Data communication apparatus and data communication method

The present invention relates to a data communication device and a data communication method for storing data other than voice in a voice packet and performing communication in a voice communication system that sends and receives voice packets via an IP network to perform a call.

In recent years, so-called VoIP (Voice over Internet Protocol) communication systems, which send and receive voice packets via the IP network to make calls, are rapidly becoming widespread, but conventional telephones targeting PSTN will continue to be used in VoIP communication systems. IP telephone adapter devices that have been made available for use are known, and by using such IP telephone adapter devices, VoIP communication systems can also be used in data communication devices such as facsimile devices for PSTN. It will be possible.

On the other hand, in the VoIP communication system, a jitter buffer is provided in the receiving unit in order to absorb fluctuations in the packet arrival interval due to delay in the IP network, and retransmission processing is not performed in order to ensure real-time performance of voice reproduction. When an overflow occurs in the jitter buffer, the data is discarded, and when an underflow occurs, the data is interpolated based on the immediately preceding data (see Patent Document 1).<patcit num="1"><text>Japanese Unexamined Patent Publication No. 11-88533</text></patcit>

<p> However, although such discarding and interpolation processing in the event of overflow and underflow is effective in ensuring communication quality in the original telephone application, data faithfulness is achieved in data communication such as facsimile communication. It is desirable to avoid it because it hinders playback and adversely deteriorates communication quality.</p><p> However, since the fluctuation of the packet arrival interval due to the delay in the IP network is a non-cumulative fluctuation, it can be absorbed by a certain amount of buffer capacity, but if the change in the buffer accumulation amount is cumulative, Jitter buffer alone is not enough.</p><p> That is, there is a slight difference in the clock frequency between the transmitting side device and the receiving side device due to the individual difference of the clock generator, which causes a difference in the processing timing between the transmitting side device and the receiving side device, for example, the transmitting side. If the processing timing of the device is earlier than that of the receiving device, data gradually accumulates in the jitter buffer of the receiving device, and finally an overflow occurs. On the contrary, if the processing timing of the transmitting side device is later than that of the receiving side device, the data in the jitter buffer of the receiving side device gradually decreases, and finally an underflow occurs. Overflow and underflow caused by such a shift in processing timing are not desirable because there is a limit to how much the buffer capacity can handle, and a large increase in capacity leads to an increase in manufacturing cost.</p><p> The present invention has been devised to solve such problems in the prior art, and its main purpose is to reliably avoid overflow and underflow in the jitter buffer to faithfully reproduce data. To provide data communication devices and methods configured to enable.</p>

<p> In order to solve the above problems, in the present invention, as shown in claim 1, in a voice communication system that sends and receives voice packets via an IP network to make a call, data other than voice is stored in the voice packets. A data communication device that performs communication, a storage means that temporarily stores data obtained from received voice packets, a data processing means that performs required processing on data that is sequentially output from the data, and the storage. It is assumed that the means has a control means for adjusting the operation timing of the output processing of the storage means and the data processing of the data processing means according to the size of the stored data amount of the means.</p><p> According to this, the storage means can quickly escape from the dangerous state where overflow and underflow may occur and return to the safe state. Therefore, overflow and underflow in the storage means can be reliably prevented, and discarding and interpolation of audio frames can be avoided, whereby data can be faithfully reproduced.</p><p> In this case, it is desirable to adopt a lossless compression voice coding method, for example, ITU-T Recommendation G.711.</p><p> In the data communication device, as shown in claim 2, the data processing means is digitally connected to a codec unit that decodes the data of the received voice packet and the codec unit, and is output from the codec unit. It has a modem unit that demolishes the PCM data, and these codec units and the modem unit can have a configuration in which the operation timing is adjusted by a reference clock that synchronizes the operations with each other. According to this, the operation timing can be adjusted accurately with a relatively simple configuration.</p><p> As shown in claim 3, the data communication device has a detection means for detecting a buffer full state in which the amount of accumulated data of the storage means exceeds a predetermined upper limit value, and the control means is buffered by the detection means. When the full state is detected, the operation timing of the output processing of the storage means and the data processing of the data processing means can be accelerated. According to this, by performing packet processing at an interval shorter than the packet arrival interval, the amount of accumulated data of the storage means can be reduced, and thus overflow in the storage means can be avoided.</p><p> Further, in the data communication device, as shown in claim 4, in the storage means, the upper limit value serving as a detection reference of the buffer full state is a fixed delay region that absorbs fluctuations in the packet arrival interval due to delay in the network. It is possible to take a configuration in which the upper limit is set and a congestion-corresponding area corresponding to a state in which voice packets arrive intensively at short intervals due to congestion in the network is secured. According to this, even if packets stagnant at a router or the like arrive at one time due to congestion in the network, overflow in the storage means can be avoided. In this case, it is desirable to secure a large capacity of the congestion handling area in order to reliably avoid overflow.</p><p> Further, as shown in claim 5, the data communication device includes a detection means for detecting a buffer empty state in which the amount of accumulated data of the storage means is less than a predetermined lower limit value, and the control means is the detection means. When the buffer empty state is detected, the operation timing of the output processing of the storage means and the data processing of the data processing means can be delayed. According to this, the amount of accumulated data of the accumulating means can be increased, and thus the underflow in the accumulating means can be avoided.</p><p> In this case, the adjustment of the operation timing may not be performed until the buffer empty state is resolved. In addition, although the processing is performed in the buffer empty state, the processing rate is lowered to reduce the packet arrival interval. It is also possible to configure packet processing at longer intervals.</p><p> Further, in the data communication device, as shown in claim 6, the data stored in the voice packet can be configured as facsimile data generated according to the facsimile communication procedure.</p><p> Further, in the present invention, as shown in claim 7, in a voice communication system in which voice packets are transmitted / received via an IP network to make a call, a data communication method in which data other than voice is stored in the voice packets for communication. Therefore, the output processing of the storage means and the data sequentially output from the storage means are processed according to the size of the storage data amount of the storage means for temporarily storing the data obtained from the received voice packet. The operation timing of the data processing of the data processing means to be performed is adjusted.</p><p> According to this, overflow and underflow in the storage means can be reliably prevented, and discarding and interpolation of audio frames can be avoided, whereby data can be faithfully reproduced.</p><p> In the data communication method, as shown in claim 8, when a buffer full state in which the accumulated data amount of the storage means exceeds a predetermined upper limit value is detected, the output processing of the storage means and the data of the data processing means are detected. The operation timing of the processing is advanced, and when a buffer empty state in which the amount of accumulated data of the storage means is lower than a predetermined lower limit value is detected, the operation timing of the output processing of the storage means and the data processing of the data processing means is delayed. Can be.</p>

