table of FIG. 4 (steps S306 and S307). Instead of directly specifying the PCM codec 103, a PCM buffer may be specified.

  Subsequently, cross point control is performed. That is, the cross point switch is controlled so as to form a path between the PCM codecs 103 registered in the exchange status management table (step S308). As a result, communication is performed between the PCM codec 103 adapted to the configuration and setting of the adapter 2 on the LAN side and the WAN side, and consequently, voice communication between the telephones 31 via the IP network becomes possible. .

  In the case of voice packet communication, the RTP IP address, port number, PCM buffer ID, etc. are controlled based on the information obtained by call control, the process of creating and transmitting the RTP packet, the received RTP packet As a process of demodulating, a LAN side RTP packet generation / demodulation process and a WAN side RTP packet generation / demodulation process are performed (steps S309 to S312), thereby enabling a voice call.

  In addition, this invention is not limited to the said embodiment, A various deformation | transformation and application are possible.

  For example, the VoIP router 100 may include an adapter function included in the adapter 2. Further, a method for setting a call is not limited to SIP, and a method for modulating voice is not limited to PCM.

  In the above-described embodiment, the description has been made assuming that the voice modulation method and the voice packet transmission method are unified in the communication network 5. However, each adapter 2 can adopt a different modulation scheme and transmission scheme.

  Note that the information processing apparatus for realizing the VoIP router 100 according to the embodiment of the present invention can be realized using a normal computer system, not a dedicated system. For example, the VoIP router 100 that executes the above-described processing can be configured by installing the program from a medium (such as a CD-ROM) that stores the program for executing the above-described processing in a general-purpose computer.

  When the OS shares a part of the above processing, or when the OS constitutes a part of the constituent elements of the present invention, a program excluding the part may be stored in the recording medium. Good. Also in this case, it is assumed that the recording medium stores a program for executing each function or each step executed by the computer.

  It is also possible to superimpose a program on a carrier wave and distribute it via a communication network. For example, the program may be posted on a bulletin board (Bulletin Board System: BBS) of a communication network and distributed through the network. Then, the above-described processing may be executed by starting this program and executing it under the control of the OS in the same manner as other application programs.

It is a block diagram which shows the structure of the IP telephone terminal system concerning embodiment of this invention. It is a block diagram which shows the structure of the VoIP router of FIG. It is a figure which shows the example of the information described in the "call management table" which the central processing part of FIG. 2 produces | generates. It is a figure which shows the example of the information described in the "exchange status management table" which the crosspoint control part of FIG. 2 produces | generates. It is a figure for demonstrating the example 1 of operation | movement of the IP telephone terminal system concerning this Embodiment. It is a figure for demonstrating the structure of the packet transmitted / received by the IP telephone terminal system concerning this Embodiment. It is a figure for demonstrating the example 2 of operation | movement of the IP telephone terminal system concerning this Embodiment. It is a figure for demonstrating the example 3 of operation | movement of the IP telephone terminal system concerning this Embodiment. It is a figure for demonstrating the internal operation | movement of a VoIP router.

Explanation of symbols

DESCRIPTION OF SYMBOLS 1 IP telephone terminal system 2 Adapter 31, 32 Telephone 5, 6 Communication network 7 SIP server 8 Gateway 100 VoIP router 101 Central processing part 102 Crosspoint control part 103 PCM codec 105,106,107,108 Communication control part

Claims (14)

