JP2009182653A - Wireless communication device - Google Patents

Abstract

<P>PROBLEM TO BE SOLVED: To provide a wireless communication device performing handovers to different wireless communication networks without reducing reproduction quality and immediacy. <P>SOLUTION: When a handover is performed from a first wireless communication network 15 to a second wireless communication network 16, the reproduction speed of an application is calculated by a reproduction speed calculation part 56 on the basis of the preparation time until the handover, respective delay times of the first wireless communication network 15 and second wireless communication network 16, and the amount of data in a jitter buffer 47 at the point of time at which the start of the handover preparation is determined, and the reproduction speed of the running application is controlled on the basis of the calculated reproduction speed and an evaluation value stored in a reproduction speed evaluation part 57. <P>COPYRIGHT: (C)2009,JPO&INPIT

Description

  The present invention relates to a radio communication apparatus capable of performing handover between different radio communication networks.

  In recent years, the Internet Engineering Task Force (IETF) has developed an IP mobility technology that enables seamless movement by enabling handover between a plurality of different wireless communication networks such as a mobile phone network and a wireless LAN in order to realize a ubiquitous environment. Is being considered. Specific protocols in this IP mobility technology include Mobile IPv4 and Mobile IPv6 (hereinafter collectively referred to as Mobile IP) that support movement of individual communication terminals, and support movement in units of networks. There is NEMO (Network Mobility).

  By the way, when an application having real-time characteristics such as VoIP (hereinafter, abbreviated as an application as appropriate) is executed via a wireless communication network, the allowable bandwidth of the wireless communication path changes depending on a propagation environment such as fading. Then, the arrival interval of packets received by the communication terminal also changes according to the change in the allowable bandwidth.

  For this reason, generally, a jitter buffer is provided in a communication terminal, and received packets are temporarily stored in the jitter buffer.Then, packets are read out from the jitter buffer at an interval according to the application and played back, thereby causing packet fluctuations. That is, a packet reproduction interval shift due to arrival interval shift (jitter) is absorbed to prevent a decrease in reproduction quality such as reproduction sound quality. Also, if the jitter is large and there are no packets in the jitter buffer and silence occurs, or if a large amount of packets are received in a short time and the packets do not fit in the jitter buffer, the playback speed is increased. It changes, discards received packets, and changes the size of the jitter buffer.

  On the other hand, the downlink absolute delay time of the packet received by the communication terminal, that is, the time (delay time) required until the packet transmitted from the partner communication terminal is received via the wireless communication network depends on the wireless communication network. Different. For this reason, in the case of a wireless communication apparatus in which a communication terminal moves, when handover is performed to a different wireless communication network, for example, if the downlink absolute delay time of the handover destination is longer than the downlink absolute delay time of the handover source, the difference As a result, a packet reception idle time is generated.

  In such a case, for example, when the packet reading interval from the jitter buffer is set to a constant interval according to the application and the packet is reproduced at a constant reproduction speed, the last packet received from the handover source wireless communication network is When the reception idle time is longer than the time required for reading from the jitter buffer (that is, the jitter buffer standard delay time), the packet in the jitter buffer is emptied for the longer time. As a result, since the packet is not played back at least during this time period, the sound becomes silent and the playback quality deteriorates.

  FIG. 14 is a diagram for explaining a control method of the jitter buffer in this case. FIG. 14A shows the number of packets received by the jitter buffer per unit time, and FIG. The packet playback speed (reading interval), FIG. 14C shows the number of packets in the jitter buffer. FIG. 15 shows the packet flow in this case, where “transmission” is the packet transmission timing by the counterpart communication terminal, “reception” is the reception timing of the packet received by the jitter buffer of the wireless communication device, “Reproduction” indicates packet reproduction timing (packet read timing from the jitter buffer) by the wireless communication apparatus. Here, in each of the handover source radio communication network A and the handover destination radio communication network B, it is shown that there is no fluctuation (shift in arrival interval) in the received packet.

  As apparent from FIGS. 14 and 15, the downlink absolute delay time TddnB in the handover destination radio communication network B is longer than the downlink absolute delay time TddnA in the handover source radio communication network A, and (TddnB−TddnA). Is longer than the jitter buffer standard delay time Ta received by the received packet when the standard number of packets is accumulated in the jitter buffer, the time of Tn = {(TddnB−TddnA) −Ta} is Packets are not played back. In addition, in this case, when a packet is received from the wireless communication network B that is the handover destination, the packet is immediately reproduced, so that jitter cannot be absorbed.

  For example, if the packet reception status is monitored and the packet cannot be received at a normal reception interval, reading of the packet from the jitter buffer, that is, the packet A jitter buffer control method for controlling the reproduction speed has been proposed (see, for example, Patent Document 1).

  FIG. 16 is a diagram for explaining the jitter buffer control method disclosed in Patent Document 1. FIGS. 16A to 16C are similar to FIGS. 14A to 14C, respectively. The number of received packets per unit time, the playback speed, and the number of packets in the jitter buffer are shown. FIG. 17 shows the packet flow in this case.

  In FIG. 16 and FIG. 17, the packet is received at the reception interval until then, as in the case of handover from the wireless communication network A having the downlink absolute delay time TddnA to the wireless communication network B having the downlink absolute delay time TddnB longer than TddnA. If the packet cannot be received, the playback speed of the packets in the jitter buffer is gradually reduced as the reception interval increases, and if the reception interval returns to normal after that, it depends on the number of packets in the jitter buffer. Thus, the playback speed is controlled to gradually increase to the normal playback speed.

JP 2006-238445 A

  However, in the jitter buffer control method disclosed in Patent Document 1, if a packet cannot be received at the previous reception interval, the playback speed of the packets currently accumulated in the jitter buffer is gradually decreased. It just controls it. For this reason, when the downlink absolute delay time TddnB at the handover destination is relatively long, there is a concern that the low speed change of the reproduction speed becomes large and the reproduction quality is deteriorated. Therefore, for example, in VoIP, the playback speed greatly changes from the original voice speed, so that the quality of the playback sound is greatly reduced, making it difficult for the user to hear.

