KR100738914B1 - Method, system and server for providing multimedia ring back tone service for use in global system for mobile communication network and mobile station therefor - Google Patents

Abstract

A method, a system and a server for providing a multimedia RBT substitute sound service in a GSM(Global System for Mobile telecommunication) network and a mobile terminal therefor are provided to minimize the change of an established GSM network by encoding an RBT substitute sound source through an audio codec and transmitting the encoded result to a mobile terminal in a bypass mode. A wireless communication processing unit(510) performs wireless communication with a GSM network. An audio codec unit(540) decodes audio data. A voice codec unit(550) decodes voice data. An output unit(560) outputs voice and audio sound. A controller(570) controls the general operation. When plural slot data received from the GSM network is received from the wireless communication processing unit(510), a codec controller(530) analyzes the slot data to check whether a codec identifier greater than an audio threshold value is continuously detected, converts a basic codec into an audio codec, and transfers data included in the slot data to the audio codec unit(540).

Description

Method, System, Server and Providing Multimedia Ring Back Tone Service for Use in Global System for Mobile Communication Network and Mobile Station Therefor}

1 is a block diagram schematically illustrating a system for providing a multimedia ringback tone substitute sound service in a GSM network according to an embodiment of the present invention;

2 is a flowchart illustrating a method for providing a multimedia ringback tone substitute sound service in a GSM network according to an embodiment of the present invention;

3 is an exemplary diagram for explaining a data structure of slot data according to a preferred embodiment of the present invention;

4 is an exemplary view showing a codec identifier according to a preferred embodiment of the present invention;

5 is a block diagram schematically illustrating a mobile terminal for a multimedia ringback tone replacement sound service in a GSM network according to an embodiment of the present invention;

6 is a flowchart illustrating a method of applying an audio codec according to a preferred embodiment of the present invention.

<Description of Symbols for Main Parts of Drawings>

110: originating MS 112: incoming MS

120: GSM network 130: originating BSS

132: incoming BSS 140: originating MSC

142: incoming MSC 150: common line signaling network

160: HLR 170: sound source server

180: sound source providing unit 510: wireless communication processing unit

520: SIM processing unit 530: codec control unit

540: audio codec unit 550: voice codec unit

560: output unit

The present invention relates to a method for providing a multimedia ringback tone replacement sound service in a Global System for Mobile Communication (GSM) network, a system, a server, and a mobile station (MS) for the same. . More specifically, in providing a ringback tone replacement sound service to a mobile terminal in a GSM system, in order to transmit a high quality ringback tone replacement sound to a mobile terminal, the ringback tone replacement sound is encoded by an audio codec and an audio codec in a GSM network. After generating audio data by inserting an identifier for identifying the data, and changing the default codec by analyzing the method, the system, the server and the data received therefor for transmitting to the mobile terminal in bypass mode to determine whether the audio data. It relates to a mobile terminal.

With the rapid development of computer, electronic and communication technologies, mobile communication services, which are various wireless communication services using a wireless communication network, have been provided. Mobile subscribers can freely move anywhere and anytime by using mobile communication service and make voice calls with the other party, as well as various information necessary for life using wireless data service such as news, weather, sports, securities, exchange rate In addition, traffic information may be provided while moving in various forms such as text, voice, and images.

Recently, a so-called ringback tone replacement sound service, which listens to the caller's mobile terminal, which is called ringback tone, has been developed and has been in the spotlight by mobile communication subscribers. . The ringback tone substitute tone service is a service that provides a substitute tone specified by the called party or the caller to the caller's mobile terminal. Instead, a mechanical tone generated when a call is connected, such as a "following" sound, is used instead of the latest sound. It is a service that can be used as a substitute for ringback tone, as well as pop songs, birds, and water. As the ringback tone replacement sound service is provided, the caller can feel auditory satisfaction by listening to the sounds of various sound sources instead of listening to a uniform mechanical ringback tone.

The ring back tone replacement tone service is typically provided to a mobile terminal in a mobile communication network implemented by a mobile communication system such as a code division multiple access (CDMA) 2000 system, a GSM, a wideband CDMA (WCDMA) system, or the like. Here, the GSM system is a concept corresponding to the CDMA system, and is a mobile communication system using a time division multiple access (TDMA) technique and an asynchronous transmission network technique for dividing a frequency channel into time.

On the other hand, the GSM system uses a voice codec such as a Regular Pulse Excited-Long Term Predictor (RTE-LTP) codec having a transmission rate of 13.0 kbps or an Enhanced Full Rate (EFR) codec having a transmission rate of 12.2 kbps.

However, the voice codec has been developed to efficiently deliver the voice of a person, such as sampling rate is limited to a range of voices that a person can make and when there is no dialogue, silence detection is performed. However, there is a problem in that music in a high frequency region other than the human voice cannot be efficiently delivered.

That is, it is preferable to provide music other than voice using an audio codec rather than a voice codec, and apply this to a ringback tone substitute sound service to encode a ringback tone substitute sound source with an audio codec to provide a multimedia ringback tone substitute sound service. If so, the service quality of the ringback tone replacement sound service can be expected to be improved.

In addition, in order to provide the above-mentioned multimedia ringback tone replacement sound service in a GSM system, the mobile terminal should notify the mobile terminal that it is a ringback tone replacement sound using an audio codec and change the voice codec to an audio codec to control ringback tone replacement sound. This requires many retrofitting of the GSM system, so the investment for providing the service can be significant.

