KR20040012132A - Apparatus and method for transmitting voice data of group call in 1x ev-do system - Google Patents

Abstract

PURPOSE: A method of transmitting voice data for a conference call service in a mobile communication system is provided to combine forward data with reverse data, and to transmit the combined data, thereby reducing an overhead for voice packet transmission and improving bands and capacities in a wireless section. CONSTITUTION: A calling AT1 transmits an invite message of a session setup protocol to a conference call server(200). The conference call server transmits an invite request message to a DLR(Data Location Register) server(202). The DLR server transmits an invite response message to the conference call server(204). The conference call server transmits a new session setup message to each terminal according to addresses of members received from the DLR server(206). Each terminal transmits a permission message to the conference call server(208). If one of the terminals responds to a conference call, the conference call server transmits a conference call confirmation message to the calling AT1. An RTP(Real time Transport Protocol) communication channel is set(216).

Description

Apparatus and method for transmitting voice data for conference call service in mobile communication system {APPARATUS AND METHOD FOR TRANSMITTING VOICE DATA OF GROUP CALL IN 1X EV-DO SYSTEM}

The present invention relates to an apparatus and a method for a one-way conference call in a mobile communication system, and more particularly, to an apparatus and method for transmitting data in a wireless environment having different rates of forward and reverse in a mobile communication system.

In general, a mobile communication system has been developed to perform a voice call while ensuring user mobility. The mobile communication system focused on such voice calls has been developed to meet the needs of users, and has reached a level capable of performing short data services in parallel. That is, the mobile communication system basically provides a voice service, and as a data service, a short message transmission, a music file that is a ringtone of a terminal, a simple internet communication service, and a financial settlement service are provided.

As described above, in the mobile communication system, a method for providing more information to mobile subscribers at high speed is being discussed while the above data service is provided, and preparation work for commercialization is currently in progress. The method discussed as a method of providing such a high-speed data service is named "CDMA2000 1x EV-DO" (hereinafter referred to as 1x EV-DO) and is discussed as a system for transmitting high-speed data to a mobile communication terminal. It became. Accordingly, the 1x EV-DO system has different channels from the existing system, and thus, data channels for transmitting data at high speed, channels such as additional channels, signal channels, and MAC channels are defined. The structure of the channels and the various data specifications that can be transmitted on them have been discussed.

In addition, the high speed data service mobile communication system is basically a communication system mainly for data communication. However, as described above, the first purpose of the mobile communication system is to provide a voice communication service. Therefore, voice services are desired to be parallel with high-speed data services. Accordingly, a method of providing a voice service in a high speed data service mobile communication system has been disclosed. A method for this will be described with reference to FIG. 1.

In the high speed data service mobile communication method, it is a data communication method. In order to transmit voice, data-based voice transmission method (VoIP: Voice Over IP, hereinafter referred to as "VoIP") method is used. Operate Call service.

Here, VoIP (Voice over IP) based Group Call service is related to technologies such as voice encoding technology, compression, transmission and recovery. Voice coding technology is currently developing a variety of technologies including PCM (Pulse Code Modulation). Real time transport protocol (RTP), a standard packet format for real-time data transmission, is suitable for transmitting real-time data such as audio and video, and RTCP (Real Time Transport Control Protocol, real-time). Transmission Control Protocol (hereinafter referred to as "RTCP") is an RTP control protocol that periodically transmits control packets to recipients in sessions and performs Quality of Service (QoS) monitoring. RLP (Radio Link Protocol, hereinafter referred to as "RLP") is used as a protocol to handle when an error occurs in the radio section. RLP improves channel usage efficiency by using NAK based protocol which requires retransmission only for errored RLP frame.

In CDMA2000 1X EV-DO system, Push to Talk service can be used for VoIP-based conference call, and the conference call service is a communication method using half duplex mode. There are two or more listeners, and members of a predetermined group press the conference call button of the terminal to perform the conference call while receiving transmission authority.