  FIGS. 16 and 17 show examples of control in the case where the packet in the jitter buffer is empty and no silence is generated. However, how much the packet reception interval is actually free. Therefore, depending on the jitter buffer standard delay time Ta and the handover destination downlink absolute delay time TddnB, there is a concern that the packet in the jitter buffer is emptied and silence is generated. Note that the jitter buffer standard delay time Ta may be increased in order to prevent the occurrence of silence or the like. However, for example, in VoIP, there is a delay in packet reproduction from the partner terminal. Therefore, the real-time property is lacking.

  Accordingly, an object of the present invention made in view of such a point is to provide a wireless communication apparatus capable of performing handover to a different wireless communication network without deteriorating reproduction quality and real-time property.

The invention of a wireless communication device according to claim 1 that achieves the above object is as follows:
A wireless communication unit that performs wireless communication by connecting to a first wireless communication network and a second wireless communication network different from the first wireless communication network;
A jitter buffer and a jitter buffer monitoring unit that monitors a data amount of the jitter buffer, and an execution unit that executes an application of a real-time communication system via the wireless communication unit;
A communication quality acquisition unit for acquiring communication quality of a radio link in the first wireless communication network while executing the application by connecting to the first wireless communication network;
A determination unit that determines whether to start preparation for handover from the first wireless communication network to the second wireless communication network based on the communication quality acquired by the communication quality acquisition unit;
When the determination unit determines the start of handover preparation during the execution of the application, an estimation unit that estimates a handover preparation time until the handover is started based on the communication quality acquired by the communication quality acquisition unit;
When the determination unit determines the start of handover preparation, a measurement unit that measures respective delay times in the first wireless communication network and the second wireless communication network;
The handover preparation time estimated by the estimation unit, the respective delay times measured in the first wireless communication network and the second wireless communication network measured by the measurement unit, and the time when the determination unit determines the start of handover preparation Based on the amount of data in the jitter buffer by the jitter buffer monitoring unit, a playback speed calculation unit that calculates the playback speed of the application, and a playback speed evaluation unit that stores an evaluation value for each playback speed of the application, A control unit for controlling the playback speed of the application by the execution unit based on the playback speed calculated by the playback speed calculation unit and the evaluation value stored in the playback speed evaluation unit;
It is characterized by providing.

The invention according to claim 2 is the wireless communication apparatus according to claim 1,
The control unit compares the delay time in the first wireless communication network with the delay time in the second wireless communication network, and the delay time in the second wireless communication network is the delay in the first wireless communication network. The playback speed of the application by the execution unit is reduced when the time is longer than a predetermined time by a predetermined time.

The invention according to claim 3 is the wireless communication apparatus according to claim 2,
The control unit slows down the playback speed of the application by the execution unit after starting preparation for handover.

The invention according to claim 4 is the wireless communication device according to claim 1, 2 or 3,
When the playback speed calculated by the playback speed calculation unit is equal to or higher than the minimum playback speed in the maximum evaluation value stored in the playback speed evaluation unit, the control unit plays the application at the calculated playback speed. To control and
When the playback speed calculated by the playback speed calculation unit is less than the minimum playback speed in the maximum evaluation value, a composite evaluation value consisting of a combination of a plurality of playback speeds corresponding to a plurality of evaluation values is calculated by the playback speed evaluation unit. And the composite evaluation value is calculated from the calculated composite evaluation value in the period from the time when the reproduction speed calculation unit determines the start of the handover preparation to the time when data reception from the second wireless communication network starts. A combination of high playback speeds is selected, and the application is controlled to be played back at the selected combination of playback speeds.

The invention according to claim 5 is the wireless communication device according to claim 2, 3 or 4,
When the data amount of the jitter buffer monitored by the jitter buffer monitoring unit exceeds a predetermined amount after the handover to the second wireless communication network is completed, the control unit sets the playback speed of the application to a normal speed. It is characterized by returning.

  In the present invention, when handing over from the first wireless communication network to the second wireless communication network, the preparation time up to the handover, the respective delay times in the first wireless communication network and the second wireless communication network, and the handover in advance The amount of data in the jitter buffer at the time when the start of preparation is determined is acquired, and the playback speed calculation unit calculates the playback speed of the application based on the acquired information, and the calculated playback speed and playback speed The playback speed of the application is controlled based on the evaluation value for each playback speed stored in the evaluation unit. Thereby, for example, when the delay time of the second radio communication network of the handover destination is longer than the delay time of the first radio communication network of the handover source, the reproduction speed is slowed down to absorb the delay time difference. The application can be played at a highly rated playback speed while increasing the time available. Therefore, playback can be performed at a speed closer to the standard playback speed. Further, even when the delay time difference is large and playback cannot be performed at the standard playback speed, playback can be performed at a highly evaluated playback speed, so that the second wireless communication can be performed from the first wireless communication network without deteriorating the playback quality and real-time performance. Handover to the network is possible.

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

  FIG. 1 is a diagram showing a schematic configuration of a communication network that can be used by a wireless communication apparatus according to an embodiment of the present invention. In FIG. 1, it is assumed that a wireless communication device 11 that is a mobile node makes a call with a partner communication terminal 12 that is an opposite node using VoIP that is a real-time communication application. The wireless communication device 11 can be handed over between the first wireless communication network 15 and the second wireless communication network 16. The first wireless communication network 15 and the second wireless communication network 16 are coupled to the Internet 18 via a packet network 17.

  Here, the first wireless communication network 15 is composed of, for example, a wireless LAN, and the second wireless communication network 16 is composed of, for example, a cdma2000 EV-DO mobile phone network, and the delay time (downward absolute delay in the first wireless communication network 15). (Time) is shorter than the delay time (downlink absolute delay time) in the second wireless communication network 16. In FIG. 1, reference numeral 15 a indicates an access point of the first wireless communication network 15, and reference numeral 16 a indicates a base station of the second wireless communication network 16.

  For example, the partner communication terminal 12 includes a personal computer to which a handset 12a is connected and a softphone is installed, and is connected to the Internet 18 through an Internet service provider (not shown).

