JP4465525B2 - IP telephone exchange method and apparatus - Google Patents

Description

  The present invention provides an IP telephone that connects an IP telephone terminal accommodated by the apparatus and an IP telephone terminal included in an external public IP telephone network via an IP private branch exchange (hereinafter referred to as IP-PBX). The present invention relates to an exchange method and apparatus.

  As shown in FIG. 1, as one form of connecting an IP telephone accommodated in an IP-PBX via a dedicated IP network and an IP telephone of an external public IP telephone network, it responds to a call connection request between telephones. The IP-PBX main body performs the call connection procedure. After the call is set up, voice signal packets are exchanged directly via one link between telephones. Examples of the former call setting include SIP (Session Initiation Protocol), H.264, and the like. IP telephone protocols such as H.323 and Media Gateway Control Protocol (MGCP) are used, and the latter voice signal packets are packetized by the same voice compression standard using RTP (Real-time Transfer Protocol). A packet is transmitted.

  Taking an example of the development of an IP telephone network in a company that has many business offices such as branches or factories equipped with a large number of IP telephones, the IP- A call between IP telephones in a company is realized using VPN, and a call with the outside is realized by connecting to a public IP telephone network. Therefore, when connecting from an IP telephone terminal in a company to an IP telephone in a public IP telephone network, it is required that the voice signal communication between both terminals after the call connection be matched to the voice compression standard of the public IP telephone network. In particular, the public IP telephone network is G.264. When only 711 (hereinafter referred to as an uncompressed standard) is supported, packet transmission in the IP-VPN must be performed over a wide band corresponding to the uncompressed standard.

  However, the IP-VPN usage fee system is generally set according to the contracted bandwidth required for each site that is the connection point of the IP-VPN. Therefore, when there are a large number of bases and the IP telephone terminals at each base need a wide band, it is necessary to make an IP-VPN usage contract for each of the bases so that broadband communication is possible. This is a factor that increases communication costs.

  SUMMARY OF THE INVENTION An object of the present invention is to provide an IP telephone exchange method and apparatus capable of reducing communication costs in a form in which an IP telephone accommodated in an IP-PBX is connected to an IP telephone of an external public IP telephone network. is there.

An IP telephone exchange method according to the present invention is an IP telephone exchange method for making a call connection between a plurality of IP telephone terminals using different voice compression methods, and an address holding step for holding at least two free relay addresses in a free relay address table And a preparation step for associating the outgoing address of the originating IP telephone terminal with the incoming address of the incoming IP telephone terminal in response to the call connection request, and preparing the call connection between the IP telephone terminals; A route for selecting one of the vacant relay addresses as a relay address for each call connection request and setting a bidirectional voice signal packet route from the source address to the destination address via the relay address A voice signal directed to the relay address from each of the IP telephone terminals; Is converted in a compatible format to the audio signal compression method, characterized in that it comprises a relay sending toward the IP telephone terminal to the opposite.

An IP telephone exchange apparatus according to the present invention is an IP telephone exchange apparatus for making a call connection between a plurality of IP telephone terminals using different voice compression methods, and holding an at least two empty relay addresses in an empty relay address table. And a preparation means for associating the outgoing address of the originating IP telephone terminal with the incoming address of the incoming IP telephone terminal in response to the call connection request, and preparing the call connection between the IP telephone terminals; A route that selects one of the vacant relay addresses as a relay address for each call connection request, and sets a bidirectional voice signal packet route from the source address to the destination address via the relay address A voice signal packet destined for the relay address from each of the setting means and each of the IP telephone terminals; Is converted in a compatible format to the audio signal compression method, characterized in that it comprises a relay means for transmitting towards the IP telephone terminal to the opposite.

