Abstract:
A method and system to boost error recovery and synchronous tracking abilities is disclosed, using fixed code rate rules and synchronous data for connecting a voice call between mobile terminals. The system and method process packet data received from a BTS, expand the processed packet data based on a fixed code rate rule, and transmit the expanded packet data with framing information, inserted therein, to an MSC.

Description:
BACKGROUND OF THE INVENTION 
   1. Field of the Invention 
   The present invention relates to a mobile telecommunication system and, more particularly, to formatting voice data in a mobile telecommunication system. 
   2. Background of the Related Art 
     FIG. 1  illustrates a configuration of a mobile telecommunication system in the related art. Referring to  FIG. 1 , when a mobile subscriber communicates voice data with a wire subscriber, the mobile telecommunication system compresses the voice call in a mobile terminal (MT)  1   a – 1   n  and transmits the compressed voice call to a base station controller (BSC)  3   a – 3   n,  via a base transceiver station (BTS)  2   a – 2   n.  The voice call can be compressed into digital data by a compression algorithm. The BSC  3   a – 3   n  converts the compressed digital data into pulse code modulation (PCM) data and transmits the PCM data to a mobile switching center (MSC  4 . To convert the compressed digital data into PCM data, the BSC  3   a – 3   n  employs a vocoder. The MSC  4  can transmit the PCM data to a public switched telephone network (PSTN)  8 , in which the corresponding wire subscriber is matched. The corresponding wire subscriber can be selected by a switching circuit. 
   So far, the method of providing a voice call generated by the MT  1   a – 1   n  of the mobile subscriber to the wire subscriber has been explained. Similarly, it is also possible to provide the voice call from the wire subscriber to the MT  1   a – 1   n  following the same procedure described above in an inverse order. In addition, to provide the voice call from MT  1   a  to another MT  1   n,  the same procedure can be applied, again. 
   In other words, the vocoder of the BSC  3   a – 3   n  can convert the compressed digital data from the MT  1   a – 1   n  to PCM data. Unfortunately however, as the vocodet repeatedly conducts the conversion process from the compressed digital data into the PCM data, it increases the quantization error as well, deteriorating the sound quality as a result. 
   As an attempt to solve the above problem, another method has been tried. For example, when the vocoder in the BSC  3   a – 3   n  should provides the voice call between mobile subscribers, the vocoder does not convert the compressed data from the BTS  2   a – 2   n  into PCM data to deliver it to the MSC  4 . Instead, the vocoder sends the voice data to the MSC  4  after packetizing it to have a certain format. This operation can be carried out when a bypass mode is designated by a call process control of the BSC  3   a – 3   n  and the MSC  4 . Also, if the vocoder at a receiver is in the bypass mode, the vocoder can recognize the data received from the MSC  4 , through the communication time slot, as a voice packet data in the bypass format, not the PCM format. Then, the vocoder will decompose the data right away and send out a corresponding voice packet to the BTS. 
     FIG. 2  is a related art configuration of a vocoder inside of a BTS  2   a – 2   n  controller. The vocoder  20  includes a packet matcher  21 , a voice coder/decoder  22 , a bypass controller  23  and a PCM matcher  24 . In the case of connecting the voice call between mobile subscribers with this vocoder, if the voice packet is transmitted to the packet matcher  21  from the BTS  2   a – 2   n , the packet matcher  21  matches the voice packet and sends the matched voice packet to the voice coder/decoder  22 , instead of the bypass controller  23 . 
   The vocoder  22  is preferably pre-designated in the bypass mode. The bypass controller  23  transmits the voice packet to the time slot as it is. The voice packet transmitted to the time slot undergoes the packeting process in a special format and is sent out to the MSC  4  via the PCM matcher  24 . Data received from the MSC  4  proceeds in a reverse order, of the aforementioned transmitting procedure for the voice packet, and is sent to the bypass controller  23 . Later, the vocoder recognizes the data as the voice packet data, decomposes the data, and finally sends the voice packet to the BTS. 