  Further, SIP (Session Initiation Protocol) servers 21 and 22 for controlling communication are connected to the packet network 17 and the Internet 18, respectively. Further, a home agent (HA) 23 is connected to the Internet 18 for transferring a received packet addressed to the wireless communication device 11 to a wireless communication network to which the wireless communication device 11 is connected.

  In the communication network shown in FIG. 1, the home address used in the wireless communication network to which the wireless communication device 11 originally belongs is registered in the HA 23, and the care-of address of the wireless communication network 16 that is the handover destination is set at the time of handover. Registration enables handover between different wireless communication networks. Note that such IP mobility technology is well known in the above-described mobile IP and NEMO, and thus detailed description thereof is omitted here.

  In the present embodiment, it is assumed that the wireless communication network to which the wireless communication device 11 originally belongs is the first wireless communication network 15 and is handed over from the first wireless communication network 15 to the second wireless communication network 16.

  FIG. 2 is a functional block diagram showing a schematic configuration of radio communication apparatus 11 according to the present embodiment shown in FIG. The wireless communication device 11 executes a first wireless I / F (interface) 31 corresponding to the first wireless communication network 15, a second wireless I / F 32 corresponding to the second wireless communication network 16, and a VoIP application. The wireless information of the telephone function unit 33, the communication processing unit 34 that controls connection to the first wireless communication network 15 and the second wireless communication network 16, and the wireless information of the first wireless communication network 15 and the second wireless communication network 16 is acquired. A wireless information acquisition unit 35; and a handover control unit 36 that controls a handover between the first wireless communication network 15 and the second wireless communication network 16.

  The communication processing unit 34 constitutes a wireless communication unit that performs wireless communication. Between the telephone function unit 33 and the counterpart communication terminal 12, the first wireless communication network 15 or the second wireless communication network 16 is used. While making a call, the connection of the first wireless I / F 31 or the second wireless I / F 32 is controlled so as to communicate with the HA 23 under the control of the handover control unit 36.

  The wireless information acquisition unit 35 acquires the communication quality of the corresponding first wireless communication network 15 and second wireless communication network 16 from the first wireless I / F 31 and the second wireless I / F 32 as wireless information, The acquired communication quality is supplied to the handover control unit 36. Here, as the communication quality, for example, an RSSI (Received Signal Strength Indicator) representing a radio state is acquired. Therefore, in the present embodiment, the radio information acquisition unit 35 constitutes a communication quality acquisition unit that acquires the communication quality of the radio link.

  Based on the communication quality from the radio information acquisition unit 35, the handover control unit 36 generates handover information including a decision whether to schedule a handover, that is, whether to start preparation for handover, and the handover information Based on the above, handover is controlled.

  FIG. 3 is a functional block diagram showing a schematic configuration of the telephone function unit 33 of the wireless communication apparatus 11 shown in FIG. The telephone function unit 33 is composed of, for example, a soft phone, and has a button input unit 41, a screen display unit 42, a microphone 43, an encoder 44, a packet transmission unit 45, a packet reception unit 46, a jitter buffer, as in the configuration of a known soft phone. 47, a decoder 48, a speaker 49, a jitter buffer monitoring unit 50, a jitter buffer control unit 51, a SIP control unit 52, and an overall control unit 53 for controlling the overall operation.

  The overall control unit 53 acquires user operation information via the button input unit 41 and the screen display unit 42, and controls the overall operation based on the acquired information. The SIP control unit 52 controls SIP procedures for starting and ending a call. During a call, the audio data acquired from the microphone 43 is encoded by the encoder 44, and the encoded data is put into a packet from the packet transmission unit 45 and transmitted to the partner communication terminal 12 via the communication processing unit 34.

  Further, the packet from the partner communication terminal 12 received by the packet receiving unit 46 via the communication processing unit 34 is once fetched into the jitter buffer 47 and then read, and the read packet is decoded by the decoder 48 in the payload portion, Output as reproduced sound from the speaker 49. The reception status of packets in the jitter buffer 47 and the number of packets (data amount) in the jitter buffer 47 are monitored by the jitter buffer monitoring unit 50. Based on the monitoring result, the jitter buffer control unit 51 Controls the reading speed of the packet from the jitter buffer 47 and processing such as discarding the received packet.

  In the present embodiment, the telephone function unit 33 is further provided with a handover information acquisition unit 55, a playback speed calculation unit 56, and a playback speed evaluation unit 57. The handover information acquisition unit 55 monitors the handover information from the handover control unit 36 at regular intervals, and acquires information on whether or not there is a handover schedule. When there is a handover schedule, the handover information acquisition unit 55 further acquires the required handover information from the handover control unit 36 and supplies the acquired required handover information to the playback speed calculation unit 56.

  Based on the required handover information acquired from the handover information acquisition unit 55, the playback speed calculation unit 56 reads the packet reading speed of the jitter buffer 47, that is, the playback speed of the received packet (in this embodiment, the playback speed of the VoIP application) ) Is determined. As a result, in the case of control, the reproduction speed calculation unit 56 calculates the reproduction speed of the received packet based on the acquired required handover information and the monitoring result of the jitter buffer 47 by the jitter buffer monitoring unit 50. .

  The playback speed evaluation unit 57 stores an evaluation value for each playback speed of the application in advance. Here, in the case of a VoIP application, the sound quality when the playback speed is reduced from the standard playback speed (that is, the evaluation value of the sound quality when the speed changes), and the evaluation value of the sound quality of the sound alone at a constant playback speed For example, as shown in FIG. Therefore, based on the evaluation result of FIG. 4, the playback speed evaluation unit 57 stores in advance an evaluation value for each playback speed as shown in FIG. In FIG. 5, the horizontal axis represents the evaluation value when the reproduction speed changes from the standard reproduction speed to each speed, and the vertical axis represents the evaluation value of the sound alone at each reproduction speed. In the present embodiment, based on both evaluation values, reproduction is performed from an evaluation value 5 (including standard reproduction speed) close to the standard reproduction speed to an evaluation value 1 (including silence) close to the silent evaluation value 0. The speed is classified into 5 stages. The evaluation value for each reproduction speed may be set by default, or may be arbitrarily set by the user.

