JP6552035B2 - Wireless communication system - Google Patents

Description

  The present invention relates to a wireless communication system that realizes broadcast of voice in a plurality of different mobile communication networks interconnected.

A wireless communication system for Public Safety is required to be equipped with PTT (Push to Talk) and broadcast functions.
PTT is a type of half-duplex communication, and is a technology capable of achieving quick-acting communication such as a "push to talk" transceiver, and dials the other party's telephone number like a mobile phone. Instead, you can connect in a matter of seconds by selecting the person you can talk to and pressing a button.
Broadcast communication is a function of simultaneously transmitting a report of the same content to two or more specific receiving facilities, and is an indispensable technology as a communication means for transmitting emergency information to a plurality of parties.
A mechanism of a wireless communication system that realizes broadcast communication using PTT will be described with reference to FIGS.

FIG. 3 is a system configuration diagram showing an example of a conventional wireless communication system for realizing broadcast communication by PTT in the same system. FIG. 3 shows an arbitrary wireless communication system that encodes a voice signal, converts it into data, and transmits it as a voice packet. A base station 105 in the mobile communication network 101, a mobile communication terminal station (hereinafter referred to as a terminal station for short). (1) 102, terminal station (2) 103, and terminal station (3) 104, session establishment server 106, and control server 107. A broken arrow indicates a flow of control information, and a solid arrow indicates a flow of voice information (voice packet).
The wireless communication between the base station 105 and the terminal station (1) 102, the terminal station (2) 103, and the terminal station (3) 104 is assumed to be realized by an arbitrary wireless access method.

  The session establishment server 106 manages user identification information such as a telephone number and correspondence data on a network location such as an IP address, and broadcast communication between each terminal station in the mobile communication network 101 and the control server 107. It has a function of mediating calling information and the like transmitted and received with the management unit 109.

  The control server 107 is a server that performs control for realizing broadcast communication by PTT, and includes a broadcast communication management unit 109 and a voice packet copying unit 108. The broadcast communication management unit 109 is connected to each terminal station in the mobile communication network 101 via the session establishment server 106, and the call group of the broadcast communication corresponding to the connection destination requested by the source terminal station. , And instructs voice packet duplicating section 108 to duplicate voice packets.

  Next, a process of transmitting a voice packet from the terminal station (1) 102 to the terminal station (2) 103 and the terminal station (3) 104 by broadcast communication will be described using FIG. FIG. 4 is a sequence diagram showing a conventional information flow of broadcast communication in the same system. The numbers of the respective components and the symbols of the respective signal lines are the same as those in the wireless communication system configuration diagram of FIG. . Hereinafter, each processing step (S101 to S112) will be described.

  In step S101, the terminal station (1) 102 selects the telephone numbers of the terminal station (2) 103 and the terminal station (3) 104 as transmission destinations, and then presses the PTT button to transfer the information to the first connection. It is included in the request (a1) and transmitted. The first connection request (a1) passes through the base station 105 (signal (b1)), and further passes through the session establishment server 106 (signal (c1)), and the broadcast communication management unit in the control server 107 109. Here, as described above, the role of the session establishing server 106 is to provide a correspondence between a telephone number and an IP address, which is correspondence data on a network, and enables transmission within the IP network. Therefore, the control information for realizing the broadcast communication does not change even after passing through the session establishment server 106, and the contents of the information included in the signal (b1) and the signal (c1) are the same. . Regarding other control signals in the following description, it is assumed that the content of the information does not change even if it passes through the session establishment server 106.

  In step S102, when the broadcast communication management unit 109 receives the first connection request (c1), the terminal station (1) 102 is set as the calling side, and the terminal station (2) 103 and the terminal station (3) 104 are set as the receiving side. Create a call group list as. A telephone number is used as identification information of each terminal station.

  In step S103, the broadcast communication management unit 109 transmits, to the terminal station (1) 102, a registration completion notification (d1) indicating that the registration has been completed. The registration completion notification (d1) is received by the terminal station (1) 102 via the session establishment server 106 (signal (e1)) and further via the base station 105 (signal (f1)).

  In step S104, when the terminal station (1) 102 receives the registration completion notification (f1), the terminal station (1) 102 executes step S109 to be described later after a predetermined wait time considering the processing time required for steps S105 to S108 shown below and the communication time. Do.

  In step S105, the broadcast communication management unit 109 transmits the second connection request to the terminal station (2) 103 and the terminal station (3) 104. The second connection request (g1) to the terminal station (2) 103 is transmitted via the session establishment server 106 (signal (h1)) and further via the base station 105 (signal (i1)). (2) Received by 103. The second connection request (j1) to the terminal station (3) 104 passes through the session establishment server 106 (signal (k1)), and further passes through the base station 105 (signal (l1)). And received by the terminal station (3) 104.

  In step S106, upon receiving the second connection request (i1), the terminal station (2) 103 transmits a connection response (m1) to the broadcast communication management unit 109. The connection response (m1) is received by the broadcast communication management unit 109 via the base station 105 (signal (n1)) and further via the session establishment server 106 (signal (o1)).

  In step S107, when receiving the second connection request (11), the terminal station (3) 104 transmits a connection response (p1) to the broadcast communication management unit 109. The connection response (p1) is received by the broadcast communication management unit 109 via the base station 105 (signal (q1)) and further via the session establishment server 106 (signal (r1)).

  In step S108, upon receiving the connection response (o1, r1), the broadcast communication management unit 109 sends a copy instruction (s1) to the voice packet copying unit 108. In this example, since the voice packet transmitted from the terminal station (1) 102 is duplicated and transmitted to the terminal station (2) 103 and the terminal station (3) 104, 1: 2 (input: output) Instructs to perform copy processing.