     FIG. 3  illustrates a data configuration of a bypass packet of the vocoder in the related art. The bypass packet format comprises up to 40 bytes of data in total, including a maximum 32-byte encoding packet data  33  received from the BTS  2   a – 2   n,  a 4-byte message  32  for transmitting a signal, and a 4-byte preamble  31  for distinguishing a final/ending message. Additionally, the bypass packet may include a 290-bit dummy  34  and 30-bit cyclic-redundancy-code (CRC)  35 . Thus, the packet data format can include up to a total of 80-bytes of data, and the packet data format can go through the same procedure one more time before it is transmitted. After being formatted, the bypass packet is sent to the MSC  4 . 
   On the other hand, when a synchronous system is involved, 32-byte encoding packet data  33  is transmitted and received every 20 ms between the BTS  2   a – 2   n  and the BSC  3   a – 3   n . Further, between the BSC  3   a – 3   n  and the MSC  4 , a 1-byte bypass packet every 125 us (i.e., a 160-byte bypass packet every 20 ms) can be transmitted and received. 
     FIG. 4  illustrates an operational procedure of the bypass mode. The formation of the bypass packet format includes adding the 4-byte message  32  on the basis of the 32-byte encoding packet data  33  (S 41 ) and further adding the 4-byte preamble  31  (S 42 ). In addition, after adding the dummy  34  and the CRC  35  to the encoding packet data  33 , the message  32  and the preamble  31  (S 43  and S 44 ), the adding process (S 41 –S 44 ) is repeated entirely to generate a second block of data with the same length (i.e., 80 bytes) (S 45 ). Together, the two blocks of data complete the final bypass packet format (160 bytes). A final bypass packet of this form is delivered to the MSC  4 , by communicating 1 byte every 125 us (S 46 ). Meanwhile, the packet data received from the MSC  4  is decomposed in a reverse order of the above-described procedure, and the encoding packet data  33  is extracted from the packet data format and sent to the BTS  2   a – 2   n.    
   The most typical and generic data length that is effective in the related art is within a range of 32 to 36 bytes. If the preamble is added, the maximum continued data length reaches 40 bytes. Accordingly, the voice data used in practice is approximately one fourth of the total 160 bytes. Therefore, data resources are often wasted. Moreover, during the repetition of the packet data format operation, it is always possible that 80-byte data of the first half frame  30   a , in  FIG. 3 , can include an error. In this case, the BTS  2   a – 2   n  has to resend the correct data. That is to say, if the first half frame  30   a  or the second half frame  30   b  of the 80-byte packet data format contains an error, in a particular bit, it is regarded as a CRC error and the BSC  3   a – 3   n  recognizes the data as being invalid. Thereby, information on the error occurrence or a useful and effective handling method are precluded. 
   In addition, in terms of the properties of the voice data in the real time mode, no band width assigned for resending the voice data in the next 20 ms frame. Although it seems possible to regenerate and send the regenerated packet data frame, in practice the entire frame ( 30   a  and  30   b ) can be at risk of losing validity for even 1-bit error. Even worse, there is no way to correct the error. 
   If the preamble  31 , identifying a start point of the packet data, has an error during the bypass mode of operation, the valid start point is nowhere to be found. Also, when combining the preamble  31  with the encoding packet data  33 , the encoding packet might have its own preamble as well. In such a case, it is impossible to find the start point of the frame. 
   If, by any chance, a 160-byte frame is lost, not only is the voice data no longer valid but also a synchronization with the next frame is no longer valid. The synchronization loss requires a considerable amount of time to overcome, when it occurs for a large number of frames. 
   Therefore, the related art is very disadvantageous in that it fails to accomplish an original goal of improving the quality of sound, during the voice call between mobile terminals, by reducing the quantization procedure. 