1 is a diagram showing the overall structure of a mobile communication system for a one-way conference call. Hereinafter, the configuration and connection of the entire network of the mobile communication system for the conference call service and the operation of each node will be described with reference to FIG. 1. Access terminals 11, 12, and 13 of the mobile communication system for high-speed data transmission are connected to base station (ANTS: Access Network Transceiver Systems) 10a through a wireless channel discussed in the 1x EV-DO standard. , ..., 10n, 11a, ..., 11m) to perform high-speed data communication. Each of the base stations 10a, ..., 10n, 11a, ..., 11m performs data communication with the terminals 11 and 12 through the wireless channel, and access network controllers (ANCs) by wire. (20a, ..., 20p) is connected to the terminal (11, 12) to perform the processing and authentication of data transmitted and received, and the resulting signal processing. In the following description, the base stations 10a, ..., 10n, 11a, ..., 11m and the base station controllers 20a, ..., 20p are collectively called an access network (AN). The access network is wired to the general ATM switch network (GAN) 100 to perform data communication through a packet data serving node (PDSN) 60.

The gan 100 is provided with a packet control function block (PCF: not shown in Figure 1) therein to control the packet data transmitted to the terminals (11, 12, 13), Data received from the terminals 11 and 12 are packetized and delivered to the packet data service node 60. As a result, in the wired line, the subsequent stage of the gan 100 communicates with data in a packet unit regardless of the previous stage.

The gan 100 may include a base station manager (BSM) 30, an access network authentication system (AAA) 40, and a data location register (DLR) of a data terminal. 50). The BSM 30 performs functions such as hardware and software state management, monitoring of base station operation, and collecting statistical data of several base station controllers 20a, ..., 20p. In addition, the access network authentication system 40 performs a process for authentication of the terminals 11, 12, 13. In addition, the location register 50 of the data terminal, which performs a role similar to the home location register in the 1x system that mainly serves voice, provides this when the location information of the data terminal is requested from the gan 100. The location register 50 of the data terminal is also called a session control and mobility management block (SC / MM: Session Controller / Mobility Manager). The location register 50 of the data terminal may be provided outside the gan 100 as shown in FIG. 1, alternatively within the packet control function block provided in the gan 100. It can also be configured to include.

The packet data service node 60 performs authentication calculation for authentication of the packet data service of the mobile data terminal through a home agent (HA) (not shown in FIG. 1) that manages data of the mobile data terminal. It is connected to the system (AAA: Authentication, Authorization and Accounting).

The conference call server 110 edits, registers, sets up sessions, and reassembles packets for a conference call-enabled group for a conference call service and becomes a subject of a conference call. After the conference call server receives the group call request, the conference call server establishes a session with the members of the corresponding group, and then routes the RTP packet to the group members.

For 1X EV-DO, the forward speed is up to 2.457Mbps and the reverse is up to 153.6Kbps. In addition, the amount of data generated by the 8.5bit rate EVRC vocoder of the terminal is 171 bits / frame for 20ms. The voice data (Voice data-171 bits) becomes an RTP payload, and the RTP / UDP / IP / PPP header is attached thereto, and the size of the voip packet is about 68 bytes. Then, it is transmitted in frame unit of 26.67ms size in reverse direction in air link, and it is transmitted at one of five transmission speeds (9.6, 19.2, 38.4, 76.8, 153.6Kbps) according to the situation. In the forward direction, the conference call server 110 retransmits the VoIP packet received from the transmitting terminal to the members of the group. The VoIP packet is delivered to the terminal in the RLP frame in the radio section and the RLP payload is up to 122 bytes. Depending on the forward speed, 1,2,4,8 or 16 slots are used and 1.66 msec is allocated per slot. When the RLP frame is forward transmitted in the radio section, the transmission ranges from 2,457 Kbps to 38.4 Kbps. When the transmission rate is 1.228 Mbps, two MAC layer packets are included in one physical layer packet. In this case, the total RLP payload length included in two MAC layer packets is 122 bytes * 2. In addition to the pure RLP payload (122 bytes), the MAC layer packet includes an RLP sequence (22 bits), a stream header (2 bits), and a MAC tail (2 bits). This is illustrated in IS-856. See 9.1.2.3-1 for more details. As described above, when the voice data received in the mobile communication system for high-speed data transmission is transmitted as it is, that is, when the conference call server 110 transmits the received RTP packets in a forward direction without merging, a wireless section is provided. In this case, only one VoIP packet (about 68 bytes) is carried in one physical layer packet. Therefore, the utilization rate of the physical layer of the system developed for the high-speed data service is 68 * 100 / (122 * 2) is about 28%, there is a problem that the waste of bandwidth is large.

Accordingly, an object of the present invention is to provide an apparatus and method for increasing information transmission efficiency in a mobile communication system for high speed data transmission.

