Patent Publication Number: US-6985501-B2

Title: Device and method for reducing delay jitter in data transmission

Description:
BACKGROUND OF THE INVENTION 
   1. Field of Invention 
   This invention relates to a delay jitter reducing device for sequentially receiving a series of chronological data segments through a transmission path such as the Internet and delaying an individual data segment for an appropriate amount of time, thereby reducing delay jitter that has occurred in the propagation process of an individual data segment and obtaining chronological data segments from which effects of the delay jitter have been eliminated; and a delay jitter reducing method thereof. 
   2. Description of the Related Art 
   One form of data transmission is a real-time transmission that transmits a chronological sample of continuous signals such as, for example, voice signals after loading them to a plurality of consecutive packets. In such a real-time transmission, if delay time in transmitting a packet for individual packets are equal to one another, it is possible to obtain a voice signal of the same waveform as the source node by reproducing a chronological sample in a packet at the time of receiving each packet. 
   In a network such as the Internet, however, even in a case where a plurality of packet are transmitted from an unchanged source node to an unchanged destination node, the propagation delay time for individual packets are not necessarily the same as one another, and the propagation delay time varies among packets. This variation of the propagation delay time among packets is generally called delay jitter. 
   In a case where such delay jitter occurs, when a chronological sample is reproduced from received packets at the point of receiving each packet at the destination node, it is not assured that a signal of the same waveform as the original transmission signal can be reproduced from the received packets. 
   In such a case, destination nodes usually take a step of reducing delay jitter using buffers so as to obtain chronological data with effects of delay jitter eliminated. 
   This technique for reducing delay jitter will be described in detail with reference to  FIG. 12  to  FIG. 17 . 
     FIG. 12  is a block diagram showing a configuration example of a real-time voice transmission system. In the system, at a source terminal  10 , a voice signal to be transmitted is encoded by a voice encoder  11 , and chronological voice packets on which coded data of the voice signal are loaded are generated. A transmission unit  12  transmits these individual voice packets to a destination terminal  30 . Each voice packet arrives at the destination terminal  30  after passing a network  20 . At the destination terminal  30 , voice packets from the source terminal  10  are received by a receiving unit  31  and reserved in a buffer  32 . Subsequently, voice packets reserved in the buffer  32  are read from the buffer  32  in the same order as an order generated at the source node and transmitted to a voice decoder  33 . The voice decoder  33  receives voice packets transmitted in this way and decodes the voice signal from coded data included in the voice packets. 
   In the real-time voice transmission system, each voice packet generated in the source terminal  10  is sent out to the network  20  at the same transmission time interval as the generated time interval of each packet. However, as described already, propagation delay time required for these individual packets to reach the receiving terminal  30  is not fixed for each voice packet. Such being the case, the destination terminal  30  adjusts the timing for sending individual voice packets to the voice decoder  33 .  FIG. 17  shows an example of this timing adjustment. In the example shown in  FIG. 17 , voice packets P 0 , P 1 , and P 2  arrive at the destination terminal  30 , having taken a propagation delay time of d 0 , d 1 , and d 2  each. As shown, if each voice packet P 0 , P 1 , and P 2  can be delayed for D 0 , D 1 , and D 2  which is an appropriate amount of time for each, a total delay time T in turn can be fixed, where the total delay time is the amount of time required for each voice packet transmitted from the source terminal  10  to the voice decoder  33 . The buffer  32  as shown in  FIG. 12  is a device used for adjusting delays in order to fix the total delay time of each voice packet in this way. Assuming a minimum delay time of a voice packet as dmin and maximum delay time of a voice packet as dmax in the network  20 , the difference between them, D=dmax−dmin, is referred as delay jitter width as a matter of convenience. The buffer  32  in  FIG. 12  is required to adjust a variation of delay time in the range of this delay jitter width; in other words, the buffer  32  should be capable of reducing the delay jitter. 
   Hereinafter described will be on delay adjustment of a voice packet by the buffer  32  with reference to  FIGS. 13A and 13B . 
   In  FIG. 13B , there are provided four queues placed above and below in parallel, each queue consisting of a chain of nine boxes in a row. The first queue indicates a state of the buffer  32  at a certain time t 1 . The second queue indicates a state of the buffer  32  at time t 2  that is 1 s later than time t 1 . Likewise, the third and fourth queue each indicates a state of the buffer  32  at time t 3  that is 1 s later than time t 2  and at time t 4  that is 1 s later than time t 3 . 
   In the example shown in  FIG. 13B , the buffer  32  has a capacity of storing nine voice packets. Each of the nine boxes in each queue is an area for storing a voice packet, and the notation, # 1  to # 9 , in each box indicates the address of each area. 
   In the destination terminal  30 , one voice packet is read every 1 s from the buffer  32  and sent to the voice decoder  33 , where “s” is a unit such as several milliseconds and several dozen milliseconds depending on a data attribute, the unit being suitable for each data attribute. The address of an area where a voice packet is read is also updated one address every fixed time 1 s. In  FIG. 13B , an area where a voice packet is currently being read is shown at the right end of each queue, an area on the left next thereto is where the readout is performed 1 s later, and an area on the second left next thereto is where the readout is performed 2 s later. Likewise, the other areas follow; thus, the area at the leftmost of the queue is an area where a voice packet is read 8 s later. 
   In the example shown in  FIG. 13B , a voice packet is read from the area of address # 1  at time t 1 . At time t 2 , another voice packet is read from the area of address # 2 , another voice packet is read from the area of address # 3  at time t 3 , and another packet from the area of address # 4  at time t 4 . Therefore, if a voice packet received at time t 1  is written into the area of address # 4 , the voice packet is output from the buffer  32  to the voice decoder  33  at time t 4  which is 3 s later. Also, if a voice packet received at time t 1  is written into the area of address # 9 , the voice packet is output from the buffer  32  to the voice decoder  33  8 s later. In this way, controlling a write address into which a received voice packet is written enables delaying the voice packet for an arbitrary amount in the range of 0 s to 8 s. 
   Therefore, if it is possible to delay a voice packet for an amount of time followed by subtracting an absolute amount of delay time from maximum delay time to be reduced (dmax shown in  FIG. 17 ) provided that we can obtain an absolute amount of delay time since each voice packet was transmitted by the source terminal  10  till it reaches the destination terminal  30 , it would be possible to minimize as well as to fix the total delay time for each voice packet transmitted from the source terminal  30  to the voice decoder  33 . 
   However, the destination terminal  30  is not capable of finding how much propagation delay time it has taken for each voice packet to reach the destination. As a consequence, a conventional delay control for each packet is performed in the following method. For simplicity, we assume here that a series of voice packets transmitted from the source terminal  10  at a certain time interval reaches the destination terminal  30  in the same order as the transmission order. 
   First of all, the destination terminal  30 , upon receiving a first voice packet through the network  20 , writes the voice packet into an initial input location of the buffer  32  (S 1 , S 2  of  FIG. 13A ). In the example shown in  FIG. 13B , the initial input location is an area corresponding to an address whose assigned number is one larger than an area where a voice packet is read at the point of receiving the first voice packet. 
   Then, a voice packet on and after the second packet is written in an area where the readout is performed at the earliest timing among areas that are vacant at the point of receiving the subject voice packet (S 3  of  FIG. 13A ). 
