JP2009027303A - Communication apparatus and communication method - Google Patents

Abstract

<P>PROBLEM TO BE SOLVED: To sufficiently use an available band while maintaining equality in a usage band with another connection. <P>SOLUTION: A congestion determining section 63 predicts a new congestion window cwnd' by using an immediately proceeding congestion window cwnd and the history of RTT. A transmission section 64 allocates the band of the Internet by using the predicted congestion window cwnd', and transmits data in that band. <P>COPYRIGHT: (C)2009,JPO&INPIT

Description

  The present invention relates to a communication apparatus and a communication method, and more particularly, to a communication apparatus and a communication method that can fully use an available band while maintaining fairness of a used band with other connections.

  In recent years, the Internet has become widespread. Along with this, the network is becoming more complex and congestion is increasing. Since congestion causes a drop in transfer performance, a technique for avoiding congestion is important.

For example, at present, TCP (Transmission Control Protocol) is effectively the standard Internet protocol. However, in TCP, since the intermediate node is made transparent to perform communication, the intermediate node cannot know the total amount of data that flows. This
A phenomenon that the received data overflows from the buffer, that is, congestion occurs.

  When data lost due to congestion is retransmitted, the data tends to worsen because it falls into a vicious circle of congestion collapse that causes congestion again. Therefore, it is necessary to control congestion in order to avoid congestion.

  Therefore, currently, in TCP, when packet loss is detected, congestion is avoided by reducing the window size. Specifically, in the currently popular version of TCP called Reno, the window size is controlled in two states, a slow start state 11 and a congestion avoidance state 12, as shown in FIG.

  In the slow start state 11 of FIG. 1, the congestion window cwnd increases by one segment each time an ACK that is a data confirmation response is received. That is, in the slow start state 11, the congestion window cwnd increases exponentially. On the other hand, in the congestion avoidance state 12, every time an ACK is received, the congestion window cwnd increases by 1 / cwnd segment. That is, in the congestion avoidance state 12, the congestion window cwnd increases linearly.

  When the congestion window cwnd is larger than the threshold value ssthresh, the state transitions from the slow start state 11 to the congestion avoidance state 12. The initial value of the threshold value ssthresh is the maximum value of the window size, and when congestion is detected, it is set to half the window size at that time.

  Further, when congestion is detected due to timeout, if the state at that time is a congestion avoidance state, a transition is made to the slow start state, and if the state at that time is a slow start state, the state is maintained. In addition, when congestion is detected by duplicate ACK and retransmission is successful, if the current state is the slow start state, the state transits to the congestion avoidance state, and if the current state is the congestion avoidance state, Maintain state.

  The specific operation will be described below.

  Initially, the state is set to the slow start state 11 and the congestion window cwnd is set to the minimum value. Each time an ACK is received, the congestion window cwnd increases by one segment. Therefore, when an ACK for all transmitted data is received, the amount of data sent out at the same time is twice that of the previous time. As a result, in a short time, data having an amount of data exceeding the available bandwidth is transmitted, and congestion occurs.

  When congestion is detected due to timeout, the threshold ssthresh is set to half of the window size at that time, and the congestion window cwnd is returned to the minimum value. The congestion window cwnd increases again, and when the threshold value ssthresh is exceeded, the state transitions from the slow start state 11 to the congestion avoidance state 12.

  At this time, if congestion is detected by duplicate ACK, if the packet is successfully retransmitted, it is determined that the detected congestion is minor, and the congestion window cwnd is not the minimum value, and Is set to half the window size. Therefore, in this case, the congestion window cwnd immediately exceeds the threshold value ssthresh, and the state transitions from the slow start state 11 to the congestion avoidance state 12. As a result, it is possible to prevent useless transfer performance degradation due to the congestion window cwnd being returned to the minimum value.

  In the congestion avoidance state 12, the congestion window cwnd increases by 1 / cwnd every time an ACK is received. Therefore, compared with the case of the slow start state 11, the increase rate of the congestion window cwnd is reduced, and the time until the congestion occurs is increased. However, since the congestion window cwnd never decreases, congestion will eventually occur. When congestion is detected due to timeout, the threshold ssthresh is set to half of the window size at that time, and the congestion window cwnd is returned to the minimum value. The state transitions from the congestion avoidance state 12 to the slow start state 11.

  However, since congestion avoidance in TCP Reno is performed after congestion occurs, it is necessary to transmit extra packets by retransmission, and it is difficult to cope with the rapidly changing Internet. In addition, every time congestion is detected, the congestion window cwnd decreases to less than half of the window size at that time, so it is difficult to fully utilize the available bandwidth.

  Therefore, techniques for predicting the state of the network and avoiding congestion in advance have been studied. The methods of this research are roughly classified into two, and the first method is to develop a completely new high-speed and reliable protocol instead of TCP. In this method, since a new protocol is designed on the premise of broadband and high delay, in this environment, the transfer performance is higher than that of the conventional TCP. However, since it is not compatible with TCP, a new protocol and a special node to mediate TCP are required. Therefore, a considerable cost is required to spread a new protocol.

  The second method improves the congestion avoidance algorithm while maintaining compatibility with the conventional TCP. In this method, it is difficult to dramatically improve the transfer performance as in the first method, but since it is compatible with the conventional TCP, a special intermediary node or the like is not necessary for the spread. Therefore, TCP with improved congestion avoidance algorithm can be implemented at low cost and is practical.

  Therefore, Brakmo et al. Proposed a version of TCP called Vegas by the second method. In TCP Vegas, RTT (Round Trip Time) used to calculate RTO (Retransmission Time-Out) in TCP Reno is accurately measured, and expected throughput and actual throughput are calculated. Then, by comparing these two throughputs, network congestion is estimated and the window size is adjusted.

  Specifically, the method of controlling the congestion window cwnd in the slow start state and the congestion avoidance state in TCP Vegas is expressed by the following equation (1).

  In the formula (1), α and β are constants, and the constant α is smaller than the constant β. BaseRTT is the smallest RTT measured so far. According to Equation (1), the congestion window cwnd increases exponentially in the slow start state.

  Moreover, Diff in the formula (1) is a difference between the expected throughput and the actual throughput, and is expressed by the following formula (2).

  As described above, in TCP Vegas, in the congestion avoidance state, the state of the network is predicted based on the difference between the expected throughput and the actual throughput, and the congestion window cwnd is controlled. Transfer performance can be improved over Reno.

  However, in TCP Vegas, since network bandwidth is actively used, the fairness of bandwidth usage with other connections is impaired. In addition, it is difficult to measure BaseRTT in a rapidly changing network. Furthermore, the congestion avoidance algorithm may not function in severe congestion. Also, in TCP Reno, as mentioned above, the congestion window cwnd only increases and does not decrease, so when TCP Vegas communication shares bandwidth with TCP Reno communication, RTT decreases, resulting in transfer performance. It will drop significantly.

  In addition, TCP with improved congestion control algorithm solves the problem of reduced transfer performance when TCP Vegas and TCP Reno share bandwidth, and obtains information on available bandwidth by inline measurement. Some have a window size control algorithm that uses. However, these TCPs have not yet been put into practical use due to the problem of fairness in bandwidth usage with other connections.