Another object of the present invention is to provide an apparatus and method for preventing waste of bandwidth and transmitting voice data in a mobile communication system for high speed data transmission.

Still another object of the present invention is to provide an apparatus and method for efficiently transmitting voice data without changing the mobile communication system for high speed data transmission.

The method of the present invention for achieving the above object is provided with a session control and mobility management block (SC / MM: Session Controller / Mobility Manager) for managing the location and authentication information of the terminal in a mobile communication system for high-speed data transmission It provides a method for each terminal having an indicator including the number of packets that can be received in a session establishment request message and a response message transmitted and received when a conference call service request.

In addition, the conference call server may further include a step of establishing a data channel by performing connection establishment and session negotiation when a connection setting is requested from the terminal, and releasing a data channel established with the terminal when data transmission is completed. Perform further connection termination process.

In addition, the conference call server stores the number of receivable packets included in the session establishment message in a memory and adds packets correspondingly to the state of each terminal.

The server for the conference call transmits an update request message for the number of packets that can be transmitted to the session control and mobility management block when the terminal needs to update the information, and a response message notifying update success after updating the data. It may be configured to include the step of transmitting to the server for the conference call.

In addition, each terminal may be configured to include a process of displaying the number of packets that can be received in the content of an indicator defined at session establishment or change, and transmitting the received packet to the server.

1 is a diagram showing the overall structure of a mobile communication system for high-speed data transmission;

2 is a signal flow diagram for session establishment and data transmission for a conference call service according to the present invention in a mobile communication system for high-speed data transmission;

3 is a block diagram of a conference call server according to the present invention in a mobile communication system for high-speed data transmission;

4 is a flowchart illustrating session establishment and data transmission of a calling terminal according to the present invention in a mobile communication system for high speed data transmission;

5 is a flowchart illustrating session establishment and data transmission of a called terminal according to the present invention in a mobile communication system for high speed data transmission;

6 is a flowchart illustrating the operation of a conference call server according to the present invention in a mobile communication system for high-speed data transmission;

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, many specific matters such as specific messages and the like appear, which are provided to help a more general understanding of the present invention, and the present invention may be practiced without these specific matters. It is self-evident to those who have 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.

In the following description, 1x EV-DO (HRPD: High Rate Packet Data) system is referred to as a mobile communication system for high speed data transmission. In addition, in the following description, the session refers to all information used by the terminal to create and maintain a call in the radio section and the system. In addition, the network configuration and each component of the mobile communication system for high-speed data transmission according to the present invention is the same as shown in FIG. However, according to the present invention, the operation processed by the conference call server 110 is changed, and there is only a change in the operation of the mobile terminals 11, 12, 13.

The conference call server 110 may know information on the number of RTP packets that can be carried in a forward physical layer packet through a session establishment message (hereinafter referred to as a SIP message). The SIP message is divided into three types. The first part is called the Start line, which always starts and ends with the first line, and is divided into the header and the session description message from the next line. An example of using the SIP message may be shown in Table 1 below.

division Contents Start line INVITE sip: group@Samsung.com SIP / 2.0 Headers Via: SIP / 2.0 / UDP 165.213.120.101:5060To: sip: group@samsung.comFrom: Bob <sip: john@Samsung.com>; tag = 390605453Call-ID: 330277362978031178@165.213.120.101CSeq: 1 INVITEAccept: application / sdpContact: sip: timna@165.213.120.101Content-Type: application / sdpContent-Length: 142 blank SDP Massage v = 0o =-994664497110 994664497110 IN IP4 165.213.120.101s = -c = IN IP4 165.213.120.101t = 0 0m = audio 5004 RTP / AVP 8 3 0

When the conference call server 110 analyzes the message, the criterion for determining the end of the start line is a carriage return / line feed (CRLF). Therefore, when the first CRLF is received, the message analyzer (not shown) determines that the start line is over and treats the following information as information about the header. In order to be flexible, the header information is also judged to be the end of the header information when CRLF comes out twice. The analyzer considers the following information as an SDP message and analyzes it. Start Line is divided into method, Request-URI, and version. The method indicates what the current message is for, and the method has a request and a response. In case of request, there are INVITE, ACK, REGISTER, BYE, CANCEL, OPTION methods. In case of response, it responds to SIP request with status code (number format from 1xx to 6xx) and reason phrases. In addition, detailed functions can be shown using extension methods. If you want other functions, it is also possible for the provider to implement the method separately.