  A session between the terminal station (1) 102, the terminal station (2) 103, and the terminal station (3) 104 is established by the processing up to the step S108. Subsequent voice packet transmission and reception processing is performed via the voice packet copying unit 108 in the control server 107.

  In step S109, the terminal station (1) 102 transmits the voice packet (t1) to the voice packet copying unit 108 after the wait process of step S104. In practice, the voice packet is sent out by the user of the terminal station (1) 102 uttering while pressing the PTT button provided on the terminal station. The voice packet (t1) is received by the voice packet duplication unit 108 via the base station 105 (signal (u1)).

  In step S110, the voice packet copying unit 108 copies 1: 2 (input: output) to the input voice packet (u1), and outputs the voice packets (v1) and (x1) that are the output. The data is transmitted to the terminal station (2) 103 and the terminal station (3) 104.

  In step S111, the voice packet (v1) transmitted from the voice packet copying unit 108 is received by the terminal station (2) 103 via the base station 105 (signal (w1)).

  In step S112, the voice packet (x1) transmitted from the voice packet duplicating unit 108 is received by the terminal station (3) 104 via the base station 105 (signal (y1)).

  Although the description by the sequence diagram of FIG. 4 is omitted, for example, when the user of the terminal station (1) 102 releases the PTT button and then the user of the terminal station (2) 103 speaks while pressing the PTT button, the terminal A voice packet reaches the control server 107 from the station (2) 103, is duplicated by the voice packet duplicating unit 108, and is transmitted to the terminal station (1) 102 and the terminal station (3) 104. Then, when the call is over, the broadcast communication management unit 109 deletes the data of the registered call group list.

On the other hand, as a recent trend, in response to the demand for broadband communication such as public safety, coverage area expansion for commercial radio field, real-time video transmission, etc., LTE (Long Term Evolution) is a 4G (4th generation mobile communication system) communication standard. ) Movements related to new utilization of technology are intensifying. Trials are underway with the aim of applying LTE technology to Public Safety and railway systems in the US, Europe, and other regions, and the 3rd Generation Partnership Project (3GPP) is similar to LTE standards that include the Public Safety function. It will be issued in the future.
When LTE is deployed in the field of public safety and commercial radio, a wireless communication system network in a form that coexists with existing public safety and narrowband radio in the commercial radio field (hereinafter collectively referred to as existing private radio) is constructed. It is expected. The following two effects are obtained by the configuration in which the LTE network and the existing private wireless network are interconnected.

(1) By expanding the LTE system to complement the coverage area of the existing privately-owned radio system, the coverage area of the privately-owned radio can be expanded. In this case, the voice communication cover area can be expanded.
(2) The LTE system is deployed so as to overlay the cover area of the existing private wireless system, and broadband communication such as real-time video transmission is provided. Also in this case, it is assumed that the terminal station of the LTE system has a function capable of voice communication with the terminal station of the private wireless system.

  In the LTE network, the specifications of the voice communication based on the conventional circuit switching method are not stipulated, and the voice over long term is exchanged through the packet switching network in the same way as other data communication by converting voice into data. Evolution) is the standard. However, as an optional specification, CSFB (Circuit Switched FallBack: circuit-switched fallback, CS fall) that automatically connects to the conventional 3G (3rd generation mobile communication system) communication network only during voice calls and makes circuit-switched calls Back) method is prepared, and in the transition period, CSFB calls are first implemented, and it is assumed that the system will gradually transition to VoLTE.

  In addition to the AMR-NB (Adaptive Multi-Rate Narrow Band) used in 3G as a required codec, the voice coding method (voice codec) used in VoLTE is also a higher quality AMR- WB (Adaptive Multi-Rate Wide Band) is also available. The encoding rate of AMR-NB (transmission rate of encoded audio signal) is 4.75 to 12.2 kbps. The coding rate of AMR-WB is 6.60 to 23.85 kbps, and the required multirate configuration is from 6.60 kbps to 12.65 kbps. Normally, 12.65 kbps is used. A higher bit rate is used when high sound quality is required such as an environment with a lot of background noise, a combination of voice and music, and a multi-party conference.

Next, an example of an existing private radio and a voice encoding method (voice codec) used therein are shown below. Since the existing private radio is narrow band, voice communication is mainly used, but low speed data of several kbps can also be transmitted. A voice call uses a PTT that enters a voice transmission state when the transmission button is pressed.
[1] ARIB STD-T79 Prefectural and municipal digital mobile communication system (260 MHz band)
Speech coding method: The transmission rate of a coded speech signal is 6.4 kbps including an error correction code and the like. The EL-CELP method is recommended for speech coding algorithms.
[2] ARIB STD-T80 Prefecture / municipal digital mobile communication system TYPE 2 (260 MHz band)
Speech coding scheme: The speech coding algorithm in this system is ACELP. The transmission rate of the encoded audio signal is 7.2 kbps including the error correction code.
[3] ARIB STD-T102 Narrow band digital communication system (SCPC / 4-value FSK system) (400 MHz band and 150 MHz band)
Speech coding method: 4-level FSK method SCPC The transmission rate of coded speech signals in narrowband digital mobile communication systems, including error correction codes, shall be 3.6 kbps or less. For voice coding, AMBE + 2 (registered trademark) Enhanced Half-Rate (3.6 kbps: voice code 2.45 kbps, error correction code 1.15 kbps) of Digital Voice Systems, Inc. with a frame size of 20 ms (72 bits) is recommended. In addition, error correction coding is a system attached to the speech coding system.

  In a network in which different networks such as an LTE network and an existing private wireless network are interconnected, if the audio codec used in each network is different, the code is encoded by an MGW (Media Gateway) having an audio data interface function. A conversion is performed. As the code conversion method, after the coded data is input and once decoded, the coded data re-coded by the coding method on the communication partner side is output.