  When the calculated playback speed is equal to or higher than the minimum playback speed in the maximum evaluation value stored in the playback speed evaluation section 57, the playback speed calculation section 56 uses the calculated playback speed as the application playback speed to perform jitter buffer control. Supplied to the unit 51.

  On the other hand, when the playback speed calculated by the playback speed calculation unit 56 is less than the minimum playback speed in the maximum evaluation value stored in the playback speed evaluation unit 57, the playback speed evaluation unit 57 sets a plurality of evaluation values. A composite evaluation value composed of a combination of a plurality of corresponding playback speeds is calculated, and the calculation result is supplied to the playback speed calculation unit 56. The playback speed calculation unit 56 uses the composite evaluation value from the playback speed evaluation unit 57 in the period from the time when the start of handover preparation is determined to the time when the packet reception from the second wireless communication network 16 starts. A combination of playback speeds having a high evaluation value is selected, and the selected playback speed combination is supplied to the jitter buffer controller 51 as the playback speed of the application. Thereby, the jitter buffer control unit 51 controls reading of the received packet from the jitter buffer 47 so that the reproduction speed of the received packet becomes the reproduction speed from the reproduction speed calculation unit 56. Therefore, in the present embodiment, the telephone function unit 33 constitutes an execution unit that executes a real-time communication system application and a control unit that controls the playback speed of the application.

  Hereinafter, the operation of radio communication apparatus 11 according to the present embodiment will be described. First, the operation of the handover control unit 36 will be mainly described.

  The handover control unit 36 determines a handover schedule based on the communication qualities acquired from the first wireless I / F 31 and the second wireless I / F 32, respectively. For example, when a call is made by forming a wireless link with the first wireless communication network 15, the communication quality acquired from the first wireless I / F 31 becomes worse than the handover schedule determination threshold, and the second wireless I / F When the communication quality of F32 is equal to or higher than the handover schedule determination threshold, the handover control unit 36 determines the handover schedule to the second wireless communication network 16, that is, determines the start of handover preparation. The communication quality of the second wireless communication network 16 that is not used for a call is obtained by receiving broadcast information transmitted from the base station 16a, for example.

  When the handover control unit 36 determines a handover schedule, the handover preparation time T1 until the start of the handover, the handover source downlink absolute delay time Tddn1 in the currently used radio communication network (here, the first radio communication network 15) The handover destination absolute downlink delay time Tddn2 in the handover destination radio communication network (here, the second radio communication network 16) is acquired, and these information is used as necessary handover information together with information indicating that there is a handover schedule and a telephone function To the unit 33. Therefore, in the present embodiment, the handover control unit 36 determines whether to start preparation for handover, an estimation unit that estimates handover preparation time, and the first radio communication network 15 and the second radio communication. A measuring unit for measuring each delay time in the network 16 is configured.

  Next, a method for acquiring the handover preparation time T1, the handover source downlink absolute delay time Tddn1, and the handover destination downlink absolute delay time Tddn2 by the handover control unit 36 will be described.

(How to obtain handover preparation time T1)
For example, as shown in FIGS. 6A and 6B, the handover preparation time T1 is calculated based on a unit time change rate ΔRs (slope) of the radio state (Rs) for determining communication quality. Here, the rate of change ΔRs can be measured and acquired when the radio state falls below the handover schedule determination threshold and the handover schedule is determined, but in this embodiment, the handover schedule determination is performed during the call. The change rate average value ΔRsrms from the time point to a predetermined time before is acquired.

For this reason, the handover control unit 36 calculates the change rate ΔRs (t) of the unit time (Δt) of the radio state in the currently used radio communication network at a predetermined timing according to the following equation (1). A plurality of change rates ΔRs (t) up to a time before (for example, 2 seconds before) are held in the memory, and when a handover schedule is determined, the average change rate ΔRsrms up to a predetermined time held at that time calculate. Here, it is assumed that the wireless state is gradually getting worse.
[Equation 1]
ΔRs (t) = | {Rs (t) −Rs (t−Δt)} / Δt | (1)

  Thereafter, the handover control unit 36 determines whether or not the calculated change rate average value ΔRsrms is smaller than a preset change rate threshold value Rsref. As a result, in the case of ΔRsrms ≦ Rsref, that is, when the wireless state changes slowly, as shown in FIG. 6A, the handover preparation time T1 is set to a preset standard time Tref (for example, 5 sec). To do.

  On the other hand, when ΔRsrms> Rsref, that is, when the wireless state changes suddenly, for example, T1 = Tref (Rsref / ΔRsrms) is calculated, and as the rate of change ΔRsrms increases, the handover preparation time T1 Is set shorter than the standard time Tref. FIG. 6B shows a case where ΔRsrms> Rsref and the handover preparation time T1 is set to approximately half the standard time Tref (2.5 sec).

(Acquisition method of absolute delay time Tddn1, Tddn2)
The handover source downlink absolute delay time Tddn1 and the handover destination downlink absolute delay time Tddn2 are acquired by, for example, any one of first to fourth absolute delay time acquisition methods described below. In the present embodiment, since the network between the counterpart communication terminal (CN) 12 and the HA 23 is not switched, the absolute delay time between them is not considered.

(A) First Absolute Delay Time Acquisition Method After the handover schedule is determined in the handover control unit 36, the telephone function unit 33 and / or the communication processing unit 34 are controlled to synchronize the wireless communication device 11 with the HA 23. Is requested to transmit a measurement packet having a transmission time stamp, whereby the HA 23 causes the measurement packet to be transmitted to both the first wireless communication network 15 and the second wireless communication network 16. The wireless communication device 11 receives the measurement packet transmitted from the HA 23 via the corresponding first wireless I / F 31 and second wireless I / F 32, respectively, and based on the reception time and the time stamp of the measurement packet. Then, the downstream absolute delay times Tddn1 and Tddn2 of the corresponding network are measured. When the absolute downlink delay time of the handover source wireless communication network can be measured from the received packet during a call, transmission of the measurement packet to the wireless communication network can be omitted.