SIP header is classified into general header, request header, respond header and entity header. Header field is similar to HTTP header field. The header fields are used to transmit various information necessary for transmission and reception.

When adding a new header field for various applications, the defined header format must be followed, and other developers can understand the purpose of the field and decide whether to support it. Therefore, in order to know the number of packets that can be transmitted in the header of the SIP message, a new field called "Packet_num" is added so that the terminal receives the maximum number of RTP packets that can be simultaneously received in the allocated bandwidth to the conference call server. Let them know The maximum number of RTP packets can be calculated using a specific algorithm based on the DRC value in the terminal or the base station. The simplest method is as follows.

The UE accurately measures carrier to interference (C / I) and requests the DRC value of the forward data rate to the base station. For example, if the terminal requires a maximum DRC value of 12, that is, a transmission speed of 2.457 Mbps, four RLP packets are loaded in one physical layer packet, and the length of the RLP payload is 122 bytes * 4. Therefore, the number of RTP packets to be merged in the conference call server 110 is calculated as in Equation 1 below.

2 is a signal flowchart for a conference call according to the present invention in a mobile communication system for high-speed data transmission. Hereinafter, a signal flow and configuration of the signals during session establishment and voice data transmission in the mobile communication system for high speed data transmission will be described in detail with reference to FIG. 2.

The originator AT1 sends an INVITE message of the session establishment protocol including the number of RTP packets to the conference call server in step 200. If the configuration of the session establishment protocol can be illustrated as shown in Table 2 below.

INVITE sip: ptt_group2@server.com SIP / 2.0From: Bob <sip: bob@samsung.com>; tag = 390605453To: sip: ptt_group2@server.comCall-ID: 330277362978031178CSeq: 1 INVITE Packet_num: 4 Content-Type: application / sdpContent-Length: 142

Upon receiving the message as shown in Table 2, the conference call server transmits the INVITE query message to the DLR server in step 202. The INVITE query message is information for obtaining location information of a corresponding terminal. Here, the DLR server may have an address list of group members as a preconfigured database. Thus, the conference server can obtain information about the conference call members from the DLR server. In this case, the address of the group members may be configured to be previously held in the DLR server as described above, and the user may be configured to change and select the address according to the outgoing destination addresses, and transmit the same to the DLR server. Accordingly, when the list of recipients is received from the DLR server, the user selected to be excluded is excluded and the user selected to be added may be added. Alternatively, the DLR server may be configured not to have the above address. In this case, the conference call is attempted only by the terminals selected by the user. In the following description, it is assumed that a conference call is attempted using only terminals selected from a user.

Accordingly, the DLR server forwards the INVITE response message to the conference call server in step 204. Accordingly, the conference call server transmits a new session establishment message (INVITE requests) to each terminal in step 206 according to the address of the members received from the DLR server. As described above, each terminal according to the member of the selected group transmits the permission message to the conference call server as shown in step 208. In this case, the INVITE permission message received from each terminal to the conference call server is notified to the conference call server by inputting the number of RTP packets corresponding to its own forward bandwidth capacity in the header. The configuration of the RTP grant message can be illustrated as shown in Table 3 below.

SIP / 2.0 200 OKFrom: sip: ptt_group2@server.comTo: sip: john@samsung.comessagemCall-ID: 405277562478061472CSeq: 1 INVITEReferred-By: sip: bob@samsung.com Packet_num: 3 Content-Length: 0

When the response message is received by the conference call server as shown in Table 3, the session is connected and the call is available. That is, when the terminal responds to the conference call as described above, the conference call server transmits a confirmation message (200 OK) indicating that the conference call is possible to the calling terminal AT1. However, if no one answers the conference call server, the conference call server sends an unacknowledged message ("not found") to the terminal requesting the conference call. As shown in the figure, if there is a response from the terminal, the flow proceeds to step 216 and an RTP communication channel is established. That is, one-way conference call can be made by pressing the push button between group members through the RTP communication channel.

3 is a block diagram of a conference call server for a conference call according to the present invention in a mobile communication system for high-speed data transmission. Hereinafter, a configuration of a conference call server according to the present invention in a mobile communication system for high speed data transmission will be described in detail with reference to FIG. 3. The conference call server according to the present invention is composed of three parts, the media controller 310, the call controller 320, and the network manager 330. The media controller 310 copies and stores the voice data packets received from the terminal as many as the number of terminals to be transmitted, and the merging unit 312 which merges and transmits a predetermined number of RTP packets to each terminal and the same. It is composed of a buffer 312 for buffering and transmitting. The call control unit 320 confirms and stores the location of the terminal for the conference call and the call state processing unit 322 that checks the session establishment state and the session establishment state that manages the session establishment, change, and release, and manages information. It consists of the terminal processing unit 323 and the authentication server (324) performs an authentication function in conjunction with the authentication server (AAA).