  Further, as a TCP having an improved congestion control algorithm, a TCP having a congestion control algorithm for predicting congestion from a network state and controlling a window size by incorporating machine learning has been proposed. This TCP has confirmed improvements such as a quick response to changes in the network, but since the bandwidth is actively used, the fairness of the bandwidth used when sharing the bandwidth with other connections is sacrificed. .

  By the way, as a conventional congestion control method, in a communication network such as ATM (Asynchronous Transfer Mode), a packet for RTT measurement is periodically transmitted separately from actual data, and a neural network is obtained from the RTT time series of the packet. There is an autonomous distributed traffic flow control method that uses the predicted RTT to predict the next RTT and increases or decreases the transfer rate based on the predicted RTT (see, for example, Patent Document 1).

  However, in this autonomous distributed traffic flow control method, it is necessary to transmit a packet for RTT measurement, so that the amount of packet transfer increases. In addition, this autonomous distributed traffic flow control method has been devised for the purpose of controlling communication in a communication network such as ATM, and therefore, fairness with other flows is not considered.

  Also, as a conventional congestion control method, the buffer amount of the bottleneck router is estimated and the buffer is used as much as possible for the purpose of improving bandwidth utilization efficiency and fairness of bandwidth utilization with the existing TCP. There is a congestion control method for controlling the window size (see, for example, Patent Document 2).

  However, in this congestion control method, since the buffer amount is estimated by learning using only the Diff value (Diff_max) immediately before the occurrence of congestion, the learning accuracy is low. Therefore, in order to prevent congestion, the buffer amount must be estimated to be low, and it is difficult to sufficiently use the available bandwidth.

  Furthermore, as a conventional congestion control method, for the purpose of end-to-end traffic management of TCP communication of multimedia data, the line state (queue occupancy rate) determined by past packet transmission history and response history is modeled as a queue, By predicting the predicted response time series, which is the response time from 1 time ahead to m time ahead, by multi-time scale linear prediction, the state of the line determined by the predicted response time series and packet transmission history is predicted and There is a method of determining the transmission time of each packet according to the state of the line (for example, see Patent Document 3).

  However, in this method, since the prediction response time series is predicted by multi-time scale linear prediction, the accuracy of prediction is low. Therefore, in order to prevent congestion, it is necessary to predict the level of the line state to be low, and it is difficult to sufficiently use the available bandwidth.

Japanese Patent Laid-Open No. 10-23064

JP 2006-67109 A

JP 2002-368800 A

  As described above, a congestion control algorithm that fully utilizes the available bandwidth by avoiding congestion in advance while maintaining fairness of the available bandwidth with other connections has not been considered.

  The present invention has been made in view of such a situation, and makes it possible to sufficiently use the available bandwidth while maintaining the fairness of the bandwidth used with other connections.

  A communication apparatus according to an aspect of the present invention includes a prediction unit that predicts a new congestion window using a previous congestion window and RTT history in a communication apparatus that performs communication via a network, and the predicted congestion window. And transmitting means for allocating a bandwidth of the network and transmitting data in the bandwidth.

  In the communication apparatus according to the aspect of the present invention, the prediction unit can predict a new congestion window by using the previous congestion window and the history of the reciprocal of the RTT.

  In the communication apparatus according to the aspect of the present invention, the prediction unit can linearly predict a new congestion window using the previous congestion window and a history of the reciprocal of the RTT.

  A communication method according to an aspect of the present invention is a communication method of a communication apparatus that performs communication via a network, predicts a new congestion window using a previous congestion window and RTT history, and predicts the predicted congestion window. , And allocating a bandwidth of the network and transmitting data in the bandwidth.

  In one aspect of the present invention, a new congestion window is predicted using the previous congestion window and RTT history, a network bandwidth is allocated using the predicted congestion window, and data is transmitted in the bandwidth. Is done.

  Embodiments of the present invention will be described below. Correspondences between the constituent elements of the present invention and the embodiments described in the specification or the drawings are exemplified as follows. This description is intended to confirm that the embodiments supporting the present invention are described in the specification or the drawings. Therefore, even if there is an embodiment which is described in the specification or the drawings but is not described here as an embodiment corresponding to the constituent elements of the present invention, that is not the case. It does not mean that the form does not correspond to the constituent requirements. Conversely, even if an embodiment is described here as corresponding to a configuration requirement, that means that the embodiment does not correspond to a configuration requirement other than the configuration requirement. It's not something to do.

A communication device according to one aspect of the present invention includes:
In a communication device that performs communication via a network (for example, the communication device 20 of FIG. 2),
Prediction means for predicting a new congestion window using the previous congestion window and RTT history (for example, the congestion window determination unit 63 in FIG. 4);
A transmission unit (for example, the transmission unit 64 in FIG. 4) that allocates a bandwidth of the network using the predicted congestion window and transmits data in the bandwidth;

A communication method according to one aspect of the present invention includes:
In a communication method of a communication device that performs communication via a network (for example, the communication device 20 in FIG. 2),
A new congestion window is predicted using the previous congestion window and RTT history (for example, step S22 in FIG. 5),
Using the predicted congestion window, a bandwidth of the network is allocated, and data is transmitted using the bandwidth (for example, step S12 in FIG. 5).
Includes steps.

  Hereinafter, specific embodiments to which the present invention is applied will be described in detail with reference to the drawings.

  FIG. 2 shows a hardware configuration example of an embodiment of a communication apparatus to which the present invention is applied.

  In the communication device 20 of FIG. 2, a CPU (Central Processing Unit) 21, a ROM (Read Only Memory) 22, and a RAM (Random Access Memory) 23 are connected to each other by a bus 24.

  An input / output interface 25 is further connected to the bus 24. The input / output interface 25 includes a keyboard, a mouse, a microphone, an input unit 26 including a receiving unit that receives a command transmitted from a remote controller, an output unit 27 including a display and a speaker, a hard disk, a nonvolatile memory, and the like. A storage unit 28 composed of a network, a communication unit 29 composed of a network interface, and a drive 30 for driving a removable medium 31 such as a magnetic disk, an optical disk, a magneto-optical disk, or a semiconductor memory are connected.

  In the communication device 20 configured as described above, the CPU 21 stores, for example, a program stored in the storage unit 28 by loading it into the RAM 23 via the input / output interface 25 and the bus 24 and executing it. Data stored in the unit 28 or the like is transmitted to another device.

  Specifically, for example, the CPU 21 reads data to be transmitted from the storage unit 28 and supplies the data to the communication unit 29. The communication unit 29 transmits data supplied from the CPU 21 via the Internet in accordance with TCP Salt (to be described later) which is a version of TCP having compatibility with the current TCP.

  The program executed by the CPU 21 of the communication device 20 is recorded on the removable medium 31 or provided via a wired or wireless transmission medium such as a local area network, the Internet, or digital satellite broadcasting.

  The program can be installed in the storage unit 28 via the input / output interface 25 by attaching the removable medium 31 to the drive 30. The program can be received by the communication unit 29 via a wired or wireless transmission medium and installed in the storage unit 28. In addition, the program can be installed in the ROM 22 or the storage unit 28 in advance.