Here, in the LTE network, only AMR-WB with a coding rate of 6.60 to 12.65 kbps can be used as a voice codec, and only AMBE + 2 (trademark) with a coding rate of 2.45 kbps can be used in an existing private wireless network. Assuming that, there were the following problems.
(1) When a terminal station in the LTE network uses AMR-WB with a coding rate of 12.65 kbps as a voice codec, the AMR-WB is used from the LTE network to the MGW (including the radio section). Voice encoded data that has been code-converted to AMBE + 2 (trademark) with a coding rate of 2.45 kbps used in private wireless is transmitted from the MGW to the existing private wireless network. In this case, the voice quality between the terminal station in the LTE network and the existing private wireless network terminal station is limited to AMBE + 2 (trademark) quality. Therefore, it can be said that it is a waste of frequency resources to use the broadband AMR-WB in the radio section from the LTE network to the MGW.
(2) Tandem connection of voice codec (connection by code conversion) causes deterioration of voice quality. In particular, in the case of a tandem connection of an audio codec with a low encoding rate, the inferior inaudibility becomes significant. AMR-WB has an essential multi-rate configuration of 6.60 / 8.85 / 12.65 kbps, and at the normally used 12.65 kbp, the quality is sufficiently high in the LTE network to MGW section, so the AMBE + 2 (trademark) tandem is used. There is almost no inferiority due to connection (the terminal station in the LTE network--same voice quality as when using AMBE + 2 (trademark) between the existing private wireless network terminal stations). However, when 6.60kbps is used with AMR-WB when the wireless line condition is bad, the quality degradation due to AMR-WB overlaps with the quality degradation due to AMBE + 2 (trademark). The inferiority due to

  In JP 2012-105212 A (core network and communication system) of Patent Document 1, as a solution to the above problem (1), an audio codec used on the incoming side can be made an audio codec used on the outgoing side Core networks and communication systems have been proposed that can match only and use bandwidth efficiently. Specifically, the audio codec specification unit in the core network specifies the audio codec so that both audio codecs are matched as much as possible based on the information indicating the type of the audio codec sent from the transmission side and the reception side. . When there is no matching audio codec, the audio codec is designated so as to select the one with the closest encoding rate.

JP 2012-105212 A International Publication No. 2011/162349 JP 2013-251844 A JP 2009-284127 A

  However, although the technique (1) of the above-described problem can be solved by the technique of Patent Document 1, since the MGW performs code conversion when there is no matching audio codec, the above-described problem (2) is solved. I can't do it. In addition, in the prior art, a method (control sequence or the like) for realizing broadcast broadcasting of voice by PTT while solving the above-described problems (1) and (2) is not disclosed.

  The present invention has been made in view of such conventional circumstances, and by ensuring that the voice codecs used in the interconnected different types of mobile communication networks are matched, the waste of frequency resources does not occur, and Since the MGW does not need to perform code conversion, an object of the present invention is to provide a wireless communication system that can prevent deterioration of voice quality due to tandem connection and can realize voice broadcast communication using PTT. I assume.

  In order to achieve the above object, a wireless communication system according to the present invention includes a first mobile communication network and a second mobile communication network, which are different mobile communication networks to be interconnected, and identification of a user such as a telephone number. A wireless communication system comprising a session establishment server that manages correspondence information on a network such as information and an IP address and enables IP communication in the network, and a control server that performs control for broadcast communication And the control server communicates with the first mobile communication network and the second mobile communication network via the session establishment server, and is connected to the originating side belonging to the first mobile communication network. At least one voice available from the mobile communication terminal station on the first mobile communication network Information indicating a voice codec which is a deck, and an identifier such as a telephone number for identifying one or more mobile communication terminal stations of a called party belonging to the first mobile communication network or the second mobile communication network Mobile communication belonging to the second mobile communication network on the receiving side by creating a call group list in which the mobile communication terminal stations on the calling and receiving sides are registered as a call group for receiving the connection request 1 and performing broadcast communication When a second connection request is made to the terminal station and a first connection response indicating at least one voice codec available from the mobile communication terminal station belonging to the second mobile communication network on the called side is returned, After updating the call group list to reflect the contents of the first connection response, the voice codec specification unit transmits the call group list, and When the voice codec designation notification is received from the voice codec designation unit, the call group list is updated for the voice codec information to be used. When the voice codec addition notification is received from the voice codec designation unit, the voice codec addition unit receives Receiving the additional voice codec information, updating the call group list as a voice codec using the voice codec to be added, transmitting a registration completion notification including the additional voice codec information to the mobile communication terminal station on the calling side, and Transmitting a third connection request including additional voice codec information to the called mobile communication terminal station belonging to the first mobile communication network, and further to the called mobile communication terminal station belonging to the second mobile communication network A third connection request is sent and all the mobile communication terminals on the called side When the second connection response is returned, a broadcast communication management unit that transmits a copy instruction to the voice packet copy unit, and the first group indicated by the call group list input from the broadcast communication management unit At least one voice codec that can be used in a mobile communication network, at least one voice codec that can be used in the second mobile communication network, and a voice codec that can be used in an MGW ((Media Gateway) in the control server When there is a common voice codec, the common voice codec is designated as a voice codec to be used in the first mobile communication network and the second mobile communication network, and the broadcast communication management unit and the While transmitting a voice codec specification notification to the MGW, if there is no common voice codec, A voice codec addition notification is sent to the broadcast communication management unit and the MGW, and a voice codec is added to the first mobile communication network and the MGW to match the voice codec of the second mobile communication network. A voice codec addition unit that outputs a voice codec addition request to the voice codec addition unit, and stores a plurality of voice codec program information, and receives the voice codec addition request from the voice codec designation unit. A voice codec addition unit for transmitting additional voice codec information, which is information of a voice codec to be added, to the broadcast communication management unit and the MGW; and a codec specification notification from the voice codec specification unit; Set codec or An MGW (Media Gateway) that receives a voice codec addition notification from a deck designation unit and additional voice codec information from the voice codec addition unit, sets a voice codec, and performs format conversion when a voice packet is input; A voice packet duplicating unit that duplicates and outputs a voice packet upon receiving a duplication instruction from the broadcast communication management unit, and the second mobile communication network on the receiving side receives the second connection request In response to the second connection request, and transmitting the first connection response indicating at least one voice codec available in the network to the control server, When the first mobile communication network receives the additional audio codec information from the audio codec adding unit, the first mobile communication network Wherein the mobile communication terminal station within a mobile communication network to implement the speech codec program, and having an operation is to function.