Therefore, in order to provide a high quality multimedia ringback tone replacement sound service from the GSM system, the audio codec encoded sound source is transmitted to the mobile terminal with minimal change of the system, and the mobile terminal reproduces the audio codec encoded sound source. There is a need for the development of technology to enable this.

In order to solve this problem and meet the above-mentioned demands, the present invention provides a ringback tone replacement sound service to a mobile terminal in a GSM system, and to transmit a high quality ringback tone replacement sound to a mobile terminal, a ring in a GSM network. Audio data is generated by encoding the backtone replacement sound into an audio codec, inserting an identifier for identifying the audio codec, and transmitting the data to the mobile terminal in bypass mode, analyzing the system, the server, and the data received therefor. An object of the present invention is to provide a mobile terminal for changing a basic codec by checking whether it is data.

In order to achieve the above object, the present invention, in the method of receiving and playing one or more of the voice and ringback tone replacement sound source from the Global Network for Mobile Communication (GSM) network (hereinafter referred to as 'GSM') ( a) receiving a plurality of slot data from a GSM network; (b) converting the base codec into an audio codec by analyzing whether the codec identifier is continuously detected above the audio threshold by analyzing the plurality of slot data; And (c) decoding and playing the data included in the plurality of slot data with the audio codec.

In addition, according to another object of the present invention, a wireless communication processing unit for performing wireless communication with a global network (GSM) network (hereinafter referred to as 'GSM'), the audio having an audio codec to decode the audio data A mobile station (MS) comprising a codec unit, a voice codec unit having a voice codec for decoding voice data, an output unit for outputting voice and audio, and a control unit for controlling overall operations, the mobile station (MS) receiving from a GSM network When receiving a plurality of slot data from the wireless communication processing unit, by analyzing the plurality of slot data to determine whether the codec identifier is continuously detected above the audio threshold value, the basic codec is converted to an audio codec, and the plurality of slot data And a codec control unit for transmitting the included data to the audio codec unit. It provides a mobile terminal for the audio codec that applies to.

In addition, according to another object of the present invention, the ringback tone to the GSM (GSM: Global System for Mobile Communication (GSM)) network providing a ringback tone replacement sound service to a mobile station (MS) In a server providing a substitute sound source, a ringback tone substitute sound source is encoded with an audio codec to generate a plurality of audio frames, and the plurality of audio frames are mapped to a time slot to be transmitted to a mobile terminal to generate a plurality of subframes. In addition, the plurality of subframes provide a multimedia ringback tone substitute sound service providing server, characterized in that it comprises a codec identifier indicating that the ringback tone substitute sound source is encoded with an audio codec.

In addition, according to another object of the present invention, a Global System for Mobile Communication (GSM) system that provides a ringback tone replacement sound service to an outgoing mobile terminal requesting a voice call to an incoming mobile terminal (MS) An audio codec is generated by storing a ringback tone substitute sound source as an audio codec, and then stored. The codec identifier indicating that the ringback tone substitute sound source is encoded as an audio codec is inserted into a subframe of the audio codec sound source. A sound source server for generating and transmitting information on a bypass mode when a sound source is requested to be played, and transmitting an audio codec sound source; A home location register (HLR) for storing subscriber information, location information, and information on whether to subscribe to a ringback tone replacement tone service for the calling mobile terminal and the called mobile terminal; And when a voice call is requested from the calling mobile terminal, sets a call with the sound source server using the HLR, sets a bypass mode by receiving information about the bypass mode from the sound source server, and sets the audio codec sound source in the bypass mode. The present invention provides a multimedia ringback tone replacement sound providing system, comprising: a mobile switching center (MSC) for transmitting to an originating mobile terminal.

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 are used as much as possible even if displayed on different drawings. In addition, in describing the present invention, when it is determined that the detailed description of the related well-known configuration or function may obscure the gist of the present invention, the detailed description thereof will be omitted.

1 is a block diagram schematically illustrating a system for providing a multimedia ringback tone substitute sound service in a global network for mobile communication (GSM) network according to a preferred embodiment of the present invention.

Multimedia ringback tone replacement sound service providing system in a GSM network according to a preferred embodiment of the present invention includes an originating MS (Mobile Station) 110, the destination MS 112, GSM network 1200 and the sound source providing unit 180 do.

The originating MS 110 uses a GSM terminal 120 to operate in an asynchronous mode, such as a GSM phone, which performs voice calls with each mobile terminal or with a landline telephone over a public switched telephone network (not shown). MS: Mobile Station).

The originating MS 110 according to the preferred embodiment of the present invention requests a voice call to the called party's 112 using the GSM network 120 according to the caller's key manipulation, and the caller or the called party replaces the ringback tone. In the case of subscribing to the service, the ringback tone replacement sound is received from the GSM network 120 and output.

In addition, the originating MS 110 according to the preferred embodiment of the present invention requests a voice call to the destination MS 112 using the GSM network 120 and then includes a plurality of ringback tone replacement sound sources from the GSM network 120. When receiving the slot data, it analyzes a specific bit (e.g., the last 4 bits) of every slot data.If a pattern identical to the codec identifier is detected, the slot data is recognized as audio data and the basic codec is converted into an audio codec. Decode and play. If not detected, set the voice codec as the default codec to decode and play. To this end, the originating MS 110 according to the preferred embodiment of the present invention includes an audio codec as well as a voice codec.

Here, the slot data is voice data or audio data mapped to a time slot of 20 ms, and includes voice data when the ringback tone replacement sound source or voice is encoded by the voice codec in the GSM network 120, and the ringback tone replacement is performed. When the sound source is encoded by the audio codec, audio data is included. When the GSM network 120 encodes the ringback tone substitute sound source with an audio codec, the codec identifier is inserted to indicate that the audio data is encoded with the audio codec. Therefore, the originating MS 110 analyzes the slot data and the same as the codec identifier. It is checked whether the pattern is detected to determine whether the slot data is audio data or voice data.

