KR20050007990A - Method for multplexing of user traffic in a mobile communication system - Google Patents

Abstract

PURPOSE: A method for multiplexing user traffic in a mobile communication system is provided to bundle various voice data frames of one user, or to multiplex frames of various users to effectively use a TDM transmission band when voice data are transmitted, thereby more efficiently sending the data. CONSTITUTION: An MS(105) requests a voice call origination(301). A BSC(111) sends a BS ACK order to the MS(105)(302), and transmits a CM service request message to an MSCe(MSC emulator)(131) (303). The MSCe(131) requests an MGW(Media Gateway)(140) to set up a vocoder resource(304). The MGW(140) determines whether to bundle assigned frame offset information, eCIC number, and voice traffic of a corresponding user or to multiplex the information with other various users, and sends an eCIC value in accordance with the determined result to the MSCe(131)(305). The MSCe(131) transmits an assignment request message for commanding a radio channel setup to the BSC(111)(306). The BSC(111) sets up a traffic channel, and transmits an assignment complete message to the MSCe(131)(308). The MGW(140) transmits a ring back tone to the MS(105)(309).

Description

METHOOD FOR MULTPLEXING OF USER TRAFFIC IN A MOBILE COMMUNICATION SYSTEM}

The present invention relates to a multiplexing method of data in a mobile communication system, and more particularly, to a multiplexing method of traffic data provided in a mobile communication system.

In general, a mobile communication system includes a base station (BTS), a base station controller (BSC) and a mobile switching center (MSC) for transmission of voice signals. In the mobile communication system, the voice signal transmission method up to now takes the form of being transmitted in a circuit manner. That is, a channel allocated to one terminal is configured to perform communication only when a fixed channel occupies one base station, a base station control period, and a base station controller and a mobile switching period.

In general, voice signals transmitted over a wire are transmitted at a rate of 64 Kbps on a TDM-based transmission path. Therefore, in the base station, base station control period, and base station controller and mobile switching period, TDM-based transmission paths are generally used for transmission of voice signals. In this case, since transmission is performed over the wire, transmission is possible only by assigning a specific logical channel on the TDM. Do. As such, if a fixed channel is set for only one user in a TDM logical channel transmitted over a wire, voice signals of other users cannot be transmitted through the channel. In addition, the channel for transmitting the voice signal is a fixed channel used exclusively after being allocated to a user who cannot transmit other packet data. Therefore, there is a problem in that a specific channel on the wire must be allocated only for one user until the communication of the corresponding user is terminated.

As such, when one channel is fixedly used for transmission of voice signals, bandwidth is wasted. Therefore, as a solution to solve this problem in the existing mobile communication system, two major considerations are considered in the next generation mobile communication system. One does not move the transcoder located at the base station controller to the switch or media gateway, but transmits the data frame generated by the wireless vocoder, rather than transferring the voice data to the 64 kbps PCM between the base station controller and the switch or media gateway. The other is to transmit a radio vocoder frame as a packet using a packet-based transmission path such as ATM or IP, not a circuit-based TDM, between a base station controller and an exchange or a media gateway. Through this, the bandwidth can be efficiently used by transmitting the voice signal in the form of a packet. Of course, when providing a voice signal in the form of a packet, it may be difficult to provide a real-time service due to the delay of time. The solution to this problem can be solved by providing QOS (Quality of Service). However, even when a media gateway having a transcoder in a next-generation mobile communication system and a base station are connected through a TDM circuit based on a control period, the base station controller does not transmit voice data to the media gateway by 64 kbps PCM. Since it can be transmitted in the form of vocoder frame, the efficiency of band can be achieved by multiplexing voice data frames of multiple users or Bundling multiple voice data of one user even on a TDM-based circuit transmission path. The present invention is to design a way to achieve this object.

Accordingly, an object of the present invention is to multiplex voice data frames of multiple users between a media gateway having a transcoder in a mobile communication system and a media gateway in a base station control period. In other words, the present invention provides a method for bundling multiple voice data of a user.