(B) Second Absolute Delay Time Acquisition Method Once the handover control unit 36 determines the handover schedule, the telephone function unit 33 and / or the communication processing unit 34 are controlled to synchronize the wireless communication apparatus 11 with the HA 23. As a result, the measurement packet is transmitted from the HA 23 to both the first wireless communication network 15 and the second wireless communication network 16 as in the first absolute delay time acquisition method. Then, the downlink absolute delay times Tddn1 and Tddn2 of the corresponding network are measured.

(C) Third Absolute Delay Time Acquisition Method When the handover control unit 36 determines a handover schedule, the telephone function unit 33 and / or the communication processing unit 34 are controlled so that the wireless communication device 11 and the wireless communication device 11 Sending measurement packets such as PING and RTCP from both the first wireless communication network 15 and the second wireless communication network 16 to the time-synchronized HA 23, receiving the reply, and sending the corresponding network Downlink absolute delay times Tddn1 and Tddn2 are measured.

(D) Fourth Absolute Delay Time Acquisition Method When the handover control unit 36 determines a handover schedule, the absolute delay time of each wireless communication network is acquired using a handover technique studied in IEEE 802.21. The illustration is given below.
A method for obtaining the absolute delay time (Tddn1) of the first wireless communication network 15 will be described below.
The wireless communication device 11 acquires the next value stored in the first information server of the first wireless communication network 15.
From the measurement server that operates to measure the delay time (for example, connected to the backbone network of the Internet 18) to the current access point 15a connected through the first wireless communication network 15 Standard value of one-way delay time (Tn3)
Standard value of the upper and lower delay times between the access point 15a and the terminal connected to the access point 15a (downward: Trdn3, upstream: Trup3)
Further, the wireless communication device 11 transmits a measurement packet such as PING to the HA 23, receives the reply, and measures the round-trip delay time Trt1 between the wireless communication device 11 and the HA 23.
Then, the absolute delay time Tddn1 of the first wireless communication network 15 is calculated from these values according to the following equation (2). However, since the one-way delay time between the access point 15a and the HA 23 cannot be acquired, this one-way delay time is approximated by adding Tn3 and {Trt1- (Tn3 + Trdn3 + Tn3 + Trup3)} / 2. .
[Equation 2]
Tddn1 = Tn3 + Trdn3 + {Trt1- (Tn3 + Trdn3 + Tn3 + Trup3)} / 2 (2)

Next, a method for obtaining the absolute delay time (Tddn2) of the second wireless communication network 16 that is the handover destination will be described below.
The wireless communication device 11 obtains the next value stored in the second information server connected to the second wireless communication network 16 that is the handover destination via the first information server of the first wireless communication network 15. To do. Note that the location information of the wireless communication device 11 acquired by the wireless communication device 11 or the access point 15a is transmitted to the second information server.
Standard value (Tn4) of one-way delay time between the base station 16a expected to be connected to the wireless communication device 11 and the measurement server
Standard value of the upper and lower delay times between the base station 16a and the terminal connected to it (downward: Trdn4, upstream: Trup4)
Then, from these values, the absolute delay time Tddn2 of the second wireless communication network 16 is calculated according to the following equation (3). However, since the one-way delay time between the base station 16a and the HA 23 cannot be acquired, this one-way delay time is approximated by adding Tn4 and {Trt1- (Tn3 + Trdn3 + Tn3 + Trup3)} / 2. .
[Equation 3]
Tddn2 = Tn4 + Trdn4 + {Trt1- (Tn3 + Trdn3 + Tn3 + Trup3)} / 2 (3)

  As described above, the handover control unit 36 acquires the handover preparation time T1, the handover source downlink absolute delay time Tddn1, and the handover destination downlink absolute delay time Tddn2, and supplies the acquired information to the telephone function unit 33. .

  Further, when the handover control unit 36 determines the handover schedule, the handover control unit 36 controls the communication processing unit 34 to connect the second wireless I / F 32 to the second wireless communication network 16. Thereafter, when the handover preparation time T1 has passed, the handover control unit 36 transmits a Registration Request (Binding Update in NEMO) to the HA 23 via the second wireless communication network 16 that is the handover destination, and the handover destination to the HA 23 Register the care of address.

  At that time, the handover control unit 36 sets 8 bits of the Registration Request Field of the Registration Request message in the communication processing unit 34 (in NEMO, multiple care of address is used), and the first radio communication network 15 also uses the second radio. The communication network 16 can also communicate.

  After that, when receiving the Registration Reply (Binding Acknowledge in NEMO) which is the handover completion information returned from the HA 23, the handover control unit 36 cancels the registration of the care-of address of the first wireless communication network 15 that is the handover source, and connects After that, the communication processing unit 34 is controlled to continue the VoIP application via the second wireless communication network 16 that is the handover destination, and the received handover completion information is supplied to the telephone function unit 33.

  Next, the operation of the telephone function unit 33 will be described with reference to FIGS. FIG. 7 is a sequence diagram showing the operation of the main part of the telephone function unit 33, and FIG. 8 is a flowchart showing the setting operation of the playback speed.

  The handover information acquisition unit 55 of the telephone function unit 33 monitors the handover information from the handover control unit 36 at regular intervals. If the handover function acquisition information is acquired, a required information from the handover control unit 36 is further obtained. The handover preparation time T1, the handover source downlink absolute delay time Tddn1, and the handover destination downlink absolute delay time Tddn2, which are handover information, are acquired, and the acquired required handover information is supplied to the playback speed calculation unit 56.

  Based on the required handover information acquired from the handover information acquisition unit 55, the playback speed calculation unit 56 and the downlink absolute delay time (Tddn2) of the second radio communication network 16 and the downlink absolute delay time ( A difference T2 (T2 = Tddn2−Tddn1) from Tddn1) is calculated, and it is determined whether or not a predetermined value (> 0) is exceeded.