Also, the basic codec is a codec set as a basic codec for decoding encoded data among codecs included in the originating MS 110 and may be a voice codec or an audio codec. If the MS 110 analyzes the slot data and continuously detects the same pattern as the codec identifier, the source MS 110 decodes and plays back the audio data by setting the audio codec as the default codec, and if the same pattern as the codec identifier is not detected. Set the voice codec as the default codec to decode and play the voice data.

In addition, the originating MS 110 according to the preferred embodiment of the present invention may receive a signaling message from the GSM network 120 and change the default codec to a voice codec or an audio codec according to the signaling message.

The destination MS 112 according to the preferred embodiment of the present invention is a mobile terminal that receives a call request from the originating MS 110 using the GSM network 120, and responds according to the paging request of the GSM network 120.

The GSM network 120 includes an outgoing base station system (BSS) 130, an outgoing BSS 1320, an outgoing mobile communication switching center (MSC: Mobile Switching Center, hereinafter referred to as 'MSC') 140, an incoming MSC 142, Common Channel Signaling Network (CCSN: 150), Home Location Register (HLR: Home Location Register, hereinafter referred to as 'HLR', 160) and the sound source server 170, including the originating MS 110 and It is an asynchronous mobile communication network that provides voice call service and ringback tone replacement sound service to the destination MS 112.

The originating BSS 130 and the destination BSS 132 are composed of a base transceiver station (BTS) and a base station controller (BSC), and the BTS and the BSC exchange signals using an Abis interface. Here, the BTS is a device composed of a radio transceiver and communicates with the originating MS 110 and the destination MS 112 through a radio link. The BSC manages multiple BTSs and manages radio channels, protocol matching with the originating MS 110 and the destination MS 112, protocol matching with the BTS, soft handover processing, matching with the originating MSC 140 and the destination MSC 142. It is responsible for functions such as fault management, system loading, and so on.

The originating MSC 140 and the terminating MSC 142 are circuit switches that perform basic and supplementary service processing, incoming and outgoing call processing of subscribers, location registration and handoff procedures, interworking with other networks, and the like, respectively. By using the 160, the calling MS 110, the calling party which is the mobile communication subscriber of the called MS 112, the information on the called party, and location information and status information of each mobile terminal are grasped.

When the originating MSC 140 receives a call call requesting a voice call from the originating MS 110 through the originating BSS 130, the authentication procedure and encryption are performed with the originating MS 110 in conjunction with the HLR 160. (Ciphering) procedure is performed, and radio resources are allocated to the originating MS 110 to perform call setup.

In addition, the originating MSC 140 according to the preferred embodiment of the present invention is the HLR 160, the location of the destination MS 112 and additional services of the originating MS 110 or the destination MS 112, that is, ringback tone replacement tone service After inquiring whether the subscription is made to obtain routing information for the destination MSC 142 and the sound source server 170, and performing call setup with the destination MSC 1420, a ringback tone substitute sound source encoded with an audio codec, When the audio codec sound source is transmitted, the audio codec is switched to bypass mode and transmitted to the originating MS 110 without going through a voice encoder.

In addition, when the originating MSC 340 according to the preferred embodiment of the present invention is notified of the response of the terminating MS 112 by the terminating MSC 142, it releases the mutual bypass mode and responds to the terminating MS 112. The voice call is relayed to notify the originating MS 110 as alert information and to conduct a voice call between the originating MS 110 and the terminating mobile terminal 112.

When the called MSC 142 according to the preferred embodiment of the present invention is informed from the calling MSC 140 that there is a voice call request for the called MS 112, it performs a call setup with the calling MSC 140 and the called MS ( Paging 112, and receives information on the ringback tone replacement sound, information on the bypass mode, and control information from the sound source providing apparatus 170, transmits the information to the originating MSC 340, and switches to the bypass mode. Thereafter, the audio codec sound source transmitted from the sound source server 170 to the originating MSC 140 is relayed.

In addition, the destination MSC 142 according to a preferred embodiment of the present invention, if there is a response from the destination MS 112 to notify the originating MSC (140) and release the bypass mode, and the originating MS (110) and The voice call between the destination MSs 112 is relayed.

The common line signal network 150 refers to a network composed of signal points and signal links to provide a standardized common line signaling method. Here, the common line signaling scheme refers to a signaling scheme that separates signal information and network management information into voice and data and transmits them on a separate channel commonly used for signal transmission.

The HLR 160 is a database storing service profiles related to subscriber information of mobile communication subscribers. The HLR 160 performs a function such as registration recognition, registration deletion, and location checking of each mobile terminal, and uses a mobile communication subscriber using each mobile terminal. Information about the service type, such as a mobile identification number (MIN), a mobile serial number (ESN), a ringback tone replacement tone service, and routing to the sound source providing device 370, Stores information for routing.

The sound source server 170 is an IP for an intelligent network that stores the ringback tone replacement sound source. When the ringback tone replacement sound source is requested, the sound source server 170 requests the sound source authentication to the sound source providing unit 180, and when the sound source authentication succeeds, the incoming MSC 142. Transmit a ringback tone replacement sound source. In addition, the sound source server 170 stores and manages the ringback tone replacement sound source when it is transmitted from the sound source providing unit 180.