  In addition, in order to achieve the above object, a wireless communication system according to the present invention is characterized in that, in the above-described wireless communication system, any one mobile communication terminal station belonging to the second mobile communication network is an originator, other mobile stations It is characterized in that the communication terminal station is at the receiving side.

  In order to achieve the above object, a radio communication system according to the present invention is the above-described radio communication system, wherein the first mobile communication network and the second mobile communication network are LTE (Long Term Evolution) systems. And existing private wireless systems.

  According to the present invention, the frequency codec is not wasted by always matching the voice codec used in the heterogeneous mobile communication networks to be interconnected, and the MGW does not need to perform code conversion. It is possible to prevent the deterioration of the voice quality due to the tandem connection, and it is possible to realize the voice broadcast communication using the PTT.

It is a system configuration figure showing an example of a radio communications system concerning one embodiment of the present invention. It is a sequence diagram which shows the information flow in the radio | wireless communications system which concerns on one Embodiment of this invention. It is a sequence diagram which shows the information flow in the radio | wireless communications system which concerns on one Embodiment of this invention. It is a sequence diagram which shows the information flow in the radio | wireless communications system which concerns on one Embodiment of this invention. It is a system configuration | structure figure which shows an example of the radio | wireless communications system which implement | achieves the broadcast communication by PTT in the conventional same system. It is a sequence diagram which shows the information flow of the broadcast communication in the same conventional system.

Hereinafter, a radio communication system according to an embodiment of the present invention will be described.
A wireless communication system according to an embodiment of the present invention is a wireless communication system in which different mobile communication networks are interconnected, in which the frequency is matched by ensuring that the voice codecs used in the interconnected different mobile communication networks are matched. There is no waste of resources, and MGW does not need to perform code conversion, so that occurrence of voice quality deterioration due to tandem connection can be prevented, and broadcast of voice communication by PTT can be realized. It is.

[System configuration of wireless communication system]
A system configuration of a wireless communication system according to an embodiment of the present invention will be described with reference to FIG. FIG. 1 is a system configuration diagram showing an example of a wireless communication system according to an embodiment of the present invention.
FIG. 1 shows a wireless communication system that encodes voice signals, converts them into data, and transmits them as voice packets. The first mobile communication network 201 and the second mobile communication network 202 are different mobile communication networks that are interconnected. Managing the identification information of the user such as the telephone number and the correspondence data of the location on the network such as the IP address, and the session establishing server 208 enabling the IP communication in the network, and the control for the broadcast communication. And a control server 209 for performing. The mobile communication network 201 includes a base station (1) 206, a mobile communication terminal station (hereinafter referred to as a terminal station for short) (1-1) 203, and a terminal station (1-2) 204. It is assumed that a base station (2) 207 and a terminal station (2-1) 205 exist in the network 202. A broken arrow indicates a flow of control information, and a solid arrow indicates a flow of voice information (voice packet).

As a concrete example to which the configuration of the wireless communication system of the present invention is applied, LTE is expanded to public radio and private radio in the field of business radio, etc., and coexists with existing narrow band private radio to achieve interoperability of voice communication Think about the case. That is, in FIG. 1, the first mobile communication network 201 is an LTE system, and the second mobile communication network 202 is an existing private wireless system. Here, as a configuration in the first mobile communication network 201, a base station (1) 206 and a terminal station (1-1) 203, which are E-UTRAN (Evolved Universal Terrestrial Radio Access Network) of a wireless network, a terminal station ( 1-2) Only 204 is shown, and illustration and function description of EPC (Evolved Packet Core) of the core network are omitted. Further, in the following description, description of functions other than the broadcast voice communication related to the present invention is omitted.

The session establishment server 208 manages user identification information such as a telephone number and correspondence data on a location such as an IP address on the network, and each of the first mobile communication network 201 and the second mobile communication network 202 It has a function of mediation of calling information and the like transmitted and received between the terminal station and the broadcast communication management unit 210 in the control server 209.
In the following description, the terminal station (1-1) 203 in the first mobile communication network 201 will be described as a transmitting terminal station, and the other terminal stations will be described as receiving terminal stations.

  The control server 209 includes a broadcast communication management unit 210, a voice codec designation unit 213, a voice codec addition unit 214, an MGW (Media Gateway) 212, and a voice packet duplication unit 211.

The broadcast communication management unit 210 performs communication between the first mobile communication network 201 and the second mobile communication network 202 via the session establishment server 208, and the originating side belonging to the first mobile communication network 201. Information indicating at least one voice codec available for use in the first mobile communication network 201 from the terminal station (1-1) 203 of FIG. 5, and an incoming call in the first mobile communication network 201 or the second mobile communication network 202. Terminals of a calling side and a called side as a call group which receives a first connection request (a2, b2, c2) including an identifier such as a telephone number for identifying one or more terminal stations on the side and performs broadcast communication Create a call group list with stations registered.
Here, as the first connection request, the terminal output from the terminal station (1-1) 203 (a2), the one passing through the base station (1) 206 (b2), the session establishment server 208 Although what is described as (c2), the content of the information does not change even if it passes through the base station (1) 206 and the server 208 for session establishment. The same applies to other control information.

