JP2000224225A - Packet receiver, packet reception method and recording medium - Google Patents

Abstract

PROBLEM TO BE SOLVED: To enhance communication quality by suppressing aborted packets due to traffic fluctuation in a relay network. SOLUTION: A reception packet 7 received from a relay line 61 via a WAN 6 is fed to a reception buffer 41 via a CPU 2 and a CPU bus 21, and stored therein in a received sequence. A pointer pointing out the reception packet 7 stored in the reception buffer 41 is stacked to a jitter queue 303 in the order of transmission by packet processing 301 of a DSP 31 and rearranged. A jitter flow control 302 conducts recovery or scale revision processing and a decoder 304 reads voice data of a corresponding packet from the reception buffer 41 in the order of pointers of the jitter queue 303 in a read timing in response to the jitter amount, decodes the data and the recovered voice signal is sent to a reception signal line 305.

Description

DETAILED DESCRIPTION OF THE INVENTION

[0001]

BACKGROUND OF THE INVENTION 1. Field of the Invention The present invention relates to a packet receiving apparatus, a packet receiving method, and a recording medium. The present invention relates to a packet receiving apparatus and a packet receiving method for reproducing an original audio signal from a packet, and a recording medium.

[0002]

2. Description of the Related Art In a conventional jitter buffer control method, C
The PU performs the sequence interchange control and the jitter control for each DSP, and only the packets that have reached the read timing are
In this method, the data is transferred from the memory of the CPU to the jitter buffer of the corresponding DSP. Therefore, the CPU has a reception buffer, a reception queue, and a jitter queue for the number of DSPs, and performs jitter control for the number of DSPs.

Further, in the conventional jitter buffer control method, a fixed delay is added to a first received packet on the assumption that a relay path is a dedicated connection, the fluctuation is small, and the delay is constant. Was realized.

For example, when the number of relay stages in a route through which a voice call passes increases, the delay time increases, and the waiting time varies depending on the traffic volume of each relay device, so that the fluctuation also increases. Jitter buffer control is a mechanism for intentionally delaying and reproducing a received packet so that even if these delays and fluctuations occur, there is no problem in communication.

[0005]

As described above, in the conventional jitter buffer control method, a value (delay time) determined from the delay time expected at the time of installation of the device is set in the device, and a fixed delay is applied to the device. I was taking. This method is
This is effective in a company network or the like in which the number of connection stages of the device is constant and a dedicated line is connected.

However, IP (Internet)
When applying to a new technical field called protocol (telephony), the use of a public network as a relay path and communication without connection (the route through each packet is not constant) make it impossible to specify the number of connection stages. When the amount of traffic in the public network increases, the delay time in the network cannot be absorbed by the jitter buffer control, and an underflow / overflow occurs. As a result, received voice packets are discarded, and as a result, sound is interrupted. Voice quality was reduced.

Therefore, if the range of the jitter buffer control is extended to the public network and the maximum delay time considering the public network is set in the device as a fixed delay, the set time is added to the original delay time. Therefore, there is a problem that any call takes a long delay, giving a sense of incongruity.

Further, as described above, since the IP telephony is connectionless communication unlike the conventional telephone service, a route that passes through each packet is different, and the order of transmission and the order of reception occur. For this reason, there is a problem that the rearrangement of the order must be considered.

In order to solve these problems, a jitter buffer control is performed by using an optimum value (optimal delay time) at the moment of talking while anticipating the occurrence of a sound quality deterioration factor. Set the delay time to the optimum value, that is, set the delay time that is inserted unnecessarily by the jitter buffer control to the necessary minimum, although it is unavoidable to remove the fixed delay time generated in the communication path. Is required.

The present invention has been made in view of such a situation, and it is possible to improve the sound quality of a call by performing jitter buffer control based on an optimum delay time according to the call situation. Is what you do.