  Next, FIG. 3 is a state transition diagram of TCP Salt.

  As shown in FIG. 3, TCP Salt has two states, a boost state 41 and a congestion avoidance state 42. When congestion is detected, the state transitions from the boost state 41 to the congestion avoidance state 42. Further, when the available bandwidth increases and it is determined that there is room in the available bandwidth, the state transitions from the congestion avoidance state 42 to the boost state 41.

  In boost state 41, every time a normal ACK is received, the congestion window cwnd increases by one segment. On the other hand, in the congestion avoidance state 42, using the previous congestion window cwnd and the RTT available on the transmission side, the optimum congestion window cwnd ′ is linearly predicted as the current congestion window by the following equation (3). .

In Equation (3), IRTT _i is the reciprocal of RTT i (1 ≦ i ≦ n) times before, and w ₀ and w _i are weighting factors. According to Equation (3), the congestion window cwnd ′ is expressed by an inner product of the input vector represented by IRTT _i and the previous congestion window cwnd and the coefficient vector represented by the weighting coefficients w ₀ and w _i , that is, a linear combination. .

That is, since IRTT is proportional to the degree of congestion, that is, the degree of congestion of the line, IRTT is used as an expression of the line condition in Equation (3). In Expression (3), as the congestion window cwnd ′, IRTT _i that is a history of IRTT representing the state of the past line and a congestion window cwnd immediately before representing how the line is used are used. Predicted by linear combination.

Therefore, in the equation (3), the current IRTT ₀ , that is, the current line status, is obtained using IRTT _i representing the past line status and the congestion window cwnd immediately before representing how the line was used. It can be said that the congestion window cwnd ′ corresponding to the prediction is obtained.

In this way, in TCP Salt, the current line status is predicted using IRTT _i that represents the status of the past line and the congestion window cwnd immediately before that indicates how the line was used. Compared with the case where the current line status is predicted based on the current line status, it is possible to perform prediction with higher accuracy. As a result, the available bandwidth can be fully utilized.

  In addition, since the RTT in which the reciprocal number is used in the equation (3) is also used in the current communication by TCP, the TCP Salt that predicts the congestion window cwnd ′ by the equation (3) is the same as the conventional TCP. It has compatibility.

  As described above, when the congestion window cwnd ′ is predicted by the expression (3), the Internet band is allocated using the congestion window cwnd ′, and data is transmitted in the band.

The weighting factors w ₀ and w _i described above are updated (machine learning) according to the following equation (4), and are used for the prediction of the congestion window cwnd ′ described above.

  In Expression (4), the congestion window cwnd is the previous congestion window cwnd used in Expression (3). C is a predetermined learning coefficient. Furthermore, ERROR in equation (4) is obtained by the following equation (5).

In Equation (5), RTT _j is RTT _j (0 ≦ j ≦ n−1) times before, and MAX_RTT is the maximum value of RTT.

According to the equations (4) and (5) described above, when the degree of congestion decreases (RTT _{j + 1} > RTT _j ), the weighting factors w ₀ and w _i increase, and the degree of congestion increases ( RTT _{j + 1} <RTT _j ) and the weighting factors w ₀ and w _i decrease. Further, if the degree of congestion is unchanged (RTT _{j + 1} = RTT _j ), the weighting factors w ₀ and w _i do not change. As a result, the weighting factors w ₀ and w _i can increase the congestion window cwnd ′ when the degree of congestion decreases, and decrease the congestion window cwnd ′ when the degree of congestion increases.

Therefore, according to the equations (4) and (5), it is possible to learn the weighting factors w ₀ and w _i necessary for predicting the optimum congestion window cwnd ′ without congesting the line and without wasting it. it can.

As described above, in the congestion avoidance state 42, the congestion window weighting coefficients w ₀ and w _i used to predict the cwnd' is updated, the weight coefficient w ₀ and w _i, the previous congestion window cwnd, as well as the past n times The congestion window cwnd ′ is predicted from the history of IRTT _i .

In Expressions (3) and (4), IRTT _i itself is not used, but IRTT _i normalized according to the following Expression (6) is used.

  In Equation (6), MIN_RTT is the minimum value of RTT.

  FIG. 4 shows a functional configuration example of the communication unit 29 of FIG.

  The communication unit 29 in FIG. 4 includes an RTT calculation unit 61, a state determination unit 62, a congestion window determination unit 63, a transmission unit 64, and a reception unit 65. Each block that constitutes the communication unit 29 can exchange signals with each other as necessary.

  The RTT calculation unit 61 stores a time stamp supplied from the transmission unit 64 and added to data to be transmitted (hereinafter referred to as transmission data). Further, the RTT calculation unit 61 calculates the difference between the stored time stamp of the transmission data and the time stamp added to the ACK that is an acknowledgment for the transmission data supplied from the reception unit 65 as the RTT. To do. The RTT calculation unit 61 supplies the RTT to the state determination unit 62 and the congestion window determination unit 63.

  The state determination unit 62 changes the state from the boost state 41 to the congestion avoidance state 42 according to the information representing the detection of congestion supplied from the reception unit 65. Further, the state determination unit 62 transitions the state from the congestion avoidance state 42 to the boost state 41 when the variation of the RTT supplied from the RTT calculation unit 61 becomes zero. That is, since there is a relationship between the available bandwidth having a small RTT variation, the state determination unit 62 determines that the available bandwidth has a margin when the RTT variation becomes zero. Then, the state is changed to the boost state 41. Further, the state determination unit 62 supplies information indicating the current state to the congestion window determination unit 63.

  The congestion window determination unit 63 stores the RTT supplied from the RTT calculation unit 61. Further, the congestion window determination unit 63 determines the congestion window cwnd according to the information representing the current state supplied from the state determination unit 62. Specifically, when the current state is the boost state 41, the congestion window determination unit 63 sets the congestion window cwnd by one segment from the previous congestion window cwnd every time RTT is supplied from the RTT calculation unit 61. Increase this to the current congestion window cwnd.

When the current state is the congestion avoidance state 42, the congestion window determination unit 63 determines the past n times and the current RTT _j , a preset learning coefficient C, the previous weighting coefficients w ₀ and w _i , and Using the previous congestion window cwnd, the weighting factors w ₀ and w _i are updated and stored according to equations (4) and (5).

Then, the congestion window determination unit 63 uses the stored RTT _i for the past n times, the updated weighting factors w ₀ and w _i , and the previous congestion window cwnd to the above-described equation (3). Therefore, the congestion window cwnd ′ is predicted to be the current congestion window cwnd. The congestion window determination unit 63 stores the current congestion window cwnd and supplies it to the transmission unit 64.

  The transmission unit 64 adds a time stamp in segment units to the transmission data supplied from the CPU 21 (FIG. 2). Then, using the current congestion window cwnd supplied from the congestion window determination unit 63, the transmission unit 64 allocates an Internet band and transmits transmission data to which a time stamp is added in the band. Further, the transmission unit 64 supplies the time stamp added to each segment to the RTT calculation unit 61.

  The receiving unit 65 receives an ACK for the transmission data from another communication device (not shown) connected to the communication device 20 via the Internet that receives the transmission data transmitted by the transmission unit 64. Then, the reception unit 65 supplies the RTT calculation unit 61 with the time stamp added to the received ACK.