The sound source server 170 according to the preferred embodiment of the present invention performs the call setup with the incoming MSC 142, and then checks the ringback tone replacement sound source set for the destination MS 112 to obtain information on the ringback tone replacement sound. After generating the information and the control information for the bypass mode, and transmits to the destination MSC 142, and transmits the audio codec sound source to the destination MSC (142).

Here, when the sound source server 170 according to the preferred embodiment of the present invention is an audio codec sound source by checking whether the ringback tone substitute sound source set for the terminating MS 112 is an audio codec sound source encoded with an audio codec, the ring It generates information on the backtone replacement sound, information on the bypass mode, control information, and the like.

In addition, the sound source server 170 according to the preferred embodiment of the present invention encodes the ringback tone replacement sound source received and stored from the sound source providing unit 180 with an audio codec to generate an audio frame, and then transmits it through a voice channel. Subframes are generated by mapping in the form of time slots so as to be suitable, and transmitted to the originating MS 110 through each MSC. Here, the method of generating the subframe by the sound source server 170 will be described in detail in the process to be described later.

The sound source providing unit 180 generates, receives, and stores a ringback tone replacement sound source, including the Internet, a sound source providing control server, a subscriber DB, and a web server, and provides the sound source server 170.

2 is a flowchart illustrating a method for providing a multimedia ringback tone substitute sound service in a GSM network according to a preferred embodiment of the present invention.

If the originating MS 110 requests a voice call with the terminating MS 112 to the originating MSC 140 via the originating BSS 130 (S202), the originating MSC 140 works in conjunction with the HLR 160 to make the originating MS. An authentication procedure and an encryption procedure are performed with the 110 and call setup is performed by allocating a radio channel to the originating MS 110 (S204).

The originating MSC 140 requests the HLR 160 for information about the location of the terminating MS 112 and whether the originating MS 110 or the terminating MS 112 subscribes to the ringback tone replacement tone service (S206). 142 and the routing information for the sound source server 170 is received (S208).

The originating MSC 140 receiving the routing information performs call setup with the destination MSC 142 (S210). The called MSC 142 that has made the call setup with the calling MSC 140, after paging the called MS 112 (S212), performs call setup with the sound source server 170, and the calling MS 110 or the called MS ( Request a ringback tone replacement sound source set for 112 (S214), the sound source server 170 completes the call setup with the destination MSC (142) and the ringback tone set for the originating MS (110) or the destination MS (112) The replacement sound source is reproduced and transmitted to the destination MSC 142 (S216). At this time, the sound source server 170 checks the ringback tone replacement sound source set for the originating MS 110 or the terminating MS 112, and in the case of an audio codec sound source encoded with an audio codec, information about the ringback tone replacement sound, Information on the pass mode and control information are generated and transmitted to the destination MSC 142.

When the incoming MSC 142 receives the audio codec sound source from the sound source server 170, the receiving MSC 142 checks the information on the bypass mode and the control information, switches to the bypass mode, and causes the originating MSC 140 to switch to the bypass mode. After the control, the audio codec sound source is transmitted. When the source MSC 140 receives the audio codec source after switching to the bypass mode, the source MSC 140 relays the source MS 110 to the source MS 110 through the source BSS 130 without additionally passing through the voice coder (S218).

The originating MS 110 receives the audio codec sound source as slot data of a plurality of time slots, and analyzes each slot data to determine whether the same pattern as the codec identifier is continuously detected more than a predetermined number of times (S220). When the pattern is detected, the slot data is recognized as audio data, and the basic codec is changed to the audio codec (S222), and then decoded and played back (S224).

On the other hand, when the terminating MS 112 responds to the paging performed by the terminating MSC 142 in step S212 (S226), the terminating MSC 142 releases the call setup with the sound source server 170 (S228) The MSC 140 notifies the voice call connection (S230), releases the bypass mode between the originating MSC 140 and the destination MSC 142, and the originating MSC 140 notified of the voice call connection is the originating MS 110. Informs that the terminating MS 112 has responded to the paging (S232). Subsequently, when the terminating MS 112 transmits voice data for a call, the voice data is sequentially transmitted to the terminating MSC 142 and the originating MSC 140, and then coded, and then a plurality of time slots are sent to the originating MS 110. Are transmitted as slot data (S234).

The originating MS 110 receiving a plurality of slot data analyzes every slot data to determine whether the same pattern as the codec identifier is continuously detected more than a predetermined number of times (S236), and the same pattern is continuously more than a certain number of times. If it is not detected, the basic codec is converted to the voice codec to decode and reproduce the voice data with the voice codec to perform a voice call with the destination MS 112 (S240). If not detected, the default codec is not changed.

3 is an exemplary diagram for describing a data structure of slot data according to a preferred embodiment of the present invention.

In order to provide a voice call service in the GSM network 120, a RPE-LTE (Regular Pulse Excited Long Term Prediction) codec having a transmission rate of 13.0 kbps and an EFR (Enhanced Full Rate) codec having a transmission rate of 12.2 kbps are used. The size of the voice data, i.e. the voice data, allocated to the 20 ms time slot varies according to the type of the codec.

That is, 260 bits of speech data are allocated to one 20 ms time slot when the voice is encoded using the RPE-LTE codec, and 244 is allocated to one 20 ms time slot when the voice is encoded using the EFR codec. Bits of voice data are allocated. Accordingly, the size of data allocated to the ringback tone replacement sound service also varies depending on the type of the RPE-LTE codec or the EFR codec.