[0011]

According to a first aspect of the present invention, there is provided a packet receiving apparatus for receiving and decoding a packet transmitted from a transmitting side via a relay network on a receiving side. Receiving means for receiving packets, storage means for storing the packets received by the receiving means in the order in which the receiving means received the packets, and rearranging means for rearranging the packets in the order transmitted by the transmitting side; Calculating means for calculating the amount of jitter generated in the relay network from the time when the packet is received by the means; and decoding the packets rearranged by the rearranging means in accordance with the amount of jitter calculated by the calculating means. Timing control means for controlling timing;
Decoding means for decoding the packet at the timing controlled by the timing control means. Also, in the packet received by the receiving means,
An additional unit for adding address information of a storage unit in which a packet next to the packet is stored is further provided.
According to the address information, it can be stored in the storage means. Further, an address storage means for storing the address of the storage means in which the packet is stored is further provided, and the rearranging means stores the address of the storage means in which the packet is stored in the address storage means in the order in which the transmission side transmitted the address. By storing the information, the packets can be rearranged. Further, an address storage means for storing the address of the storage means in which the packet is stored is further provided, and the rearranging means stores the address of the storage means in which the packet is stored in the address storage means in the order in which the transmission side transmitted the address. By storing the packet, the packets are rearranged, and the decoding unit reads the address stored in the address storage unit according to the jitter amount calculated by the calculation unit, and stores the read address in the address of the storage unit. Packets can be read and decrypted. The packet includes at least a field for describing address information in which the next packet is stored, a field for describing a packet length, a field for describing a reception time stamp, and a transmission sequence number. It consists of a field for describing, a field for describing a transmission time stamp, and a field for describing data. At least the address information in which the next packet is stored and the reception time stamp are , And at least the transmission sequence number can be set on the transmission side. Further, an estimating means for estimating the time at which the transmitting side transmitted the packet based on the time at which the packet was received by the receiving means may be further provided. The packet receiving method according to claim 7, wherein the receiving side receives and decodes a packet transmitted from the transmitting side via the relay network, and receives the packet. A storing step of storing the packets received in the receiving step in the order in which the packets were received, and a rearranging step of rearranging the packets in the order transmitted by the transmitting side, at least,
From the time when the packet was received in the receiving step,
A calculating step of calculating the amount of jitter generated in the relay network; a timing controlling step of controlling a timing of decoding the packets rearranged in the rearranging step in accordance with the amount of jitter calculated in the calculating step; A decoding step of decoding the packet at a timing controlled in the control step. The recording medium according to claim 8,
A program capable of executing the packet receiving method according to claim 7 is recorded. In the packet receiving device and the packet receiving method according to the present invention, and in the recording medium, the packet is received, the received packet is stored in the order in which the packet was received, the packets are rearranged in the order transmitted by the transmitting side, at least, From the time at which the packet was received, the amount of jitter generated in the relay network is calculated, and the timing of decoding the rearranged packets is controlled according to the calculated amount of jitter. Decrypt.

[0012]

FIG. 1 is a block diagram showing a configuration example of an embodiment of a voice packet receiving apparatus to which the packet receiving apparatus of the present invention is applied. As shown in the figure, an audio packet receiving device 1 that receives an audio packet includes a CPU (central control unit (central pr) that controls each unit.
processing unit)) 2, CPU bus 21,
A DSP (digital signal processing unit (digital signal processor) that decodes a received voice packet and converts it into voice data.
l processor)) 31 to 3n (n is DS
P), and receiving buffers (receiving buffer and jitter buffer for each DSP) 41 to 4n for temporarily storing received voice packets.

The CPU 2 has reception buffers 41 to 4n.
Are stored. The write pointers 201 to 20n indicating the respective write addresses are stored. Also, DSP
31 to 31n each include a packet processing 301, a jitter amount control 302, a jitter queue 303, and a decoder (speech decoder) 304, and a PBX (private branch exchange).
e)) or connected to the reception signal line 305 connected to the telephone. Then, a read pointer 306 indicating the head (header read address) of the reception buffer and a replenishment pointer 307 indicating the replenishment destination of the empty buffer are stored.

The receiving buffers 41 to 4n are provided in the DSP 31
To 3n, and a memory bus 5
1 to 5n correspond to the DSPs 31 to 3n and the buffers 41, respectively.
To 4n to exchange data.

Further, this embodiment includes a WAN (relay line control unit) 6 and is connected to the outside via a trunk line 61. The WAN 6 supplies a received packet (voice packet) 7 received via the relay line 61 to the CPU 2. The timepiece 8 performs a timekeeping operation, and supplies a result of the timekeeping to each unit via the CPU bus 21 as appropriate.

FIG. 2 shows a configuration example of the jitter queue 303 of the present invention. FIG. 3 shows a loading procedure (sequence correcting procedure) of the jitter queue 303. The jitter queue 303 is a jitter queue that is also used as a transmission sequence correction queue, and #x indicates the value of the least significant byte of the queue number and the transmission sequence number.