  According to the telephone exchange method and apparatus of the present invention, a configuration for converting a compression method of a voice signal while relaying the voice signal to the IP-PBX is provided. As a result, although the voice signal path is connected between each base and the IP-PBX by a plurality of narrow-band communication paths, only the IP-PBX and the public telephone network are connected by a broadband communication path. Thereby, it is possible to reduce the communication cost even in a form in which an IP telephone accommodated in the IP-PBX is connected to an IP telephone of an external public IP telephone network.

  Embodiments of the present invention will be described in detail with reference to the accompanying drawings.

FIG. 2 shows an embodiment of the present invention and shows the configuration of an IP telephone exchange apparatus.
Here, the private branch exchange (PBX) IP-PBX 20 can be connected to an IP-VPN 10 which is a dedicated IP data network dedicated to a specific organization such as a company, and an IP telephone terminal of a general subscriber. It is connected to the public IP telephone network 30. The IP-VPN 10 is connected to the IP telephone terminal 11 as an example of a plurality of IP telephone terminals accommodated in the IP-PBX 20. An IP telephone terminal 31 is connected to the public IP telephone network 30 as one of a plurality of subscribed IP telephone terminals. The IP-PBX 20 includes a call server unit 21, an ordinary telephone accommodating interface unit (IF) 22, an IP telephone terminal accommodating interface unit (IF) 23, a public IP telephone network gateway unit (GW) 24, and a free relay address table. 26 and a codec conversion unit 25.

  The call server unit 21 is connected to each unit included in the IP-PBX 20 to control the entire PBX exchange function. The call server unit 21 performs call control of extension calls between accommodated IP telephone terminals and receives incoming calls from the public IP telephone network 30 or public It has a function of performing call control of external line calls including outgoing calls to the IP telephone network 30. The call server unit 21 also searches for a free relay address from the free relay address table 26 in response to a call connection request and sends it to the codec conversion unit 25, thereby causing the codec conversion unit 25 to perform voice packet relay processing. It has the function to make

  The ordinary telephone accommodation interface unit 22 has a function of connecting the conventional ordinary telephone 19 and has a voice IP conversion function for performing a mutual conversion between a voice signal in the conventional system and a voice signal formed into an IP packet. As a result, the ordinary telephone 19 can be handled in the same manner as an ordinary IP telephone terminal. The IP telephone terminal accommodating interface unit 23 is connected to the IP-VPN 10 and has a function of transmitting / receiving a call control signal packet and a voice signal packet to / from a plurality of IP telephone terminals using the IP protocol. The public IP telephone network gateway unit 24 is connected to the public IP telephone network 30 and has a gateway function for transferring IP packets between different networks.

  The codec conversion unit 25 has a function of mutually converting audio signal packets by different audio compression methods. Examples of different audio compression methods include ITU recommendation standard G.264. 711 as well as ITU recommendation standard G.711. 723.1 or G.I. An example is a method of performing compression such as 729a. Further, the codec conversion unit 25 receives the correspondence relationship between the relay address extracted from the free relay address table 26 based on the call connection request received by the call server unit 21, and the outgoing address and the incoming address. 27 and a relay for transmitting voice signal packets received via the IP telephone terminal accommodating interface section 23 and the public IP telephone network gateway section 24 to an appropriate counterpart according to the voice signal path setting table 27. It has a function.

  FIG. 3 illustrates the configuration of the idle relay address table and the voice signal route setting table shown in FIG. The idle relay address table 26 is a table that stores unused addresses among relay addresses that serve as relay points for relaying voice signal packets. In the example of this figure, the free relay address table 26 is set as "aaa.bbb.ccc.001 to aaa.bbb.ccc.011" and holds a plurality of free relay addresses. As for management of idle relay addresses, for example, one relay address is deleted from the table by being used in a voice signal routing table 27 described later for each individual call connection, and again with the release of the call. This is returned to the free relay address table.