A method of the present invention for achieving the above object, the switching agent (MSCe) for performing the switching operation of the voice call and a media gateway (MGW) connected to the switching agent, having a voice modulator / demodulator, It is connected to a switching agent and a media gateway on a packet-based and circuit-based transmission path, and can communicate with mobile stations on a wireless channel, and the voice signal is transmitted on a circuit-based transmission path through a media gateway. A multiplexing method of voice traffic in a mobile communication system including a base station system (BSS) capable of transmitting / receiving a wireless vocoder frame and transmitting / receiving packet data on a packet-based transmission path, the base station system being mobile When a call is requested from the terminal, whether a voice signal can be bundled or multiplexed together with the call request signal Transmitting the request to the media gateway to the media gateway when the call request signal is received, and whether the bundling or multiplexing of the voice signal is possible; Preparing the voice modulator / demodulator, determining whether to bundle or multiplex the voice signal, generating a response signal for a preparation request, and transmitting the response signal to the switching agent; Requesting call allocation; and setting up a traffic channel to the mobile station requesting the call, and providing a voice call service between the media gateway and the mobile station.

A method of the present invention for achieving the above object, the switching agent (MSCe) for performing the switching operation of the voice call and a media gateway (MGW) connected to the switching agent, having a voice modulator / demodulator, It is connected to a switching agent and a media gateway on a packet-based and circuit-based transmission path, and can communicate with mobile stations on a wireless channel, and the voice signal is transmitted on a circuit-based transmission path through a media gateway. A multiplexing method of voice traffic in a mobile communication system including a base station system (BSS) capable of transmitting / receiving a wireless vocoder frame and transmitting / receiving packet data on a packet-based transmission path, the base station system Requesting handoff to the exchange agent when the mobile terminal being handed needs a handoff; The hanger requests the media gateway to establish a bearer path, and the media gateway prepares to establish a bearer path, and the voice gateway includes voice multiplexing or bundling information in a response signal. Transmitting to the agent; and the switching agent transmitting a handoff request signal including the voice multiplexing or bundling information to the target base station system of the handoff; and the target base station system transmitting the handoff request signal. Assigning a forward traffic channel to a mobile terminal to perform handoff in response to the request, generating a response signal for the handoff request signal, and transmitting the response signal to the switching agent; Sending handoff command message to the system Or the bundle comprises the step of performing the voice call hand-off are transmitted / received.

1 is a diagram illustrating a network reference model between RAN-CNs for an IP-based next generation CDMA2000 1x legacy MS Domain (LMSD) system architecture considered in the present invention;

2 is a diagram illustrating an example of a protocol stack that may be defined in an interface between BSC and MGW, BSC, and MSCe of FIG. 1;

3 is a signal flow diagram illustrating an example of a call processing procedure of eCIC resource allocation in call origination to which the present invention is applied;

4 is a signal flow diagram illustrating an example of a call processing procedure of eCIC resource allocation in a Hard HO to which the present invention is applied;

5 is a control flowchart of eCIC resource allocation in MGW according to a preferred embodiment of the present invention.

Hereinafter, exemplary embodiments of the present invention will be described in detail with reference to the accompanying drawings. First of all, in adding reference numerals to the components of each drawing, it should be noted that the same reference numerals have the same reference numerals as much as possible even if displayed on different drawings.

In addition, in the following description, there are many specific details such as specific messages or signals, which are provided to aid the overall understanding of the present invention, and it is understood that the present invention may be practiced without these specific details. It will be self-evident to those of ordinary knowledge. In the following description of the present invention, if it is determined that a detailed description of a related known function or configuration may unnecessarily obscure the subject matter of the present invention, the detailed description thereof will be omitted.

FIG. 1 is a diagram illustrating a network reference model between a Radio Access Network (RAN) and a Core Network (CN) of an IP-based next generation CDMA2000 1x legacy MS Domain (LMSD) system architecture considered in the present invention. .

1 illustrates a base station 101, a base station controller 111, and a mobile switch 120 according to the related art. The conventional base station controller includes an SDU function 111b and a voice modulator / demodulator (Xcoder or VOCODER) 111a. In addition, signaling and data can be transmitted and received through the A1, A2, A5 interface with the mobile switch 120. However, the existing base station controller 111 is referred to as an exchange agent (MSCe: MSC emulator-"MSCe") 131 and a media gateway (MGW: "MGW") according to the present invention ( 140 may be required when the interface exists. The main reason is to be able to provide an existing service by accessing a new switching agent 131 or a media gateway 140 on an existing TDM-based circuit transmission line without any change in the existing base station controller.