   In the example shown in  FIG. 13B , the first voice packet P 1  received at time t 1  is written in the area of address # 2 , which is the initial input location. Then, no voice packet is received at time t 2 , and the voice packet P 1  is read from the area of address # 2  and sent to the voice decoder  33 . When it turns time t 3 , a second voice packet P 2  is received. It appears to have taken delay time that is 1 s longer than the voice packet P 1  for the voice packet P 2  to be transmitted. Then, the voice packet P 2  is written in an area where the readout is performed at the earliest timing among vacant areas at the receiving time t 3 , that is, the area of address # 3 . Subsequently, at time t 3 , the voice packet P 2  is read immediately after being written and is supplied to the voice decoder  33 . 
   Thus, even if the voice packet P 1  and P 2  are transmitted from the source terminal  10  at 1 s time interval between them, the difference of 1 s in propagation delay time between the two voice packets causes the arrival at the destination terminal  30  at the time interval of 2 s. However, even in such a case, determining an initial input location of the buffer  32  and applying deference by the buffer  32  as described above enables supplying the voice packet P 1  and P 2  to the voice decoder  33  at the same time interval as the transmission interval of the source terminal  10 . In other words, it is possible to reduce delay jitter as large as 1 s by allotting an initial input location for a first voice packet to an area which will be output later than the read area as of the receiving by an area equivalent to 1 s. 
   Looking at a group of serial voice packets transmitted from the source terminal  10  to the destination terminal  30 , their propagation delay time vary from the minimum value dmin to the maximum value dmax as shown for example in  FIG. 17 . In a conventional art, when a first voice packet P 1  is received at the destination terminal  30 , an initial input location is allotted to an area corresponding to an address that will be output later than the readout address as of the receiving by the number of areas equivalent to the delay jitter width D=dmax−dmin, and the voice packet P 1  is written therein. Deciding the initial input location in this way enables the complete elimination of pre-assumed delay jitters. 
   More detailed description will be given hereinafter with reference to  FIGS. 14A ,  14 B,  14 C,  15  and  16 . In the following description, it is assumed that the delay jitter width is 4 s. Also, for the sake of simplicity, we will assume a case where the minimum delay time dmin is 0 s and the delay jitter width of the network  20  is equal to the maximum delay time dmax. 
   In  FIG. 14A , the voice packets P 11  and P 12  are packets output consecutively from the voice encoder  11  of the source terminal  10 . Likewise, the voice packets P 21  and P 22  are packets output consecutively from the voice encoder  11  of the source terminal  10 .  FIG. 14B  illustrates each voice packet that has reached the receiving unit  31  of the destination terminal  30 . In the example shown, the voice packets P 11  and P 12  reach the receiving unit  33 , both being delayed the maximum delay time dmax=4 s. On the other hand, the voice packets P 21  and P 22  reach the receiving unit  31 , the former being delayed the minimum delay time dmin=0 s and the latter being delayed the maximum delay time dmax=4 s.  FIG. 14C  then illustrates each of the voice packets being supplied to the voice decoder  33  after deference being applied. 
     FIG. 15  shows how deference is performed to the packets P 11  and P 12  by the buffer  32 , and  FIG. 16  shows how deference is performed to the packets P 21  and P 22  by the buffer  32 . 
   As shown in  FIG. 15 , the voice packet P 11  that has reached the receiving unit  31  at time t 5  is written in the area of address # 5 , which is the initial input location, thereby being delayed for delay time of 4 s and output from the buffer  32  to the voice decoder  33  at time t 9 . Then, the voice packet P 12  that has reached the receiving unit  31  at time t 6  is written in the area of address # 6 , an area where a readout will be performed at the earliest timing among vacant areas as of the receiving, thereby being output from the buffer  32  at time t 10  that is the next timing of the output time for the voice packet P 11 . 
   On the other hand, deference such as follows is performed for the voice packet P 21  and P 22 . First of all, as shown in  FIG. 16 , the voice packet P 21  that has reached the receiving unit  31  at time t 1  is written in the area of address # 5 , which is the initial input location, thereby being delayed for delay time of 4 s and output from the buffer  32  at time t 5 . Then, the voice packet P 22  that has reached the receiving unit  31  at time t 6  is written in an area where a readout will be performed at the earliest timing among vacant areas as of the receiving, thereby being output immediately from the buffer  32 . 
   As described so far, if an initial input location is set to an area of address which will be output later than the read address as of the receiving by the number of areas equivalent to the delay jitter width D=dmax−dmin, it becomes possible to reduce every delay jitter in the range of the minimum value dmin and the maximum value dmax. 
   However, in the conventional art described above, that a first voice packet received by the destination terminal  10  is delayed for delay time which is equivalent to the delay jitter width D means that the same amount of delay time will be applied for the succeeding voice packets. If it is assumed that delay time required for the first voice packet to pass a network is d 0  here, the total delay time T will be D+d 0 , the total delay time T designating the amount of time required for each voice packet to reach the voice encoder  33  of the destination terminal  30  since the point of being output from the voice encoder  11  of the source terminal  10 . However, the delay time of the first voice packet varies from the minimum value dmin to the maximum value dmax, which in turn makes the total delay time T depended on the delay time d 0  of the first voice packet. That means that, in the case of the delay time d 0  of the first voice packet being the minimum delay time dmin, the total delay time T can be made short. However, in a case where the delay time of the first voice packet is as long as the maximum delay time dmax, the total delay time T results in a long period of time two times the maximum delay time dmax. In recent years, the prevalence of such as an Internet telephony using VoIP (Voice over IP) technique has caused a call for high-quality communication, which requires the shortening in the total delay time. Thus, it is unfavorable that the total delay time T becomes long for the sake of reducing delay jitter. 
   SUMMARY OF THE INVENTION 
   This invention is made for solving the above-mentioned problem and aims at providing a delay jitter reducing device capable of shortening the total delay time and a delay jitter reducing method thereof. 
   In order to solve the above-mentioned problem, this invention provides a delay jitter reducing device, comprising: a receiving unit sequentially receiving chronological data segments through a network; a time detecting unit for obtaining a reception time of each data segment received by said receiving unit; transmission time estimating means for estimating transmission time of each data segment received by said receiving unit; a delay time estimating unit for estimating a delay time required for transmitting each data segment based on said reception time and said transmission time of each data segment; a minimum delay time estimating unit for estimating a minimum delay time in transmitting a data segment through the network from the estimated values of delay time of a plurality of data segments obtained from said delay time estimating unit; relative delay time computing means for obtaining a relative delay time by subtracting said minimum delay time from the estimated value of delay time of a data segment estimated by said delay time estimating unit; and delay means for obtaining an amount of holding time corresponding to each data segment by subtracting the relative delay time of each data segment from a maximum delay time to be reduced, and outputting each data segment after delaying each data segment for the amount of holding time corresponding to each data segment. 
   Such a delay jitter reducing device enables the estimation of a minimum value of delay time required for transmitting data segments such as packets, thereby determining holding time of deference for reducing delay jitter based on the minimum value. As a result, delay jitter of a group of received data segments is reduced as well as the total delay time thereof is shortened. 
   The embodiments of the present invention include an embodiment such as of producing and selling a device which reduces delay jitter as disclosed in the above-mentioned embodiments as well as an embodiment of distributing through a telecommunication line a program for making a network-connected computer function as a delay jitter reducing device as disclosed in the above embodiments and an embodiment of distributing such a program recorded in a computer-readable recording medium. 