The network manager 330 is connected to a log data processing unit 331 for managing system log data, a billing processing unit 333 for collecting, storing, and processing billing information, and a remote control system for enabling remote control and state management. The control processing unit 332 is configured.

In addition, the database 340 is commonly connected to store and share data necessary for operation and configuration, and to store and load shapes and operation files required for the system.

4 is a control flowchart when calling from a terminal for a conference call of a mobile communication system for high-speed data transmission according to the present invention. Hereinafter, with reference to Figure 4 will be described in detail the control process of the terminal for the conference call in the mobile communication system for high-speed data transmission according to the present invention.

The terminal collects information about the position of the terminal when the movement of the position occurs or the power is turned on. That is, step 400 of FIG. 4 illustrates a process of obtaining location information of the terminal. This is generally to indicate that the terminal has location information. When the conference call is requested by the user in the terminal having the location information as described above, the number of receivable packets can be calculated using the DRC value received from the base station. The DRC value may be classified into a case in which it is received at the request of the terminal, a case in which calculation can be performed according to a channel condition, and a case in which the base station continuously transmits. In the present invention, in any case, it is enough that the terminal can acquire the number of packets that can be transmitted.

When the terminal acquires the number of packets that can be transmitted in step 402, the terminal proceeds to step 404 to generate a SIP INVITE message including the number of packets that can be obtained and delivers the message to the conference call server. It then waits for a response message from the conference server. In this case, the response message may be divided into a confirmation message 200 ok and a not found message as described above with reference to FIG. 2. In the present invention, since it is a description of the case where the terminal performs a conference call, a description of the non-acknowledgement message will be omitted.

The terminal receives an acknowledgment message (200 ok) when a call is made in response to any one terminal among terminals required for a conference call from the conference call server. Therefore, when receiving the response signal of the test result confirmation message 200 ok (step 406), the process proceeds to step 408 and generates and transmits a response message (Ack) to the conference call server. Then, as shown in step 216 of FIG. 2, an RTP channel for a conference call is established. That is, as shown in step 410, it is possible to send and receive RTP packets via the conference call server. In this manner, during the conference call, the terminal proceeds to step 412 to check whether the DRC is changed. If the DRC is changed as a result of the check, the flow proceeds to step 414 in which the information change message according to the DRC change is transmitted to the conference call server to change the number of received packets.

On the contrary, if the DRC change is not made as a result of the check in step 412 according to a predetermined period, the process proceeds to step 416 to check whether a session termination message is received. The session end message is a message received from the mobile communication system when the communication ends. In other words, it checks whether the communication is terminated. When the test result communication is terminated, the routine of FIG. 4 ends. However, if communication is not terminated, the process proceeds to step 410 to continue transmission and reception of the RTP packet. Through this method, the terminal may perform a voice call in a mobile communication system for high speed data transmission.

5 is a control flowchart when a call is received in a terminal for a conference call of a mobile communication system for high speed data transmission according to the present invention. Hereinafter, referring to FIG. 4, when a call is received to a terminal of a mobile communication system for high-speed data transmission according to the present invention, a control process for setting up a conference call and performing a conference call will be described in detail.

Since steps 500 and 502 of FIG. 5 correspond to steps 400 and 402 of FIG. 4, detailed descriptions thereof will be omitted. In step 504, the terminal checks whether a call for voice communication is received. If a call for voice communication is received as a result of the check, the process proceeds to step 506 to generate and transmit a response message to the conference call server. In this case, the response message is a confirmation message 200 ok as described above. Thereafter, the terminal proceeds to step 508 to check whether a response message (Ack) indicating that the call is connected from the conference call server is received. If the test result response message is received, the process proceeds to step 510 where voice communication is performed through the RTP channel. In addition, step 510 of FIG. 5 corresponds to step 410 of FIG. 4, step 512 corresponds to step 412 of FIG. 4, step 514 corresponds to step 414 of FIG. 4, and step 516 of FIG. 4. It corresponds to step 416 of. Therefore, the above process will not be described any more.