The base station controller (BSC) 113 included in the present invention does not include a voice modulator / demodulator (Xcoder or VOCODER). The mobile exchange according to the present invention is composed of an exchange agent 131 and a media gateway 140. And the voice modulator / demodulator 141 configured not to be included in the base station controller 113 in the MGW 140. Through this, the MGW 141 according to the present invention and the base station controller 113 according to the present invention are configured to transmit voice signals as packet data. The base station controller 113 may receive another packet data service through the PCF 160, and the PCF 160 is connected to the PDSN 170.

The base station controller 113 according to the present invention cannot communicate with the conventional mobile switch 120, and also the base station controller 111 according to the prior art is MSCe / MGC (Media Gateway Control) 131 according to the present invention. Cannot communicate with). Next, the interface of the above-described components will be described.

The signal corresponding to the existing A1 interface between the base station controller (BSC) 113 and the MSCe 131 uses an 'A1p 48' interface. In addition, an 'A2p 27' interface corresponding to the existing A2 interface exists between the BSC 113 and the MGW 140. Also, optionally, 'Amp' may be considered for out-of-band signaling for bearer setup and maintenance between BSC and MGW. Functions defined in this 'Amp' interface may be provided in-band signaling at the FP of A2p. It should be noted that these 'A1p', 'A2p', and 'Amp' interfaces are considered mainly based on Packet (ATM or IP) rather than existing circuit based. Of course, it is possible to further consider the interface circuit based protocols so that A1p or A2p can also operate on the circuit basis.

An existing A1 interface may be used between the existing base station controller 111 and the switching agent 131 according to the present invention, or the circuit-based protocol may be further considered in the A1p interface. The base station controller 111 and the media gateway 140 may introduce a new interface instead of the existing A2 interface or may add a circuit-based protocol to the A2p.

FIG. 2 is a diagram illustrating an example of a protocol stack that may be defined in an interface between the BSC and the MGW, and the BSC and the MSCe of FIG. To date, since there is no standard for this part, the protocol stack shows what is considered in the present invention. In case of using Packet Transport, Case 1 is mainly considered among protocol stacks. When using TDM transport, consider Case 3, the same protocol stack as the existing IOS A1. The protocol stack in case of using the TDM transmission path is added to the packet-based protocol A1p. In the A2p Protocol Stack, Case 1 and 2 are considered when using Packet Transport. Case 1 is mainly considered. Among them, RTP (Real time Transport Protocol) * and GRE (Generic Route Encapsulation) * mean that the functions of existing RTP and GRE can be changed slightly, mainly for multiplexing multiple users on one port. When using TDM transport, consider Case 3, a protocol stack that is different from the existing IOS A2, and consider the new Framing Protocol, also known as FPoT, defined on TDM as in the case of Packet Transport.

'Amp' is an interface for out-of-band signaling in the control procedure provided by the frame protocol, and constitutes a separate interface (hereinafter, referred to as Amp). The protocol stack in the case where the Amp interface is matched to the media gateway 140 via the exchange agent 131 uses the Session Control Transmission Protocol (SCTP).

Next, a description will be given of an efficient eCIC (expanded circuit identity code) allocation processing apparatus of the present invention. 3 is a signal flow diagram illustrating an example of a call processing procedure of eCIC resource allocation at the call origination to which the present invention is applied. In FIG. 3, the Frame Protocol used in A2p uses the contents included in the previously filed patent.

When the voice call is needed, the MS 105 sends an origination to the BSC 111 in step 301 to request the voice call. Then, the BSC 111 sends a BS Ack Order in step 302 to inform the MS whether to receive the Origination message. Thereafter, the BSC 111 transmits a CM Service Request message to the MSCe 131 through the A1p interface in step 303. At this time, the BSC 111 inserts a Preferred Frame Offset and an eCIC (expanded Circuit Identity Code) value preferred by the BSC 111 into the CM Service Request and transmits it. In addition, the BSC 111 may request whether to bundle user traffic and how many frames to bundle simultaneously if bundled, and may also indicate whether to multiplex with voice traffic of different users.