A pointer (HP) 2002 in FIG. 2 is a pointer indicating the head of the jitter queue 303 and is represented by a queue number. The virtual ruler 2003 indicates the jitter amount and the voice packet reading point. The pointer 2004 indicates a pointer of the reception buffers 41 to 4n (FIG. 3).
100, 200,. . . , 900).

The voice packet 2005 is a voice packet received from the relay network via the relay line 61, and #x indicates a transmission sequence number. In addition, it is stored in a predetermined location of the reception buffers 41 to 4n indicated by the pointer 2004. The audio data 2006 represents reproduced audio data transmitted to the reception signal line 305.

The jitter queue 303 has, for example, a queue length of 12 so that a fluctuation of up to ± 600 ms can be realized in a packet cycle C (= 10 milliseconds (ms)).
It is set to 8. The jitter queue 303 has a configuration adopting a ring format corresponding to the lower 7 bits of the transmission sequence number i, and a pointer (HP) 2002 indicating the head transmission sequence number i of the jitter queue 303 is set. The transmission sequence number # 128 is the jitter queue 3
After one round of 03, it is loaded into # 0.

For example, when the HP 2002 points to the transmission sequence number # 0, if an audio packet with the transmission sequence number # 128 is received, since the audio packet with the transmission sequence number # 0 still remains in the jitter queue 303, the transmission is performed. It is interpreted that the voice packet with the sequence number # 128 arrives too early, and this voice packet is discarded and not reproduced. Conversely, HP 2002 transmits the transmission sequence number #
When the voice packet with the transmission sequence number # 0 is received when pointing to 128, the voice packet with the transmission sequence number # 0 is interpreted as having arrived too late, and this packet is discarded and not reproduced.

The jitter queue 303 has a virtual ruler 2003 indicating the amount of jitter in synchronization with the HP 2002. Although the depth of the jitter queue 303 is 128, the amount of jitter that is adaptively controlled is within the range indicated by the virtual ruler 2003. As a result of the jitter buffer control, the jitter queue 3
The number of voice packets loaded in 03 is always before and after the midpoint.

FIG. 4 shows an actual packet flow. FIG. 5 illustrates a packet flow after correcting the clock skew in the receiving apparatus 3004 in FIG. 4 and an algorithm for calculating the amount of jitter.

As shown in FIG. 4, voice (transmitted sound) spoken by telephone 3001 is digitized / encoded by transmitting apparatus (voice packet transmitting apparatus) 3002 and has a constant period (10 ms in the present embodiment). ) And sent to the relay network 3003 as a transmission packet. Relay network 3
The packet transmitted via 003 is received by a receiving apparatus (voice packet receiving apparatus) 3004, and is decoded and converted to analog through jitter buffer control.
It is delivered to telephone 3005. In the case of this example, the transmission packets are received in the order of # 0, # 1, # 3, # 4, # 2 and are regarded as reception packets. Also, the receiving device 3004
Has the same configuration as the voice packet receiving device 1 shown in FIG.

FIG. 6 shows an audio packet format on the reception buffer 4x (x is 1 to n). In the field of the next buffer pointer 4001, a pointer indicating the start address of the next buffer is set. In the field of the command (voice packet reception) 4002,
A command number indicating voice packet reception is set.
In the field of the voice packet length 4003, the length of the voice packet is set. Reception time stamp (Ri)
In the field 4004, a time stamp indicating the reception time is set. The command 4002, the voice packet length 4003, and the reception time stamp 4004
Has been made to be set by.

In the fields of the other information (1) 4005 and the fields of the other information (2) 4008 among the fields of the audio header, header information not related to the jitter buffer control processing is set. In the field of the transmission sequence number 4006, a transmission sequence number i indicating the order of transmission is set. The transmission time stamp (Si) indicating the transmission time is set in the field of the transmission time stamp (Si) 4007, but is not an essential element. In the field of the audio data 4009,
The digitized audio data is set.

The receiving buffer 4x is configured based on a pointer chain system indicating the head address of the next buffer.
Has a write pointer 201, and when writing is completed,
The next buffer pointer 4001 of the completed buffer is
The write pointer 201 is set.

The DSP 31 has a read pointer 30
6, when reading is completed, the next buffer pointer 4001 of the completed buffer is set to the reading pointer 306. By doing so, the receiving order is maintained. The case of the write pointers 202 to 20n of the CPU 2 and the case of the read pointer 306 of the DSPs 32 to 3n are also the same as those described above, and thus description thereof is omitted here.