Here, when the delay time difference T2 exceeds the predetermined value, the playback speed calculation unit 56 determines the playback speed V based on the acquired required handover information and the monitoring result of the jitter buffer 47 by the jitter buffer monitoring unit 50. Is calculated (step S11). In the present embodiment, the following is performed so that the number of packets in the jitter buffer 47 becomes 0 from the time when the information indicating that there is a handover schedule is acquired to the time when reception of packets from the second wireless communication network 16 that is the handover destination starts. From the equation (4), the reproduction speed V of the received packet of the jitter buffer 47 is calculated. In equation (4), Vn represents a standard reproduction speed, and Tb represents a time corresponding to the number of packets (data amount) of the jitter buffer 47 at the time when information indicating that a handover is scheduled is received. The reproduction speeds V and Vn are expressed as a time ratio (time / time), for example, Vn = 1.
[Equation 4]
V = (Tb + T1 × Vn) / (T1 + T2) (4)

  When the playback speed calculation unit 56 calculates the playback speed V by the above equation (4), the playback speed evaluation unit 57 calculates the minimum playback speed (in this embodiment, the evaluation value 5) from the minimum playback speed (in this embodiment, V5) is acquired, and the calculated reproduction speed V is compared with the reproduction speed V5 corresponding to the evaluation value 5 (step S12).

  As a result, when the playback speed V is equal to or higher than the playback speed V5, the calculated playback speed V is supplied to the jitter buffer control unit 51 so that the application is played back at the calculated playback speed V from the jitter buffer 47. The reading of the received packet is controlled (step S13).

On the other hand, when the playback speed V is less than the playback speed V5, the playback speed calculation unit 56 further receives from the playback speed evaluation unit 57 each minimum playback speed Vm (Vm = V4, V3, V2, V2). V1). Then, with reference to the minimum playback speed V5 of the evaluation value 5 that is the maximum evaluation value, the playback time tm at each playback speed Vm when playback is performed using each minimum playback speed Vm in the other evaluation values acquired, While calculating from the following equation (5), the playback time t5 at the reference playback speed V5 is calculated from the calculation result (step S14). If the playback time tm exceeds (T1 + T2), the playback speed combination is not used.
[Equation 5]
tm = {(T1 + T2) × V5-T1 × Vn-Tb} / (V5-Vm) (5)

  For example, when T1 = 380 msec, T2 = 380 msec, Ta = 160 msec, Tb = 160 msec, Vn = 1 (100%), V5 = 0.75 (75%), the playback speed V1 (0 %) Playback time t1 is t1 = {(380 + 380) × 0.75-380 × 1-160} / (0.75-0) = 40 (msec), and playback time t5 at playback speed V5 is t5 = 380 + 380-40 = 720 (msec). Further, in combination with the playback speed V2, the playback time t2 of the playback speed V2 (50%) is t2 = {(380 + 380) × 0.75-380 × 1-160} / (0.75-0.5) = 120 (msec ), And the playback time t5 at the playback speed V5 is t5 = 380 + 380-120 = 640 (msec). In combination with the playback speed V3, the playback time t3 of the playback speed V3 (60%) is t3 = {(380 + 380) × 0.75-380 × 1-160} / (0.75-0.6) = 200 (msec ), And the playback time t5 at the playback speed V5 is t5 = 380 + 380-200 = 560 (msec). Furthermore, in combination with the playback speed V4, the playback time t4 of the playback speed V4 (71.4%) is t4 = {(380 + 380) × 0.75-380 × 1-160} / (0.75-0.714) = 833.3 (msec ). In this case, since the reproduction time t4 exceeds (T1 + T2), this combination is not used.

  9 to 12 show the combination results of the respective reproduction speeds. FIG. 9 shows the combination results of the reproduction speed V5 (evaluation value 5) and the reproduction speed V1 (evaluation value 1), and FIG. 10 shows the reproduction speed. The combination result of V5 and reproduction speed V2 (evaluation value 2) is shown, and FIG. 11 shows the combination result of reproduction speed V5 and reproduction speed V3 (evaluation value 3). FIG. 12 shows a case where the playback time t4 at the playback speed V4 (evaluation value 4) exceeds T1 + T2. In the case of FIG. 12, since the playback time t4 exceeds T1 + T2, even if playback is continued at the playback speed V4 from the beginning, there are no packets at the end. As a result, the playback speed V4 and playback speed V1 (silence) ). 9 to 12, (a) shows the number of packets received by the jitter buffer 47 per unit time, and (b) shows the combined reproduction speed (reading interval) and its time (period). (C) shows the number of packets in the jitter buffer 47, and (d) shows an evaluation value.

  Upon completion of the above calculation, the playback speed calculation unit 56 supplies the playback speed evaluation unit 57 with the calculated combinations of effective playback speeds and playback times based on the playback speeds. Thus, the playback speed evaluation unit 57 calculates a composite evaluation value based on the playback speed combination information from the playback speed calculation unit 56 (step S15 in FIG. 8).

  For example, from the playback speed calculation unit 56, the combination information of the playback speed V5 and the playback speed V1, the combination information of the playback speed V5 and the playback speed V2, and the combination information of the playback speed V5 and the playback speed V3 described above, Is input, the reproduction speed evaluation unit 57 performs the following calculation based on the combination information to calculate each composite evaluation value.

  That is, for the combination of the playback speed V5 and the playback speed V1, 5 × 720 + 1 × using the evaluation value 5 of the playback speed V5, the evaluation value 1 of the playback speed V1, and the playback time of each playback speed. The composite evaluation value of 40 = 3640 is calculated. For the combination of playback speed V5 and playback speed V2, the composite evaluation value of 5 × 640 + 2 × 120 = 3440 is calculated, and playback speed V5 and playback speed V3 are calculated. For the combination of, a composite evaluation value of 5 × 560 + 3 × 200 = 3400 is calculated.