  The receiving unit 65 determines whether a normal ACK that does not overlap with the previously received ACK is received for the transmission data within a predetermined time after the transmission data is transmitted. When it is determined that a normal ACK has not been received within a predetermined time, the receiving unit 65 detects congestion and supplies information indicating the detection of congestion to the state determining unit 62. On the other hand, when it is determined that a normal ACK has been received within a predetermined time, the receiving unit 65 does not detect congestion.

  Next, a determination process for determining the congestion window cwnd by the communication unit 29 in FIG. 4 will be described with reference to the flowchart in FIG. This determination process is started, for example, when transmission data is supplied from the CPU 21 of FIG.

  In step S <b> 11, the transmission unit 64 adds a time stamp in segment units to the transmission data supplied from the CPU 21, and supplies the time stamp to the RTT calculation unit 61. In step S12, the transmission unit 64 allocates the Internet band using the current congestion window cwnd determined by the congestion window determination unit 63, and transmits the transmission data to which the time stamp is added in the band to other communication devices. To do.

  In step S13, the receiving unit 65 determines whether or not an ACK for the transmission data transmitted in step S12 has been received. When it is determined in step S13 that ACK has not been received, in step S14, the receiving unit 65 determines whether or not a predetermined time set in advance has elapsed. If it is determined in step S14 that the predetermined time has not elapsed, the process returns to step S13, and the processes in steps S13 and S14 are repeated until ACK is received or until the predetermined time elapses. .

  On the other hand, when it is determined in step S14 that the predetermined time has elapsed, that is, when a timeout occurs, the receiving unit 65 detects congestion and supplies information indicating the detection of congestion to the state determining unit 62. . Then, the process proceeds to step S20.

  If it is determined in step S13 that an ACK has been received, in step S15, the receiving unit 65 determines whether the received ACK is a normal ACK, that is, whether it is an ACK that does not overlap with the previously received ACK. Determine. If it is determined in step S15 that the received ACK is not a normal ACK, the reception unit 65 detects congestion and supplies information indicating the detection of congestion to the state determination unit 62. Then, the process proceeds to step S20.

  On the other hand, if it is determined in step S15 that the received ACK is a normal ACK, the receiving unit 65 does not detect congestion and supplies the RTT operation unit 61 with the time stamp added to the received ACK. . Then, the process proceeds to step S16. In step S16, the RTT calculation unit 61 calculates the difference between the time stamp supplied from the transmission unit 64 in step S11 and the time stamp supplied from the reception unit 65 in step S15 as an RTT. Then, the RTT calculation unit 61 supplies the RTT to the state determination unit 62 and the congestion window determination unit 63.

  In step S <b> 17, the state determination unit 62 determines whether or not there is a change in the RTT, that is, the difference between the latest RTT supplied from the RTT calculation unit 61 and the RTT supplied from the previous RTT calculation unit 61. Determine if there is no. If it is determined in step S17 that there is no RTT variation, the state determination unit 62 sets the state to the boost state 41 in step S18. Then, the state determination unit 62 supplies information indicating that the current state is the boost state 41 to the congestion window determination unit 63.

  In step S19, the congestion window determination unit 63 increases the congestion window cwnd by one segment according to the information supplied from the state determination unit 62 and indicating that the current state is the boost state. The congestion window determination unit 63 stores the congestion window cwnd as the current congestion window cwnd and supplies the congestion window cwnd to the transmission unit 64. This current congestion window cwnd is used in the next process of step S12.

  On the other hand, if it is determined in step S17 that there is a variation in RTT, the process proceeds to step S20. In step S <b> 20, the state determination unit 62 changes the state to the congestion avoidance state 42. Then, the state determination unit 62 supplies information indicating that the current state is the congestion avoidance state 42 to the congestion window determination unit 63.

In step S21, the congestion window determination unit 63 determines the past n times and the current RTT _j , the preset learning coefficient C, and the previous weight according to the information indicating that the current state is the congestion avoidance state 42. Using the coefficients w ₀ and w _i and the previous congestion window cwnd, the weight coefficients w ₀ and w ₁ are updated and stored in accordance with the above-described equations (4) and (5).

In step S22, the congestion window determination unit 63 determines the RTT _i stored in the past n times and the weighting factor w updated in step S21 according to the information indicating that the current state is the congestion avoidance state 42. _{Using 0} and w _i and the previous congestion window cwnd, the congestion window cwnd ′ is predicted according to the above-described equation (3). The congestion window determination unit 63 stores the congestion window cwnd ′ as the current congestion window cwnd and supplies the congestion window cwnd ′ to the transmission unit 64. This current congestion window cwnd is used in the next process of step S12.

  After step S19 or S22, in step S23, the transmission unit 64 determines whether all transmission data supplied from the CPU 21 has been transmitted. If it is determined in step S23 that all transmission data has not yet been transmitted, the process returns to step S11, and the above-described process is repeated. On the other hand, if it is determined in step S23 that all transmission data has been transmitted, the process ends.

  Hereinafter, the operation of communication using the above-described TCP Salt will be described using simulation results. This simulation is performed using the network simulator NS-2, where the learning coefficient C in TCP Salt is 0.6 and n, which is the number of RTT histories, is 5.

  First, the first simulation will be described.

  FIG. 6 shows the network topology of the first simulation.

  In the network topology of FIG. 6, two separate nodes, a node 80 and a node 84, and two separate nodes, a node 83 and a node 85, are connected across two intermediate nodes 81 and 82. Then, data conforms to UDP (User Datagram Protocol) and flows from the node 80 through the intermediate nodes 81 and 82 to the node 83, and data conforms to TCP Salt or TCP Reno and the data from the node 84 to the intermediate node 81. And 82 to the node 85.

  Note that the receiving terminal for TCP Salt and TCP Reno data flows is TCP Sink, and the maximum window size of TCP Salt and TCP Reno is 33 segments. This is the same in the simulation described later.

  Further, in the network topology of FIG. 6, the total bandwidth is 100 Mbps, the flow from the node 80 to the intermediate node 81, the flow from the node 84 to the intermediate node 81, the flow from the intermediate node 82 to the node 83, and the intermediate The flow from the node 82 to the node 85 causes a delay of 5 ms in the data. Further, the flow from the intermediate node 81 to the intermediate node 82 causes a delay of 40 ms in the data.

  In the first simulation, out of the total bandwidth of 100 Mbps, data flows in compliance with TCP Salt or TCP Reno with the 98.5 Mbps bandwidth being compressed by a UDP-compliant data flow (hereinafter referred to as UDP flow). . Specifically, a UDP flow is started 4 seconds after the first simulation is started, and a data flow conforming to TCP Salt or TCP Reno (hereinafter referred to as TCP Salt flow or TCP Reno flow) is started 5 seconds later. The Then, the TCP Salt flow or the TCP Reno flow is stopped after 50 seconds, the UDP flow is stopped after 51 seconds, and the first simulation itself is stopped after 55 seconds.

  Next, the result of the first simulation will be described with reference to FIGS.