When the DSP 31 completes the reproduction of the audio packet, the empty buffer sets the next buffer pointer 4001 to 0 (a code indicating that there is no next pointer) and connects it to the position after the supplementary pointer 307. That is, the refill pointer 3
Next buffer pointer 400 of the buffer pointed to by 07
The address of the buffer to be refilled is set in both 1 and the refill pointer 307.

Next, the operation of receiving a voice packet will be described with reference to the flowcharts of FIGS. 1, 6, and 7 to 10 by taking a packet received via the Internet network as an example. 7 to 10 show the DSP 3x (x is 1
It is a flowchart for demonstrating the audio | voice packet receiving process in a through n. First, the reception buffer 4x (x
Are 1 to n) and the jitter queue 303 is cleared to 0.

Write pointer 201, read pointer 30
6. The replenishment pointer 307 indicates the start (least significant) address of the reception buffer 41. Prior to communication, the DSP 31 writes the chain destination buffer address to the next buffer pointer 4001 of each of the reception buffers 41 to 4n (each of the reception buffers 41 to 4n is referred to as one block of the reception buffer) using the supplementary pointer 307. , A pointer chain is generated in advance. After the pointer chain is generated, the supplementary pointer 307 is stored in the reception buffer 41.
4n, the next buffer pointer 4001 of the last receive buffer 4n is set to “0” indicating that there is no chain destination.

The voice packet receiving apparatus 1 includes a trunk line 61
When the voice packet 7 is received, the WAN 6
After performing the above processing, the received voice packet (hereinafter, the received voice packet is appropriately referred to as a received packet) 7
Is transferred to the CPU 2.

The CPU 2 performs a header process on the received packet 7 and outputs an IP (intern) included in the received packet 7.
After confirming that the received packet 7 is addressed to the own device (speech packet receiving device) 1 based on the “E.protocol” header, the UDP (User Datagram)
Protocol), the receiving DSP is determined from the TSAP identifier indicated by the header. Here, it is assumed that the DSP 31 is determined to be the receiving DSP. Then, the voice packet (see FIG. 6) with the reception time stamp (Ri) added to the buffer address indicated by the write pointer 201 of the reception buffer 41 for the determined DSP 31 is transferred, and the write pointer 201 is The next buffer pointer 4001 indicated at the head of the buffer is rewritten.

Since the CPU 2 does not check the transmission sequence number of the audio header, the reception buffer 41 stores the audio packets in the order of reception.

The DSP 31 periodically refers to the buffer indicated by the head pointer (read pointer) 306 of the reception buffer 41, and when confirming the reception of a new packet, as shown in the flowchart of FIG. The packet is read (step S1), and a process such as a check of a voice header is performed in a packet process 301 (step S2).
If it is determined that there is a jitter, a process of loading the value of the read pointer 306 into the jitter queue 303 also serving as the transmission sequence correction queue (see FIGS. 2 and 3) is performed (step S).
4) The value of the head pointer (read pointer) 306 is
The voice packet is updated to the value of the next buffer pointer 4001 indicated at the head of a predetermined block of the reception buffers 41 to 4n from which the voice packet has been read (step S6).

Next, the jitter amount control 302 is activated to calculate the jitter amount n and the read timing R from the value of the reception time stamp (Ri) 4004 and the value of the transmission time stamp (Si) 4007 of the received packet 7. (Step S7) (see FIGS. 4 and 5).

On the other hand, if it is determined in step S3 that there is no jitter, the packet is discarded (step S3).
5) Then, the process proceeds to step S6, and the processes after step S6 are executed.

When a reproduction instruction is received from the jitter amount control 302, as shown in the flowchart of FIG. 9, a pointer is read from the position indicated by the HP 2002 in the jitter queue 303 (step 21), and the reception buffers 41 to 4n pointed to by the read pointer. Is read out from the audio data (Voice data) stored in the predetermined block (step S).
22), decode (step S23), and
5 (step S24) and output to the HP 200
0 (empty code) is set at the position indicated by 2, and the reception buffer is released (step S25). Next, the supplement pointer is updated (step S26), the jitter buffer is released (step S27), and the value of the HP 2002 is incremented by 1 (step S28).

Next, with reference to FIGS. 2 and 3, a description will be given of the audio packet sequence correction processing and the jitter queue loading / reproduction processing. FIG. 2 shows an initial state of the jitter queue 303. FIG. 3 shows a state in which the received packet 7 is loaded in the jitter queue 303 also for sequence correction.