  After completing the above calculation, the reproduction speed evaluation unit 57 supplies the calculated composite evaluation values to the reproduction speed calculation unit 56. The playback speed calculation unit 56 selects a combination of playback speeds having the highest composite evaluation value from the composite evaluation values input from the playback speed evaluation unit 57, and each playback speed and playback in the selected combination of playback speeds. The time is supplied to the jitter buffer controller 51 to control the reading of the received packet from the jitter buffer 47 so that the application is reproduced at each selected reproduction speed and reproduction time (step S16 in FIG. 8).

  For example, in the above case, a combination of the playback speed V5 and the playback speed V1 is selected, and the playback speed V5 (75%) is initially played for 720 msec, and then for 40 msec, the playback speed V1 (0% = The jitter buffer 47 is controlled so that there is no sound.

  Here, the playback speed control of the received packet by the jitter buffer control unit 51 is executed by, for example, one of the first playback speed control method and the second playback speed control method described below.

(A) First playback speed control method Calculate the packet reading interval from the jitter buffer 47 with respect to the standard playback speed Vn as TR1, and calculate (Tb + T1 × Vn) / (T1 + T2) as k in the above equation (4). When the reading interval of packets from the jitter buffer 47 corresponding to the playback speed V is TR, TR = TR1 / k. For example, at the standard playback speed Vn, in the case of a VoIP application that reads and plays back packets in the jitter buffer 47 at intervals of 20 msec, when the playback speed V is 75% (k = 0.75) of the standard playback speed Vn. The packet reading interval TR from the jitter buffer 47 is TR = 20 / 0.75 (msec).

(B) Second Playback Speed Control Method When playback speed control for handover is started, the time stamp of the packet (first packet) played back immediately after that is recorded in combination with the playback time. Subsequent packets are read from the jitter buffer 47 and reproduced at time Tv shown by the following equation (6). In Equation (6), TD is a delay time, and the initial value is 0.
[Equation 5]
Tv = (packet time stamp−first packet time stamp) + (first packet playback time + TD) (6)

  Here, when the packet is read from the jitter buffer 47, the [{Vn / (Vn-V)}-1] -th read packet is copied and stored in the memory in the decoder 48, and the copy source packet is stored. After reproduction, the copied packet is read and reproduced at the next reproduction timing. For example, when the reproduction speed V is set to 75% of the standard reproduction speed Vn, as shown in FIG. 13, the sequential three packets P1 to P3 in the jitter buffer 47 are sequentially read and reproduced, The third packet P3 is copied, and the copied packet P3 ′ is reproduced at the next reproduction timing after reproducing the copy source packet P3. Thereafter, when reading the packet P4 from the jitter buffer 47, the TD of the above equation (6) is increased by the reproduction interval time by copying. If the [{Vn / (Vn−v)} − 1] -th packet to be read has not arrived or is discarded and is not in the jitter buffer 47, the packet at the next reproduction timing is The same processing is performed.

  As described above, the reproduction speed control of the received packet by the jitter buffer control unit 51 is executed. Thereafter, when the handover information acquisition unit 55 acquires the handover completion information from the handover control unit 36, the playback speed calculation unit 56 acquires the packet reception interval time from the jitter buffer monitoring unit 50 at regular intervals, and acquires the received reception. An average value of the interval time in a predetermined time is calculated, and it is monitored whether or not a difference between the calculated packet reception interval average value and a standard reception interval in the VoIP application is within a threshold value.

  As a result, if it falls within the threshold value, the playback speed calculation unit 56 determines that the packet from the handover destination has been received, and determines the number of packets (data amount) in the jitter buffer 47 at that time as the jitter buffer. It is acquired from the monitoring unit 50, and it is determined whether or not the acquired number of packets exceeds a predetermined amount.

  As a result, when the number of packets in the jitter buffer 47 does not exceed the predetermined amount, the normal reproduction speed after t = (Ta−Tb) / (Vn−V5), where the number of packets exceeds the predetermined amount. The jitter buffer control unit 51 is instructed to return to the control. Here, Ta is a jitter buffer standard delay time corresponding to the standard number of packets in the jitter buffer 47. On the other hand, when the number of packets in the jitter buffer 47 exceeds a predetermined amount, the jitter buffer controller 51 is instructed to immediately return to the normal reproduction speed control. That is, when the playback speed calculation unit 56 determines that a packet from the handover destination has been received, the jitter buffer control unit 51 returns the standard playback speed Vn to the standard playback speed Vn when the number of packets in the jitter buffer 47 exceeds a predetermined amount. The reading of the jitter buffer 47 is controlled.

  As described above, in the present embodiment, when a handover is performed from the first wireless communication network 15 having a short absolute delay time Tddn1 to the second wireless communication network 16 having a long absolute delay time Tddn2, preparation for the handover is performed in advance. Obtain the time T1, the absolute delay times Tddn1 and Tddn2, and the number of packets (data amount) of the jitter buffer 47 at the time of determining the handover schedule, and calculate the playback speed of the application based on the obtained information, The playback speed of the application is controlled based on the calculated playback speed and an evaluation value for each playback speed set in advance. If the calculated playback speed does not reach the minimum playback speed at the maximum evaluation value, a composite evaluation consisting of a combination of a plurality of playback speeds in the period from the handover schedule determination to the start of packet reception from the handover destination. Select the playback speed combination with the highest value and play the app.

  Therefore, according to the present embodiment, the delay time difference between the first wireless communication network 15 that is the handover source and the second wireless communication network 16 that is the handover destination can be absorbed over a long time, and playback is performed at a highly evaluated playback speed. Therefore, handover can be performed without degrading the reproduction quality and real-time property. For example, in the case of the combination of reproduction speeds as shown in FIGS. 9 to 12 described above, the combination of reproduction speeds in FIG. 9 having the highest composite evaluation value is selected. In this case, even if silence occurs for a short time at the playback speed V1, the playback speed V5 up to that time is a playback speed of an evaluation value 5 close to the standard playback speed selected in consideration of the listener's evaluation. Therefore, as compared with the case where the playback time with a low evaluation value is lengthened as shown in FIGS. 10 and 11 or the evaluation value is lowered to “4” as shown in FIG. Can be highly evaluated.