The voice codec is defined by the voice coding scheme, coding rate, and specifications of the error correction code to be applied. After that, the broadcast communication management unit 210 transmits the second connection request (d2, e2, f2) to the terminal station (2-1) 205 in the second mobile communication network 202 on the receiving side, 2-1) When the first connection response (g2, h2, i2) indicating at least one voice codec available from 205 is returned, the content is updated to reflect the contents and the voice group list is updated, When the call group list information (k2) is transmitted to the codec designation unit 213, and then the voice codec designation notification (g3) is received from the voice codec designation unit 213, the call group list is updated (the voice codec to be used) Register information).
On the other hand, the broadcast communication management unit 210, when receiving the audio codec addition notification (g3) from the audio codec specifying unit 213, adds additional audio codec information (method name, program, frame format) from the audio codec adding unit 214. Etc.) (o2) is received and the call group list is updated as a voice codec using it. Thereafter, a registration completion notification (p2, q2, r2) is transmitted to the terminal station (1-1) 203 on the transmission side.

  Here, the broadcast communication management unit 210 completes registration of additional audio codec information (method name, program, frame format, etc.) when the audio codec addition notification (g3) is received from the audio codec designation unit 213. The notification (p2, q2, r2) is included in the notification (p2, q2, r2) and transmitted to the terminal station (1-1) 203 on the transmission side. Then, third connection requests (s2, t2, u2), (v2, w2, x2) are sent to the receiving terminal station (1-2) 204 and terminal station (2-1) 205 registered in the call group list. ). In addition, when the broadcast codec management unit 210 receives the audio codec addition notification (g3) from the audio codec designation unit 213, the broadcast communication management unit 210 receives the additional audio codec information (method name, program, frame format, etc.) as the third code. The connection request (s2, t2, u2) is included in the connection request (s2, t2, u2) and transmitted to the terminal station (1-2) 204.

  As described above, when the broadcast codec addition notification (g3) is received from the speech codec designation unit 213, the broadcast communication management unit 210 adds the speech codec to the terminal station in the first mobile communication network 201. Send information for. In addition, the broadcast communication management unit 210 sends second connection responses (y2, z2, a3), (b3, c3, d3) from the terminal station (1-2) 204 and the terminal station (2-1) 205 on the receiving side. ) Is sent, the copy instruction (j2) is sent to the voice packet copy unit 211.

The voice codec designating unit 213 is configured to perform at least one voice codec usable in the first mobile communication network 201 indicated by the call group list information (k2) input from the broadcast communication control unit 210, and a second transfer. If there is a common audio codec among at least one audio codec that can be used in the communication network 202 and an audio codec that can be used by the MGW 212 in the control server 209, the common audio codec and the second mobile communication network 201 It designates as a voice codec to be used in the mobile communication network 202, and sends voice codec specification notices (g3) and (12) including voice codec information to the broadcast communication management unit 210 and MGW 212, respectively.
Here, when there are a plurality of common speech codecs, for example, the speech codec with the lowest coding rate is selected. On the other hand, if there is no common audio codec, the audio codec addition notifications (g3) and (l2) are transmitted to the broadcast communication management unit 210 and the MGW 212, and the audio codec addition request (m2) is sent to the audio codec addition unit. Output to 214. Here, the audio codec used in the first mobile communication network 201 and the MGW 212 is added so as to match the audio codec available in the second mobile communication network 202. When a plurality of voice codecs available in the second mobile communication network 202 exist, a voice codec to be used as a standard is selected. Information on the voice codec used as a standard is assumed to be included in the first connection response (g2).

  The voice codec adding unit 214 can store and update a plurality of voice codec programs, and can include voice codec information (scheme name, program, frame format, etc.) defined in all existing private wireless standards to which application is assumed. keeping. Here, the storage format of the audio codec program is arbitrary. When the audio codec addition request (m2) is received from the audio codec specifying unit 213, the additional audio codec information (method name, program, frame format, etc.) indicated therein is sent to the broadcast communication management unit 210 and the MGW 212 (o2), ( Send as n2). The transmission format of the voice codec program is arbitrary.

When the MGW 212 receives a codec designation notification (l2) from the voice codec designation unit 213 at the time of session establishment, the MGW 212 sets the voice codec accordingly. When a codec addition notification (l2) is received from the audio codec specifying unit 213, additional audio codec information (method name, program, frame format, etc.) (n2) from the audio codec adding unit 214 is received and the audio codec is set. To do. Here, the MGW 212 uses only the frame format information of the received additional audio codec information (method name, program, frame format, etc.) (n2). In this embodiment, since the voice codecs used in the first mobile communication network 201 and the second mobile communication network 202 are controlled to always coincide with each other, it is necessary based on the above-mentioned frame format information of the encoded data. The frame format conversion process is performed according to the above. After the session is established, when voice packets (g3) and (h3) are input, format conversion is performed as necessary, and (i3) and (k3) are output.
The voice packet copying unit 211 copies the input voice packet (f3) in response to the copy instruction (j2) from the broadcast communication management unit 210, and outputs it as voice packets (g3) and (h3).