  FIG. 7 is a graph showing temporal changes in the congestion window cwnd in the TCP Salt flow and TCP Reno flow in the first simulation. In FIG. 7, the horizontal axis represents the time (sec) from the start of the first simulation, and the vertical axis represents the congestion window cwnd (segment).

As shown in FIG. 7, in the TCP Salt flow, the congestion window cwnd decreases when the boost state 41 first transits to the congestion avoidance state 42, and thereafter gradually converges to a relatively large value. That is, when the transition to the congestion avoidance state 42 is made for the first time, the weighting factors w ₀ and w _i are not yet learned, so the TCP Salt flow cannot accurately predict the congestion window cwnd. Since the weight coefficients w ₀ and w _i are learned, the congestion window cwnd can be accurately predicted using the weight coefficients w ₀ and w _i . As a result, the congestion window cwnd gradually converges.

  On the other hand, in the TCP Reno flow, as shown in FIG. 7, the congestion window cwnd decreases at a constant cycle even after the first transition from the slow start state 11 to the congestion avoidance state 12. That is, in the TCP Reno flow, the congestion window cwnd cannot be reduced. Therefore, congestion occurs due to compression by the UDP flow, and the congestion window cwnd decreases each time the congestion is detected.

  FIG. 8 is a graph showing changes in throughput over time in the TCP Salt flow and TCP Reno flow in the first simulation. In FIG. 8, the horizontal axis represents time (sec) after the first simulation is started, and the vertical axis represents throughput (Mbps).

  As shown in FIG. 8, in the TCP Salt flow, the throughput is reduced only when the boost state 41 first transits to the congestion avoidance state 42, and thereafter, the throughput is the maximum value of the available bandwidth. It is stable at around 1.5Mbps. As a result, it is understood that the optimum congestion window cwnd can be predicted in the TCP Salt flow.

  On the other hand, as shown in FIG. 8, in the TCP Reno flow, the throughput decreases many times after the first transition from the slow start state 11 to the congestion avoidance state 12. That is, as shown in FIG. 7, in the TCP Reno flow, every time congestion occurs due to compression of the UDP flow, the congestion window cwnd is lowered, so that the throughput is also lowered.

  FIG. 9 shows average throughput, maximum throughput, and transfer data amount in the TCP Salt flow and TCP Reno flow of the first simulation.

  As shown in FIG. 9, the average throughput of the TCP Reno flow is 1.084 Mbps, and the average throughput of the TCP Salt flow is 1.428. Therefore, the improvement rate of the average throughput of the TCP Salt flow with respect to the TCP Reno flow is 31.7%.

  As shown in FIG. 9, the maximum throughput of the TCP Reno flow is 2.030 Mbps, and the maximum throughput of the TCP Salt flow is 2.030 Mbps, which is the same as that of the TCP Reno flow. Therefore, the improvement rate of the maximum throughput of the TCP Salt flow with respect to the TCP Reno flow is 0.0%.

  Further, as shown in FIG. 9, the transfer data amount of the TCP Reno flow is 6.096 Mb, and the transfer data amount of the TCP Salt flow is 8.030 Mb. Therefore, the improvement rate of the transfer data amount of the TCP Salt flow with respect to the TCP Reno flow is 31.7%.

  As described above, in the first simulation, it can be seen that the average throughput and the amount of transfer data in the TCP Salt flow are improved as compared with the TCP Reno flow.

  Next, the second simulation will be described.

  The network topology of the second simulation is the same as the network topology of the first simulation shown in FIG. In the second simulation, unlike the first simulation, the bandwidth pressed by the UDP flow is not always 98.5 Mbps, but from 20 seconds to 35 seconds after the start of the second simulation. For 15 seconds, the bandwidth squeezed by the UDP flow drops to 50 Mbps.

  FIG. 10 illustrates a time change of the throughput in the TCP Salt flow and the TCP Reno flow in the second simulation. In FIG. 10, the horizontal axis represents time (sec) since the start of the second simulation, and the vertical axis represents throughput (Mbps).

  As shown in FIG. 10, in the TCP Salt flow, compared to the TCP Reno flow, the compressed bandwidth is reduced. The throughput is immediately improved 20 seconds after the second simulation is started. This is because the bandwidth that is compressed by the UDP flow decreases and the available bandwidth of the TCP Salt flow increases, so the RTT does not fluctuate. Therefore, in the TCP Salt flow, the state immediately changes from the congestion avoidance state 42 to the boost state. This is because the rate of increase in the congestion window cwnd increases.

  On the other hand, in the TCP Reno flow, the state remains the congestion avoidance state 12, and therefore the increase rate of the congestion window cwnd is smaller than that in the TCP Salt flow.

  As described above, in the TCP Salt flow, the state of the current band is grasped by the fluctuation of RTT, so that it is possible to react more quickly to a change in the compressed band than in the TCP Reno flow.

  In addition, in the TCP Salt flow and the TCP Reno flow, the throughput stabilizes around 2.75 Mbps for 15 seconds from 20 seconds to 35 seconds after starting the second simulation in which the compressed bandwidth is reduced. At this time, the throughput of the TCP Salt flow and the TCP Reno flow is smaller than the available 50 Mbps bandwidth, and thus congestion does not occur.

  In addition, it is not 15 seconds from 20 seconds to 35 seconds after the start of the second simulation, but 33 seconds after 18 seconds when the throughput of TCP Reno flow has dropped most around 20 seconds. When the band compressed by the UDP flow is reduced to 50 Mbps for 15 seconds until, the difference in the response speed to the change of the compressed band becomes more remarkable as shown in FIG.

  Next, the third simulation will be described.

  FIG. 12 shows the network topology of the third simulation.

  In the network topology of FIG. 12, two separate nodes, node 100 and node 104, and two separate nodes, node 103 and node 105, are connected across two intermediate nodes 101 and 102. Then, as a flow compliant with any version of the first TCP (hereinafter referred to as TCP flow), data is transferred from the node 100 to the node 103 through the intermediate nodes 101 and 102 in accordance with TCP Salt or TCP Reno. As a second TCP flow, a TCP Salt flow or a TCP Reno flow is passed from the node 104 through the intermediate nodes 101 and 102 to the node 105.

  In the network topology of FIG. 12, the bandwidth between the intermediate nodes 101 and 102 is 2.5 Mbps, and the other bandwidth is 100 Mbps. Further, the flow from the node 100 to the intermediate node 101, the flow from the node 104 to the intermediate node 101, the flow from the intermediate node 102 to the node 103, and the flow from the intermediate node 102 to the node 105 cause a delay of 5 ms in the data. appear. Furthermore, the flow from the intermediate node 101 to the intermediate node 102 causes a 40 ms delay in the data.

  In the third simulation, the first TCP flow and the second TCP flow are TCP Salt flows. Specifically, the first TCP Salt flow as the first TCP flow is started 1 second after the first simulation is started, and the second TCP Salt flow as the second TCP flow is started 5 seconds later. Be started. Then, the first TCP Salt flow is stopped after 45 seconds, the second TCP Salt flow is stopped after 50 seconds, and the third simulation itself is stopped after 55 seconds.

  FIG. 13 is a graph showing changes in throughput over time in the first and second TCP Salt flows of the third simulation. In FIG. 13, the horizontal axis represents the time (sec) after the start of the third simulation, and the vertical axis represents the throughput (Mbps).