For this reason, according to a preferred embodiment of the present invention, the ringback tone substitute sound source in the GSM network 120 is referred to as an Advanced Audio Codec Plus (AAC +) codec, adaptive multiple ratio (AMR: Adaptive). Even when encoding using an audio codec such as a multi rate (hereinafter, referred to as an 'AMR') codec, the data is allocated with the size of data allocated to one 20 ms time slot.

That is, when the voice codec used in the GSM network 120 is an RPE-LTE codec, an audio codec sound source is also assigned 260 bits in one 20 ms time slot, and when the voice codec is an EFR codec, one audio codec sound source is also used. 244 bits are allocated to a 20 ms time slot in. The subframe structure of the audio codec sound source according to the type of voice codec used in the GSM network 120 is illustrated in FIG. 4.

3A shows a subframe structure of an audio codec sound source when encoding the voice using the RPE-LTE codec in the GSM network 120. Although 3A shows two audio frames (N-1 th audio frame and N th audio frame) for convenience of description, the audio frame of the audio codec sound source may actually consist of a plurality of more audio frames.

Each audio frame is mapped to a 20 ms time slot in consideration of a wireless channel for transmitting a voice or sound source to the originating MS 110 of the GSM network 120 and is generated as a plurality of subframes. In the case of encoding the voice using the RPE-LTE codec, since 260 bits of data are allocated to one 20 ms time slot, the subframe also includes 260 bits. At this time, since it is efficient to transmit an audio frame in byte units due to the characteristics of an audio codec, a total of 260 bit subframes are generated to include 32 bytes, that is, 256 bits of audio data and 4 bits of a codec identifier.

That is, the sound source server 170 generates an audio frame by encoding the ringback tone substitute sound using an audio codec, and takes the form of a 20 ms timeslot in consideration of a wireless channel to transmit the audio frame to the originating MS 110. By mapping, a plurality of subframes having a size of 260 bits are generated, and audio data is inserted into only 256 bits of the subframes, and codec identifiers are generated to recognize the originating MS 110 as audio data in the remaining 4 bits. Insert it.

3B shows a subframe structure of an audio codec sound source when encoding the voice using the EFR codec in the GSM network 120. Although 3B shows two audio frames (N-1 th audio frame and N th audio frame) for convenience of description, the audio frames of the audio codec sound source may actually constitute a plurality of more audio frames.

Each audio frame is mapped to a 20 ms time slot in consideration of a wireless channel for transmitting a voice or sound source to the originating MS 110 of the GSM network 120 and is generated as a plurality of subframes. When the voice is encoded using the EFR codec, since 244 bits of data are allocated to one 20 ms time slot, the subframe also includes 244 bits. At this time, since the audio frame is efficiently transmitted in byte units due to the characteristics of the audio codec, a total of 244 bit subframes are generated to include 30 bytes, that is, 240 bits of audio data and 4 bits of codec identifier.

That is, the sound source server 170 generates an audio frame by encoding the ringback tone substitute sound using an audio codec, and takes the form of a 20 ms timeslot in consideration of a wireless channel to transmit the audio frame to the originating MS 110. By mapping, a plurality of subframes having a size of 244 bits are generated, and audio data is inserted into only 240 bits of the subframes, and codec identifiers are generated and inserted into the remaining 4 bits to recognize that the originating MS 110 is audio data. do.

As described above (when the RTE-LTE codec is used as the voice codec in the GSM network 120 or when the EFR codec is used), the generated subframes are carried in a plurality of 20 ms time slots in the originating BSS 130. A plurality of slot data is transmitted to the originating MS 110, and the originating MS 110 analyzes the received plurality of slot data to determine whether a codec identifier exists in each slot data and the codec identifier is above a preset audio threshold. It checks whether it appears repeatedly and recognizes it as an audio codec sound source, converts the default codec to an audio codec, and decodes and plays the audio codec sound source with an audio codec. Here, the audio threshold refers to the maximum number of times set to be recognized by the audio codec when the codec identifier appears continuously in a plurality of slot data.

4 is an exemplary view showing a codec identifier according to a preferred embodiment of the present invention.

4A shows a bit value when the codec identifier indicates only whether the audio codec is applied or not. When the codec identifier indicates only whether the audio codec is applied or not, the bit value of the codec identifier is defined as two values. That is, in the case of applying the audio codec, the bit value of the codec identifier is set to '0000' and has a value of '0', so that the value of 4 bits in the codec identifier of the slot data in the originating MS 110 is '0000'. If it is detected that the slot data is audio data and to be encoded with an audio codec.

In addition, when the audio codec is not applied, the bit value of the codec identifier is set to one of '0001' to '1111' instead of '0000', so that the value of the codec identifier in the slot data in the originating MS 110 is' If detected as one of '0001' to '1111', the corresponding slot data is recognized as not audio data. Here, the values of '0000' and '0001' to '1111' are shown by way of example only, and bit values of the codec identifier according to the preferred embodiment of the present invention may be in various forms (for example, hexadecimal '000F', etc.). )

4B shows a bit value when the codec identifier indicates the type of codec. When the codec identifier indicates not only whether the audio codec is applied but also the type of the corresponding codec, the audio codec type or the voice codec type corresponding to each bit value of 4 bits may be designated. That is, as shown in the figure, when the bit value of the codec identifier is '0000', the codec encoding the corresponding slot data is an AAC + codec, and when '0001', the codec encoding the corresponding slot data is an AMR-WB + codec. In case of '1111', the codec encoding the corresponding slot data is an AAC + codec, and the type of the codec can be designated according to the bit value of the codec identifier. Here, the values of '0000' and '0001' to '1111' are shown by way of example only, and bit values of the codec identifier according to the preferred embodiment of the present invention may be in various forms (for example, hexadecimal '000F', etc.). )

In addition, when generating each subframe, the sound source server 170 may insert the 3-bit codec identifier without inserting the 4-bit codec identifier as described above. That is, 4 bits except 256 bits of 260 bits or 1 bit in 4 bits except 240 bits of 244 bits are used for other purposes (for example, when one subframe is lost due to transmission error of audio data, the corresponding subframe is lost). In order to prevent an error from occurring in the entire audio frame including a, a 3-bit codec identifier may be inserted using a header of the audio frame to another subframe and a header identifier for indicating the same).