6 is a control flowchart of call setup and communication of a conference call in a conference call server of a mobile communication system for high-speed data transmission according to the present invention. Hereinafter, a call setup and a control process during communication in a conference call server of a mobile communication system for high speed data transmission according to the present invention will be described in detail with reference to FIG. 6.

In step 600, the conference call server checks whether a session establishment message for requesting a conference call is received from a predetermined terminal. If the session establishment message requesting the conference call is received as a result of the check, after receiving the location information of the terminal requesting the call to the DRL server, the process proceeds to step 602 and transmits the session establishment message to the called terminal. That is, the process of step 206 of FIG. 2 is performed. In step 604, it is checked whether a response message, that is, a confirmation message 200 ok, is received within a predetermined time. If a response message is received from one terminal as a result of the check, the process proceeds to step 608. However, if all of the terminals requesting the conference call do not have a response within a predetermined time, the process proceeds to step 606 to generate and transmit a Not Found message to the terminal requesting the call.

When even one of the called terminals receives the response message (Ack), the conference call server proceeds to step 608 and generates and delivers a confirmation message (200 ok) informing the calling terminal of the conference call setup. In step 610, it is checked whether a response message (Ack) is received from the calling terminal. When the test result response message is received, the conference call server proceeds to step 612 and stores the number of receivable packets of each terminal where the call is performed. When the number of transmittable packets of the RTP packet set in step 612 is greater than or equal to "1", the conference call server performs step 618 and the terminal in step "620" performs step 620. That is, the terminal capable of receiving one or more packets proceeds to step 618 to integrate and reassemble the packets to be transmitted to the terminal. However, in the case of the terminal capable of receiving only one packet, the received packet is transmitted as it is in step 620.

In this way, it is possible to transmit more than one voice data packet to a specific terminal at the same time. Assuming that the number of RTP voice data packets transmitted as such is four, the configuration of the RTP header transmitted according to the present invention can be illustrated as shown in Table 4 below.

0 One 2 3 4 5 6 7 8 9 0 One 2 3 4 5 6 7 8 9 0 One 2 3 4 5 6 7 8 9 0 One V = 2 P X CC M payload sequence number timestamp (in sample rate units) synchronization source (SSRC) identifier contributing source (CSRC) identifiers length voice data ... voice data ... length voice data ... voice data ...

In Table 4, the header adds 8 bits indicating the payload size of the original header and the sub-packet. Therefore, if the number of RTP packets is 4, the RTP header can be reduced from a total of 48 bytes to 16 bytes. If you change the number of RTP packets during a session, you can use the INFO method of SIP. The INFO method does not change the state of the SIP session or call. It can only pass additional information to the server. When the server receives an INFO request, it must do a 200 OK response. In this case, if the number of RTP packets is changed from 2 to 2, the terminal inserts a "Packet_num: 2" field into the body of the SIP INFO message and sends it to the server. The server then responds with a 200 OK response and sends two RTP packets into one packet.

When the information change message is received from the terminal as described above, the process proceeds to step 624 to generate and transmit a response signal to the confirmation message (200 ok response) of the terminal. The above process is continuously performed until the end of the session in step 628, that is, until the call is terminated.

As described above, in the mobile communication systems having different forward and reverse transmission capacities, data that transmits the same data in the forward direction can be merged and transmitted, thereby reducing the overhead added for voice packet transmission. It can improve the band and performance.

Claims (3)

A method for providing a conference call service with packet data in a high speed data mobile communication system using a session establishment message including a number of receivable packets, the method comprising: Inspecting the session establishment message received from each terminal when the conference call service is set, and setting the number of packets that can be received by each receiving terminal; Preparing a buffer having a size according to the size set for each terminal; And receiving the data set in each of the buffers and merging them to the corresponding terminal. The method of claim 1, Changing the buffer size of the terminal upon receiving a capacity change message for the forward link from any terminal during the conference call. A conference call server apparatus for providing a conference call service with packet data in a high-speed data mobile communication system using a session establishment message including a number of receivable packets, comprising: A call controller for inspecting the session establishment message, determining the number of receivable packets of the corresponding terminal, and outputting a control signal for determining a buffer size according to the determined number of packets; And a media control unit for allocating a buffer according to the buffer control signal received from the call control unit and merging and transmitting packets to be transmitted according to the size of the buffer allocated to each terminal.