WO2014083961A1 - Packet transfer control device and communications system - Google Patents

Packet transfer control device and communications system Download PDF

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Publication number
WO2014083961A1
WO2014083961A1 PCT/JP2013/078399 JP2013078399W WO2014083961A1 WO 2014083961 A1 WO2014083961 A1 WO 2014083961A1 JP 2013078399 W JP2013078399 W JP 2013078399W WO 2014083961 A1 WO2014083961 A1 WO 2014083961A1
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Prior art keywords
transfer control
packet
packet transfer
control device
congestion
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PCT/JP2013/078399
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French (fr)
Japanese (ja)
Inventor
一範 小澤
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日本電気株式会社
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Publication of WO2014083961A1 publication Critical patent/WO2014083961A1/en

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    • HELECTRICITY
    • H04ELECTRIC COMMUNICATION TECHNIQUE
    • H04WWIRELESS COMMUNICATION NETWORKS
    • H04W28/00Network traffic management; Network resource management
    • H04W28/02Traffic management, e.g. flow control or congestion control
    • H04W28/0205Traffic management, e.g. flow control or congestion control at the air interface
    • HELECTRICITY
    • H04ELECTRIC COMMUNICATION TECHNIQUE
    • H04LTRANSMISSION OF DIGITAL INFORMATION, e.g. TELEGRAPHIC COMMUNICATION
    • H04L47/00Traffic control in data switching networks
    • H04L47/10Flow control; Congestion control
    • H04L47/26Flow control; Congestion control using explicit feedback to the source, e.g. choke packets

Definitions

  • the present invention relates to a packet transfer control device that performs packet transfer while controlling communication quality for each packet, and a communication system including the same.
  • LTE Long Term Evolution
  • EPC Evolved Packet Core
  • circuit switching for voice calls and videophone calls and packet switching for sending data are configured as separate systems.
  • voice call data, videophone data, content distribution data, and so-called data signals flow together on the same packet communication path.
  • data signals application data, document data, photo data, etc.
  • Patent Document 1 discloses a packet transfer control device that performs packet transfer rate control.
  • the packet transfer control device disclosed in Patent Document 1 includes a line congestion state determination unit, a transfer rate control determination unit, and a packet processing unit.
  • the line congestion state determination unit determines whether the backbone line is congested based on the accumulated packet total amount that is the accumulated value of the packet size for a plurality of packets.
  • the transfer rate control determination unit selects one or more IP (Internet Protocol) flows having a hop count value lower than the threshold value.
  • the packet processing unit determines whether the IP flow selected by the transfer rate control determination unit is a TCP (Transmission Control Protocol) packet.
  • TCP Transmission Control Protocol
  • the packet processing unit performs the following three types of packet processing. Apply. Specifically, 1) In the case of an outgoing packet from the server, the CE (Consultation Experience) bit of ECN (Explicit Connection Notification) is set in the TCP header. 2) In the case of a reply packet returned from the client, the advertisement window size of the TCP header is reduced and changed. 3) In the case of an acknowledgment (Ack) packet, the transmission timing of the packet to the backbone line is delayed. If it is not a TCP packet, the packet is discarded.
  • JP 2004-320452 A ([0051] to [0057])
  • QCI Quality Class Id
  • S-GW Packet data network Gateway
  • QoS Quality of Service
  • Parameters such as (Maximum Bit Rate) and GBR (Guaranteed Bit Rate) are set, and QoS is controlled for each packet.
  • QCI Quality Class Identifier
  • MBR Maximum Bit Rate
  • GBR Guard Bit Rate
  • Patent Document 1 discloses a technical idea of selecting one or more IP flows having a hop count value lower than a threshold when the backbone line is congested. Not too much. That is, Patent Document 1 does not recognize the above-described problem relating to the deterioration of QoE.
  • An object of the present invention is to provide a packet transfer control device capable of avoiding QoE degradation.
  • One aspect of the present invention is a packet transfer control device for connecting a terminal via a network and transferring a packet storing media data in response to a request from the terminal, the congestion detecting unit detecting congestion of the network, A transfer control unit for notifying at least one of the partner terminal and the own terminal of a request to reduce the rate of the media data when the congestion detection unit detects congestion.
  • FIG. 1 is a block diagram showing a connection configuration of a communication system according to the first embodiment of the present invention.
  • FIG. 2 is a block diagram showing a configuration of a packet transfer control device used in the communication system shown in FIG.
  • FIG. 3 is a block diagram showing a configuration of a rate control unit used in the packet transfer control device shown in FIG.
  • FIG. 4 is a block diagram showing a connection configuration of a communication system according to the second embodiment of the present invention.
  • FIG. 5 is a block diagram showing a configuration of a packet transfer control device used in the communication system shown in FIG.
  • FIG. 6 is a block diagram showing a configuration of a mobile terminal used in the communication system shown in FIG.
  • FIG. 1 is a block diagram showing a configuration of a communication system according to the first embodiment of the present invention.
  • a packet transfer control device 190 shows a configuration using P-GW (Packet data network Gateway) or S-GW (Serving Gateway) or both.
  • the mobile terminal is assumed to be a so-called Galapagos mobile phone, a smartphone, or a tablet.
  • the communication system of FIG. 1 shows an example in which user A communicates with a partner user (not shown).
  • a user A uses a portable terminal 170 ⁇ / b> _ ⁇ b> 1 via a mobile network 150 and an IMS (IP Multimedia Subsystem) network 130 via a counterpart network (not shown). (Not shown) and VoIP (Voice Over IP) voice communication.
  • IMS IP Multimedia Subsystem
  • VoIP Voice Over IP
  • a TV phone may be used in place of the voice communication (voice call), but in FIG.
  • FIG. 1 shows a configuration for detecting congestion by receiving congestion information by ECN (Explicit Congestion Notification) for an upstream packet sent from a terminal.
  • ECN Exlicit Congestion Notification
  • the outdoor LTE radio base station apparatus (eNodeB apparatus) 194 detects a congestion state in the wireless network
  • the ECN field of the IP (Internet protocol) header portion of the uplink packet from the eNodeB apparatus to the packet transfer control apparatus 190 It is assumed that the packet transfer control device is notified of the congestion state by setting a predetermined value for the packet transfer control device.
  • connection request signal is ,
  • SIP Session Initiation Protocol
  • IMS IP Multimedia Subsystem
  • PCRF Policy and Charging Rules Function
  • the mobile terminal 170_1 adds at least one parameter of parameters such as voice call traffic, desired QoS class, MBR (Maximum Bit Rate), GBR (Guaranteed Bit Rate) to the connection request signal, and the parameter is added.
  • the connection request signal thus made can be notified to at least one of the SIP server 110 and the PCRF device 191 via the packet transfer control device 190.
  • the SIP server 110 receives a connection request signal for a voice call and sends a connection request signal to a partner terminal (not shown) via a partner network (not shown).
  • the SIP server 110 When the SIP server 110 receives the Ack signal from the counterpart terminal, the SIP server 110 sends the Ack signal to the portable terminal 170_1 via the packet transfer control device 190 and the eNodeB device 194. When this is received by the portable terminal 170_1, control signals for voice call are exchanged.
  • control signals for voice call are exchanged.
  • MBR Maximum Bit Rate
  • GBR Guard Bit Rate
  • the PCRF device 191 inputs the voice call traffic, the IP address of the portable terminal 170_1, and the port number from the packet transfer control device 190 for at least one of the upstream and downstream directions. If necessary, the PCRF device 191 also inputs parameters such as a desired QoS class, MBR (Maximum Bit Rate), GBR (Guaranteed Bit Rate), etc. from the packet transfer control device 190 as QoS information. Next, the PCRF device 191 generates a QoS parameter for QoS control.