  The voice signal path setting table 27 sets a path that gives to which IP address the voice signal packet should be transmitted according to the destination address included in the voice signal packet received by the IP-PBX 20. It is a table. The audio signal path setting table 27 is managed by being divided into a plurality of relay addresses 271a and 271b provided for each call connection. For example, the relay address 271a is further associated with the setting table 272a, and similarly the relay address 271b is further associated with the corresponding setting table (not shown). In the setting table 272a, the IP address on the IP-PBX side and the IP address on the public IP telephone network side to which voice signal packets are communicated using the relay address as a relay point are set.

  The setting table 272a in the example of the figure shows the IP address of the IP telephone terminal 11 on the IP-PBX side and the public IP telephone network side to which voice signal packets are communicated with the relay address "aaa.bbb.ccc.001" as a relay point. The IP address of the IP telephone terminal 31 is set. Thus, the voice signal packet transmitted from the IP telephone terminal 11 to the IP-PBX 20 is transmitted to the IP telephone terminal 31 once via the relay address “aaa.bbb.ccc.001” of the codec conversion unit 25. . On the other hand, the voice signal packet transmitted from the IP telephone terminal 31 to the IP-PBX 20 is transmitted to the IP telephone terminal 11 once via the relay address “aaa.bbb.ccc.001” of the codec conversion unit 25.

  The setting table 272a also includes a voice signal compression method between the relay address and the IP telephone terminal on the IP-PBX side, and a voice signal compression method between the relay address and the IP telephone terminal on the public telephone network side. Contains specification information that specifies. Based on the designation information, the codec conversion unit 25 appropriately converts the audio signal packet and executes the relay function.

  In the example of this figure, the compression method of the audio signal between the relay address “aaa.bbb.ccc.001” and the IP telephone terminal 11 on the IP-PBX side is designated as “G.729a (compression)”. The compression method of the audio signal between the relay address “aaa.bbb.ccc.001” and the IP telephone terminal 31 on the public telephone network side is designated as “G.711 (uncompressed)”. Therefore, the audio signal packet transmitted from the IP telephone terminal 11 to the IP-PBX 20 is transmitted to the G.C. 729a (compressed) to G.I. After being converted to 711 (uncompressed), it is transmitted to the IP telephone terminal 31. On the other hand, the voice signal packet transmitted from the IP telephone terminal 31 to the IP-PBX 20 is transmitted to the G.C. 711 (uncompressed) to G.I. After being converted to 729a (compressed), it is transmitted to the IP telephone terminal 11.

  In the IP address specification in the above description, along with the IP address specification, additional address information such as a port number may be used in combination as a subaddress required when the same IP address is shared.

  FIG. 4 shows a processing procedure of the IP telephone exchange method according to the present invention. The processing procedure will be described with reference to the components shown in FIG.

  First, the IP telephone terminal 11 makes a call with the IP telephone terminal 31 as a destination (step S01). Next, in response to a call from the IP telephone terminal 11, the call server unit 21 of the IP-PBX 20 searches for a free relay address from the free relay address table 26 (step S02). Next, the call server unit 21 transmits the relay address to the public IP telephone network 30 using the transmission side voice signal termination point (step S03). In response to the call, the public IP telephone network 30 receives the incoming call to the IP telephone terminal 31 that is the destination (step S04). When the IP telephone terminal 31 responds to the incoming call (step S05), the public IP telephone network 30 responds to the IP-PBX 20 with the incoming address of the IP telephone terminal 31 as the incoming voice signal termination point (step S06). Thereby, the call connection between the IP telephone terminal 11 and the IP telephone terminal 31 is prepared.

  In response to the response, the call server unit 21 of the IP-PBX 20 causes the codec conversion unit 25 included in the IP-PBX 20 to transmit a voice signal including the designation of each address of the IP telephone terminal 11 and the IP telephone terminal 31 and the relay address. A relay request is sent (step S07). In response to this, the codec conversion unit 25 creates an audio signal path setting table 27 based on each designated address (see FIG. 3), and responds to the call server unit 21 with the completion thereof (step S08). In response to the response, the call server unit 21 responds to the IP telephone terminal 11 using the relay address as the terminating voice signal termination point (step S09), so that the IP telephone terminal 11 and the IP telephone terminal 31 are connected. A voice signal packet path is set as the speech path.