The first mobile communication network 201 includes a base station (1) 206, a terminal station (1-1) 203, and a terminal station (1-2) 204, and the terminal station (calling side) ( 1-1) 203 indicates information indicating at least one voice codec available in the first mobile communication network 201 when the PTT button is pressed, and the terminal station (1-2) 204 and the terminal on the receiving side The first connection request (a2, b2, c2) including an identifier such as a telephone number for identifying the station (2-1) 205 is transmitted to the broadcast communication control unit 210 in the control server 209, and as a response, registration is performed. When completion notification (p2, q2, r2) is returned, if the content includes additional voice codec information (method name, program, frame format, etc.), the voice codec to be used and To add Te. Thereafter, after a predetermined wait time has elapsed, that is, after the session is established, the voice packet (e3, f3) is transmitted to the voice packet copying unit 211 in the control server 209 via the base station (1) 206.
The terminal station (1-2) 204 on the receiving side receives the third connection request (s2, t2, u2) from the broadcast communication management unit 210, and adds to the contents the additional voice codec information (method name, program, Frame format, etc.) is added as a voice codec to be used, and a second connection response (y2, z2, a3) indicating that a call is possible is returned. After the session is established, the terminal station (1-2) 204 receives the voice packet (i3, j3) transmitted from the MGW 212 in the control server 209 via the base station (1) 206.

  Here, the terminal station (1-1) 203 and the terminal station (1-2) 204 in the first mobile communication network 201 comply with all the audio codec specifications (sampling frequency, audio signal band) assumed to be applied. When the base station (1) 206 receives additional audio codec information (method name, program, frame format, etc.) (q2) and (t2) from the broadcast communication management unit 210, The additional audio codec information is transmitted to and set in the terminal station (1-1) 203 and the terminal station (1-2) 204 to be operable. Between the base station (1) 206, the terminal station (1-1) 203, and the terminal station (1-2) 204, the additional voice codec information can be transmitted in a short time using an LTE high-speed line.

  The second mobile communication network 202, which is the call receiving side, comprises the base station (2) 207 and the terminal station (2-1) 205, and at the time of session establishment, the second connection request (d2, e2, f2) A first connection response (g2, h2, i2) received from the broadcast management unit 210 in the control server 209 and responsive thereto indicating at least one voice codec available in the second mobile communication network 202 ) Also, the third connection request (v2, w2, x2) is received from the broadcast communication management unit 210, and in response thereto, a second connection response (b3, c3, d3) indicating that a call is possible is received. Send back. After the session is established, the terminal station (2-1) 205 receives, via the base station (2) 207, the voice packet (k3, l3) transmitted from the MGW 212 in the control server 209.

[Information flow in wireless communication system]
Next, an information flow in the wireless communication system according to the embodiment of the present invention will be described with reference to FIG. 2 (FIGS. 2A, 2B, and 2C). FIG. 2 is a sequence diagram showing an information flow in a wireless communication system according to an embodiment of the present invention, and shows an information flow of broadcast communication in a wireless communication system in which different mobile communication networks are interconnected. In FIG. 2, the number of each component and the symbol of each signal line coincide with the configuration diagram of the wireless system in FIG.
In the following description, the process of transmitting voice packets from the terminal station (1-1) 203 to the terminal station (1-2) 204 and the terminal station (2-1) 205 by broadcast communication will be described. The respective processing steps (S201 to S225) will be described below.

  As shown in FIG. 2A, in step S201, the terminal station (1-1) 203, which is the calling side, includes information indicating at least one voice codec available in the first mobile communication network 201, and A first connection request (a2) including an identifier such as a telephone number for identifying the terminal station (1-2) 204 and the terminal station (2-1) 205 is transmitted to the broadcast communication management unit 210 in the control server 209. . The first connection request (a2) passes through the base station (1) 206 (signal (b2)), and further passes through the session establishment server 208 (signal (c2)) and broadcasts in the control server 209 Received by the communication manager 210. Here, as described above, the role of the session establishment server 208 is to provide a correspondence between a telephone number and an IP address, which is correspondence data on the network, and enables transmission within the IP network. Therefore, as the control information for realizing the broadcast communication, the contents of the information included in the signal (b2) and the signal (c2) are the same without passing through the session establishment server 208 and changing. . Regarding other control signals in the following description, it is assumed that the content of the information does not change even if it passes through the session establishment server 208.

  In step S202, upon receiving the first connection request (c2), the broadcast communication management unit 210 sets the terminal station (1-1) 203 to the calling side, the terminal station (1-2) 204, and the terminal station (2- 1) Create a call group list with 205 as the called party. A telephone number is used as identification information of each terminal station.

  In step S203, the broadcast communication management unit 210 transmits a second connection request (d2) to the terminal station (2-1) 205 in the second mobile communication network 202. The second connection request (d2) passes through the session establishment server 208 (signal (e2)), and further passes through the base station (2) 207 (signal (f2)), and the terminal station (2-1) 205.

  In step S204, the terminal station (2-1) 205 returns a first connection response (g2) indicating at least one voice codec available in the second mobile communication network 202. The first connection response (g2) is received by the broadcast communication management unit 210 via the base station (2) 207 (signal (h2)) and via the establishment server 208 (signal (i2)). Ru.

  As shown in FIG. 2B, in step S205, when the broadcast communication management unit 210 receives the first connection response (i2), it updates the call group list to reflect the content, and then the voice codec designating unit. Call group list information (k 2) is sent to S 213.

  In step S <b> 206, the voice codec designation unit 213 can use at least one voice codec that can be used in the first mobile communication network 201 indicated by the call group list information (k <b> 2) and the second mobile communication network 202. It is determined whether or not there is a common voice codec among at least one voice codec and a voice codec that can be used by the MGW 212 in the control server 209. If there is a common voice codec, processing step S207 is performed. If it does not exist, the process proceeds to processing step S208.