   
     BRIEF DESCRIPTION OF THE DRAWINGS 
       FIG. 1  is a block diagram showing an overall configuration of a real-time voice transmission system with respect to a first embodiment of the present invention. 
       FIG. 2  is a block diagram showing a configuration of a destination terminal in the embodiment. 
       FIG. 3  is a block diagram showing a configuration of a delay unit in the embodiment. 
       FIG. 4  is a block diagram showing a configuration of a voice packet in the embodiment. 
       FIG. 5  is a time chart illustrating an operation of the destination terminal in the embodiment. 
       FIG. 6  is a block diagram showing a configuration of a destination terminal in a second embodiment of the present invention. 
       FIG. 7  is a diagram showing a packet notifying the start of a non-voice section. 
       FIG. 8  is a time chart showing an operation of the embodiment. 
       FIG. 9  is a flow chart illustrating an operation of the embodiment. 
       FIGS. 10A ,  10 B, and  10 C are time charts illustrating an operational example of the embodiment. 
       FIGS. 11A and 11B  illustrate an effect of the embodiment. 
       FIG. 12  is a block diagram showing a configuration example of a real-time voice transmission system. 
       FIG. 13A  is a flow chart illustrating an operation of the system. 
       FIG. 13B  is a time chart illustrating the operation of the system. 
       FIGS. 14A ,  14 B, and  14 C are time charts illustrating an example of the system. 
       FIG. 15  is an operational example of the system. 
       FIG. 16  is an operational example of the system. 
       FIG. 17  is an operational example of the system. 
   