  • MBR Maximum Bit Rate
  • GBR Guard Bit Rate
  • the QoS parameter for QoS control is at least one of QCI (Quality Class Identifier) which is a value for identifying a QoS class, ARP (Allocation and Retention Priority) indicating the priority of resource reservation and retention, MBR, and GBR. is there.
  • QCI Quality Class Identifier
  • ARP Allocation and Retention Priority
  • MBR Resource reservation and retention
  • GBR GBR
  • the MBR and the GBR are used as they are when received from the packet transfer control device 190, and are generated by the PCRF device 191 when there is no reception.
  • the PCRF device 191 generates at least one of these four types of QoS parameters for each of the uplink direction and the downlink direction, and sends the generated parameter to the packet transfer control device 190.
  • QCI 1 (Conversational Voice)
  • ARP 2
  • GBR 12
  • an AMR-NB Adaptive Multi-Rate Narrowband
  • an AMR-WB Adaptive Multi-Rate Wideband
  • the value of GBR can be changed.
  • the packet transfer control device 190 includes a packet transfer unit 176, a transfer control unit 188, a congestion detection unit 200, a control unit 211, and a rate control unit 230.
  • the control unit 211 relays a control signal from the mobile terminal 170_1 to the SIP server 110, and relays a control signal and an Ack signal from the SIP server 110 to the mobile terminal 170_1.
  • the control unit 211 inputs at least one of four types of QoS parameters of QCI, ARP, MBR, and GBR for each traffic data from the PCRF device 191.
  • the control unit 211 controls the uplink and downlink directions of the voice call traffic. At least one of the four types of QoS parameters and at least one of the four types of QoS parameters for the downlink direction of the download data traffic are input from the PCRF device 191. The control unit 211 sends these QoS parameters to the rate control unit 230.
  • the congestion detection unit 200 checks the ECN (Explicit Congestion Notification) field of the IP header for the upstream packet sent from the eNodeB device 194 via the packet transfer unit 176.
  • the congestion detection unit 200 recognizes that the downlink direction from the eNodeB device 194 to the portable terminal is in a congestion state, and controls downlink congestion detection information indicating the recognition result. Output to the unit 211 and the rate control unit 230.
  • the control unit 211 sends the congestion detection information to the PCRF device 191 in FIG.
  • the PCRF device 191 in FIG. 1 checks the QoS parameter for the portable terminal 170_1 currently connected to the eNodeB device 194.
  • the parameter and the change flag are sent to the control unit 211 of the packet transfer control device 190.
  • the rate control unit 230 inputs congestion detection information from the congestion detection unit 200, and inputs a QoS parameter and a change flag from the control unit 211.
  • the rate control unit 230 includes a congestion identification unit 231, a flag identification unit 232, a QoS parameter identification unit 233, and a rate change setting unit 234.
  • the congestion identification unit 231 receives the congestion detection information, identifies that congestion has occurred in the downstream direction, and sends the identification result to the rate change setting unit 234.
  • the flag identifying unit 232 inputs a change flag related to the QoS parameter for the mobile terminal 170_1, recognizes that there is a change in the QoS parameter for the mobile terminal 170_1, and instructs the rate change setting unit 234 to change the rate setting for the mobile terminal 170_1.
  • the QoS parameter identification unit 233 inputs the QoS parameter for the mobile terminal 170_1, and recognizes that the GBR and MBR are changed (reduced) among the QoS parameters. Then, the QoS parameter identification unit 233 sends these QoS parameters to the rate change setting unit 234.
  • the rate change setting unit 234 creates an instruction to reduce the rate based on the voice communication based on the input values from the congestion control unit 231, the flag identification unit 232, and the QoS parameter identification unit 233. Specifically, for the uplink direction, the rate change setting unit 234 gives an instruction to change the rate of the audio codec transmitted from the mobile terminal 170_1 to a rate equal to or less than the rate of the uplink GBR by using a downlink signal to the mobile terminal 170_1. It sends it to the transfer control unit 188 (FIG. 2) for notification.
  • the rate after reduction is set to 6.7 kbps.
  • the rate change setting unit 234 sends an instruction to change to a rate equal to or lower than the downlink GBR to the uplink to the counterpart terminal.
  • the data is sent to the transfer control unit 188 (FIG. 2) for notification by a signal.
  • the rate after reduction is set to 6.7 kbps.
  • the rate change setting unit 234 outputs these uplink and downlink rate reduction requests to the transfer control unit 188 in FIG.
  • the transfer control unit 188 in FIG. 2 sends an instruction signal for controlling transfer to the packet transfer unit 176 in accordance with the changed GBR and MBR for the downstream packet to the mobile terminal 170_1.
  • the transfer control unit 188 instructs the mobile terminal 170_1 to use a downstream packet so as to reduce the audio codec transmission rate to 6.7 kbps.
  • a downstream packet so as to reduce the audio codec transmission rate to 6.7 kbps.
  • SIP / SDP Session Description Protocol
  • 6.7 kbps can be set in the CMR (Codec Mode Request) field in the RTP payload format header of the RTP packet.
  • the transfer control unit 188 sends to the packet transfer unit 176 an instruction signal for controlling transfer in accordance with the changed GBR and MBR for the uplink packet.
  • the transfer control unit 188 gives an instruction to the partner terminal using the uplink packet so as to reduce the transmission rate of the voice codec to 6.7 kbps.
  • 6.7 kbps can be set in the CMR field in the RTP payload format header of the RTP packet.
  • the packet transfer unit 176 instructs the mobile terminal 170_1 to use the downstream packet so as to reduce the audio codec transmission rate to 6.7 kbps.
  • 6.7 kbps can be set in the CMR field in the RTP payload format header of the RTP packet.
  • the packet transfer unit 176 gives an instruction to the partner terminal using the uplink packet so as to reduce the transmission rate of the voice codec to 6.7 kbps.
  • 6.7 kbps can be set in the CMR field in the RTP payload format header of the RTP packet.
  • the function of the PCRF device 191 can be built in the packet transfer device 190.
  • the mobile network 150 may be a 3G network
  • the packet transfer control device 190 may be an SGSN (Serving GPRS Support Node) or a GGSN (Gateway GPRS Support Node).
  • the packet transfer control device 190 can be realized by a program executed by a computer. That is, the packet transfer control device 190 may be composed of a packet transfer control processor (not shown) and a storage device (not shown).
  • the storage device stores a packet transfer control program. In this case, the packet transfer control processor performs the above-described packet transfer control operation according to the packet transfer control program stored in the storage device.
  • the conventional QoS parameter control method avoids this. It becomes possible to avoid the congested state that was difficult. As a result, there is an effect that it is possible to avoid QoE (Quality of Experience) degradation such as sound being interrupted or screen being frozen at a terminal for a preferential user. Furthermore, even if a service such as a high-quality VoIP or a high-resolution TV phone is started in the future using the packet communication path of the LTE / EPC system, it is possible to avoid deterioration of QoE on the terminal side. .
  • QoE Quality of Experience
  • FIG. 5 is a block diagram showing a configuration of the packet transfer control device 190A.
  • the constituent elements having the same numbers as those in FIG. 2 perform the same operations as those in FIG.
  • the packet transfer control device 190A is the same as the packet transfer control device 190 shown in FIG.
  • the bandwidth measuring unit 205 calculates at least one bandwidth of the network to which the mobile terminal 170_1 is connected at the time of session connection or at predetermined time intervals.
  • the bandwidth measuring unit 205 uses the transmission packet from the packet transfer unit 176 and the return packet from the mobile terminal 170 to determine the upstream bandwidth and the downstream bandwidth for the network connected to the mobile terminal 170. At least one is measured. In the present embodiment, a configuration in the case of measuring both upstream and downstream bands will be described.