  In step S207, if there is a common audio codec, the audio codec designating unit 213 designates it as an audio codec to be used in the first mobile communication network 201 and the second mobile communication network 202, and broadcast communication. The voice codec specification notices (g3) and (l2) are transmitted to the management unit 210 and the MGW 212, respectively. Here, when there are a plurality of common audio codecs, for example, the audio codec with the lowest coding rate is selected.

  In step S208, when there is no common audio codec, the audio codec specification unit 213 transmits the broadcast communication management unit 210 and the MGW 212 audio codec addition notification (g3) and (12).

  In step S209, when there is no common audio codec, the audio codec specifying unit 213 transmits an audio codec addition request (m2) to the audio codec adding unit 214. Here, the audio codec used in the first mobile communication network 201 and the MGW 212 is added so as to match the audio codec available in the second mobile communication network 202. When a plurality of voice codecs available in the second mobile communication network 202 exist, a voice codec to be used as a standard is selected. Information on the voice codec used normally is included in the second connection response (g2).

  In step S210, the audio codec adding unit 214 has a function of storing / updating a plurality of audio codec programs. When the audio codec addition request (m2) is received from the audio codec specifying unit 213, the additional audio codec indicated therein is received. Information (scheme name, program, frame format, etc.) is transmitted to the broadcast communication management unit 210 and the MGW 212 as (o2) and (n2), respectively.

  In step S211, when the MGW 212 receives the voice codec designation notification (l2) from the voice codec designation unit 213, the MGW 212 sets the voice codec accordingly. When the audio codec addition notification (l2) is received from the audio codec designation unit 213, the additional audio codec information (method name, program, frame format, etc.) (n2) from the audio codec addition unit 214 is received and the audio codec is selected. Set Here, the MGW 212 uses only the frame format information of the received additional audio codec information (method name, program, frame format, etc.) (n2).

  As shown in FIG. 2C, in step S212, when the broadcast communication management unit 210 receives the audio codec specification notification (g3) from the audio codec specification unit 213, the broadcast communication management unit 210 updates the call group list based on that (audio used Register codec information). When the audio codec addition notification (g3) is received from the audio codec specifying unit 213, the additional audio codec information (method name, program, frame format, etc.) (o2) from the codec adding unit 214 is received and used. Update the call group list as an audio codec.

  In step S213, the broadcast communication management unit 210 transmits a registration completion notification (p2) to the terminal station (1-1) 203 on the transmission side. Here, if the voice codec addition notification (g3) has been received from the voice codec specification unit 213, the additional voice codec information (method name, program, frame format, etc.) is included in the registration completion notification (p2) and the transmission side To the terminal station (1-1) 203 of FIG. The registration completion notification (p2) passes through the session establishment server 208 (signal (q2)), and further passes through the base station (1) 206 (signal (r2)) to the terminal station (1-1) 203. Sent to

  In step S214, when the terminal station (1-1) 203 receives the registration completion notification (r2), if the content includes additional voice codec information (method name, program, frame format, etc.), the voice to be used is used. Add as a codec.

  In step S215, the terminal station (1-1) 203 waits for a predetermined time set in consideration of the time required for session establishment (time required in the processing steps S216 to S220).

  In step S216, the broadcast communication management unit 210 sends a third connection request (s2) to the terminal station (1-2) 204 and terminal station (2-1) 205 registered in the call group list. ) And (v2) are transmitted. Here, when the voice codec addition notification (g3) has been received from the voice codec designation unit 213, the additional voice codec information (method name, program, frame format, etc.) is included in the third connection request (s2). It transmits toward the terminal station (1-2) 204. As described above, when the audio codec addition notification (g3) is received from the audio codec specifying unit 213, information for adding the audio codec is transmitted to the terminal station in the first mobile communication network 201. . The third connection request (s2) passes through the session establishment server 208 (signal (t2)), and further passes through the base station (1) 206 (signal (u2)) to the terminal station (1-2). 204. Also, the third connection request (v2) passes through the session establishment server 208 (signal (w2)), and further passes through the base station (2) 207 (signal (x2)), and the terminal station (2- 1) transmitted to 205;

  In step S 217, the terminal station (1-2) 204 receives the third connection request (u 2) from the broadcast communication management unit 210, and adds additional voice codec information (method name, program, frame format, etc.) to its contents. If it is included, add it as a voice codec to use.

  In step S218, the terminal station (1-2) 204 transmits a second connection response (y2) indicating that a call is possible to the broadcast communication management unit 210. The second connection response (y2) is sent to the broadcast communication management unit 210 via the base station (1) 206 (signal (z2)) and further via the session establishment server 208 (signal (a3)). Will be sent.

  In step S219, the terminal station (2-1) 205 transmits, to the broadcast communication management unit 210, a second connection response (b3) indicating that communication is possible. The second connection response (b3) passes through the base station (2) 207 (signal (c3)), and further passes through the session establishment server 208 (signal (d3)) to the broadcast communication management unit 210. Will be sent.

  In step S220, the broadcast communication management unit 210 returns the second connection responses (a3) and (d3) from the terminal station (1-2) 204 and the terminal station (2-1) 205 on the receiving side. The duplication instruction (j2) is transmitted to the voice packet duplication unit 211.

  Through the processing up to step S220, a session between the terminal station (1-1) 203, the terminal station (1-2) 204, and the terminal station (2-1) 205 is established. Subsequent voice packet reception processing is performed via the voice packet duplication unit 211 and the MGW 212 in the control server 209.

  In step S221, the terminal station (1-1) 203 transmits the voice packet (e3) to the voice packet duplication unit 211 in the control server 209 via the base station (1) 206 (signal (f3)). Actually, the voice packet (e3) is transmitted when the user of the terminal station (1-1) 203 speaks while pressing the PTT button equipped in the terminal station.