First, the reception buffer pointer "100"
Since the voice packet with the transmission sequence number # 0 has been received, "100" is written in the jitter queue 303 at # 0.
Next, since the transmission sequence number # 1 has been received by the reception buffer pointer “200”,
Write "200" to Next, the reception buffer pointer "
Since the transmission sequence number # 3 is received at "300", "300" is written into # 3 of the jitter queue 303. Similarly, the reception buffer pointer is set to the transmission sequence number of the jitter queue 303 in the order of reception. Write in the corresponding position.

Next, the reproduction process will be described. First,
As will be described later, the read timing of the jitter queue 303 is calculated based on the algorithm of FIG. The packet received immediately after the read timing is determined is loaded with a pointer in the jitter queue 303 and simultaneously with the jitter amount control 3.
At 02, a timer is started at this read timing, and when a timeout occurs, a reproduction instruction is generated. Thereafter, a reproduction instruction is generated for each packet cycle C.

When a reproduction instruction is received, a pointer is taken out from the jitter queue 303 indicated by the HP 2002, and audio data is read out from the reception buffer 41. The read audio data is decoded by the decoder 304, and the decoded audio data is transmitted to the PBX, the telephone network, or the telephone via the reception signal line 305.

After taking out the pointer from the jitter queue 303 and performing reproduction processing, the HP 2002 is advanced by one. If there is no change in the adaptive jitter amount, thereafter, the voice packet is reproduced by taking out the pointer from the jitter queue 303 in accordance with the fixed packet period C.

In the case of the example shown in FIG. 3, when reproducing the voice packet of the transmission sequence number # 2, since the transmission sequence number # 2 has not been received, the interpolation frame or
Generate and send silence frames. That is, HP2002
If the content of the jitter queue indicated by is "0", it indicates that the voice packet of the sequence number has not been received yet. Also, in FIG. 3, since the audio packet of the transmission sequence number # 6 is received after the audio packet of the transmission sequence number # 6 is reproduced (interpolated or transmitted without sound), the audio packet of the transmission sequence number # 6 is received. Interprets the arrival to be late and discards this voice packet.

When there is a change in the adaptive jitter amount, there is a ruler change instruction from the jitter amount control 302.
The jitter amount is changed as described later with reference to the flowchart of FIG.

Next, the algorithm of the jitter amount control 302 will be described with reference to FIG. 4, FIG. 5, and FIG.

The time from when the voice packet is transmitted from the transmitting apparatus 3002 to when it is reproduced by the receiving apparatus 3004 is determined by the delay / fluctuation time in the relay network 3003 and the processing time of the receiving apparatus 3004 (packet processing, fluctuation (Time required for absorption). Since the packet processing time and the voice encoding time of the receiving device 3004 are given as constants if the device can be specified, this value is defined as β.

The fluctuation absorption time of the receiving apparatus is determined by the
It is desirable to calculate an adaptive value from the delay / fluctuation time in 03 and set dynamically.

As shown in FIG. 6, a reception time stamp for notifying the reception time is defined in the reception packet. Since the transmission time stamp given by the transmission device 3002 is given at a constant interval depending on the packet cycle, it can be calculated on the receiving side. The reception time stamp of the transmission sequence number i is defined as Ri, and the transmission time stamp is defined as Si. The reception time stamp Ri is the reception time stamp Ri
The receiving device 3004 is provided by the CPU 2 of FIG.
Has the same configuration as that of the voice packet receiving device 1).

The DSP 31 uses the time stamp of the packet first received at the start of voice communication as the transmission device 3002
And a clock that may be out of sync with the receiving device 3004. A transmission time stamp corresponding to the clock 8 of the receiving device 3004 is defined as RSi, and a clock offset is defined as X.
Since the exact value of the clock lag between the transmitting device 3002 and the receiving device 3004 is not known, it is assumed that the delay time of the first packet (i = 0) received after the start of the voice communication is 0, and Define X = R0-S0. R0
Is the reception timestamp of the first packet received, S
0 is the transmission time stamp of the first received packet. Accordingly, the transmission time stamp RSi generated by the receiving device 3004 can be represented by the following equation.

RSi = Si + X

Here, in a system in which the transmission time stamp Si is not defined, Si = i × C and X = R0. If C = 10 ms, i = 0 is Si =
0, i = 1 is Si = 10, i = 2 is Si = 20, and Si goes up at equal intervals with respect to the sequence number i.
Since there is no guarantee that the reception will be performed in the order of the sequence number i, it must be calculated by Si = sequence number i × packet cycle C every time reception is performed. Therefore, RSi may use the following equation.