  • the bandwidth measuring unit 205 instructs the packet transfer unit 176 to send a plurality of specific packets (probe packets) at a predetermined timing.
  • the packet transfer unit 176 sends a plurality of specific packets to the portable terminal 170_1 at a timing when the instruction is received, and continuously transmits in a predetermined order in time.
  • the plurality of packets indicates two or more packets.
  • the sending order is a predetermined order. For example, packets are sent in order from a packet having a small data size to a packet having a large data size. Further, the time interval between the packet and the next packet is a predetermined time interval.
  • the packet transfer unit 176 receives a reply packet from the portable terminal 170_1 as a result of sending the plurality of packets.
  • the reply packet includes, for example, the packet number received by the portable terminal 170_1 below the threshold with respect to the differential delay, the data size of the packet, the time when the packet was transmitted from the server, and the packet. Include information such as the time received by the mobile terminal.
  • the packet transfer unit 176 obtains, from the return signal received from the mobile terminal 170_1, the packet number that the mobile terminal 170_1 has received below the delay difference threshold, the transmission time from the server, and the reception time information at the mobile terminal.
  • the bandwidth measuring unit 205 receives the above information from the packet transfer unit 176 and calculates the network bandwidth.
  • the bandwidth measuring unit 205 receives the packet number, the packet data size, the transmission time of the packet from the server, and the reception time of the packet at the portable terminal at the portable terminal 170_1 below the delay difference threshold.
  • N and D (N) indicate the number and data size of a packet that can be received by the first portable terminal 170_1 below the delay difference threshold value, respectively.
  • R (N) is the reception time at the portable terminal 170_1 of the Nth packet sent from the server
  • R (N-1) is the first portable of the (N-1) th packet sent from the server.
  • the reception times at terminal 170_1 are respectively shown.
  • the bandwidth measuring unit 205 smoothes the downstream bandwidth measurement value W_d temporally according to the following equation (2) to calculate B (n) _d.
  • B (n) _d (l ⁇ ) * B (n ⁇ 1) _d + ⁇ W_d (2)
  • B (n) _d indicates a downstream band measurement value after smoothing at the nth time
  • is a constant in a range of 0 ⁇ ⁇ 1. Note that the time direction smoothing according to the equation (2) may not be performed if unnecessary.
  • the mobile terminal 170_1 sends a plurality of specific packets to the packet transfer control device 190A in a predetermined order at any timing after the connection request or when the reply signal is transmitted.
  • the upstream band is measured in the packet transfer control device 190A.
  • the plurality of packets indicates two or more packets.
  • the sending order is a predetermined order.
  • packets are sent in order from a packet having a small data size to a packet having a large data size.
  • the time interval between the packet and the next packet is a predetermined time interval.
  • RTP / UDP / IP is used as the protocol.
  • the bandwidth measuring unit 205 receives a plurality of packets sent from the portable terminal 170_1 from the packet transfer unit 176, and receives the packet number that is received below the threshold of the delay difference, the data size of the packet, the mobile phone of the packet Information on the transmission time from the terminal and the reception time of the packet at the packet transfer unit 176 is input, and the upstream bandwidth W_u is calculated according to the following equation (3).
  • the bandwidth measuring unit 205 smoothes the bandwidth measurement value W_u in the upstream direction temporally according to the following equation (4) to calculate B (n) _u.
  • B (n) _u (l ⁇ ) * B (n ⁇ 1) _u + ⁇ W_u (4)
  • the packet transfer unit 176 sends information necessary for network bandwidth measurement to the portable terminal 170_1 only at the time of session connection or at predetermined time intervals starting from the time of session connection and the time of session connection. A response signal is received from the mobile terminal.
  • the bandwidth measuring unit 205 calculates B (n) _d and B (n) _u, for example, at predetermined time intervals, and sends them to the congestion detection unit 210.
  • the congestion detection unit 210 performs the following determination to detect congestion. 1) Downlink bandwidth calculation B (n) _d ⁇ If GBR + ⁇ in the downlink direction of a voice call by the mobile terminal 170_1, the congestion detection unit 210 does not change any of the downlink QoS parameters for the mobile terminal 170_1. This is transmitted to the PCRF device 191 in FIG. 4 via the control unit 211.
  • is a margin, and a predetermined value is used.
  • the congestion detection unit 210 detects congestion and notifies the PCRF device 191 of the congestion information.
  • the congestion detection unit 210 determines the same as in 1) and 2) above, and notifies the PCRP device 191 whether or not congestion has been detected.
  • the PCRP apparatus 191 checks at least one of the user profile information and the QoS parameter for the portable terminal 170_1.
  • FIG. 6 is a block diagram illustrating a configuration of the mobile terminal 170_1.
  • the portable terminal 170_1 includes a packet transmission / reception unit 250, an audio codec 253, a rate setting unit 254, and a delay difference determination unit 255.
  • the packet transmitting / receiving unit 250 generates a reply packet for the received probe packet and sends the generated reply packet to the network.
  • the reply packet is generated as follows, for example.
  • the packet transmission / reception unit 250 receives each of the plurality of probe packets transmitted from the packet transfer unit 176 in FIG. 5 and outputs the received packet to the delay difference determination unit 255.
  • the delay difference determination unit 255 measures the delay time T (n) for each packet by the following equation (5).
  • T (n) R (n) -S (n) (5)
  • T (n), R (n), and S (n) indicate the delay time of the nth packet, the reception time of the nth packet, and the transmission time of the nth packet, respectively.
  • the delay difference determination unit 255 calculates the delay difference ⁇ (n) between the packets by the following equation (6).
  • ⁇ (n) T (n) ⁇ T (n ⁇ 1) (6)
  • ⁇ (n) represents the delay difference of the nth packet.
  • the delay difference determination unit 255 uses ⁇ (n) to determine whether or not the delay difference exceeds a predetermined threshold value.
  • the delay difference determination unit 255 determines that the delay difference has exceeded the threshold value in the nth packet.
  • Th3 is a predetermined threshold value. Then, the delay difference determining unit 255 outputs the packet number N immediately after the delay difference exceeds the threshold value to the packet transmitting / receiving unit 250.
  • the packet transmission / reception unit 250 receives the packet number N from the delay difference determination unit 255, the packet number N immediately after the delay difference exceeds the threshold, the data size of the Nth packet, the (N ⁇ 1) th
  • the data size of each packet, the reception time and transmission time of each packet are stored in the payload of the reply packet, and then transmitted to the packet transfer control device 190A via the eNodeB device 194 in FIG.
  • the threshold value Th3 may be determined in advance, or may be determined each time after looking at a series of values of ⁇ (n). Also, other methods can be used as the discrimination method. For example, T (n) may be compared with a threshold value, and n when the threshold value is exceeded may be set to N.
  • the rate setting unit 254 receives the reduced audio codec rate from the packet transmission / reception unit 250 and sets the rate in the encoder of the audio codec 253.
  • the rate setting unit 254 reads the numerical value of 6.7 kbps set in the CMR field of the SDP or RTP payload header, and sets 6.7 kbps in the audio codec 253.
  • AMR is used as the audio codec 253.
  • the bandwidth measuring unit 205 calculates the bandwidth based on the response signal from the mobile terminal, but the bandwidth measuring unit 205 measures the amount of delay with the mobile terminal and based on this, It can also be calculated.
  • the bandwidth measuring unit 205 does not measure the bandwidth by sending a probe packet for bandwidth measurement from the packet transfer unit, but uses a response signal from the mobile terminal without using the probe packet. It can also be used. In this case, the delay difference determination unit 255 in FIG. 6 is unnecessary.