  As shown in FIG. 13, the first TCP flow has a throughput of 2.5 Mbps from 1 second after the start of the first TCP Salt flow to 5 seconds after the start of the second TCP Salt flow. The available bandwidth is utilized to the maximum extent.

  Similarly, the throughput of the second TCP Salt flow is 2.5 Mbps from 45 seconds after the first TCP Salt flow is stopped to 50 seconds after the second TCP Salt flow is stopped. The available bandwidth is utilized to the maximum extent.

  Furthermore, as shown in FIG. 13, the available bandwidth is used fairly even though there are some fluctuations from 5 seconds to 45 seconds after both the first and second TCP Salt flows are performed. Has been. That is, when TCP Salt flows share a bandwidth, the available bandwidth is used fairly.

  Next, the fourth simulation will be described.

  The network topology of the fourth simulation is the same as the network topology of the third simulation shown in FIG. Note that, in the fourth simulation, unlike the third simulation, data flows as the second TCP flow in conformity with TCP Reno. That is, in the fourth simulation, the first TCP flow is a TCP Salt flow, and the second TCP flow is a TCP Reno flow.

  FIG. 14 is a graph showing changes in throughput over time in the TCP Salt flow and TCP Reno flow of the fourth simulation. In FIG. 14, the horizontal axis represents the time (sec) since the start of the fourth simulation, and the vertical axis represents the throughput (Mbps).

  As shown in FIG. 14, the TCP Salt flow has a throughput of 2.5 Mbps from 1 second after the start of the TCP Salt flow to 5 seconds after the start of the TCP Reno flow. Is used to the maximum.

  Similarly, from 45 seconds after the TCP Salt flow is stopped to 50 seconds after the TCP Reno flow is stopped, the TCP Reno flow has a throughput of 2.5 Mbps and the maximum available bandwidth Limited use.

  Furthermore, between 5 seconds and 45 seconds after both TCP Salt flow and TCP Reno flow are performed, as shown in FIG. 14, the available bandwidth is used fairly even though there is some variation. Yes. That is, even when the TCP Salt flow and the TCP Reno flow share a bandwidth, the available bandwidth is used fairly.

  From the above first to fourth simulations, it is understood that the available bandwidth can be sufficiently used in the TCP Salt flow while maintaining the fairness of the bandwidth used with other connections. It can also be seen that the TCP Salt flow can quickly respond to changes in the available bandwidth.

Next, transition of the weighting factors w ₀ and w _i used when predicting the congestion window cwnd ′ in TCP Salt will be described using simulation results.

First, the transition of the weighting factors w _{0 to} w ₅ during the first simulation will be described with reference to FIG. In FIG. 15, the horizontal axis represents the time (sec) from the start of the first simulation, and the vertical axis represents the values of the weighting factors w _{0 to} w ₅ . The same applies to FIG. 16 described later.

As shown in FIG. 15, in the first simulation, the weighting factors w _{0 to} w ₅ converge at a constant time. In addition, in the first simulation, a large weighting factor w ₁ for the most recent past of IRTT ₁ is the best, the weighting factor w ₀ for the last time of the congestion window cwnd is not very large.

Next, FIG. 16 shows the transition of the weighting factors w _{0 to} w ₅ during the second simulation. In FIG. 16, in the second simulation, the bandwidth pressed by the UDP flow is reduced to 50 Mbps for 15 seconds from 20 seconds to 35 seconds after the second simulation is started.

As described above, in the second simulation, congestion does not occur for 15 seconds from 20 seconds to 35 seconds after the bandwidth compressed by the UDP flow is reduced to 50 Mbps, and the boost state 41 is set in the TCP Salt flow. Therefore, as shown in FIG. 16, the weighting factors w _{0 to} w ₅ do not change. In the second simulation, similar to the first simulation, larger weighting coefficients w ₁ with respect to the most recent past irtt ₁ is best, the weight coefficient w ₀ for the previous congestion window cwnd is not very large.

As described above, the weighting factor w ₀ for the congestion window cwnd is not so large. Therefore, the necessity of the previous congestion window cwnd used when the congestion window cwnd ′ is predicted will be described below using simulation results.

  First, the fifth simulation will be described.

  The fifth simulation is not a TCP Salt flow or a TCP Reno flow in the first simulation, but a flow conforming to the first or second modification of the TCP Salt (hereinafter referred to as a first modified TCP Salt flow or a second modified TCP). (Salt flow) or TCP Salt flow.

Here, the first modification of TCP Salt is one in which the weight coefficient w _{0 is set} to 0 in TCP Salt and n, which is the number of RTT histories, is set to 5, and the second modification is in TCP Salt. The weight coefficient w ₀ is 0 and n is 6.

  FIG. 17 is a graph showing changes in throughput over time in the first modified TCP Salt flow, the second modified TCP Salt flow, and the TCP Salt flow of the fifth simulation. In FIG. 17, the horizontal axis represents time (sec) since the start of the fifth simulation, and the vertical axis represents throughput (Mbps).

  As shown in FIG. 17, in the fifth simulation, there is no difference in throughput between the first modified TCP Salt flow, the second modified TCP Salt flow, and the TCP Salt flow.

  Next, the sixth simulation will be described.

  The sixth simulation uses the first modified TCP Salt flow, the second modified TCP Salt flow, or the TCP Salt flow instead of the TCP Salt flow or the TCP Reno flow in the second simulation.

  FIG. 18 is a graph showing changes in throughput over time in the first modified TCP Salt flow, the second modified TCP Salt flow, and the TCP Salt flow of the sixth simulation. In FIG. 18, the horizontal axis represents time (sec) after the sixth simulation is started, and the vertical axis represents throughput (Mbps).

  As shown in FIG. 18, even in the sixth simulation, there is no difference in throughput between the first modified TCP Salt flow, the second modified TCP Salt flow, and the TCP Salt flow.

  Next, the seventh simulation will be described.

  FIG. 19 shows a network topology of the seventh simulation.

  In the network topology of FIG. 19, three separate nodes, node 120, node 124, and node 126, and three separate nodes, node 123, node 125, and sandwiched between two intermediate nodes 121 and 122. Node 127 is connected.

  As a UDP flow, in accordance with UDP, data flows from the node 120 through the intermediate nodes 121 and 122 to the node 123. As the first TCP flow, the first modification of TCP Reno, TCP Vegas, and TCP Salt In accordance with the second variation of TCP Salt, or TCP Salt, data flows from node 124 through intermediate nodes 121 and 122 to node 125. Further, as the second TCP flow, in accordance with TCP Reno, TCP Vegas, the first modification of TCP Salt, the second modification of TCP Salt, or TCP Salt, the node 127 passes from the node 126 through the intermediate nodes 121 and 122 to the node 127. Data is streamed to

  In the network topology of FIG. 19, the entire band is 100 Mbps. Also, the flow from the node 120 to the intermediate node 121, the flow from the node 124 to the intermediate node 121, the flow from the node 126 to the intermediate node 121, the flow from the intermediate node 122 to the node 123, and the flow from the intermediate node 122 to the node 125 The flow and the flow from the intermediate node 122 to the node 127 cause a 5 ms delay in the data. Further, the flow from the intermediate node 121 to the intermediate node 122 causes a delay of 40 ms in the data.