Then, the MSCe 131 receiving the CM Service Request requests the MGW 140 to set the Vocoder resource in step 304. At this time, the Preferred Frame-offset value and the eCIC value received from the BSC 111 are inserted into the MEGACO (MEedia GAteway COntrol) ADD message in the form of Session Description Protocol (SDP) Information and transmitted. Then, the MGW 140, which receives the MEGACO ADD from the MSCe 131, determines the Preferred Frame-offset and the eCIC request requested by the BSC 111, and finally assigns the assigned frame offset information and the eCIC number determined by the Vocoder stage. It decides whether to bundle the user's voice traffic or multiplex with other users. The value determined as described above is assigned to the eCIC value, and the SDP information is inserted into the MEGACO Reply in step 305 and transmitted to the MSCe 131. Thereafter, the MGW 140 prepares the necessary Vocoder resources.

Thereafter, the MSCe 140 transmits an Assignment Request message to the BSC 111 instructing wireless channel setting, including the eCIC value received from the MGW 140 and the Assigned Frame Offset value determined in the MGW 140 in step 306. The BSC 111 then establishes a Traffic Channel between the MS 105 and the BSC 111 in the wireless channel. After the wireless TCH is configured as described above, the BSC 111 transmits an Assignment Complete message to the MSCe 131 in step 308 to inform that the channel setup is completed. At this time, the BSC 111 adds a Bearer ID for voice data transmission. Then, the ring back tone is transmitted from the MGW 140 to the corresponding MS 105.

3 described above illustrates an example of allocating an eCIC resource applied when a call is sent from the MS, and is commonly applied to a case where a voice call is received by the MS. In this case, the eCIC is inserted into the Page Response message from the BSC to the MSCe and transmitted, and the transmission from the MSCe to the MGW through the MEGACO is the same as the MS voice call origination. In the case of voice call reception, only the application example is different, and since the actual operation is processed in the same way as the voice call origination, the description of the call scenario is not described in the present invention.

4 is a signal flow diagram illustrating an example of a call processing procedure of eCIC resource allocation in a Hard HO to which the present invention is applied. Next, a process of signal flow according to an example of a call processing procedure of eCIC resource allocation in Hard HO to which the present invention is applied will be described in detail with reference to FIG. 4. In addition, in the following description, the source base station (S-BS) and the target base station (T-BS) are configured with a BSC and a BTS of the same type as the BSC 111 in FIG.

When the signal strength reported from the UE exceeds the signal strength defined in the network, the S-BS 180 performs hard handoff to one or more cells below the T-BS 190 in step 401. Recommended to do. At this time, the S-BS 180 constructs a Handoff Required message together with a list of corresponding cells and transmits the message to the MSCe 131. Upon receiving the HO Required, the MSCe 131 sends an ADD message of MEGACO to the MGW 140 in order to request an OCW allocation corresponding to a Vocoder resource and a bearer path from the T-BS 190 to the MGW 140. send. Then, the MGW 140, which receives the MEGACO ADD Request from the MSCe 131, specifies an eCIC value for setting a bearer with the T-BS 190 in step 403, and assigns an eCIC value to a frame offset to be used by the current MS 105. MEGACO Reply is sent with the value for.

Since the MSCe 131 indicates a hard handoff because the hard handoff bit in the Handoff Required message received from the S-BS 180 is set to 1, the MSCe 131 indicates the current TIA / EIA to the T-BS 190 in step 404. -95 Handoff Request message including Channel Identity element, eCIC, and Frame Offset value determined by MGW is transmitted. Then, upon receiving the Handoff Request message from the MSCe 131, the T-BS 190 allocates the appropriate radio resource described in the message and connects the call. The T-BS 190 transmits a Null forward traffic channel frame to the terminal.

As described above, the T-BS 190 transmitting the Null forward traffic channel frame to the UE sends a Handoff Request Acknowledge message to the MSCe 131 by inserting the eCIC in step 406. The MSCe 131 prepares to switch from the S-BS 180 to the T-BS 190, and sends a HandoffCommand message to the S-BS 180 in step 407. In operation 408, the S-BS 180 transmits a handoff direction message, which is one of a general handoff direction message, an extended handoff direction message, and a universal handoff direction message, to the MS 105. Accordingly, in step 409, the MS 105 sends an MS Ack Order to the S-BS 180 as an acknowledgment of receipt of the Handoff Direction message.