  • the bandwidth measurement unit 205, the congestion detection unit 210, and the rate control unit 230 are built in the packet transfer control device 190A, but at least one of them may be an external device. it can.
  • the packet transfer control device 190A can be realized by a program executed by a computer. That is, the packet transfer control device 190A may be composed of a packet transfer control processor (not shown) and a storage device (not shown). The storage device stores a packet transfer control program. In this case, the packet transfer control processor performs the above-described packet transfer control operation according to the packet transfer control program stored in the storage device.
  • the conventional control method using the QoS parameters can be avoided. It becomes possible to avoid the congested state that was difficult. As a result, there is an effect that it is possible to avoid QoE (Quality of Experience) degradation such as sound being interrupted or screen being frozen at a terminal for a preferential user. Furthermore, even if a service such as a high-quality VoIP or a high-resolution TV phone is started in the future using the packet communication path of the LTE / EPC system, it is possible to avoid deterioration of QoE on the terminal side. .
  • QoE Quality of Experience
  • a packet transfer control device for connecting a terminal via a network and transferring a packet storing media data in response to a request from the terminal, A congestion detection unit for detecting congestion of the network; When the congestion detection unit detects congestion of the network, a transfer control unit that notifies a request to reduce the rate of the media data to at least one of the partner terminal and the own terminal;
  • a packet transfer control device comprising: (Appendix 2) The packet transfer control device according to appendix 1, wherein the congestion detection unit extracts congestion information from an upstream packet and detects congestion of the network. (Appendix 3) The packet transfer control device according to appendix 2, wherein the congestion detection unit extracts the congestion information by checking an ECN field of an IP header portion of the uplink packet.
  • (Appendix 4) Based on a reply response signal from the terminal to the packet sent from the packet transfer control device, further comprising a bandwidth measuring unit that calculates the bandwidth of the network as a calculated value, The packet transfer control device according to appendix 1, wherein the congestion detection unit detects congestion of the network based on the calculated value. (Appendix 5) The packet transfer control device according to appendix 4, wherein the bandwidth measuring unit calculates the bandwidth of the network at the time of session connection or at a predetermined time interval. (Appendix 6) The packet transfer control device according to any one of appendices 1 to 5, wherein the media data includes at least one of audio data, video data, and audio.
  • (Appendix 7) A communication system including the packet transfer control device according to any one of supplementary notes 1 to 6 and the terminal connected to the packet transfer control device via the network.
  • (Appendix 8) A PCRF device for changing the QoS parameter for the terminal and generating a changed QoS parameter and a change flag when the congestion detection unit detects congestion of the network;
  • the packet transfer control device further includes a rate control unit that creates an instruction to reduce a rate in response to the changed QoS parameter and the change flag, 8.
  • Appendix 9 The communication system according to appendix 8, wherein the QoS parameter is at least one of QCI, ARP, MBR, and GBR.
  • Appendix 10 A packet transfer control method in a packet transfer control device for connecting a terminal via a network and transferring a packet storing media data in response to a request from the terminal, A congestion detection step of detecting congestion of the network; A transfer control step of notifying at least one of the partner terminal and the own terminal of a request to reduce the rate of the media data when congestion of the network is detected; A packet transfer control method including: (Appendix 11) 11. The packet transfer control method according to appendix 10, wherein the congestion detection step detects congestion of the network by extracting congestion information from an upstream packet. (Appendix 12) 12.
  • Appendix 13 Based on a reply response signal from the terminal to the packet transmitted from the packet transfer control device, further comprising a bandwidth measurement step of calculating the bandwidth of the network as a calculated value, The packet transfer control method according to appendix 10, wherein the congestion detection step detects congestion of the network based on the calculated value.
  • Appendix 14 14.
  • the packet transfer control method according to any one of appendices 10 to 14, wherein the media data includes at least one of audio data, video data, and audio.
  • Appendix 16 A computer-readable recording medium recorded with a packet transfer control program for transferring a packet storing media data in response to a request from the terminal to a computer connected to the terminal via a network, the packet transfer control program comprising: To the computer, A congestion detection procedure for detecting congestion of the network; A transfer control procedure for notifying at least one of the partner terminal and the own terminal of a request to reduce the rate of the media data when congestion of the network is detected; A computer-readable recording medium for executing
  • SYMBOLS 110 SIP server 130 IMS network 140 Internet network 145 Web server 150 Mobile network 170_1 Portable terminal 176 Packet transfer part 188 Transfer control part 190, 190A Packet transfer control apparatus 191 PCRF apparatus 194 eNodeB apparatus 205 Bandwidth measurement part 200, 210 Congestion detection part 211 Control unit 230 Rate control unit 250 Packet transmission / reception unit 253 Audio codec 254 Rate setting unit 255 Delay difference determination unit
  • This application is based on Japanese Patent Application No. 2012-260636 filed on November 29, 2012 And the entire disclosure is incorporated herein.

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Abstract

A packet transfer control device that, in order to avoid deterioration in Quality of Experience (QoE), connects a terminal via a network, and transfers packets storing media data, by using a request from the terminal, said packet transfer control device comprising a congestion detection unit that detects network congestion and a transfer control unit that, if the congestion detection unit has detected network congestion, notifies at least either a partner terminal or the terminal itself of a request to reduce the rate for media data.

Description

パケット転送制御装置および通信システムPacket transfer control device and communication system
 本発明は、パケット毎に通信品質を制御しながらパケット転送を行なうパケット転送制御装置およびそれを備えた通信システムに関する。 The present invention relates to a packet transfer control device that performs packet transfer while controlling communication quality for each packet, and a communication system including the same.
 近年、モバイルネットワークでも大容量化、高速化が進展し、LTE(Long Term Evolution)やEPC(Evolved Packet Core)などのシステムが導入開始されている。
 従来の通信システムでは、音声通話やTV電話を行なう回線交換と、データを流すパケット交換とは別々のシステムから構成されていた。それに対して、LTE/EPCシステムでは、同じパケット通信路に、音声通話データやTV電話データやコンテンツ配信データなどといわゆるデータ信号(アプリケーションデータ、ドキュメントデータ、写真データなど)とを一緒に流すことを特徴とする。さらに、携帯端末としては、従来型のいわゆるガラパゴス携帯だけでなく、スマートフォンやタブレットなどの、いわゆるスマートデバイスの普及が加速化している。
 これにより、LTE/EPCシステムでは、従来の通信システムとは比較にならない膨大なデータ量のパケットが、パケット通信路を流れることになる。
 そのため、LTE/EPCシステムにおいては、パケット転送レート制御を行う必要がある。例えば、特開2004−320452号公報(特許文献1)は、パケット転送レート制御を行うパケット転送制御装置を開示している。
 この特許文献1に開示されたパケット転送制御装置は、回線輻輳状態判定部と、転送レート制御決定部と、パケット加工処理部とを備える。回線輻輳状態判定部は、複数パケットに対しパケットサイズの累積値である累積パケット総量にもとづき、バックボーン回線が輻輳かどうかを判別する。転送レート制御決定部は、バックボーン回線が輻輳状態との判定のときは、しきい値を下回るホップカウント値を有する1以上のIP(Internet Protocol)フローを選択する。パケット加工処理部は、転送レート制御決定部で選択されたIPフローに対し、TCP(Transmission Control Protocol)パケットか否かを判別し、TCPパケットの場合は、次に述べる3種のパケット加工処理を施す。
 具体的には、1)サーバからの発信パケットの場合は、TCPヘッダにECN(Explicit Congestion Notification)のCE(Congestion Experience)ビットをセットする。2)クライアントから返信される返信パケットの場合は、TCPヘッダの広告ウィンドゥサイズを縮小変更する。3)確認応答(Ack)パケットの場合は、当該パケットのバックボーン回線に対する送出タイミングを遅らせる。なお、TCPパケットでない場合は、パケットを廃棄する。
In recent years, the capacity and speed of mobile networks have been increased, and systems such as LTE (Long Term Evolution) and EPC (Evolved Packet Core) have begun to be introduced.