  Next, by a call between the IP telephone terminal 11 and the IP telephone terminal 31, the voice signal packet is exchanged between the transmission address and the relay address of the IP telephone terminal 11 while converting the voice compression method (step). S10). On the other hand, the voice signal packet is exchanged between the incoming address and the relay address of the IP telephone terminal 31 while performing the voice compression conversion (step S11). Between these, the IP-PBX 20 performs mutual conversion of compression methods of audio signals from both sides (step S12).

  In the above embodiment, when the IP telephone terminal under the IP-PBX performs voice communication with the public IP telephone network, an uncompressed voice signal packet from the public IP telephone network is configured by including a codec conversion unit in the IP-PBX. Is converted into a compressed voice signal packet and transferred to an IP telephone terminal under the IP-PBX. On the other hand, uncompressed voice signal packets from IP telephone terminals under the IP-PBX are converted into compressed voice signal packets and transferred to the public IP telephone network. Thereby, the contract bandwidth of the IP-VPN at the base where the IP telephone terminal under the IP-PBX is located can be reduced, and the cost of the IP-VPN can be reduced.

It is explanatory drawing explaining the conventional IP telephone exchange method and apparatus. 1 is a block diagram showing an overall configuration including an IP telephone exchange apparatus according to the present invention, showing an embodiment of the present invention. FIG. It is explanatory drawing explaining the structure of the idle relay address table and audio | voice signal path | route setting table which are shown by FIG. It is a sequence diagram which shows the process sequence of the IP telephone exchange method by this invention.

Explanation of symbols

10 IP-VPN
11, 31 IP phone terminal 19 Normal phone 20 IP-PBX
21 Call Server Unit 25 Codec Conversion Unit 26 Free Relay Address Table 27 Voice Signal Routing Table 30 Public IP Telephone Network

Claims (4)

An IP telephone exchange method for making a call connection between a plurality of IP telephone terminals using different voice compression methods,
An address holding step for holding at least two empty relay addresses in the empty relay address table;
In response to the call connection request, a preparation step for associating the outgoing address of the IP telephone terminal on the outgoing side with the incoming address of the IP telephone terminal on the incoming side to prepare for the call connection between the IP telephone terminals;
A route that selects one of the vacant relay addresses as a relay address for each call connection request and sets a bidirectional voice signal packet route from the originating address to the terminating address via the relay address Configuration steps;
A voice signal packet directed to the relay address is received from each of the IP telephone terminals, converted into a format suitable for the voice signal compression method of the opposite IP telephone terminal in the call, and the opposite IP telephone terminal Relay step to send to
An IP telephone exchange method comprising:   2. The different audio compression methods include at least a method of compressing an audio signal and encoding it into an audio signal packet, and a method of encoding an audio signal into an audio signal packet without compression. The described IP telephone exchange method. An IP telephone exchange apparatus for call-connecting a plurality of IP telephone terminals using different voice compression methods,
Address holding means for holding at least two free relay addresses in the free relay address table;
In response to the call connection request, the originating address of the originating IP telephone terminal is associated with the incoming address of the terminating IP telephone terminal, and preparing means for preparing a call connection between the IP telephone terminals;
A route that selects one of the vacant relay addresses as a relay address for each call connection request and sets a bidirectional voice signal packet route from the originating address to the terminating address via the relay address Setting means;
Voice signal packets destined for the relay address are received from each of the IP telephone terminals, converted into a format suitable for the voice signal compression method of the opposite IP telephone terminal in the call, and sent to the opposite IP telephone terminal Relay means to send to,
An IP telephone exchange apparatus comprising:   4. The different audio compression methods include at least a method of compressing an audio signal and encoding it into an audio signal packet, and a method of encoding an audio signal into an audio signal packet without compression. The IP telephone exchange device described.

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