  In the seventh simulation, the first TCP flow and the second TCP flow are TCP Reno flows.

  FIG. 20 is a graph showing changes in throughput over time in the first and second TCP Reno flows of the seventh simulation. In FIG. 20, the horizontal axis represents time (sec) since the start of the seventh simulation, and the vertical axis represents throughput (Mbps).

  As shown in FIG. 20, in the seventh simulation, the solid line representing the throughput in the first TCP Reno flow and the dotted line representing the throughput in the second TCP Reno flow do not overlap very much. That is, when TCP Reno flows share a bandwidth, the available bandwidth is not used fairly.

  Next, the eighth to fifteenth simulations will be described.

  The network topologies of the eighth to fifteenth simulations are the same as the network topology of the seventh simulation shown in FIG. In the eighth simulation, unlike the seventh simulation, the first and second TCP flows are data flows compliant with TCP Vegas (hereinafter referred to as TCP Vegas flows).

  FIG. 21 is a graph showing changes in throughput over time in the first and second TCP Vegas flows of the eighth simulation.

  As shown in FIG. 21, in the eighth simulation, of the first TCP Vegas flow and the second TCP Vegas flow, the one that starts first has an advantage in bandwidth utilization. That is, when TCP Vegas flows share a bandwidth, the available bandwidth is not used fairly.

  Next, the ninth to eleventh simulations will be described.

  In the ninth simulation, the first and second TCP flows are the first modified TCP Salt flow, in the tenth simulation, the second modified TCP Salt flow, and in the eleventh simulation, the TCP Salt flow.

  22 to 24 show the first and second first modified TCP Salt flows, the first and second modified TCP Salt flows, and the first and second TCP Salts of the ninth to eleventh simulations. It is a graph which shows the time change of the throughput in each of the flows.

  As shown in FIGS. 22 to 24, the fairness of band utilization is high in the order of the eleventh simulation, the ninth simulation, and the tenth simulation. That is, the fairness is the highest when the TCP Salt flows share the bandwidth, and the fairness is the second highest when the first modified TCP Salt flows share the bandwidth, and the second modified TCP Salt flows Sharing bandwidth is the least fair.

  Next, the twelfth to fourteenth simulations will be described.

  In the twelfth simulation, the first TCP flow is the first modified TCP Salt flow, in the thirteenth simulation, the second modified TCP Salt flow, and in the fourteenth simulation, the TCP Salt flow. In the twelfth to fourteenth simulations, the second TCP flow is a TCP Reno flow.

  FIGS. 25 to 27 are graphs showing temporal changes in throughput in the first modified TCP Salt flow, the second modified TCP Salt flow, and the TCP Salt flow and the TCP Reno flow in the twelfth to fourteenth simulations. It is.

  As shown in FIG. 25, in the twelfth simulation, the first modified TCP Salt flow is superior in bandwidth utilization compared to the TCP Reno flow. Also, as shown in FIG. 26, in the thirteenth simulation, the second modified TCP Salt flow is superior in bandwidth utilization compared to the TCP Reno flow, as in the first modified TCP Salt flow. . Furthermore, as shown in FIG. 27, in the fourteenth simulation, the bandwidths used for the TCP Salt flow and the TCP Reno flow are almost fair.

  Next, the fifteenth simulation will be described.

  In the fifteenth simulation, the first TCP flow is a TCP Vegas flow, and the second TCP flow is a TCP Reno flow.

  FIG. 28 is a graph showing changes in throughput over time in the TCP Vegas flow and TCP Reno flow of the fifteenth simulation.

  As shown in FIG. 28, in the fifteenth simulation, the fairness of the used bandwidth of the TCP Vegas flow and the TCP Reno flow is relatively good, but the throughput in the TCP Vegas flow and the TCP Reno flow is small.

  FIG. 29 is a table showing transfer amounts (bytes) in the seventh, eighth, and fifteenth simulations, and FIG. 30 is a table showing transfer amounts (bytes) in the ninth to eleventh simulations. FIG. 31 is a table showing transfer amounts (bytes) in the twelfth to fourteenth simulations. Further, FIG. 32 is a graph showing the transfer amount (byte) in the seventh to fifteenth simulations.

  As shown in FIG. 29, in the seventh simulation, the transfer amount in the TCP Reno flow as the first TCP flow is 3410 bytes, and the transfer amount in the TCP Reno flow as the second TCP flow is 4104 bytes. . Therefore, as shown in FIGS. 29 and 32, in the seventh simulation, the sum of the transfer amounts in the first TCP flow and the second TCP flow is 7514 bytes, and the absolute value of the difference is 694.

  In the eighth simulation, as shown in FIG. 29, the transfer amount in the TCP Vegas flow as the first TCP flow is 7851 bytes, and the transfer amount in the TCP Vegas flow as the second TCP flow is 2176 bytes. . Therefore, as shown in FIGS. 29 and 32, in the eighth simulation, the sum of the transfer amounts in the first TCP flow and the second TCP flow is 1,0027 bytes, and the absolute value of the difference is 5675.

  In the fifteenth simulation, as shown in FIG. 29, the transfer amount in the TCP Vegas flow as the first TCP flow is 3320 bytes, and the transfer amount in the TCP Reno flow as the second TCP flow is 3700 bytes. . Therefore, as shown in FIGS. 29 and 32, in the fifteenth simulation, the sum of the transfer amounts in the first TCP flow and the second TCP flow is 7020 bytes, and the absolute value of the difference is 380.

  Also, as shown in FIG. 30, in the ninth simulation, the transfer amount in the first first modified TCP Salt flow as the first TCP flow is 3849 bytes, and the second TCP flow as the second TCP flow. The transfer amount in the first modified TCP Salt flow is 4397 bytes. Therefore, as shown in FIGS. 30 and 32, in the ninth simulation, the sum of the transfer amounts in the first TCP flow and the second TCP flow is 8246 bytes, and the absolute value of the difference is 548.

  In the tenth simulation, as shown in FIG. 30, the transfer amount in the first second modified TCP Salt flow as the first TCP flow is 3818 bytes, and the second second as the second TCP flow. The transfer amount in the modified TCP Salt flow is 4462 bytes. Therefore, as shown in FIGS. 30 and 32, in the tenth simulation, the sum of the transfer amounts in the first TCP flow and the second TCP flow is 8280 bytes, and the absolute value of the difference is 644.

  In the eleventh simulation, as shown in FIG. 30, the transfer amount in the first TCP Salt flow as the first TCP flow is 3960 bytes, and the transfer in the second TCP Salt flow as the second TCP flow is performed. The amount is 4273 bytes. Therefore, as shown in FIGS. 30 and 32, in the eleventh simulation, the sum of the transfer amounts in the first TCP flow and the second TCP flow is 8233 bytes, and the absolute value of the difference is 313.