In this case, however, the audio threshold must be increased in order to reduce the number of bits of the codec identifier so that the originating MS 110 cannot accurately identify the codec. That is, when only 3 bits are inserted as the codec identifier, the audio threshold is set to be larger than when 4 bits are inserted as the codec identifier, so that the originating MS 110 continuously detects the codec identifier in the slot data more and more times. Only in one case, the slot data is recognized as audio data, thereby ensuring stability of codec switching.

5 is a block diagram schematically illustrating a mobile terminal for a multimedia ringback tone replacement sound service in a GSM network according to an embodiment of the present invention.

A mobile terminal for a multimedia ringback tone replacement service in a GSM network according to an embodiment of the present invention includes a wireless communication processor 510, a subscriber indetification module (SIM) processor 520, a codec controller 530, and an audio codec unit. 540, a voice codec unit 550, an output unit 560, and a controller 570. Here, since the mobile terminal for the multimedia ringback tone replacement sound service in the GSM network 120 according to the preferred embodiment of the present invention may be implemented as the outgoing MS 110 shown in FIG. It is called the originating MS 110 for.

The wireless communication processing unit 510 performs a mobile communication service including a voice call service and a ringback tone replacement tone service provided by the GSM network 120 by the originating MS 110 communicating wirelessly with the GSM network 120. Wireless communication processing means for performing. The wireless communication processing unit 510 according to the preferred embodiment of the present invention receives a plurality of slot data from the GSM network 120 and transmits it to the codec controller 530.

When the SIM card storing the subscriber information of the mobile subscriber is inserted, the SIM processing unit 520 receives the subscriber information from the SIM card, performs a predetermined authentication procedure, and then checks whether the calling MS 110 is available. In addition, the mobile communication telephone number of the calling MS 110 is provided to the controller 570 so that the wireless communication processor 510 communicates with the GSM network 120 using the mobile communication telephone number.

When the codec controller 530 receives a plurality of slot data from the wireless communication processor 510, the codec controller 530 analyzes each slot data, and determines whether the codec identifier is inserted and whether the codec identifier is continuously detected beyond the audio threshold. If the codec identifier is continuously detected from the plurality of slot data by more than the audio threshold, the controller recognizes that the slot data is audio data, and converts the basic codec into an audio codec to convert the audio data of the slot data into the audio codec unit 540. If it is detected continuously below the audio threshold value, and recognizes that the slot data is the voice data, the default codec is set as the voice codec to transmit the voice data to the voice codec unit 550.

The audio codec unit 540 includes an audio codec implemented in software or hardware. When audio data is delivered, the audio codec unit 540 decodes the audio data using the audio codec and then delivers the audio to the output unit 560.

The voice codec unit 550 includes a voice codec implemented by software or hardware. When the voice data is transmitted, the voice codec unit 550 decodes the voice data using the voice codec, and then transfers the voice to the output unit 560.

The output unit 560 outputs audio and voice transmitted from the audio codec unit 540 or the voice codec unit 550 through a speaker.

The control unit 570 is a control means for controlling the overall operation of the originating MS 110, the wireless communication processing unit 510, SIM processing unit 520, codec control unit 530, audio codec unit according to a predetermined algorithm provided 540, the voice codec unit 550, the output unit 560 controls the originating MS 110 receives the slot data from the GSM network 120, and analyzes the slot data to determine whether the slot data is audio data or voice. It checks whether it is data, and sets an appropriate codec, that is, an audio codec or a voice codec, as a default codec to decode and play audio or voice.

6 is a flowchart illustrating a method of applying an audio codec according to a preferred embodiment of the present invention.

The originating MS 110 according to the preferred embodiment of the present invention should check whether the slot data received from the GSM network 120 is voice data or audio data, and then decode and play the codec accordingly. To this end, the originating MS 110 must have a codec counter to count the number of times the codec identifier has been detected in the slot data.

That is, when the default codec is set as the voice codec in the originating MS 110 and the codec count is counted as 0 (S602), when the slot data is received from the GSM network 120 (S604), the slot data is analyzed. If the codec identifier is not detected in the slot data (S606), if the codec identifier is not detected, the slot data is recognized as voice data, and the codec count is reset without switching the basic codec to the audio codec. After setting to '0', the next slot data is received (S608).

On the other hand, the originating MS 110 stores the increased codec count by increasing the codec count by '+1' when the codec identifier is detected as a result of checking in step S606 (S610), that is, the codec count is '0'. In this case, the increased codec count is stored as '1', and when the codec count is '1', the increased codec count is stored as '2'.

The originating MS 110, which has increased the codec count, checks whether the stored codec count is greater than or equal to the audio threshold (S612), and if it is not greater than or equal to the audio threshold, proceeds to step S604 to receive the next slot data, and the audio threshold If the value is larger than the default codec, the audio codec is switched (S614). That is, when the audio threshold value is set to '4', when the codec count is '4' or more, the default codec is switched to the audio codec.