Thereafter, the S-BS 180 transmits a Handoff Commenced message to the MSCe 131 in step 410 to inform that the terminal is ready to move to the channel of the T-BS 190. Accordingly, the MS 105 sends a reverse traffic channel frame or a traffic channel preamble to the T-BS 190 in step 411. Thereafter, the MS 105 transmits a handoff completion message to the T-BS 190 in step 412. The T-BS 190 then wirelessly sends a BS Ack Order to the MS 105. In operation 414, the T-BS 190 sends a handoff complete message to the MSCe 131 indicating that the MS 105 has successfully hard handed off. Accordingly, the MSCe 131 transmits a clear command to the S-BS 180 in step 415. In operation 416, the S-BS 180 transmits a Clear Complete message to the MSCe 131 indicating that the release is successfully completed.

5 is a control flowchart of eCIC resource allocation in MGW according to an embodiment of the present invention. Hereinafter, a control process for allocating an eCIC resource according to the present invention will be described in detail with reference to FIG. 5.

The MGW 140 waits for the reception of the MEGACO message including the preferred eCIC that the BSC prefers for voice processing of the terminal. If the MEGACO message is received in this standby state (step 500), the MGW 140 proceeds to step 502. In this case, the BSC may transmit whether bundling or multiplexing with voice traffic of another user is possible when the voice traffic of the corresponding user is transmitted through the TDM. When the MGW 140 receives the MEGACO message, in step 502, the MGW 140 checks whether bundling is requested by the BSC and whether multiplexing with other user voice traffics is possible. At this time, only one of Bundling or Multiplexing is selected, and there is no Bundling as default and no multiplexing. One eCIC is used exclusively by one user. If bundling is possible in step 502, the process proceeds to step 503; otherwise, the process proceeds to step 504.

First, the case proceeds to step 503. In step 504, the MGW 140 determines the maximum number of bundling frames available from the BSC, and prepares an internal process and a buffer to process bundling. In addition, the Vocoder resource is reserved and prepared to transmit voice data to the reserved Vocoder when voice is transmitted through the voice transmission bearer path of the corresponding terminal. Thereafter, the MGW 140 notifies the BSC of the MEGACO message response via the MSCe 131 for the allocated Bundling frame maximum number.

On the other hand, when proceeding to step 504, that is, when performing multiplexing MGW 140 determines whether the multiplexing from the BSC. At this time, after confirming the frame-offset of the user, an eCIC value of another user using the same frame-offset under the same BSC and the corresponding Multiplexing ID are allocated. When multiplexing with other users who do not use the same frame-offset, the MGW prepares a buffer of a certain size (ex, 20ms) and assigns an eCIC value and the corresponding Multiplexing ID. In order to process multiplexing, an initialization of an internal process and a frame protocol is prepared, a Vocoder resource is reserved, and a voice is transmitted to a reserved Vocoder when the voice is delivered to a voice transmission bearer path of a corresponding terminal. Thereafter, the MGW 140 notifies the BSC of the MEGACO message response through the MSCe whether the requested multiplexing is allowed and the corresponding Multiplexing ID.

Through this process, multiplexing of voice traffic can be performed in the MGW.

Next, we look at the message used for eCIC allocation and the information elements of the message. The messages to be examined below are mainly used for A1 or A1p, and the parts of A2 and A2p to which actual voice data is delivered are handled in FPoT.

<Table 1> and <Table 2> are information elements indicating newly required eCIC and eCICE (expanded CIC Extension). The other fields except BUNDLE_Indicator, Number of Bundling Frames, MUC_Indicator, and MUX_ID, which are fields added to existing CIC and CICE, have the same configuration as existing CIC and CICE.

The fields for Table 1 and Table 2 are defined as follows.

(1) Length: Specifies the value for the length of the fields after Length.

(2) CIC (Circuit Identity Code): 2 bytes indicating TDM transmission path where user's voice traffic is transmitted between BSC and MGW. This field is interpreted differently in each case of the call processing scenario. For example, if it is set in BSC and delivered to MSCe (eg, voice call / outgoing), it means the preferred value (eg, currently idle CIC value) in BSC. When delivered from the MSCe to the BSC, it means the CIC value finally allocated by the MGW or already provisioned to the MSCe, and thus finally allocated by the MSCe. The value is specified in CIC (decimal). The format of this value is the same as in the existing Inter-Operability Specification (IOS) 4.3. For example, it is as shown in Table 3 below.

In Table 3, the a-k bits mean PCM multiplexer values and the xxxxx bits mean timeslots.

(3) BUNDLE_Indicator: Set as '1' if bundling is possible and '0' if not. The default value is '0'.