   The above references are incorporated by reference herein where appropriate for appropriate teachings of additional or alternative details, features and/or technical background. 
   SUMMARY OF THE INVENTION 
   An object of the invention is to solve at least the above problems and/or disadvantages and to provide at least the advantages described hereinafter. 
   It is, therefore, an object of the present invention to provide a method for formatting voice data in a mobile telecommunication system in order to prevent any loss of voice data. 
   It is another object of the present invention to improve sound quality by improving a data format, when connecting a voice call between mobile terminals. 
   To achieve the above objects, a preferred embodiment of the present invention provides a method for formatting voice data in a mobile telecommunication system. The method includes processing packet data received from a BTS, expanding the processed packet data using fixed code rate rules, inserting framing information into the expanded packet data, and transmitting the packet data with the framing information to a MSC. 
   According to the method for formatting voice data in a mobile telecommunication system, the fixed code rate rule can be made by repeating each bit of the processed packet data at the equivalent value. 
   Further, in accordance with the method for formatting voice data in a mobile telecommunication system of the present invention, the framing information represents synchronous information, and a plurality of bits configured from 0 or 1 are generated on a regular time basis. Here, the combination order of the plural bits can be pre-designated. 
   Preferably, the method for formatting voice data in a telecommunication system further includes extracting framing information from the packet data received from the MSC, wherein the packet data is expanded by as many times as the pre-designated times that has been set up based on the repetition of the equivalent value in each bit; restoring the size of the packet data to the original size of the packet data; and transmitting the restored and processed packet data to the BTS. 
   Particularly, restoring the size of the packet data to the original size of the packet data further includes judging whether a repeated times at the equal value based on each bit of the packet data is in accord with the pre-designated times; converting each bit of the packet data into the majority values, if a repeated times at the equal value based on each bit of the packet data is not in accord with the pre-designated times; and diminishing the size of each bit to be the same size with the inverse number of the pre-designated times. 
   Another preferred embodiment of the present invention provides a mobile telecommunication system, which includes a first BSC for converting voice data received from the first BTS into a fixed packet format and for transmitting the converted voice data to a MSC. A second BSC for converts the voice data received from the MSC into a fixed packet format and transmits the converted voice data to the second BTS. In addition, there is provided a method for transmitting voice data in a mobile telecommunication according to the subject embodiment of the present invention, wherein the first BSC expands the voice data from the first BTS plus pre-designated information, using a fixed code rate rule; inserts synchronous information into the expanded voice data; and transmits the voice data with the synchronous information to the MSC. The second base controller extracts the synchronous information based on the voice data received from the MSC, whose bit has been repeatedly expanded according to the pre-designated times; restores the original voice data by adjusting the bit size of the voice data, to correspond to the inverse number of the pre-designated times; processes the restored voice data; and transmits the processed voice data to the second BTS. 
   The objects of the present invention may be further achieved in whole or in part by a communication method, including encoding original information as encoded information, comprising the original information and redundant information, using a coding scheme, generating transport information by overwriting portions of the encoded information with framing information, transmitting the transport information to a receiver, extracting the framing information from the transport information, synchronizing the encoded information at the receiver using the extracted framing information, and regenerating the original information by decoding the encoded information, wherein the redundant information of the encoded information is used to regenerate the original information lost due to the overwriting of the encoded information with the framing information. 
   The objects of the present invention may be further achieved in whole or in part by a communication system, including an encoder that encodes original information as encoded information, comprising the original information and redundant information, using a coding scheme, an information framing unit that generates transport information by overwriting portions of the encoded information with framing information, a transmitter that transmits the transport information to a receiver, a framing information extraction unit that extracts the framing information from the transport information received by the receiver, a synchronization unit that synchronizes the encoded information at the receiver using the extracted framing information obtained by the framing information extraction unit, and a decoder that regenerates the original information by decoding the encoded information, wherein the redundant information of the encoded information is used to regenerate the original information lost due to the overwriting of the encoded information with the framing information. 