  Further, as shown in FIG. 31, in the twelfth simulation, the transfer amount in the first modified TCP Salt flow as the first TCP flow is 4743 bytes, and the transfer amount in the TCP Reno flow as the second TCP flow. Is 3352 bytes. Therefore, as shown in FIGS. 31 and 32, in the twelfth simulation, the sum of the transfer amounts in the first TCP flow and the second TCP flow is 8095 bytes, and the absolute value of the difference is 1391.

  In the thirteenth simulation, as shown in FIG. 31, the transfer amount in the second modified TCP Salt flow as the first TCP flow is 4783 bytes, and the transfer amount in the TCP Reno flow as the second TCP flow is 3456. It is a byte. Therefore, as shown in FIGS. 31 and 32, in the thirteenth simulation, the sum of the transfer amounts in the first TCP flow and the second TCP flow is 8239 bytes, and the absolute value of the difference is 1327.

  In the fourteenth simulation, as shown in FIG. 31, the transfer amount in the TCP Salt flow as the first TCP flow is 4316 bytes, and the transfer amount in the TCP Reno flow as the second TCP flow is 3824 bytes. . Therefore, as shown in FIGS. 31 and 32, in the fourteenth simulation, the sum of the transfer amounts in the first TCP flow and the second TCP flow is 8140 bytes, and the absolute value of the difference is 492.

  As shown in FIGS. 29 to 32, in the eighth simulation, the sum of the transfer amount in the TCP Vegas flow as the first TCP flow and the transfer amount in the TCP Vegas flow as the second TCP flow is the maximum. However, the difference in the transfer amount is also the largest. That is, when TCP Vegas flows share a bandwidth, the overall transfer amount is large, but the fairness is poor.

  In the seventh and fifteenth simulations, the difference between the transfer amount in the TCP Reno flow or TCP Vegas flow as the first TCP flow and the transfer amount in the TCP Reno flow as the second TCP flow is small. The sum of the transfer amount is also small. That is, when the TCP Reno flows, or the TCP Reno flow and the TCP Vegas flow share the bandwidth, the fairness is relatively good, but the overall transfer amount is reduced.

  On the other hand, in the ninth to fourteenth simulations, the transfer amount in the first modified TCP Salt flow, the second modified TCP Salt flow, or the TCP Salt flow as the first TCP flow, and the second TCP flow In the first modified TCP Salt flow, the second modified TCP Salt flow, the TCP Salt flow, or the TCP Reno flow, the difference in transfer amount is small, but the sum of these transfer amounts is relatively large.

  That is, each of the first modified TCP Salt flow, the second modified TCP Salt flow, and the TCP Salt flow, the first modified TCP Salt flow, the second modified TCP Salt flow, and the TCP Salt flow and the TCP Reno flow have different bandwidths. When shared, the bandwidth used is fair and the overall transfer amount is large.

  In addition, when the first modified TCP Salt flow, the second modified TCP Salt flow, and the TCP Salt flow share a bandwidth, and each of the first modified TCP Salt flow, the second modified TCP Salt flow, and the TCP Salt flow There is almost no difference in the total transfer amount when the TCP Reno flow shares the bandwidth, so even if the current TCP Reno does not migrate to TCP Salt at one time, but gradually migrates, It turns out that there is no problem.

Furthermore, in the ninth to fourteenth simulations, the sum of the transfer amounts in the first TCP flow and the second TCP flow is almost the same, but there is a difference in the difference. Specifically, the difference in the eleventh simulation is smaller than that in the ninth and tenth simulations, and the difference in the fourteenth simulation is smaller than in the twelfth and thirteenth simulations. That is, among the first modified TCP Salt flow, the second modified TCP Salt flow, and the TCP Salt flow, the TCP Salt flow has the best fairness. Therefore, it can be seen that the weighting factor w ₀ is necessary to maintain the fairness of the used bandwidth.

  When distributing multimedia data, the communication device 20 allocates Internet bandwidth based on the congestion window cwnd determined according to TCP Salt, and uses the control method described in Patent Literature 3 to control the Internet. The transmission interval of individual transmission data may be determined according to the state, and the transmission data may be transmitted using the allocated band at the transmission interval.

  Further, in this specification, the step of describing the program stored in the program recording medium is not limited to the processing performed in time series in the described order, but is not necessarily performed in time series. Or the process performed separately is also included.

  Furthermore, the embodiments of the present invention are not limited to the above-described embodiments, and various modifications can be made without departing from the gist of the present invention.

It is a state transition diagram of TCP Reno. It is a figure which shows the structural example of the hardware of one Embodiment of the communication apparatus to which this invention is applied. It is a state transition diagram of TCP Salt. It is a figure which shows the functional structural example of the communication part 29 of FIG. It is a flowchart explaining the determination process by a communication part. It is a figure which shows the network topology of a 1st simulation. It is a graph which shows the time change of the congestion window of the 1st simulation. It is a graph which shows the time change of the throughput of the 1st simulation. It is a figure showing the average throughput of 1st simulation, the maximum throughput, and the amount of transfer data. It is a graph which shows the time change of the throughput of the 2nd simulation. It is a graph which shows the time change of the throughput of other 2nd simulations. It is a figure which shows the network topology of a 3rd simulation. It is a graph which shows the time change of the throughput of the 3rd simulation. It is a graph which shows the time change of the throughput of the 4th simulation. It is a graph which shows transition of the weighting coefficient in the 1st simulation. It is a graph which shows transition of the weighting coefficient in the 2nd simulation. It is a graph which shows the time change of the throughput of the 5th simulation. It is a graph which shows the time change of the throughput of the 6th simulation. It is a figure which shows the network topology of a 7th simulation. It is a graph which shows the time change of the throughput of the 7th simulation. It is a graph which shows the time change of the throughput of the 8th simulation. It is a graph which shows the time change of the throughput of the 9th simulation. It is a graph which shows the time change of the throughput of the 10th simulation. It is a graph which shows the time change of the throughput of the 11th simulation. It is a graph which shows the time change of the throughput of the 12th simulation. It is a graph which shows the time change of the throughput of the 13th simulation. It is a graph which shows the time change of the throughput of the 14th simulation. It is a graph which shows the time change of the throughput of the 15th simulation. It is a figure which shows the transfer amount in 7th, 8th, and 15th simulation. It is a figure which shows the transfer amount in 9th thru | or 11th simulation. It is a figure which shows the transfer amount in the 12th thru | or 14th simulation. It is a graph which shows the transfer amount in the 7th thru | or 15th simulation.

Explanation of symbols

  20 communication device, 63 congestion window determination unit, 64 transmission unit

Claims (4)

In a communication device that performs communication via a network,
A prediction means for predicting a new congestion window using the previous congestion window and RTT history,
A communication apparatus comprising: a transmission unit that allocates a bandwidth of the network using the predicted congestion window and transmits data in the bandwidth. The communication apparatus according to claim 1, wherein the prediction unit predicts a new congestion window using the immediately preceding congestion window and a history of the reciprocal of the RTT. The communication apparatus according to claim 2, wherein the prediction unit linearly predicts a new congestion window using the immediately previous congestion window and a history of the reciprocal of the RTT. In a communication method of a communication device that performs communication via a network,
Predict a new congestion window using the previous congestion window and RTT history,
A communication method including the steps of allocating a bandwidth of the network using the predicted congestion window and transmitting data in the bandwidth.

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