Therefore, if the codec identifier is not detected in the situation where the codec count is '3', since the codec count is reset and set back to '0' as in step S608, the default codec is maintained as the preset voice codec, and 4 Only when the codec identifier is detected in the at least one slot data, the codec count becomes '4', that is, the audio threshold value or more, and the default codec is switched to the audio codec.

In summary, when the originating MS 110 receives each of a plurality of slot data, it analyzes each slot data to determine whether the codec identifier is inserted, and if the codec identifier is not inserted, does not switch the default codec. When the slot data is received and the codec identifier is inserted, it is checked whether the slot code is continuously inserted into the next slot data. When the codec identifier is continuously inserted above the audio threshold value, the basic codec is switched to the audio codec.

In step S614, the originating MS 110 that converts the basic codec into the audio codec decodes and plays the audio data of the received slot data with the audio codec set as the default codec, resets the codec count to '0' (S616). ).

In addition, the source MS 110 having reset the codec count receives the next slot data again (S618), checks whether the codec identifier is inserted into the received slot data (S620), and inserts the codec identifier into the slot data. If it is recognized that the slot data is audio data, the flow proceeds to step S618 to receive the next slot data, and if the codec identifier is not inserted, the codec count is increased by '+1' to store the increased codec count ( S622).

The originating MS 110 storing the increased codec count checks whether the codec count is greater than or equal to the voice threshold (S624), and if the codec count is not greater than or equal to the voice threshold, proceeds to step S618 to receive the next slot data. If the codec count is greater than or equal to the voice threshold, the default codec is switched to the voice codec (S626), and the codec count is reset and set to '0' (S628). Here, when the codec count is not equal to or greater than the voice threshold value, the reset of the codec count and not set to '0' as in step S608 means that the slot data is voiced if it is detected for a predetermined number or more even if the codec identifier is not continuously detected in the slot data. This is because it may be recognized as data. On the other hand, the originating MS 110, which has converted the basic codec into the voice codec in step S626, decodes the voice data of the slot data to reproduce the voice.

As described above, when the originating MS 110 receives the slot data from the GSM network 120, the procedure of steps S602 to S616 and S618 to S628 is repeated to determine whether the data included in the slot data is voice data or audio. It judges whether it is data and plays a voice or audio codec sound source.

The above description is merely illustrative of the technical idea of the present invention, and those skilled in the art to which the present invention pertains may make various modifications and changes without departing from the essential characteristics of the present invention. Therefore, the embodiments disclosed in the present invention are not intended to limit the technical idea of the present invention but to describe the present invention, and the scope of the technical idea of the present invention is not limited by these embodiments. The protection scope of the present invention should be interpreted by the following claims, and all technical ideas within the equivalent scope should be interpreted as being included in the scope of the present invention.

As described above, according to the present invention, when a ringback tone replacement sound service is provided to a mobile terminal in a GSM network, a high quality ringback tone replacement sound source encoded by an audio codec can be provided to the mobile terminal, thereby providing a quality of service. Can improve.

In addition, since the ringback tone substitute sound source is encoded by the audio codec and then transmitted to the mobile terminal through the bypass mode, in order to provide a high quality multimedia ringback tone substitute sound service, it is possible to change the existing GSM network. It can be minimized.

In addition, when encoding a voice or audio in the GSM network, a codec identifier may be inserted to specify the type of the codec encoding the voice or audio, so that the codec designated by the mobile terminal may be specified without further controlling the mobile terminal in the GSM network. It can play voice or audio with basic codec and can provide stable service.

In addition, when the mobile terminal switches the audio codec to the audio codec, the audio codec is called only when the codec identifier is detected for a predetermined number of consecutive times, so that the voice or audio can be stably reproduced.

Claims (26)