An IP telephone terminal system comprising an adapter for connecting a telephone, and a router device whose uplink side is connected to an IP (Internet Protocol) network and whose downlink side is connected to the adapter via the network,
The router device is
Call accepting means for accepting a call setting request addressed to the self from the adapter;
Destination specifying means for specifying a destination connected to the uplink from the call setting request received by the call receiving means;
Call transmission means for creating and transmitting a second call setting request to the destination specified by the destination specifying means based on the call setting request;
Response reception means for receiving a response to the second call setting request created by the call transmission means;
When the response received by the response receiving means is information indicating that an uplink-side call has been set based on the second call setting request transmitted by itself, between the adapter and the destination Means to set up a call path for
An IP telephone terminal system comprising: The destination specifying means includes
Determine whether a call destination is set in the call setting request addressed to the self,
When it is determined that a call destination is set, the destination is the destination,
The IP telephone terminal system according to claim 1, wherein when it is determined that a call destination is not set, the call destination is a predetermined destination. The router device
First voice packet receiving means for receiving voice packets from the adapter;
First conversion means for converting the voice packet received by the first voice packet receiving means by a predetermined method;
First voice packet transmission means for transmitting the voice packet converted by the first conversion means toward the uplink;
Second voice packet receiving means for receiving voice packets from the uplink;
Second conversion means for converting the voice packet received by the second voice packet receiving means by a predetermined method;
Second voice packet transmitting means for transmitting the voice packet converted by the second converting means to the adapter;
The IP telephone terminal system according to claim 1, further comprising: The call setting request includes first method designation information for designating a method for converting a voice packet;
The router device
Comprising a first extraction means for extracting the first method designation information from the call setting request received by the call reception means;
The first converting means is for converting based on the first method designation information extracted by the first extracting means;
The IP telephone terminal system according to claim 3. The response includes second method designation information for designating a method for converting a voice packet,
The router device
Second extraction means for extracting the second method designation information from the call setting request received by the call reception means;
The second converting means is for converting based on the second method designation information extracted by the second extracting means;
The IP telephone terminal system according to claim 4. The IP telephone terminal system is capable of connecting the uplink side to a public telephone network,
The destination specifying means includes
When a call destination is set in the call setting request addressed to the self, it is determined whether the destination is for the public line network,
If the call destination set in the call setting request addressed to the self is for the public line network, the predetermined destination is the destination for the public line network,
When the call destination set in the call setting request addressed to the self is for an IP network, the predetermined destination is the destination for the IP network,
The router device includes signal transmitting means for generating and transmitting a signal for setting a call when the destination specified by the destination specifying means is for the public line network;
The IP telephone terminal system according to any one of claims 1 to 5, wherein: A router device connected to an uplink and a downlink connectable to a global IP network,
Call accepting means for accepting a call setting request addressed to the self from the downlink;
Destination specifying means for specifying a destination connected to the uplink from the call setting request received by the call receiving means;
Call transmission means for creating and transmitting a second call setting request to the destination specified by the destination specifying means based on the call setting request;
Response reception means for receiving a response to the second call setting request created by the call transmission means;
When the response received by the response receiving means is information indicating that the call on the uplink side is set based on the second call setup request transmitted by the response receiving means, the downlink and the destination A means to set up a call path between,
A router device comprising: The destination specifying means includes
Determine whether a call destination is set in the call setting request addressed to the self,
When it is determined that a call destination is set, the destination is the destination,
The router device according to claim 7, wherein when it is determined that a call destination is not set, the call destination is a predetermined destination. The router device
First voice packet receiving means for receiving voice packets from the downlink;
First conversion means for converting the voice packet received by the first voice packet receiving means by a predetermined method;
First voice packet transmission means for transmitting the voice packet converted by the first conversion means toward the uplink;
Second voice packet receiving means for receiving voice packets from the uplink;
Second conversion means for converting the voice packet received by the second voice packet receiving means by a predetermined method;
Second voice packet transmission means for transmitting the voice packet converted by the second conversion means toward the downlink;
The router device according to claim 7 or 8, further comprising: The call setting request includes first method designation information for designating a method for converting a voice packet;
The router device
Comprising a first extraction means for extracting the first method designation information from the call setting request received by the call reception means;
The first converting means is for converting based on the first method designation information extracted by the first extracting means;
The router device according to claim 9. The response includes second method designation information for designating a method for converting a voice packet,
The router device
Second extraction means for extracting the second method designation information from the call setting request received by the call reception means;
The second converting means is for converting based on the second method designation information extracted by the second extracting means;
The router device according to claim 10. The router device is capable of connecting the uplink side to a public telephone network,
The destination specifying means includes
When a call destination is set in the call setting request addressed to the self, it is determined whether the destination is for the public line network,
If the call destination set in the call setting request addressed to the self is for the public line network, the predetermined destination is the destination for the public line network,
If the call destination set in the call setting request addressed to the self is for an IP network, the predetermined destination is the destination for the IP network;
The router device according to any one of claims 7 to 11, wherein: A router device connected to the uplink and the downlink that can be connected to the global IP network,
From the downlink, a call acceptance step for accepting a call setting request addressed to itself;
A destination identification step for identifying a destination connected to the uplink from the call setting request received in the call reception step;
A call transmission step of creating and transmitting a second call setting request to the destination specified in the destination specifying step based on the call setting request;
A response reception step of receiving a response to the second call setting request created in the call transmission step;
When the response received in the response reception step is information indicating that the call on the uplink side has been set based on the second call setting request transmitted by itself, the link between the downlink and the destination Setting up a call path between,
A call control method comprising: To computers connected to the uplink and downlink that can be connected to the global IP network,
From the downlink, a call acceptance procedure for accepting a call setup request addressed to itself,
A destination identification procedure for identifying a destination connected to the uplink from the call setup request received in the call acceptance procedure;
A call transmission procedure for creating and transmitting a second call setting request to the destination specified in the destination specifying procedure based on the call setting request;
A response reception procedure for receiving a response to the second call setting request created in the call transmission procedure;
When the response received in the response reception procedure is information indicating that the call on the uplink side has been set based on the second call setup request transmitted by itself, the downlink and the destination To set up a call path between
A program characterized by having executed.

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