   Therefore, the method for formatting voice data in a mobile telecommunication, described above, enables the synchronization of the extracted data to be determined based on the combination order. 
   Additional advantages, objects, and features of the invention will be set forth in part in the description which follows and in part will become apparent to those having ordinary skill in the art upon examination of the following or may be learned from practice of the invention. The objects and advantages of the invention may be realized and attained as particularly pointed out in the appended claims. 

   
     BRIEF DESCRIPTION OF THE DRAWINGS 
     The invention will be described in detail with reference to the following drawings in which like reference numerals refer to like elements wherein: 
       FIG. 1  illustrates a configuration of a mobile telecommunication system in accordance with the related art; 
       FIG. 2  illustrates a configuration of a vocoder inside the BSC of  FIG. 1 , in accordance with the related art; 
       FIG. 3  illustrates a configuration of bypass packet data of a vocoder, in accordance with the related art; 
       FIG. 4  illustrates a flow chart of a vocoder bypass operation, in accordance with the related art; 
       FIG. 5  illustrates a configuration of bypass packet data of a vocoder, in accordance with a preferred embodiment of the present invention; and 
       FIG. 6  and  FIG. 7  illustrate a data processing procedure during a voice call between mobile terminals, in accordance with a preferred embodiment of the present invention. 
   

   DETAILED DESCRIPTION OF PREFERRED EMBODIMENTS 
     FIG. 5  is a configuration of a bypass packet of a vocoder according to the present invention. The bypass packet  50  includes a preamble  51 , a message  52 , and encoding packet data  53 . The bypass packet  50 , unlike a bypass packet in the related art, does not include a dummy and a cyclic redundancy code (CRC). Instead of including the dummy and the CRC, which are somewhat useless in the present invention, the bypass packet  50  leaves them out and utilizes the recovered bandwidth. As explicitly shown in  FIG. 5 , the bypass packet  50  can be expanded up to 160 bytes through a series of procedures, and framing information  54  can be inserted in fixed positions of the 160 bytes. The framing information  54  includes a plurality of bits and can be generated on a regular time basis (e.g., 20 ms). 
   A method for transceiving the bypass packet illustrated in  FIG. 5  is now explained with reference to  FIG. 6  and  FIG. 7 .  FIG. 6  illustrates the method for transmitting the converted encoding packet data to the mobile switching station. 
   Referring to  FIG. 6 , when the encoding packet data  53  (32 bytes) from the BTS  2   a – 2   n  is received, the vocoder in the BSC can add the pre-designated preamble  51  (4 bytes) and the message  52  (4 bytes) to the encoding packet data  53  (S 61  and S 62 ). Since the procedure applied above is identical with that of the related art, further details on the subject procedure will not be provided here. The vocoder conducts a channel coding on the basis of the packet data  50  (40 bytes), including the preamble  51  and the message  52  (S 63 ). Here, the channel coding can be performed using the fixed code rate rule. In other words, what the channel coding rule means is that the process is performed repeatedly, until each bit of the packet data is reproduced the pre-designated number of times. 
   For example, suppose that the pre-designated number of times is 4 and a certain bit value of the packet data is ‘1’. Applying the code rate rule the value ‘1’ is converted to ‘1111’. Of course, the encoding packet data having a ‘0’ bit value is converted into ‘0000’. Therefore, if the channel coding is carried out, the packet data  50  (40 bytes) with the preamble  51  and the message  52  is expanded to 160 bytes. In short, the channel coding procedure expands the packet data to a certain size, without difficulties. 