  In step S222, when the voice packet duplicating unit 211 receives the voice packet (f3), the voice packet duplicating unit 211 performs two broadcast transmissions to the terminal station (1-2) 204 and the terminal station (2-1) 205 on the receiving side. The voice packets (g3) and (h3) are copied and output to the MGW 212.

  In step S223, when the voice packets (g3) and (h3) are input, the MGW 212 performs format conversion as necessary, and converts the results (i3) and (k3) into the terminal station (1-2). ) And the terminal station (2-1) 205.

  In step S224, the voice packet (i3) transmitted from the MGW 212 is received by the terminal station (1-2) 204 via the base station (1) 206 (signal (j3)).

  In step S225, the voice packet (k3) transmitted from the MGW 212 is received by the terminal station (2-1) 205 via the base station (2) 207 (signal (l3)).

Although the description of the sequence diagram of FIG. 2 is omitted, for example, the user of the terminal station (1-1) 203 releases the PTT button, and then the user of the terminal station (2-1) 205 presses the PTT button. Speaking, the terminal station (2-1) 205 is the transmitting side, and the voice packet passes through the voice packet copying unit 211 and the MGW 212 in the control server 209, and the terminal station (1-1) 203 and the terminal station (1-2 ) Is transmitted.
In addition, when the call ends, the broadcast communication management unit 210 deletes the registered call group list data.

  In the above description, the first mobile communication network 201 has been described as an LTE system, and the second mobile communication network 202 has been described as an existing private wireless system. However, the scope of the present invention is not limited to these systems. Absent.

  As described above, according to the wireless communication system according to an embodiment of the present invention, the frequency codec is wasted by ensuring that the voice codecs used in different types of interconnected mobile communication networks are matched. In addition, since it is not necessary to perform code conversion in the MGW, it is possible to prevent the deterioration of the voice quality due to the tandem connection, and it is possible to realize the voice broadcast communication using the PTT.

  It should be noted that the configuration and operation of the above-described embodiment are examples, and it goes without saying that the embodiment can be appropriately modified and executed without departing from the spirit of the present invention.

101: mobile communication network, 102: terminal station (1), 103: terminal station (2), 104: terminal station (3), 105: base station, 106: server for session establishment, 107: control server, 108: voice Packet replication unit 109: Broadcast communication management unit 201: First mobile communication network 202: Second mobile communication network 203: Terminal station (1-1) 204: Terminal station (1-2) 205: Terminal station (2-1), 206: Base station (1), 207: Base station (2), 208: Server for session establishment, 209: Control server, 210: Broadcast communication management unit, 211: Voice packet Duplicating unit, 212: MGW, 213: audio codec specifying unit, 214: audio codec adding unit.

Claims (4)

A first mobile communication network and a second mobile communication network which are different mobile communication networks interconnected;
A server for session establishment that manages user identification information and IP addresses;
A wireless communication system comprising a control server that controls broadcast communication, comprising:
The control server is
Information indicating a voice codec, which is at least one voice codec available in the first mobile communications network, from a mobile communications terminal station on the calling side belonging to the first mobile communications network; and the first mobile communications network Or receiving a connection request including an identifier identifying a called mobile station or mobile stations belonging to the second mobile communication network,
The broadcast communication management unit creates a call group list in which the at least one voice codec is specified, and the mobile communication terminal stations on the calling side and the called side are registered as a call group for performing broadcast communication ,
When the call group list is updated, the call group list is transmitted to the voice codec designation unit,
If the voice codec designation is received from the voice codec designation unit, the call group list is updated for the voice codec information to be used.
If there is a common voice codec available in the first mobile communication network, the second mobile communication network, and the MGW (Media Gateway) in the control server, the common voice codec is selected as the first voice codec. Designated as a voice codec to be used in the mobile communication network and the second mobile communication network,
When there is no common voice codec, a voice codec addition notification is transmitted to the broadcast management unit and the MGW, and the first mobile communication is matched with the voice codec of the second mobile communication network. A wireless communication system, wherein a voice codec is added to a network and the MGW.   2. The radio communication system according to claim 1, wherein said first mobile communication network and said second mobile communication network employ an LTE (Long Term Evolution) system and an existing private radio system, respectively. Communications system. A first mobile communication network and a second mobile communication network which are different mobile communication networks interconnected;
A server for session establishment that manages user identification information and IP addresses;
A wireless communication method of a wireless communication system comprising a control server for controlling broadcast communication,
The control server is
Information indicating a voice codec, which is at least one voice codec available in the first mobile communications network, from a mobile communications terminal station on the calling side belonging to the first mobile communications network; and the first mobile communications network Or receiving a connection request including an identifier identifying a called mobile station or mobile stations belonging to the second mobile communication network;
Designating the at least one voice codec, and creating a call group list in which mobile communication terminal stations on the calling side and the call receiving side are registered as call groups for performing broadcast communication ;
When the call group list is updated, transmitting the call group list to a voice codec designation unit;
When receiving a voice codec designation from the voice codec designation unit, updating the call group list for voice codec information to be used;
If there is a common voice codec available in the first mobile communication network, the second mobile communication network, and the MGW (Media Gateway) in the control server, the common voice codec is selected as the first voice codec. Designating a voice codec to be used in the mobile communication network and the second mobile communication network;
When there is no common voice codec, the first mobile communication is transmitted so as to transmit a voice codec addition notification to the broadcast communication management unit and the MGW, and to match the voice codec of the second mobile communication network. A wireless communication method comprising a step of adding a voice codec to a network and the MGW. 4. The wireless communication method according to claim 3, wherein the first mobile communication network and the second mobile communication network respectively apply wireless communication by a long term evolution (LTE) system and wireless communication by an existing private wireless system. A wireless communication method.