   DETAILED DESCRIPTION 
   An embodiment of the present invention will be described hereinafter with reference to the drawings. 
   A. First Embodiment 
     FIG. 1  is a block diagram showing a configuration of a real-time voice transmission system that is a first embodiment of the present invention. In the real-time voice transmission system, there are provided a source terminal  10  with a voice encoder  11  and a transmission unit  12  as in the conventional art. The source terminal  10  and a destination terminal  100  are both VoIP terminals. This real-time voice transmission system is for providing an Internet telephone service to a user. 
     FIG. 2  is a block diagram showing a configuration of the destination terminal  100 . In this figure, a receiving unit  101  is a device which receives voice packets from the source terminal  10  through the Internet  20 . A packet terminating unit  102  is a device that terminates a protocol of the Internet  20 . A voice packet received by the receiving unit  101  is transmitted through the packet terminating unit  102  to a time stamp detecting unit  108  and a delay time estimating unit  106 . Also, the packet terminating unit  102  fetches coded voice data from the payload section of the received voice packet and supplies the data to a delay unit  103 . 
   An internal clock generator  107  generates an internal clock of a certain frequency and supplies the generated clock to the delay time estimating unit  106  and a delay unit  103 . 
   The delay unit  103  is supplied with data of holding time from a holding time setting unit  104 . The description will be given later on how to generate the data of holding time. The delay unit  103  is a device that supplies a voice decoder  110  after holding coded voice data that have been supplied from the packet terminating unit  102 . The delay unit  103 , as shown for example in  FIG. 3 , comprises a RAM  103 A, a write circuit  103 B for writing coded voice data supplied from the packet terminating unit  102  into the RAM  103 A, and a read circuit  103 C for reading out coded voice data from the RAM. The read circuit  103 C counts an internal clock supplied from the internal clock generator  107 , supplies the counted value to the RAM  103 A as a read address, reads out coded voice data from an area in the RAM  103 A corresponding to the read address, and outputs the data to the voice decoder  110 . When coded voice data of a voice packet is output from the packet terminating unit  102 , the write circuit  103 B obtains a write address based on a read address that is output from the read circuit  103 C as of that point and data of holding time that is output from the holding time setting unit  104 . Then, the write address is supplied to the RAM  103 A, and the coded data of the voice packet is written into an area corresponding to the write address of the RAM  103 A. The coded voice data written in the RAM  103 A, when time corresponding to the data of holding time elapses at a later time, are read from the ROM  103 A and output to the voice decoder  110 . 
   The voice decoder  110  is a device which decodes voice data from coded data that are output from the delay unit  103 . 
   The time stamp detecting unit  108 , the delay time estimating unit  106 , a minimum delay time estimating unit  105 , and the holding time setting unit  104  cooperate to form a means for generating data of holding time. 
   As described already, the time stamp detecting circuit  108  is supplied with voice packets received by the receiving unit  101 . The source terminal  10  ( FIG. 1 ), where the voice packets are originated, contains a counter that counts a clock of predetermined frequency and outputs time data designating a current time and reads the time data from the counter at the point of generating a voice packet, so that the time data is included in the header of the voice packet as a time stamp.  FIG. 4  is an example of voice packets with such a time stamp in the header. The time stamp detecting circuit  108  fetches the time stamp from the received voice packet and send it to the delay time estimating unit  106 . 
   The internal clock generator  107  outputs an internal clock of the same frequency as that of the clock used in the source terminal  10 . The delay time estimating unit  106  counts an internal clock that is output from the internal clock generator  107  and generates time data designating a current time. This time data almost coincides with the time data generated in the source terminal  10 , but there is no assurance of the complete coincidence. However, both time data units are generated by counting a clock whose frequency is identical to each other. Therefore, the difference in time between both time data units is fixed. The delay time calculating circuit  106 , when a time stamp of a voice packet is supplied from the time stamp detecting circuit  108 , obtains an estimated value of delay time required for the transmission of a voice packet by subtracting the time stamp from the time data of the receiving time of the voice packet. 
   The minimum delay time estimating unit  105  is a device for estimating a minimum delay time required for the transmission of a voice packet. The minimum delay time estimating unit  105  sequentially receives from the delay time estimating unit  106  estimated values of delay time of voice packets that have been received in sequence by the receiving unit  101 . Every time the minimum delay time estimating unit  105  receives an estimated value, it selects the smallest value among estimated values of delay time up to that point and regards the selected value as an estimated value of the minimum delay time. 
   The holding time setting unit  104  is a device which, every time a voice packet Pi (i=0,1,2, . . . ) is received, computes data of holding time da corresponding to the voice packet Pi from the equation below:
 