In conventional communication systems, circuit switching for voice calls and videophone calls and packet switching for sending data are configured as separate systems. On the other hand, in the LTE / EPC system, voice call data, videophone data, content distribution data, and so-called data signals (application data, document data, photo data, etc.) flow together on the same packet communication path. Features. Furthermore, as mobile terminals, the spread of so-called smart devices such as smartphones and tablets as well as conventional so-called Galapagos mobiles is accelerating.
Thereby, in the LTE / EPC system, a packet with a huge amount of data that cannot be compared with the conventional communication system flows through the packet communication path.
Therefore, in the LTE / EPC system, it is necessary to perform packet transfer rate control. For example, Japanese Patent Laying-Open No. 2004-320452 (Patent Document 1) discloses a packet transfer control device that performs packet transfer rate control.
The packet transfer control device disclosed in Patent Document 1 includes a line congestion state determination unit, a transfer rate control determination unit, and a packet processing unit. The line congestion state determination unit determines whether the backbone line is congested based on the accumulated packet total amount that is the accumulated value of the packet size for a plurality of packets. When determining that the backbone line is in a congested state, the transfer rate control determination unit selects one or more IP (Internet Protocol) flows having a hop count value lower than the threshold value. The packet processing unit determines whether the IP flow selected by the transfer rate control determination unit is a TCP (Transmission Control Protocol) packet. In the case of a TCP packet, the packet processing unit performs the following three types of packet processing. Apply.
Specifically, 1) In the case of an outgoing packet from the server, the CE (Consultation Experience) bit of ECN (Explicit Connection Notification) is set in the TCP header. 2) In the case of a reply packet returned from the client, the advertisement window size of the TCP header is reduced and changed. 3) In the case of an acknowledgment (Ack) packet, the transmission timing of the packet to the backbone line is delayed. If it is not a TCP packet, the packet is discarded.
特開2004−320452号公報([0051]~[0057])JP 2004-320452 A ([0051] to [0057])
 パケット転送制御装置(例えばEPCのP−GW:Packet data network GateWayやS−GW:Serving GateWay)では、これまでは、QoS(Quality of Service)を制御するパラメータとして、QCI(Quality Class Identifier)、MBR(Maximum Bit Rate)、GBR(Guaranteed Bit Rate)などのパラメータを設定して、パケット毎にQoSを制御している。
 しかしながら、LTE/EPCシステム全体のネットワークの帯域幅は、トラヒック量の時間的な変動に依存して時間的に変動するため、QCI(Quality Class Identifier)、MBR(Maximum Bit Rate)、GBR(Guaranteed Bit Rate)などのパラメータ値の設定による転送制御は十分ではなく、最悪の場合は端末で画面がフリーズする、音が途切れる、などといった、QoE(Quality of Experience)の劣化に関する課題が発生していた。
 さらに、今後、LTE/EPCシステムのパケット通信路を用いて高音質VoIP(Voice Over IP)や高解像度TV電話などのリアルタイム通信サービスとして開始される状況において、ネットワークが輻輳すると、最悪の場合には、VoIPによる音声通話の際に端末で音が途切れたり、TV電話の際に端末で映像が乱れたりあるいはフリーズしたりする、といった、端末側のQoEの劣化に関する課題が発生する恐れがあった。
 尚、特許文献1に開示されたパケット転送制御装置は、バックボーン回線が輻輳状態のときに、しきい値を下回るホップカウント値を有する1以上のIPフローを選択する技術思想を開示しているに過ぎない。すなわち、特許文献1では、上述したQoEの劣化に関する課題の認識がない。
 本発明の目的は、QoEの劣化を回避することができる、パケット転送制御装置を提供することにある。
Until now, in packet transfer control devices (for example, EPC P-GW: Packet data network Gateway or S-GW: Serving Gateway), QCI (Quality Class Id) is used as a parameter for controlling QoS (Quality of Service). Parameters such as (Maximum Bit Rate) and GBR (Guaranteed Bit Rate) are set, and QoS is controlled for each packet.
However, since the network bandwidth of the entire LTE / EPC system varies temporally depending on the temporal variation of the traffic volume, QCI (Quality Class Identifier), MBR (Maximum Bit Rate), GBR (Guaranteed Bit Rate) Transfer control by setting parameter values such as (Rate) is not sufficient, and in the worst case, a problem relating to quality of experience (QoE) degradation such as the screen being frozen at the terminal or the sound being interrupted has occurred.
Furthermore, in the future, when the network is congested in a situation where it is started as a real-time communication service such as a high-quality voice VoIP (Voice Over IP) or a high-resolution videophone using the packet communication path of the LTE / EPC system, There is a risk of problems regarding QoE degradation on the terminal side, such as sound being interrupted at the terminal during a voice call by VoIP, or video being disturbed or frozen at the terminal during a videophone call.
The packet transfer control device disclosed in Patent Document 1 discloses a technical idea of selecting one or more IP flows having a hop count value lower than a threshold when the backbone line is congested. Not too much. That is, Patent Document 1 does not recognize the above-described problem relating to the deterioration of QoE.
An object of the present invention is to provide a packet transfer control device capable of avoiding QoE degradation.
 本発明の一形態は、ネットワークを介して端末を接続し、前記端末からの要求によりメディアデータを格納したパケットを転送するパケット転送制御装置であって、前記ネットワークの輻輳を検出する輻輳検出部と、前記輻輳検出部が輻輳を検出した場合に、前記メディアデータに対しレートを削減する要求を相手端末および自端末の少なくとも一方に通知する転送制御部と、を備えることを特徴とする。 One aspect of the present invention is a packet transfer control device for connecting a terminal via a network and transferring a packet storing media data in response to a request from the terminal, the congestion detecting unit detecting congestion of the network, A transfer control unit for notifying at least one of the partner terminal and the own terminal of a request to reduce the rate of the media data when the congestion detection unit detects congestion.
 本発明によれば、トラヒックが統計値に比べ大幅に変化した場合でも、輻輳状態を回避することが可能となる。 According to the present invention, it is possible to avoid a congestion state even when the traffic changes significantly compared to the statistical value.
 図1は、本発明の第1の実施形態に係る通信システムの接続構成を示すブロック図である。
 図2は、図1に示した通信システムに使用される、パケット転送制御装置の構成を示すブロック図である。
 図3は、図2に示したパケット転送制御装置に使用される、レート制御部の構成を示すブロック図である。
 図4は、本発明の第2の実施形態に係る通信システムの接続構成を示すブロック図である。
 図5は、図4に示した通信システムに使用される、パケット転送制御装置の構成を示すブロック図である。
 図6は、図4に示した通信システムに使用される、携帯端末の構成を示すブロック図である。
FIG. 1 is a block diagram showing a connection configuration of a communication system according to the first embodiment of the present invention.
FIG. 2 is a block diagram showing a configuration of a packet transfer control device used in the communication system shown in FIG.
FIG. 3 is a block diagram showing a configuration of a rate control unit used in the packet transfer control device shown in FIG.
FIG. 4 is a block diagram showing a connection configuration of a communication system according to the second embodiment of the present invention.
FIG. 5 is a block diagram showing a configuration of a packet transfer control device used in the communication system shown in FIG.