RSi = i × C + R0

The second and subsequent received packets can be converted to the clock of the receiving device 3004 by using RSi for the transmission time stamp. Therefore, the reception delay time based on the difference between Ri and RSi, that is, the packet reception without fluctuation, The difference between the time actually received and the time is calculated, and this is calculated as 16
When calculated as a moving average value in packet units, the average reception delay B can be expressed by the following equation.

Average reception delay B = B + ((Ri−RSi) −B) ÷ 16

Using the average reception delay B calculated here, an average reception timing RTi is obtained.

Average reception timing RTi = RSi + B

If packets are transmitted at equal intervals and received at equal intervals, there is no fluctuation and the value of the average reception delay B is zero. However, in practice, fluctuations occur depending on the traffic volume and route of the relay network 3003. This is called inter-arrival jitter and can be expressed by the following equation (inter-arrival jitter is defined by the international standard ITU-T H.2).
Use the contents specified in 25.0. It is recommended that the interarrival jitter be calculated in the order of reception (without considering the transmission sequence number)).

Fluctuation D (i−1, i) = (R _i −R _i−1 ) − (S _i −S _i−1 ) = (R _i −R _i−1 ) − (RS _i −RS _{i−) 1} )

Interarrival jitter J = J + (| D
(I-1, i) | -J) ÷ 16

The received packet is received with the average reception delay B, and after the reception processing time β, the interarrival jitter J
Since the fluctuation absorption processing of × α is performed, the optimal time for reproducing the packet is when the clock 8 of the receiving device 3004 indicates the average read timing RAi. The average read timing RAi can be represented by the following equation.

Average read timing RAi = RSi + B + J × α + β

Here, by providing α as a fluctuation coefficient, the relay network 3 used by a provider or the like can be used.
The value of the coefficient α can be set according to the characteristic of 003, and the adaptability can be further improved. For example,
The value of the coefficient α is set to a small value in the case of a dedicated line connection with little fluctuation, and the value of the coefficient α is set to a large value in the case of using a provider whose traffic varies greatly.

However, since voice packets are processed as fixed-length packets, once the initial value RA0 of the read timing is once determined, the jitter queue 303 is set at equal intervals.
And play it out. This is defined as jitter read timing RBi. When this value increases or decreases in the jitter amount n, a process of subtracting this time is necessary. When the increase / decrease of the jitter amount is defined as the jitter amount change number nD, the jitter read timing RBi can be expressed by the following equation.

Jitter read timing RBi = RA0 +
i × C + nD × C

The difference between the average read timing RAi and the jitter read timing RBi is determined for each packet cycle C or for each timing review cycle. If the difference is equal to or longer than the packet cycle C, the jitter amount change number nD is expressed by ( RA
An integer value obtained by rounding the value calculated by i-RBi) / C is used. ND = 0 other than the timing review cycle
Set to.

In other words, the packet of sequence number i is received by the clock 8 of the receiving device 3004 at the average reception timing RTi, and the average read timing RA
Since it is ideal to be reproduced at i, the jitter amount nA required for fluctuation absorption can be expressed by the following equation.

[0067]

Even if there is no fluctuation, one is required for the packet processing, so one is added to the calculation result to calculate. Since the required jitter amount nA is processed by an integer, the calculation result is rounded off.

On the other hand, the currently adaptively controlled jitter buffer length is defined as a jitter amount n. Jitter amount n
Acts in two directions, plus and minus, across the midpoint. This is the virtual ruler 2003. With this,
The jitter amount change number nD = nA-n.

The initial value RA0 of the read timing may be set in the configuration, but in the case of the present embodiment, a value calculated from a ringback tone or a silent packet which flows first when voice communication is started is used. Then, from the beginning of the call, control can be performed with an adaptive amount of jitter buffer.

When the value calculated by the present algorithm is used as the initial value, the received packet 7 at the time of determining the read timing RA0 is loaded into the jitter queue 303, and the HP20
02 is set as the transmission sequence number i of the packet.
When the clock 8 of the receiving apparatus 3004 indicates the read timing RA0, the reproduction instruction is output, and thereafter, the packet cycle C
Each time, a reproduction instruction or a ruler change instruction is output.