 da=d min+ D−di   (1)
 
   where di is delay time of a voice packet Pi estimated by the delay time estimating unit  106 , dmin is a minimum delay time of all the voice packets up to the voice packet Pi, and D is a pre-set maximum delay time. 
   The data of holding time da is used in computing a write address for writing a coded voice data unit of a voice packet into the RAM  103 A, as described above. 
     FIG. 5  is a diagram showing an operation of the present embodiment. An operation of the present embodiment will be described with reference to the figure. 
   In the destination terminal  100 , when a first voice packet P 0  is received, the delay time estimating unit  106  will calculate an estimated value of delay time according to the following equation from reception time c 0  and time t 0  designated by the time stamp fetched from the voice packet P 0 :
 
 d   0 = c   0 − t   0   (2)
 
   from which, in the example shown, delay time of the first voice packet P 0  is found out to be 7 s. 
   Then, the minimum delay estimating unit  105  regards the d 0 =7 s as an initial estimated value of the minimum delay time dmin. 
   Subsequently, the holding time setting unit  104  obtains data of holding time da corresponding to the voice packet P 0  as follows: 
                   da   =       d   ⁢           ⁢   min     +   D   -   d0                 =       7   ⁢   s     +     12   ⁢   s     -     7   ⁢   s                   =     12   ⁢   s                   (   3   )             
 
where D is set 12 s in this example.
 
   The data of holding time da obtained by the holding time setting unit  104  is sent to the delay unit  103 . The delay unit  103  delays the coded voice data of the voice packet P 0  for an amount of time equivalent to the data of holding time da to supply the coded data to the voice decoder  110 . 
   When a subsequent voice packet Pi is received at a later time, the delay time estimating unit  106  calculates an estimated value of delay time according to the following equation from reception time data ci and time ti designated by a time stamp fetched from the voice packet Pi.
 
 di=ci−ti   (4)
 