FIG. 6 is a block diagram showing a configuration of a mobile terminal used in the communication system shown in FIG.
 以下、図面を参照して、本発明の実施形態と動作について詳細に説明する。
[第1の実施形態]
 図1は本発明の第1の実施形態に係る通信システムの構成を示すブロック図である。ここでは、ネットワークとしてはモバイルLTE/EPCパケットネットワーク150を用いる場合の構成を示している。
 また図1の通信システムにおいては、後述するパケット転送制御装置190は、P−GW(Packet data network GateWay)またはS−GW(Serving GateWay)またはそれらの両者を用いる場合の構成を示している。また携帯端末は、いわゆるガラパゴス携帯、スマートフォン、タブレットを想定している。
 図1の通信システムでは、ユーザAが相手のユーザ(図示せず)と通信を行なう例を示している。
 図1の通信システムでは、ユーザAが携帯端末170_1を用いて、モバイルネットワーク150およびIMS(IP Multimedia Subsystem)網130を介して、相手先ネットワーク(図示せず)を経由して相手先の端末(図示せず)とVoIP(Voice Over IP)音声通信を行なう。音声通信(音声電話)の代わりにTV電話でもよいが、図1では音声通信とする。
 図1では、輻輳検出の一例として、端末から送出された上り方向のパケットに対しECN(Explicit Congestion Notification)による輻輳情報を受信して輻輳を検出する構成について示す。なお、ここでは、屋外型LTE無線基地局装置(eNodeB装置)194が無線ネットワークでの輻輳状態を検出すると、eNodeB装置からパケット転送制御装置190に対する上りパケットのIP(internet protocol)ヘッダ部のECNフィールドに所定の値をたてることにより、パケット転送制御装置に輻輳状態であることを通知することを想定している。
 図1の通信システムにおいて、携帯端末170_1は、音声通話の接続要求信号として、携帯端末170_1から相手先端末のIPアドレス、RTP(real−time transport protocol)ポート番号を送出すると、その接続要求信号は、eNodeB装置194ならびにパケット転送制御装置190を経由して、IMS(IP Multimedia Subsystem)網130に配置されたSIP(Session Initiation Protocol)サーバ110ならびに、PCRF(Policy and Charging Rules Function)装置191、の少なくとも一方に転送される。さらに、携帯端末170_1は、接続要求信号に音声通話トラヒック、希望QoSクラス、MBR(Maximum Bit Rate)、GBR(Guaranteed Bit Rate)などのパラメータのうちの少なくとも一つのパラメータを追加し、そのパラメータが追加された接続要求信号をパケット転送制御装置190経由で、SIPサーバ110ならびにPCRF装置191の少なくとも一方に通知することもできる。
 SIPサーバ110は、音声通話の接続要求信号を受け取り、相手先端末(図示せず)に対し相手先ネットワーク(図示せず)を経由して接続要求信号を送出する。そして、SIPサーバ110は、相手先端末からAck信号を受け取ると、当該Ack信号をパケット転送制御装置190ならびにeNodeB装置194を経由して携帯端末170_1に送出する。携帯端末170_1でこれを受信することで、音声通話のための制御信号のやりとりが行なわれる。ここで、相手先端末からは、携帯端末170_1のIPアドレス、RTPポート番号だけでなく、音声通話トラヒック、希望QoSクラス、MBR(Maximum Bit Rate)、GBR(Guaranteed Bit Rate)のパラメータの少なくとも一つを追加して送出することもできる。これらのパラメータは、SIPサーバ110だけでなくPCRF装置191に伝えることできる。
 PCRF装置191は、上りならびに下り方向の少なくとも一方に対して、音声通話トラヒック、携帯端末170_1のIPアドレス、ポート番号をパケット転送制御装置190から入力する。さらに必要であれば、PCRF装置191は、希望QoSクラス、MBR(Maximum Bit Rate)、GBR(Guaranteed Bit Rate)などのパラメータもQoS情報としてパケット転送制御装置190から入力する。
 次に、PCRF装置191は、QoS制御のためのQoSパラメータを生成する。QoS制御のためのQoSパラメータは、QoSクラスを識別する値であるQCI(Quality Class Identifier)、リソースの確保と保持の優先度を表すARP(Allocation and Retention Priority)、MBR、GBRの少なくとも一つである。ここで、MBRとGBRは、パケット転送制御装置190から受信する場合はそのまま使い、受信がない場合はPCRF装置191が生成する。
 PCRF装置191は、上り方向ならびに下り方向の各々に対し、これら4種類のQoSパラメータの少なくとも一つを生成し、生成したパラメータをパケット転送制御装置190に送出する。携帯端末170_1に対しては、トラヒックが音声通話であることから、QoSパラメータの値は、具体的には、例えば、上り、下りともに、QCI=1(Conversational Voice)、ARP=2、GBR=12.2kbps、MBR=22.8kbps、と設定する。ここでは、一例として携帯端末でAMR−NB(Adaptive Multi−Rate Narrowband)音声コーデックを用いるものと想定して、上記のパラメータ値を用いることとする。別の構成として、AMR−WB(Adaptive Multi−Rate Wideband)音声コーデックを用いることもできるが、この場合は、GBRの数値を変更することができる。
 次に、図2を参照して、パケット転送制御装置190の構成について説明する。図2に示されるように、パケット転送制御装置190は、パケット転送部176と、転送制御部188と、輻輳検出部200と、制御部211と、レート制御部230とから構成される。
 図2において、制御部211は、携帯端末170_1からの制御信号をSIPサーバ110に中継するとともに、SIPサーバ110からの制御信号やAck信号を携帯端末170_1へ中継する。また、制御部211は、PCRF装置191から、トラヒックデータ毎に、QCI、ARP、MBR、GBRの4種のQoSパラメータの少なくとも一つを入力する。
 ここで、本第1の実施の形態では、音声通話トラヒックならびに、Webコンテンツのダウンロードデータトラヒックの2種類のトラヒックが発生するので、制御部211は、音声通話トラヒックの上り方向ならびに下り方向の各々に対する4種類のQoSパラメータの少なくとも一つ、及び、ダウンロードデータトラヒックの下り方向に対する4種類のQoSパラメータの少なくとも一つを、PCRF装置191から入力する。制御部211はこれらのQoSパラメータをレート制御部230に送出する。
 輻輳検出部200は、パケット転送部176を介して、eNodeB装置194から送出された上りパケットに対し、IPヘッダ部のECN(Explicit Congestion Notification)フィールドをチェックする。輻輳検出部200は、ECNフィールドに所定の値がたっている場合に、eNodeB装置194から携帯端末までの下り方向が輻輳状態であると認識し、その認識結果を示す下り方向の輻輳検出情報を制御部211とレート制御部230とに出力する。制御部211は前記輻輳検出情報を図1のPCRF装置191に送出する。
 図1のPCRF装置191は、図2の制御部211から輻輳検出情報を入力すると、現在eNodeB装置194に接続している携帯端末170_1に対しQoSパラメータをチェックする。
 携帯端末170_1のQoSパラメータは、前述のように、上り、下りともに、QCI=1(Conversational Voice)、ARP=2、GBR=12.2kbps、MBR=22.8kbpsである。