Even when the reproduction of the packet is started, the calculation of the average read timing RAi and the jitter amount change number nD is continuously performed. Then, as shown in the flowchart of FIG.
In step S11, the calculated jitter amount change number nD
Is determined to be 0 or not. If the calculated jitter amount change number nD ≠ 0 (if not 0), a ruler change instruction is output instead of the reproduction instruction (step S13). On the other hand, when the calculated jitter amount change number nD = 0 (in the case of 0), a reproduction instruction is output (step S12).

For changing the ruler, as shown in the flowchart of FIG. 10, it is determined in step S31 how to change the amount of jitter. If the amount of jitter is to be increased, the process proceeds to step S32, where the amount of jitter n = After n + 1 (increased by 1), in step S33, 1
Interpolation for the packet Cms or silent data is reproduced.

On the other hand, when the amount of jitter is to be reduced, the process proceeds to step S34, where the amount of jitter is set to n = n-1 (reduced by 1). In step S35, the leading packet of the jitter queue 303 is discarded. Pretend that playback is complete. When this process is performed, the jitter read timing RBi is also updated. Then, step S
At 36, HP 2002 is incremented by one and returns.

If the fluctuation of the jitter amount n occurs every time a packet is received, packet discarding and silent transmission are repeated, and the voice quality is rather deteriorated. Therefore, the fluctuation check of the jitter amount is performed every few packets or the jitter amount is reduced. In this case, the effect can be improved by combining the logic of returning gradually with the addition of hysteresis control.

Specifically, as shown in FIG.
When the amount of jitter n is increased, it is increased according to the calculation result of the number of changes in jitter amount nD. FIG. 11 shows an example of calculating the jitter amount. As preconditions, the fluctuation coefficient α is 1.5, the processing time β of the device is 30 ms, and the packet period is 10 ms. FIG. 11A shows an example of calculating the amount of jitter when there is no fluctuation, and FIG. 11B shows an example of calculating the amount of jitter when there is fluctuation. FIG. 11 (1)
As shown in the figure, when there is no fluctuation, the increase / decrease of the jitter amount is zero.
And the jitter amount n does not change.

Based on the algorithm described above,
By calculating the optimum fluctuation time at the moment of a call and controlling the jitter queue 303, adaptive jitter buffer control becomes possible.

As described above, according to the present invention, the voice packet transmitting / receiving apparatus performs the optimum jitter buffer control based on the delay time and the fluctuation time of the voice call, so that the following effects can be obtained.

The first effect is that the occurrence of underflow / overflow of the jitter buffer due to sudden traffic fluctuation can be suppressed. Therefore, it is possible to suppress a decrease in voice quality such as a break in sound due to packet discarding in the apparatus.

The second effect is that the call service user can receive the call service with the best delay time at that moment regardless of the connection destination and the traffic volume.

A third effect is that the call service provider (operator) can save the trouble of changing the setting of the jitter buffer to a small value during the daytime when traffic is small, and to change the setting of the jitter buffer to a large value at nighttime when traffic is heavy. It is.

The fourth effect is that the jitter buffer control is controlled by CP
By moving from U to DSP and using both the receive buffer and the jitter buffer, the load on the CPU is distributed, the processing capacity is increased, and the resources for buffer control are reduced, so that the number of devices to be accommodated can be increased. In addition, it is possible to cope with various additional services.

In the above embodiment, the case where packets including voice data are transmitted / received has been described. However, the present invention can be applied to the case where packets including other data are transmitted / received.

The specific numerical values used in the above embodiment are examples, and the present invention is not limited thereto.

[0085]

As described above, according to the packet receiving apparatus, the packet receiving method, and the recording medium according to the present invention, a packet is received, the received packet is stored in the order in which the packet was received, and the packet is transmitted. Rearranged in the order in which the packets were transmitted, and at least, based on the time at which the packet was received, calculate the amount of jitter generated in the relay network, and according to the calculated amount of jitter, determine the timing of decoding the rearranged packets. Control, and decodes the packet at a controlled timing, so that the amount of jitter can be automatically changed based on the optimum delay time according to the communication condition, suppressing packet discarding, Communication quality can be improved.

[Brief description of the drawings]

FIG. 1 is a block diagram showing a configuration example of an embodiment of a voice packet receiving apparatus to which the present invention is applied.

FIG. 2 is a diagram illustrating a configuration example of a jitter queue 303 in FIG. 1;

FIG. 3 is a diagram for explaining a jitter queue loading procedure.

FIG. 4 is a diagram for explaining an algorithm for calculating a jitter buffer amount.