(4) Number of Bundling frames: When Bundling_Indicator is designated as '1', it means the maximum number of frames that can be bundled and interpreted as decimal. If Bundling_Indicaor is set to '0', it is ignored.

(5) MUX_Indicator: Set as '1' when multiplexing is available, and set as '0' when not. The default value is '0'.

(6) MUX_ID: ID value for distinguishing voice traffic of the user from the eCIC.

(7) Circuit Mode: Circuit Mode in eCICE is a value that indicates full rate transmission when using circuit data rather than voice data. Use the same values as in IOSv4.3.

The information elements shown in Tables 1 and 2 are inserted in the following messages used during the following call processing scenarios. Each inserted is shown in the following <Table 4> to <Table 9>. <Table 4> shown below shows how eCIC is inserted into an existing CM Service Request message, and <Table 5> shows how eCIC is inserted into an existing Page Response message. > Indicates how eCIC is inserted into an existing Assignment Request message. <Table 7> shows how the eCIC is inserted into the existing Handoff Request message, and <Table 8> shows how the eCIC is inserted into the existing Additional Service Request message. It shows how eCIC is inserted to be added to existing MEGACO message.

The contents shown in Tables 4 to 9 can be summarized as follows.

(1) Call origination: CM Service Request, Assignment Request

(2) Call incoming: Page Response, Assignment Request

(3) Hard HO: Handoff Request (eCICE)

(4) Concurrent Service Voice Call / Send: Additional Service Request, Assignment Request

As described above, when transmitting voice data of a user's Selectable Mode Vocoder (SMV) or Enhanced Variable-Rate Codec (EVRC) vocoder between BSC and MGW based on the existing TDM transmission path, for efficient use of the TDM transmission band By bundling multiple frames of voice data of one user or multiplexing frames of multiple users, data can be efficiently transmitted.

Claims (2)

A switching agent (MSCe) for performing a voice call switching operation and a media gateway (MGW) connected to the switching agent and having a voice modulator / demodulator, and a packet-based and circuit-based communication with the switching agent and the media gateway. It is connected on a transmission path and can communicate with mobile stations in a wireless channel, and the voice signal can transmit / receive a wireless vocoder frame of a mobile terminal on a circuit-based transmission path through a media gateway and is packet-based. A multiplexing method of voice traffic in a mobile communication system including a base station system (BSS) capable of transmitting / receiving packet data on a transmission path of When the call is requested from the mobile terminal, transmitting the request signal of the call to the switching agent whether a bundling or multiplexing of a voice signal is possible; The switching agent, when receiving a request signal of a call, transmitting the preparation request of the voice modulator / demodulator and whether the voice signal can be bundled or multiplexed to the media gateway; The media gateway prepares the voice modulator / demodulator, determines whether the voice signal can be bundled or multiplexed, and generates a response signal for the preparation request and delivers the response signal to the switching agent; The switching agent requesting allocation of the requested call to the base station system; The base station system comprises the step of establishing a traffic channel to the mobile terminal requesting the call, and providing a voice call service between the media gateway and the mobile terminal. A switching agent (MSCe) for performing a voice call switching operation and a media gateway (MGW) connected to the switching agent and having a voice modulator / demodulator, and a packet-based and circuit-based communication with the switching agent and the media gateway. It is connected on a transmission path and can communicate with mobile stations in a wireless channel, and the voice signal can transmit / receive a wireless vocoder frame of a mobile terminal on a circuit-based transmission path through a media gateway and is packet-based. A multiplexing method of voice traffic in a mobile communication system including a base station system (BSS) capable of transmitting / receiving packet data on a transmission path of The base station system is a process for requesting handoff to the switching agent when the mobile terminal in communication needs handoff, and The switching agent requesting establishment of a bearer path to the media gateway; The media gateway prepares to establish a bearer path, and transmits to the switching agent including voice multiplexing or bundling information in a response signal thereto; The switching agent transmitting a handoff request signal including the voice multiplexing or bundling information to a base station system to which the handoff target is located; The target base station system assigns a forward traffic channel to a mobile terminal to perform handoff in response to the handoff request signal, generates a response signal to the handoff request signal, and transmits the response signal to the switching agent; The switching agent includes a step of transmitting a handoff command message to the base station system requesting the handoff to perform a handoff of a mobile call multiplexed or bundled and transmitted / received. .