 一方、輻輳情報として下り方向が輻輳検出と出ているので、この情報をもとにして、PCRF装置191は、携帯端末170_1に対するQoSパラメータを変更することと判断し、変更フラグ=1と設定する。ここでは一例として、携帯端末170_1に対し、QCI=1(Conversational Voice)、ARP=2、GBR=6.7kbps、MBR=12.2kbpsと変更し、そして、PCRF装置191は、携帯端末170_1に関するQoSパラメータならびに変更フラグをパケット転送制御装置190の制御部211に送出する。
 図2に戻って、レート制御部230は、輻輳検出部200から輻輳検出情報を入力し、制御部211からQoSパラメータと変更フラグとを入力する。
 ここで、図3を参照して、レート制御部230の構成について説明する。レート制御部230は、輻輳識別部231と、フラグ識別部232と、QoSパラメータ識別部233と、レート変更設定部234とから構成されている。
 図3において、輻輳識別部231は、輻輳検出情報を入力し、下り方向で輻輳が発生していることを識別し、その識別結果をレート変更設定部234に送出する。
 フラグ識別部232は、携帯端末170_1に対するQoSパラメータに関する変更フラグを入力し、携帯端末170_1に対するQoSパラメータについて変更があることを認識し、携帯端末170_1に対するレート設定を変更する指示をレート変更設定部234に送出する。
 QoSパラメータ識別部233は、携帯端末170_1に対するQoSパラメータを入力し、QoSパラメータのうち、GBRとMBRが変更(削減)されていることを認識する。そして、QoSパラメータ識別部233は、これらのQoSパラメータをレート変更設定部234に送出する。
 レート変更設定部234は、輻輳制御部231、フラグ識別部232、QoSパラメータ識別部233からの入力値にもとづき、音声通信であることから、レートを削減する指示を作成する。具体的には、上り方向については、レート変更設定部234は、携帯端末170_1から送出する音声コーデックのレートを上りのGBRと同等以下のレートに変更する指示を、携帯端末170_1への下り信号で通知するために転送制御部188(図2)に送出する。ここでは、一例として、削減後のレートを6.7kbpsとする。
 さらに、相手端末から携帯端末170_1にむけて送出する音声コーデックのレートも削減する必要があり、レート変更設定部234は、下りのGBRと同等以下のレートに変更する指示を、相手端末への上り信号で通知するために転送制御部188(図2)に送出する。ここでは、一例として、削減後のレートを6.7kbpsとする。
 このように、レート変更設定部234は、これらの上り方向ならびに下り方向のレート削減要求を図2の転送制御部188に出力する。
 図2の転送制御部188は、携帯端末170_1への下り方向のパケットに対して、変更後のGBRおよびMBRに従い、転送を制御するような指示信号をパケット転送部176に送出する。さらに、転送制御部188は、携帯端末170_1に対し、音声コーデックの送出レートを6.7kbpsに削減するよう、下り方向のパケットを使って指示をだす。これは、たとえば、SIP/SDP(Session Description Protocol)を用いている場合は、SDPの部分の”b=”フィールドを6.7kbpsに設定する。あるいは、RTPパケットのRTPペイロードフォーマットヘッダにあるCMR(Codec Mode Request)フィールドに6.7kbpsを設定することもできる。
 一方、転送制御部188は、相手端末に対しては、上り方向のパケットに対して、変更後のGBRおよびMBRに従い、転送を制御するような指示信号をパケット転送部176に送出する。さらに、転送制御部188は、相手端末に対し、音声コーデックの送出レートを6.7kbpsに削減するよう、上り方向のパケットを使って指示をだす。これは、たとえば、SIP/SDPを用いている場合は、SDPの部分の”b=”フィールドを6.7kbpsに設定する。あるいは、RTPパケットのRTPペイロードフォーマットヘッダにあるCMRフィールドに6.7kbpsを設定することもできる。
 図2のパケット転送部176は、転送制御部188から指示信号を入力し、携帯端末170_1に対しては変更後のGBR=6.7kbpsよびMBR=12.2kbpsに従い、転送を制御する。さらに、パケット転送部176は、携帯端末170_1に対し、音声コーデックの送出レートを6.7kbpsに削減するよう、下り方向のパケットを使って指示をだす。これは、たとえば、SIP/SDPを用いている場合は、SDPの部分の”b=”フィールドを6.7kbpsに設定する。あるいは、RTPパケットのRTPペイロードフォーマットヘッダにあるCMRフィールドに6.7kbpsを設定することもできる。
 さらに、パケット転送部176は、上り方向のパケットに対しては、GBR=6.7kbps、MBR=12.2kpbsに従い、パケットを転送する。さらに、パケット転送部176は、相手端末に対し、音声コーデックの送出レートを6.7kbpsに削減するよう、上り方向のパケットを使って指示をだす。これは、たとえば、SIP/SDPを用いている場合は、SDPの部分の”b=”フィールドを6.7kbpsに設定する。あるいは、RTPパケットのRTPペイロードフォーマットヘッダにあるCMRフィールドに6.7kbpsを設定することもできる。
 以上で本発明の第1の実施形態の構成の説明を終えるが、種々の変形が可能である。
 本第1の実施の形態では、音声通話を行なう場合について示したが、例えば、TV電話に対しても同じ構成で対応することができる。また、オーディオ信号にも適用できる。
 なお、上記第1の実施形態では、輻輳の検出はECN情報を用いたが、他の情報を用いることも出来る。
 輻輳検出部200ならびにレート制御部230はパケット転送制御装置190に内蔵させたが、これらを外付けの装置とすることもできる。
 また、PCRF装置191の機能をパケット転送装置190に内蔵させることもできる。
 また、モバイルネットワーク150は、3Gネットワークとすることもできるし、パケット転送制御装置190に、SGSN(Serving GPRS Support Node)やGGSN(Gateway GPRS Support Node)を用いることも出来る。
 尚、パケット転送制御装置190は、コンピュータによって実行されるプログラムによって実現され得る。すなわち、パケット転送制御装置190は、パケット転送制御プロセッサ(図示せず)と記憶装置(図示せず)とから構成されてよい。記憶装置は、パケット転送制御プログラムを記憶する。この場合、パケット転送制御プロセッサは、記憶装置に記憶されたパケット転送制御プログラムに従って上述したパケット転送制御動作を行う。
 次に、本発明の第1の実施形態の効果について説明する。
 本発明の第1の実施形態によれば、LTE/EPCネットワークの帯域幅が、時間的なトラヒック量の変動に依存して時間的に変動した場合に、従来のQoSパラメータによる制御方式では回避が難しかった輻輳状態を回避することが可能となる。その結果、優先的なユーザに対しては、端末で、音が途切れる、画面がフリーズする、といったQoE(Quality of Experience)の劣化を回避することができるという効果がある。さらに、今後、LTE/EPCシステムのパケット通信路を用いて高音質VoIPや高解像度TV電話などのサービスがして開始されても、端末側のQoEの劣化を回避することができるという効果がある。
[第2の実施形態]
 次に、図4を参照して、本発明の第2の実施形態に係る通信システムの構成について説明する。図示の通信システムは、後述するように、パケット転送制御装置のみの構成が異なり、他の構成は図1に示す第1の実施形態に係る通信システムの構成と同じである。従って、パケット転送制御装置に190Aの参照符号を付してある。
 図5は、パケット転送制御装置190Aの構成を示すブロック図である。図5において、図2と同じ番号を付した構成要素は図2と同じ動作を行うので、説明は省略する。
 パケット転送制御装置190Aは、帯域計測部205が追加されていると共に、輻輳検出部の動作が後述するように変更されている点を除いて、図2に示したパケット転送制御装置190と同様の構成を有する。従って、輻輳検出部に210の参照符号を付してある。