FIG. 5 is a diagram for explaining an algorithm for calculating a jitter buffer amount.

FIG. 6 is a diagram showing a format of an audio packet.

FIG. 7 is a flowchart illustrating a procedure for receiving a packet.

FIG. 8 is a flowchart illustrating a procedure for issuing a reproduction instruction or a ruler change instruction in accordance with the number of changes in the amount of jitter.

FIG. 9 is a flowchart illustrating a procedure of a reproduction instruction.

FIG. 10 is a flowchart illustrating a procedure of a ruler change instruction.

FIG. 11 is a diagram illustrating an example of calculating a jitter amount.

[Explanation of symbols]

 Reference Signs List 1 voice packet receiving device 2 CPU 6 WAN 7 received packet 8 clock 21 CPU bus 31 to 3n, 32 to 3n DSP 41 to 4n receive buffer 51 to 5n memory bus 61 relay line 201 to 20n, 202 write pointer 301 packet processing 302 Jitter amount control 303 Jitter queue 304 Decoder 305 Receive signal line 306 Read pointer 307 Supplement pointer 2002 Pointer (HP) 2003 Virtual ruler 2004 Pointer 2005 Voice packet 2006 Voice data 3001, 3005 Telephone 3002 Transmitter 3003 Relay network 3004 Receiver 4001 Next Buffer pointer 4002 command 4003 voice packet length 4004 reception time stamp 4005 other information (1) 4006 transmission sequence number 4007 transmission time stamp 4008 other information (2) 4009 voice data

Claims (8)

[Claims] 1. A packet receiving apparatus for receiving and decoding a packet transmitted from a transmitting side via a relay network on a receiving side, comprising: receiving means for receiving the packet; and receiving by the receiving means. Storage means for storing the received packets in the order in which the receiving means has received the packets; reordering means for rearranging the packets in the order transmitted by the transmitting side; and at least the packet is received by the receiving means. A calculating unit that calculates a jitter amount generated in the relay network from a time, and a timing for decoding the packets rearranged by the rearranging unit according to the jitter amount calculated by the calculating unit. Timing control means for controlling; and recovering the packet at a timing controlled by the timing control means. And a decoding means for decoding. 2. The apparatus according to claim 1, further comprising an adding unit that adds address information of the storage unit that stores a packet next to the packet to the packet received by the receiving unit. The packet receiving device according to claim 1, wherein the packet is stored in the storage unit. 3. An address storage unit for storing an address of the storage unit in which the packet is stored, wherein the rearranging unit transmits an address of the storage unit in which the packet is stored, and a transmission side transmits the address of the storage unit. 2. The packet receiving apparatus according to claim 1, wherein the packets are rearranged by storing them in the address storage unit in the order in which the packets are stored. 4. An address storage unit for storing an address of the storage unit in which the packet is stored, wherein the rearranging unit transmits an address of the storage unit in which the packet is stored, and a transmission side transmits the address of the storage unit in which the packet is stored. The packet is rearranged by storing the packet in the address storage unit in the order given, and the decoding unit is stored in the address storage unit according to the jitter amount calculated by the calculation unit. 2. The packet receiving apparatus according to claim 1, wherein the address is read, and the packet stored in the address of the storage unit is read and decoded. 5. The packet includes at least a field for describing address information in which the next packet is stored, a field for describing a packet length, a field for describing a reception time stamp, It consists of a field for describing a transmission sequence number, a field for describing a transmission time stamp, and a field for describing data. At least address information in which the next packet is stored, and a reception time stamp The packet receiving apparatus according to claim 1, wherein is set on the receiving side, and at least the transmission sequence number is set on the transmitting side. 6. The packet receiving apparatus according to claim 1, further comprising estimating means for estimating a time at which a transmitting side transmitted the packet based on a time at which the packet was received by the receiving means. . 7. A packet receiving method for receiving and decoding a packet transmitted from a transmitting side via a relay network at a receiving side, comprising: a receiving step of receiving the packet; The packet is
A storing step of storing the packets in the order in which they are received; a rearranging step of rearranging the packets in the order transmitted by the transmitting side; and A calculating step of calculating a jitter amount, a timing control step of controlling a timing of decoding the packets rearranged in the rearranging step in accordance with the jitter amount calculated in the calculating step, and the timing control A decoding step of decoding the packet at a timing controlled in the step. 8. A recording medium on which a program capable of executing the packet receiving method according to claim 7 is recorded.