   Then, the minimum delay time estimating unit  105  compares the di against an estimated value of the minimum delay time dmin as of that point, and maintains the current estimated value dmin of the minimum delay time when it is found di≧dmin; when it is found di&lt;dmini, dmin is replaced with a value of di. 
   The holding time setting unit  104  computes data of holding time da corresponding to the voice packet Pi from the aforementioned equation (1). Then, the delay unit  103  delays the coded voice data of the voice packet Pi for an amount of time equivalent to data of holding time da to supply the coded data to the voice decoder  110 . 
   The above operation is performed for all the voice packets. 
   In the beginning of a session, an estimated value of the minimum delay time dmin is updated relatively often. However, the more voice packets are received and the more times the minimum delay time is estimated, the closer the estimated value of the minimum delay time dmin becomes to a true value of the minimum delay time. Therefore, as a time interval for updating the estimated value of the minimum delay time dmin becomes longer, the estimated value of the minimum delay time dmin becomes stabilized. In the example shown, an estimated value of the minimum delay time dmin changes in a way such as becoming 7 s at the point of receiving the voice packet P 0 , 6 s at the point of receiving the voice packet P 2 , 4 s at the point of receiving the voice packet P 6 , and 3 s at the point of receiving the voice packet P 12 . 
   Total delay time T since a voice packet was output from the voice encoder  11  of the source terminal  10  until coded voice data thereof are output to the voice decoder  110  of the destination terminal  110  is obtained from the following equation: 
                   T   =     di   +   da                 =     di   +     d   ⁢           ⁢   min     +   D   -   di                 =       d   ⁢           ⁢   min     +   D                   (   5   )             
 
   As shown for example in  FIG. 5 , as more voice packet are received, the estimated value of the minimum delay time dmin gradually converges into a small value. As a result, the total delay time T also gradually converges into a small value. 
   Since total delay time T depends on an estimated value of the minimum delay time, the total delay time T changes relatively often in the beginning of a session. However, the more voice packets are received, the longer a time interval for updating the total delay time T becomes, and the value of total delay time T finally reaches a minimum value. 
   B. Second Embodiment 
     FIG. 6  is a block diagram showing a configuration of a destination terminal  100  with respect to a second embodiment of the present invention. The destination terminal  100  in this embodiment further contains a non-voice section detecting unit  109  in addition to the components of the destination terminal  100  for the first embodiment. The non-voice section detecting unit  109  monitors the payload of voice packets received in sequence and detects non-voice sections. To describe further in detail, a source terminal  10  in the present embodiment, when a user of the terminal  10  stops vocalization and a non-voice section in which there is no voice to be transmitted begins, transmits to the destination terminal  100  a voice packet which includes information designating the start of the non-voice section in the payload as shown in  FIG. 7 . The non-voice section detecting unit  109  of the destination terminal  100 , by receiving this voice packet, detects the start of a non-voice section. When the destination terminal  100  receives a voice packet including some kind of coded voice data in the payload at a later time, the non-voice section detecting unit  109  detects the end of the non-voice section. 
   Subsequently, a holding time setting unit  104  in the present embodiment, when the end of the non-voice section is detected by the non-voice section detection unit  109 , computes data of holding time da from an estimated value of delay time for a first voice packet of a voice section obtained from a delay time estimating unit  106 , an estimated value of the minimum delay time obtained from a minimum delay time estimating unit  105  at that point, and a known delay jitter width, the result being output to a delay unit  103 . The computing of data of holding time and the supplying of the data to the delay unit  103  are performed every time non-voice section begins. 
     FIG. 8  is a time chart showing an operation of the destination terminal  100  with respect to the present embodiment, and  FIG. 9  is a flowchart showing an operation of the destination terminal  100  with respect to the present embodiment. The operation of the present embodiment will be described hereinafter with reference to these figures. 
   When a phone-to-phone conversation between the source terminal  10  and the destination terminal  100  is initiated, a voice section and non-voice section are repeated alternately as shown in  FIG. 8 , the voice section being a period where voice packets representing the voice of a caller are received by the destination terminal  100  and the non-voice section being a period where no voice packets are received. 
   As in the first embodiment, every time a voice packet is received by the receiving unit  101 , the delay time estimating unit  106  obtains an estimated value of delay time for the voice packet (step S 101  and S 102 ). 
   In a first voice section SP 0 , the minimum delay time estimating unit  105  considers an estimated value of delay time for a first voice packet P 0  to be an estimated value of the minimum delay time dmin (step S 103  and S 104 ). As for each of the received voice packets in the first voice section SP 0 , data of holding time da is computed from the aforementioned equation (1), and a result thereof is set to the delay unit  103  (step S 105 ). In the delay unit  103 , a write address is found out from the data of holding time da and a read address of a RAM  103 A as of that point. Then, coded voice data of a voice packet is written into an area of the RAM  103 A corresponding to the write address. The coded voice data, after time has elapsed by an amount of time equivalent to the data of holding time da, are read from the RAM  103 A and supplied to a voice decoder  110  (step S 106 ). 
   Then, when a voice packet as illustrated in  FIG. 7  is received by the receiving unit  101 , the non-voice section detecting unit  109  detects the start of a non-voice section NP 0  . Instead of transmitting a packet for notifying the start of a non-voice section from the source terminal  10  to the destination terminal  100  in such a way, it is also possible to detect the start of a non-voice period when a voice packet is not received over a certain period at the destination terminal  100 . 
   We assume that the voice section SP 0  changes to the non-voice section NP 0  and that a subsequent voice section SP 1  begins at a later time. When a first voice packet P 0  of the voice section SP 1  is received by the receiving unit  101 , the delay time estimating unit  106  finds out an estimated value of delay time d 0  of the voice packet P 0  (S 101  and S 102  in  FIG. 6 ). 
   Subsequently, the minimum delay time estimating unit  105  estimates a minimum delay time dmin from among estimated values of delay time for all the voice packets that have been received up to that point (step S 104 ). In the present embodiment, an estimated value of the minimum delay time can be updated only when a first voice packet of a voice section is received. In other words, once a voice section begins, the estimated value of the minimum delay time is not updated even if a value of delay time is estimated to be smaller than that of the minimum delay time at the beginning. It is when the voice section ends to turn to a non-voice section and another voice section begins that the update can be made. 
   At the point of receiving the first voice packet P 0  of the voice section SP 1 , the holding time setting unit  104  obtains the estimated value of the minimum delay time dmin from the minimum delay time estimating unit  105  (step SP 104 ). 
   Subsequently, the holding time setting unit  104  computes data of holding time da from the aforementioned equation (1), and supplies a result to the delay unit  103  (step S 105 ). 
   In the delay unit  103 , a write address is found from the data of holding time da and a read address of the RAM  103 A of that point. Then, coded voice data of the voice packet P 0  are written in an area of the RAM  103 A corresponding to the write address (step S 106 ). 
   In the voice section SP 1  as in the voice section SP 0 , an estimated value of delay time di is calculated as to a voice packet Pi received by the receiving unit  101  (step S 102 ). The estimated value of delay time di obtained in the voice section SP 1  is used for estimating a minimum delay time when a voice section SP 2  is started at a later time (step S 103  and S 104 ). 
   The operation of the present embodiment will be described further in detail with concrete examples shown. 
     FIG. 10A  shows voice packets that are output in sequence from the voice encoder  11  of the source terminal  10 .  FIG. 10B  shows voice packets that are received in sequence by the receiving unit  101  of the destination terminal  100 .  FIG. 10C  shows voice packets that are output in sequence to the voice decoder  110 . As shown in  FIG. 10B , voice packets P 0 , P 1 , P 2 , and P 3  serially output from the voice encoder  11  reach the receiving unit  101 , each having delayed d 0  (=3 s), d 1  (=4 s), d 2  (=2 s), and d 3  (=2 s). During this period, an estimated value of delay time di output by the delay time estimating unit  106  and an estimated value of maximum delay time dmin in the minimum delay time estimating unit  105  will change as follows: 
   