In the method of receiving and playing one or more of the voice and ringback tone replacement sound source from the Global System for Mobile Communication (GSM) network, (a) receiving a plurality of slot data from the GSM network; (b) converting a basic codec into an audio codec by analyzing whether the codec identifier is continuously detected above an audio threshold value by analyzing the plurality of slot data; And (c) decoding and playing data included in the plurality of slot data with the audio codec; Audio codec application method comprising a. The method of claim 1, wherein step (b) comprises: analyzing whether the codec identifier is detected by analyzing slot data of one of the plurality of slot data; (b2) if the codec identifier is detected as a result of the checking in step (b1), increasing the codec count by '+1' and storing the codec count; (b3) checking whether the codec count is greater than or equal to the audio threshold; And (b4) converting the default codec into the audio codec when the codec count is greater than or equal to the audio threshold as a result of the checking in step (b3); Audio codec application method comprising a. The method of claim 2, wherein the method is After the step (b2), (b21) If the codec identifier is not detected as a result of checking in step (b1), resetting the codec count to set to '0' and proceeding to step (b1). Audio codec application method characterized in that it further comprises. The method of claim 2, wherein the method is After the step (b4), (b5) proceeding to step (b1) if the codec count is less than the audio threshold; Audio codec application method characterized in that it further comprises. The method of claim 1, wherein the method is After step (c), (d) receiving a plurality of slot data from the GSM network; (e) analyzing the plurality of slot data to determine whether the codec identifier is not detected above a voice threshold and converting the default codec into a voice codec; And (f) decoding and playing data contained in the plurality of slot data with the voice codec; Audio codec application method comprising a. The method of claim 5, wherein step (e) (e1) determining whether the codec identifier is detected by analyzing one slot data of the plurality of slot data; (e2) if the codec identifier is not detected as a result of the checking in step (e1), increasing the codec count by '+1' and storing the codec count; (e3) checking whether the codec count is greater than or equal to the voice threshold value; And (e4) converting the basic codec into the voice codec when the codec count is greater than or equal to the voice threshold as a result of the checking in step (e3); Audio codec application method comprising a. The method of claim 6, wherein the method is After the above step (e2), (e21) if the codec identifier is detected as a result of the checking in step (e1), proceeding to step (e1) Audio codec application method characterized in that it further comprises. The method of claim 6, wherein the method is After the above step (e4), (e5) proceeding to step (e1) when the codec count is less than the voice threshold value; Audio codec application method characterized in that it further comprises. The method of claim 1, wherein the data, The audio codec application method characterized in that the voice data is a voice encoded with a voice codec or audio data that is a ringback tone replacement sound source encoded by the audio codec. The method of claim 1, wherein the codec identifier is And the data included in the plurality of slot data indicates whether the data is encoded by the audio codec. The method of claim 1, wherein the codec identifier is And a type of codec encoding data included in the plurality of slot data. A wireless communication processing unit for performing wireless communication with a Global System for Mobile Communication (GSM) network, an audio codec unit for decoding audio data with an audio codec, and a voice codec In the mobile terminal (MS) including a voice codec unit for decoding the control unit, an output unit for outputting voice and audio, and a control unit for controlling the overall operation, When receiving a plurality of slot data received from the GSM network from the wireless communication processing unit, by analyzing the plurality of slot data to determine whether the codec identifier is continuously detected above the audio threshold value to convert the default codec to an audio codec And a codec controller for transferring data included in the plurality of slot data to the audio codec unit. Mobile terminal for applying an audio codec, characterized in that it comprises a. The method of claim 12, wherein the codec control unit, And a codec counter for counting the number of times the codec identifier is detected in the plurality of slot data. The method of claim 13, wherein the codec control unit, Analyze each slot data of the plurality of slot data to determine whether the codec identifier is detected. If the codec identifier is detected, the codec count is increased by '+1' using the codec counter and the codec count is increased. And if the codec count is greater than or equal to the audio threshold value, and if the codec count is greater than or equal to the audio threshold value, converts the basic codec into the audio codec. The method of claim 14, wherein the codec control unit, If the codec identifier is not detected, the codec count is reset and set to '0'. The method of claim 12, wherein the codec control unit, After converting the basic codec to the audio codec, by analyzing the plurality of slot data, it is determined whether the codec identifier is not detected above a voice threshold, and the basic codec is converted to a voice codec and the plurality of slots The mobile terminal for applying the audio codec, characterized in that for transmitting the data contained in the data to the voice codec unit. The method of claim 12, wherein the data, And the audio data which is the voice data which is the voice encoded by the voice codec or the ringback tone substitute sound source which is encoded by the audio codec. The method of claim 12, wherein the codec identifier, The mobile terminal for applying an audio codec, characterized in that whether the data included in the plurality of slot data is encoded with the audio codec. The method of claim 12, wherein the codec identifier, A mobile terminal for applying an audio codec, characterized in that it represents a type of codec encoding data included in the plurality of slot data. In the server for providing a ringback tone replacement sound source to the Global System for Mobile Communication (GSM) network that provides a ringback tone replacement sound service to a mobile station (MS), A function of generating a plurality of audio frames by encoding the ringback tone substitute sound source with an audio codec, and mapping the plurality of audio frames to time slots to be transmitted to the mobile terminal to generate a plurality of subframes, And the plurality of subframes include a codec identifier indicating that the ringback tone substitute sound source is encoded by the audio codec. The method of claim 20, wherein the server, When the voice codec applied to the mobile communication service provided to the mobile terminal in the GSM network is a RPE-LTE (Regular Pulse Excited Long Term Prediction) codec, each subframe of the plurality of subframes is generated with a size of 260 bits. Multimedia ring back tone replacement sound service providing server, characterized in that. The method of claim 21, wherein each subframe, And a 256 bit audio data and a 4 bit codec identifier. The method of claim 20, wherein the server, When the voice codec applied to the mobile communication service provided to the mobile terminal in the GSM network is an enhanced full rate (EFR) codec, each subframe of the plurality of subframes is generated with a size of 244 bits. Multimedia ringback tone replacement sound service providing server. The method of claim 23, wherein each subframe, And a 240 bit audio data and a 4 bit codec identifier. The method of claim 20, wherein the codec identifier, The multimedia ringback tone substitute sound service providing server, further comprising a type of codec encoding the ringtone substitute sound source. In the Global System for Mobile Communication (GSM) system, which provides a ringback tone replacement sound service to an outgoing mobile terminal requesting a voice call to an incoming mobile terminal (MS), Generate and store an audio codec sound source by encoding a ringback tone substitute sound source with an audio codec, and insert a codec identifier indicating that the ringback tone substitute sound source is encoded with the audio codec in a subframe of the audio codec source, and insert the audio codec. A sound source server for generating and transmitting information on a bypass mode when receiving a codec sound source and transmitting the audio codec sound source; A home location register (HLR) for storing subscriber information, location information, and information on whether to subscribe to the ringback tone replacement tone service for the calling mobile terminal and the called mobile terminal; And When the voice call is requested from the originating mobile terminal, a call is set up with the sound source server using the HLR, and the bypass mode is set by receiving information about the bypass mode from the sound source server. An outgoing mobile switching center (MSC) for transmitting the audio codec sound source to the outgoing mobile terminal in a bypass mode Multimedia ringback tone replacement sound providing system comprising a.

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