   When the channel coding is conducted, the framing information generated on a regular time basis can be inserted into the channel coded packed data (S 64  and S 65 ). The synchronous framing information is used for chasing frame synchronization, later, and includes a plurality of bits configured as 0 or 1. The combination order of the plural bits within the framing information can be designated beforehand. An important thing to be aware of is that each bit of the framing information can be inserted into any place of the pre-generated, 160-byte encoding packet data. This is so for the following reason. If one bit of the framing information is inserted in a certain location, causing a bit value of the original packet data to change the value of the original bit changed by this operation, is restored when the receiver changes each received bit value of the packet data to the value indicated by a majority of the corresponding redundant coding bits. Accordingly, the packet data in which the framing information is inserted can be sent out to the MSC  4  through a transmitter (S 66 ). 
     FIG. 7  illustrates a procedure of receiving the encoded packet data through the receiver of the BSC  3   a – 3   n.  When the encoded packet data is received from the MSC  4 , the vocoder in the BSC  3   a – 3   n  extracts framing information from it (S 71 ). The coded packet data is the expanded 16-byte packet whose every bit is repeated over and over the pre-designated number times by another BSC  3   a – 3   n,  supporting the other party&#39;s mobile terminal. 
   The packet synchronization of the packet data is decided based on whether the combination order of each bit of the extracted framing information is in accord with the pre-designated combination order. In the case that the received combination order and the pre-designated combination order are identical, the packet data is regarded as normal and the normally processed packet data can be transmitted to the BTS  2   a – 2   n.  On the other hand, if the received combination order differs from the pre-designated combination order, due to a potential packet data error, the BSC  3   a – 3   n  can request that the MSC  4  retransmit the 160-byte expanded frame. 
   The vocoder restores the packet data size to the original data size through the channel decoding procedure (S 72 ). Before performing the channel decoding process, the vocoder judges whether the number of times the packet data is repeated is accord with the pre-designated number of times. 
   If the number of times each bit is repeated is not in accord with the pre-designated number of times, each bit of the packet data can be changed to the value indicated by a majority of the corresponding redundant values. For instance, suppose that the original packet data is ‘1’ and is channel coded to ‘1111’, by the transmitting BSC  3   a – 3   n  of the other party, on the way to the receiving BSC via the MSC  4 . If the second coded bit value is converted into ‘0’, that is, the packet data is received as ‘1011’, the vocoder recognizes the majority of the redundant bits have the value ‘1’ and converts the 4 bits ‘1011’ into ‘1111’. The procedure can be applied to the rest of the bits of the coded packet data to be converted. 
   In addition, the bit size can be diminished to the size of the inverse number of the designated times by the channel decoding. In other words, the redundant bits can be removed so that the 160-byte coded packet data, of the previous example, is reduced to its prior un-encoded size of 40 bytes. Complying with the code rate rule, the packet data of the present invention has been expanded to 4 times the original size, meaning that the packet data can be diminished to one fourth of the original size by the channel decoding. That is to say, ‘1111’ can be diminished to ‘1’. Similarly, other bits of the packet data can be diminished as well. 
   Later, the vocoder deletes the preamble and the message from the channel decoded packet data and transmits the remaining packet data to the BTS  2   a – 2   n  (S 73  and S 74 ). 
   In conclusion, the present invention to transceives the packet data using the code rate rule and the framing information, thereby overcoming any possible problems generated frequently during the bypass mode operation in the related art. In addition, the method of the present invention advantageously improves the data-recovery rate upon the occurrence of errors, which consequently prevents any voice data loss in the voice call between mobile terminals, and further provides much improved mobile telecommunication service. 
   In addition, the present invention makes use of the bandwidth, which the dummy and CRC used in the related art, for the channel coding, by which it can more effectively use the band. 
   The foregoing embodiments and advantages are merely exemplary and are not to be construed as limiting the present invention. The present teaching can be readily applied to other types of apparatuses. The description of the present invention is intended to be illustrative, and not to limit the scope of the claims. Many alternatives, modifications, and variations will be apparent to those skilled in the art. In the claims, means-plus-function clauses are intended to cover the structures described herein as performing the recited function and not only structural equivalents but also equivalent structures.