     
       
         
             
             
             
           
             
                 
             
             
               received 
               estimated 
               estimated minimum 
             
             
               packet Pi 
               delay time di 
               delay time dmin 
             
             
                 
             
           
          
             
               P0 
               3s 
               3s 
             
             
               P2 
               2s 
               2s 
             
             
               P1 
               4s 
               2s 
             
             
                 
             
          
         
       
     
   
   Because an estimated value of the minimum delay time dmin is not available in the first voice section SP 0 , the addition of the network delay jitter width D and is 1 s used as data of holding time da. Therefore, supposing that the delay jitter width D is 3 s, the data of holding time will be 4 s. Given that d 0 =3 s in the example shown, the total delay time of serial voice packets P 0  to P 2  turns out to be d 0 +da=3 s+4 s=7 s. 
   To the contrary, in the next voice section SP 1 , the minimum delay time dmin (=2 s) is obtained from estimated values of delay time obtained up to that point, and based on the dmin (=2 s) the holding time will be determined. 
   Hence, supposing that voice packet P 3  is transmitted with delay d 3 =1 s in the voice section SP 1  as shown in  FIG. 10A and 10B , the data of holding time will be as follows: 
             da   =     (       d   ⁢           ⁢   min     +   D   -   d3     )                 =       2   ⁢   s     +     3   ⁢   s     -     1   ⁢   s                   =     4   ⁢     s   .                 
 