<p> As described above, according to the present invention, the storage means causes overflow and underflow in order to adjust the operation timing of the output processing of the storage means and the data processing of the data processing means according to the size of the stored data amount of the storage means. You can quickly get out of a potentially dangerous state and return to a safe state, ensuring that overflow and underflow in the storage means can be prevented and audio frame discarding and interpolation can be avoided. Since the data can be faithfully reproduced, a great effect can be obtained in improving the quality of data communication such as facsimile communication.</p>

Hereinafter, embodiments of the present invention will be described with reference to the drawings.

FIG. 1 is a block diagram showing a facsimile machine with a built-in IP telephone adapter, which is an example of a data communication device according to the present invention. This facsimile machine 1 stores facsimile data in a voice packet in a VoIP communication system that sends and receives voice packets via an IP network to make a call, and is a partner device, that is, a facsimile machine 2 having the same configuration as the present facsimile machine 1. , Or the IP telephone adapter device (VoIPTA) 4 to which the facsimile machine 3 of the conventional configuration is connected, performs deemed voice communication, and the transmission / reception unit 11, the VoIP processing unit 12, the facsimile processing unit 13, and the facsimile processing unit 13. It has a control unit 14.

At the time of transmission of the facsimile apparatus 1, the image data generated by reading the transmitted document by the scanning unit 21 of the facsimile processing unit 13 is input to the modem unit 23 via the CPU 22, and the modem unit 23 performs the G3 facsimile communication procedure. Therefore, the facsimile data is converted into PCM audio data. This PCM audio data is encoded by the encoder 32 of the code code unit 31 of the VoIP processing unit 12 by the audio coding method specified in ITU-T Recommendation G.711, and then RTP (Real-time) is performed by the RTP processing unit 33. It is assembled into a voice packet according to the procedure of Transport Protocol) and sent to the IP network via the transmission / reception unit 11.

On the other hand, at the time of reception, when the transmission / reception unit 11 receives the voice packet containing the facsimile data from the IP network, the RTP processing unit 33 of the VoIP processing unit 12 decomposes the packet according to the RTP procedure, and also data. Is analyzed to determine packet replacement. The audio frames output from the RTP processing unit 33 are temporarily stored in the jitter buffer (storage means) 35 of the jitter buffer processing unit 34 and then sequentially output to the codec unit 31, and in the codec unit 31, the audio interpolation unit The audio frame interpolation processing in 36 and the decoding processing in the decoder 37 are performed respectively, and the PCM audio data is output. This PCM audio data is sent to the facsimile processing unit 13 and demodulated by the modem unit 23, and the image data acquired thereby is recorded on a recording medium (paper or the like) by the recording unit 24.

The codec unit 31 operates on the reference clock generated by the internal clock generation unit 42, and the output processing of the audio frame from the jitter buffer 35 is also performed in synchronization with the reference clock of the clock generation unit 42. Further, the codec unit 31 and the modem unit 23 are digitally connected to each other, and the modem unit 23 performs modulation / demodulation processing in synchronization with the reference clock of the clock generation unit 42 in the codec unit 31. The clock generation unit 42 can adjust the oscillation timing of the clock signal according to the instruction signal from the control unit (control means) 14, thereby processing the output of the jitter buffer 35 and the codec unit 31 and the modem unit 23 ( The operation timing of the data processing of the data processing means) can be adjusted.