  In addition, this invention is not limited only to the said embodiment, Many deformation | transformation or a change is possible. For example, the present invention is not limited to the case of executing a VoIP application, but can be effectively applied to the case of executing a real-time communication application such as streaming reproduction of multimedia data such as video and music. In this case, the application execution unit may be configured by a multimedia function unit having a similar jitter buffer control function instead of the telephone function unit.

It is a figure which shows schematic structure of the communication network which can use the radio | wireless communication apparatus which concerns on one embodiment of this invention. It is a block diagram which shows schematic structure of the radio | wireless communication apparatus shown in FIG. It is a functional block diagram which shows schematic structure of the telephone function part of the radio | wireless communication apparatus shown in FIG. It is a figure which shows the evaluation value of the sound quality by a speed change, and the evaluation value of the sound quality in a fixed reproduction speed. It is a figure which shows the evaluation value for every reproduction | regeneration speed memorize | stored in the reproduction | regeneration speed evaluation part shown in FIG. FIG. 3 is a diagram for explaining a method for calculating a handover preparation time by the handover control unit shown in FIG. 2. It is a sequence diagram which shows operation | movement of the principal part of the telephone function part shown in FIG. It is a flowchart which shows the reproduction speed setting operation | movement by the telephone function part shown in FIG. It is a figure which shows the combination result of the evaluation value 5 and the evaluation value 1 explaining the reproduction speed setting operation | movement by the telephone function part shown in FIG. It is a figure which shows the combination result of the evaluation value 5 and the evaluation value 2 explaining the reproduction speed setting operation | movement by the telephone function part shown in FIG. It is a figure which shows the combination result of the evaluation value 5 and the evaluation value 3 explaining the reproduction speed setting operation | movement by the telephone function part shown in FIG. It is a figure which shows the combination result of the evaluation value 4 and the evaluation value 1 explaining the reproduction speed setting operation | movement by the telephone function part shown in FIG. FIG. 4 is a diagram for explaining an example of a reception packet reproduction rate control method by a jitter buffer control unit shown in FIG. 3. It is a figure for demonstrating an example of the control method of the conventional jitter buffer. It is a figure which shows the flow of the packet by the control method shown in FIG. It is a figure for demonstrating the other example of the control method of the conventional jitter buffer. It is a figure which shows the flow of the packet by the control method shown in FIG.

Explanation of symbols

DESCRIPTION OF SYMBOLS 11 Wireless communication apparatus 12 Partner communication terminal 12a Handset 15 1st wireless communication network 15a Access point 16 2nd wireless communication network 16a Base station 17 Packet network 18 Internet 21,22 SIP server 23 Home agent (HA)
31 First wireless I / F
32 Second wireless I / F
33 Telephone function unit 34 Communication processing unit 35 Wireless information acquisition unit 36 Handover control unit 47 Jitter buffer 50 Jitter buffer monitoring unit 51 Jitter buffer control unit 55 Handover information acquisition unit 56 Playback speed calculation unit 57 Playback speed evaluation unit

Claims (5)

A wireless communication unit that performs wireless communication by connecting to a first wireless communication network and a second wireless communication network different from the first wireless communication network;
A jitter buffer and a jitter buffer monitoring unit that monitors a data amount of the jitter buffer, and an execution unit that executes an application of a real-time communication system via the wireless communication unit;
A communication quality acquisition unit for acquiring communication quality of a radio link in the first wireless communication network while executing the application by connecting to the first wireless communication network;
A determination unit that determines whether to start preparation for handover from the first wireless communication network to the second wireless communication network based on the communication quality acquired by the communication quality acquisition unit;
When the determination unit determines the start of handover preparation during the execution of the application, an estimation unit that estimates a handover preparation time until the handover is started based on the communication quality acquired by the communication quality acquisition unit;
When the determination unit determines the start of handover preparation, a measurement unit that measures respective delay times in the first wireless communication network and the second wireless communication network;
The handover preparation time estimated by the estimation unit, the respective delay times measured in the first wireless communication network and the second wireless communication network measured by the measurement unit, and the time when the determination unit determines the start of handover preparation Based on the amount of data in the jitter buffer by the jitter buffer monitoring unit, a playback speed calculation unit that calculates the playback speed of the application, and a playback speed evaluation unit that stores an evaluation value for each playback speed of the application, A control unit for controlling the playback speed of the application by the execution unit based on the playback speed calculated by the playback speed calculation unit and the evaluation value stored in the playback speed evaluation unit;
A wireless communication apparatus comprising:   The control unit compares the delay time in the first wireless communication network with the delay time in the second wireless communication network, and the delay time in the second wireless communication network is the delay in the first wireless communication network. The wireless communication apparatus according to claim 1, wherein when the execution unit is longer than the time by a predetermined time, the playback speed of the application by the execution unit is slowed down.   The wireless communication apparatus according to claim 2, wherein the control unit slows down a reproduction speed of the application by the execution unit after starting preparation for handover. When the playback speed calculated by the playback speed calculation unit is equal to or higher than the minimum playback speed in the maximum evaluation value stored in the playback speed evaluation unit, the control unit plays the application at the calculated playback speed. To control and
When the playback speed calculated by the playback speed calculation unit is less than the minimum playback speed in the maximum evaluation value, a composite evaluation value consisting of a combination of a plurality of playback speeds corresponding to a plurality of evaluation values is calculated by the playback speed evaluation unit. And the composite evaluation value is calculated from the calculated composite evaluation value in the period from the time when the reproduction speed calculation unit determines the start of the handover preparation to the time when data reception from the second wireless communication network starts. The wireless communication apparatus according to claim 1, 2 or 3, wherein a combination of playback speeds with a higher playback speed is selected and the application is played back with the selected playback speed combination.   When the data amount of the jitter buffer monitored by the jitter buffer monitoring unit exceeds a predetermined amount after the handover to the second wireless communication network is completed, the control unit sets the playback speed of the application to a normal speed. The wireless communication apparatus according to claim 2, 3 or 4, wherein the wireless communication apparatus is returned.

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