   The total delay amount for each voice packet of the voice section SP 1  starting from voice packet P 3 , in turn, becomes d 3 +da=1 s+4 s=5 s. 
     FIGS. 11A and 11B  show an effect of the present embodiment. Supposing that delay time for a first voice packet is d 0  in a first voice section SP 0 , the total delay time d 0 +D of each voice packet for voice section SP 0  will be d 0 +D. 
   When a voice packet is received with the minimum delay time dmin=3 in voice section SP 0 , and in a subsequent voice section SP 1  the holding time determined based on this minimum delay time is applied. As a result, the total delay time will be d 3 +D. 
   In a conclusion, the present embodiment enables the reduction of the total delay time by deciding the amount of holding time based on a minimum delay time estimated based on estimated delay time of received packets. Also, updating the minimum delay time at the point of receiving a first voice packet after a non-voice section keeps voice quality from deteriorating. For these reasons, the delay jitter reducing device and reducing method are well suited for an application requires real-timeliness and high voice quality such as the Internet telephony. 
   C: Modifications 
   The present invention is not limited to the above-described embodiments, but various modifications such as are exemplified below are possible. 
   (1) In the above-described embodiments, the present invention is applied to a device that receives data segments such as packets through the Internet. However, the present invention may be applied to a device that receives data segments through a wide-area network such as, for example, a frame relay, not being limited to the Internet. The present invention can also be applied to a device that receives data segments through a network where delay jitter is produced in a wireless section as in the mobile network.
 
(2) In the above-described embodiments, a packet is shown as an example of a data segment. However, a form of data segment is not limited to a packet. Data segments may be anything that includes transmission time or any clue information for finding the transmission time. Data segments may be in any unit such as frames and cells depending on a transmission path or a protocol to be used. Protocols may be the VoIP such as is described above or such things as the Voice over Frame Relay (VoFR).
 
(3) In the above-described embodiment, the present invention is applied for a device that receives voice packets through a network. However, the present invention is well-suited for transmission of not only voice but video and information requiring real-time transmission.
 
(4) In the above-described second embodiment, the present invention is applied to a real-time voice transmission in which a voice section and a non-voice section is alternately repeated, where in the voice section voice packets are consecutively transmitted and in the non-voice section the transmission of voice packets is not performed for a consecutive period of time. In this embodiment, the holding time of a voice section is decided based on an estimated value of the minimum delay time acquired in a first previous voice section. However, the application of the present invention is not limited thereto. For example, another form of data transmission is that a first section and a second section repeat by turns, where in the first section information requiring continuity such as motion pictures are transmitted and in a second section information not requiring continuity such as still pictures are transmitted. The present invention can be applied to such a form of data transmission. In this application, the following procedure for reducing the delay jitter will be performed at the destination device:
     i) during a period of receiving data segments including information of a second section not requiring continuity, delay time of each data segment and a minimum delay time are estimated;   ii) when receiving a first data segment of a first section right after the second section, delay time of the first data segment is estimated; and   iii) based on the above estimated value of the minimum delay time and the estimated value of delay time for the first data segment acquired in the above ii), data of holding time for the first data segment is computed. The computing method is same as what has been described in each of the above embodiments.
 
(5) In the above-described second embodiment, no packets are transmitted in a non-voice section, but it is also possible to keep transmitting data that designates it being a non-voice section.
 
(6) In each of the above-described embodiments, the delay jitter width is a fixed value acquired by measuring the value in advance. However, when the delay jitter width turns out to be bigger than the initially supposed amount, it is possible to update the delay jitter width D to be used for computing data of holding time so that such a large delay jitter can be reduced. In the above second embodiment, for example, we assume that it follows from equation (1) that the data of holding time is −3 s, the result being computed based on an estimated value of delay time d 0  of a first packet in a voice section SPk, an estimated value of the minimum delay time acquired in the previous voice section SPk−1, and the delay jitter width D. This is because an actual delay jitter width is at least 3 s larger than the initially supposed delay jitter width D. Therefore, the delay jitter width D is to be incremented 3 s, so that the data of holding time becomes 0 s. This renewed delay jitter width D is used for computing data of holding time from equation (1) in the subsequent voice section SPk+1.
 
(7) A device for reducing delay jitter with respect to the present invention can be provided with a relay device of a network or a router, for example. This modification is for the sake of reducing delay jitter in the middle of a transmission path because a long transmission path leads to a long delay jitter width.
 
(8) The minimum delay time maybe estimated in a certain limited period. To illustrate, the following example can be conceived. First, in the beginning of a session, before initiating the voice packet transmission, a training packet including a time stamp is repeatedly transmitted from a source terminal to a destination terminal. At the destination terminal, a minimum delay time dmin is estimated from estimated values of delay time for these individual training packets. Data of holding time da applied to a subsequent voice packet is obtained from the aforementioned equation (1) using the dmin.
 
(9) In the above-mentioned embodiment, a transmission time of a packet is estimated from a time stamp. However, in a case where a time stamp is not included in a packet, it is possible to estimate the transmission time from such things as serial numbers included in a packet.
 
(10) The embodiments of the present invention include an embodiment such as of producing and selling a device which reduces delay jitter as disclosed in the above-mentioned embodiments as well as an embodiment of distributing through a telecommunication line a program for making a network-connected computer function as a delay jitter reducing device as disclosed in the above embodiments and an embodiment of distributing such a program recorded in a computer-readable recording medium.