The operation timing is adjusted by the control unit 14 according to the amount of data stored in the jitter buffer 35. That is, the jitter buffer processing unit 34 has a comparator (detection means) 41 that detects a buffer full state in which the amount of accumulated data in the jitter buffer 35 exceeds a predetermined upper limit value and a buffer empty state in which the accumulated data amount falls below a predetermined lower limit value. However, when the buffer full state is detected by the comparator 41, the operation timing of the output processing of the jitter buffer 35 and the data processing of the codec 31 and the modem 23 is accelerated, while the buffer full state is detected by the comparator 41. This delays the operation timing of the output processing of the jitter buffer 35 and the data processing of the codec 31 and the modem 23.

FIG. 2 is a conceptual diagram of the jitter buffer shown in FIG. FIG. 2 (A) shows the buffer full state, and FIG. 2 (B) shows the buffer empty state. In the jitter buffer 35, the lowest accumulated audio frames are sequentially output, and the audio frames are sequentially moved from top to bottom accordingly, and the input audio frames are stored at the top of the accumulated data. The jitter buffer 35 secures a fixed delay area of a predetermined capacity (for example, 200 msec) that absorbs fluctuations in the packet arrival interval due to delays in the IP network and delays due to processing such as packet replacement in the RTP processing unit 33. , When the data reaches the upper limit of the fixed delay area, the transfer of the data to the codec unit 31 is started.

Further, in the jitter buffer processing unit 34, the upper limit value (first threshold value) that serves as a reference for detecting the buffer full state by the comparator 41 is set to the upper limit of the fixed delay area, and above that, the router is set due to congestion in the network. A congestion-corresponding area with a predetermined capacity (for example, 200 msec) corresponding to a state in which packets stagnant due to such reasons arrive intensively at abnormally short intervals is secured. The lower limit value (second threshold value) that serves as a reference for detecting the buffer empty state by the comparator 41 is set to a predetermined value (for example, 100 msec) in the fixed delay region.

FIG. 3 is a timing diagram of data processing in the facsimile machine shown in FIG. Packet transfer via the IP network is performed at a standard predetermined transfer rate in the VoIP communication system, and in the proper state where the amount of data stored in the jitter buffer 35 is between the upper limit value and the lower limit value, Fig. 3 (A). As shown in the above, the VoIP processing unit 12 and the modem unit 23 perform processing at a predetermined time interval T (for example, 10 msec).

On the other hand, when the accumulated data amount of the jitter buffer 35 reaches the buffer full state exceeding the upper limit value, the operation timing of the output processing of the jitter buffer 35 and the data processing of the codec 31 and the modem 23 is accelerated, as shown in FIG. 3 (B). As shown, processing is performed at a predetermined time interval T'(for example, 5 msec). As a result, packet processing is performed at intervals shorter than the packet arrival interval, the amount of accumulated data in the jitter buffer 35 can be quickly reduced and returned to the proper state, and overflow occurs in the jitter buffer 35. it can.

On the other hand, when the accumulated data amount of the jitter buffer 35 becomes a buffer empty state below the lower limit value, the operation timing of the output processing of the jitter buffer 35 and the data processing of the codec 31 and the modem 23 is delayed. Here, as shown in FIG. 3C, the output processing of the jitter buffer 35 and the data processing of the codec 31 and the modem 23 are temporarily stopped until the buffer empty state is resolved. As a result, the amount of accumulated data in the jitter buffer 35 can be quickly increased and returned to an appropriate state, and underflow can be avoided in the jitter buffer 35.

In this embodiment, the data communication device (facsimile device) includes an IP telephone adapter device including a transmission / reception unit 11, a control unit 14, an RTP processing unit 33, a jitter buffer processing unit 34, and a codec unit 31. Although the above example is shown, it is also possible to make the IP telephone adapter device provided with the above-mentioned parts separate from the data communication device and digitally connect the IP phone adapter device to the data communication device. Further, data can be exchanged between the codec unit 31 and the modem unit 23 by using a digital (PCM) modulated wave signal or by serial transfer.

The data communication device and data communication method according to the present invention have an advantage that overflow and underflow in the storage means can be reliably prevented and data can be faithfully reproduced, and voice can be reproduced via an IP network. It is useful when data other than voice is stored in voice packets for communication in a so-called VoIP communication system in which packets are sent and received to make a call.

<figref num="1">A block diagram showing a facsimile machine with a built-in IP telephone adapter, which is an example of a data communication device according to the present invention.</figref><figref num="2">Conceptual diagram of the jitter buffer shown in Fig. 1.</figref><figref num="3">Timing diagram of data processing in the facsimile machine shown in Fig. 1.</figref>

Code description

1 Facsimile device according to the present invention 2 Facsimile device according to the present invention 3 Conventional facsimile device 4 IP telephone adapter device 12 VoIP processing unit 13 Facsimile processing unit 14 Control unit (control means) 23 Modem unit (data processing means) 31 Codec unit (data) Processing means) 34 Jitter buffer processing unit 35 Jitter buffer (storage means) 36 Voice interpolation unit 41 Comparer (detection means) 42 Clock generator