 図5において、帯域計測部205は、セッション接続時、またはあらかじめ定められた時間間隔毎に、携帯端末170_1が接続されるネットワークの少なくとも一方の帯域を算出する。
 帯域計測部205は、パケット転送部176からの送信パケットならびに携帯端末170からの返信パケット、の両者を用いて、携帯端末170に接続されるネットワークに対し、上り方向の帯域及び下り方向の帯域の少なくとも一方を計測する。本実施の形態では、上りと下りの両方の帯域を計測する場合の構成を説明する。
 帯域計測部205は、まず、あらかじめ定められたタイミングで、パケット転送部176に対し、複数個の特定のパケット(プローブパケット)を送出する指示を出す。パケット転送部176は、前記指示に従い、指示を受けたタイミングで複数個の特定のパケットを携帯端末170_1に向け、あらかじめ定められた順番で、時間的に連続して送出する。
 ここで、複数個のパケットとは2パケット以上を示す。また、送出する順番はあらかじめ定められた順番とし、例えば、データサイズの小さいパケットからデータサイズの大きなパケットまで、順番に送出するものとする。さらに、パケットと次のパケットの間の時間間隔はあらかじめ定められた時間区間とする。ここで、プロトコルとしては、例えばRTP/UDP(user datagram protocol)/IPを使用する。
 次に、パケット転送部176は、前記複数個のパケットを送出した結果として、携帯端末170_1から返信パケットを受信する。
 ここで、返信パケットには、例えば、携帯端末170_1において、差分遅延に対してしきい値以下で受信できたパケットの番号、当該パケットのデータサイズ、当該パケットをサーバから送信した時刻、当該パケットを携帯端末で受信した時刻、などの情報を含めておく。
 パケット転送部176は、携帯端末170_1から受信した返信信号から、携帯端末170_1が遅延差分のしきい値以下で受信できたパケットの番号、サーバからの送信時刻、携帯端末での受信時刻の情報を抽出し、これらを帯域計測部205に出力する。
 帯域計測部205は、パケット転送部176から上記の情報を入力し、ネットワークの帯域を算出する。
 帯域計測部205は、携帯端末170_1において、遅延差分のしきい値以下で受信できたパケットの番号、当該パケットのデータサイズ、当該パケットのサーバからの送信時刻、当該パケットの携帯端末での受信時刻の情報を入力し、次の(1)式に従い、下り方向の帯域W_dを算出する。
 D(N)/W_d = R(N) − R(N−1)     (1)
 上記(1)式で、D(N)は、パケット転送部176からN番目に送出したパケットのデータサイズを示す。ここで、N、D(N)は、それぞれ、第1の携帯端末170_1において遅延差分のしきい値以下で受信できたパケットの番号、データサイズを示す。また、R(N)は、サーバからN番目に送出したパケットの携帯端末170_1での受信時刻を、R(N−1)はサーバから(N−1)番目に送出したパケットの第1の携帯端末170_1での受信時刻を、それぞれ、示す。
 次に、帯域計測部205は、下り方向の帯域計測値W_dを下記の式(2)により時間的に平滑化し、B(n)_dを算出する。
 B(n)_d = (l−β)*B(n−1)_d + βW_d   (2)
ここで、B(n)_dは、第n時刻の平滑化後の、下り方向の帯域計測値を示し、βは0<β<1の範囲の定数である。なお、式(2)よる時間方向平滑化は、不要な場合は、施さなくても良い。
 次に、携帯端末170_1からは、接続要求後、または前記返信信号送出時のいずれかのタイミングで、複数個の特定のパケットを、パケット転送制御装置190Aに向け、あらかじめ定められた順番で、時間的に連続して送出することにより、パケット転送制御装置190Aにおいて上り方向の帯域を計測する。ここで、複数個のパケットとは2パケット以上を示す。また、送出する順番はあらかじめ定められた順番とし、例えば、データサイズの小さいパケットからデータサイズの大きなパケットまで、順番に送出するものとする。さらに、パケットと次のパケットの間の時間間隔はあらかじめ定められた時間区間とする。ここで、プロトコルとしては、例えばRTP/UDP/IPを使用する。
 帯域計測部205は、パケット転送部176から、携帯端末170_1が送出した複数個のパケットを受け取り、遅延差分のしきい値以下で受信できたパケットの番号、当該パケットのデータサイズ、当該パケットの携帯端末からの送信時刻、当該パケットのパケット転送部176での受信時刻の情報を入力し、次の(3)式に従い、上り方向の帯域W_uを算出する。
 D(M)/W_u = P(M) − P(M−1)   (3)
 次に、帯域計測部205は、上り方向の帯域計測値W_uを下記の式(4)により時間的に平滑化し、B(n)_uを算出する。
 B(n)_u = (l−β)*B(n−1)_u + βW_u   (4)
 パケット転送部176は、ネットワーク帯域計測に必要な情報を、例えばセッション接続時のみ、またはセッション接続時ならびにセッション接続時を基点としあらかじめ定められた時間間隔毎に携帯端末170_1に送出するとともに、定期的に携帯端末から応答信号を受信する。
 帯域計測部205は、例えばあらかじめ定められた時間間隔毎に、B(n)_dとB(n)_uを算出し、これらを輻輳検出部210に送出する。
 輻輳検出部210は、以下の判別を行って、輻輳を検出する。
 1) 下り方向の帯域算出B(n)_d ≧ 携帯端末170_1による音声通話の下り方向のGBR+α の場合、輻輳検出部210は、携帯端末170_1に対する下り方向のQoSパラメータは何も変更しないことを、制御部211を介して、図4のPCRF装置191に伝える。ここで、αはマージンでありあらかじめ定められた値を用いる。
 2) B(n)_d<音声通話の下り方向のGBR+α の場合、輻輳検出部210は、輻輳を検出し、輻輳である情報をPCRF装置191に通知する。
 3) 上り方向についても、輻輳検出部210は、上記1)および2)と同様に判別し、輻輳検出したか否かをPCRP装置191に通知する。
 図4にもどって、PCRP装置191は、図5の輻輳検出部210から輻輳検出情報を入力すると、携帯端末170_1について、ユーザプロファイル情報ならびにQoSパラメータの少なくとも一方をチェックする。
 携帯端末170_1に対するQoSパラメータは、前述のように、上り、下りともに、QCI=1(Conversational Voice)、ARP=2、GBR=12.2kbps、MBR=22.8kbpsであるが、輻輳検出情報を入力しているので、例えば、PCRP装置191は、上り、下りともに、GBR=6.7kbps、MBR=12.2kbpsに変更し、変更したQoSパラメータをパケット転送制御装置190Aに送出する。具体的には、PCRP装置191は、変更したQoSパラメータを図5の制御部211を経由してレート制御部230に送出する。
 図6は、携帯端末170_1の構成を示すブロック図である。携帯端末170_1は、パケット送受信部250と、音声コーデック253と、レート設定部254と、遅延差分判別部255とを備える。
 図6において、パケット送受信部250は、受信したプローブパケットに対し、返信パケットを生成し、生成した返信パケットをネットワークに送出する。ここで、返信パケットは、例えば、以下のように生成する。
 パケット送受信部250は、図5のパケット転送部176から送出された複数個のプローブパケットの各々を受信し、遅延差分判別部255に出力する。
 遅延差分判別部255は、各パケットに対する遅延時間T(n)を下記の式(5)により計測する。
 T(n) = R(n) − S(n)      (5)
ここで、T(n)、R(n)、S(n)は、それぞれ、第n番目のパケットの遅延時間、第n番目のパケットの受信時刻、第n番目のパケットの送信時刻、を示す。
 さらに、遅延差分判別部255は、各パケット間の遅延差分τ(n)を下記の式(6)により算出する。
 τ(n)=T(n) − T(n−1)      (6)
ここで、τ(n)はn番目のパケットの遅延差分を示す。
 次に、遅延差分判別部255は、τ(n)を用いて、遅延差分があらかじめ定められたしきい値を超えるか否かを判別する。もし、τ(n) ≧ Th3の場合、遅延差分判別部255は、n番目のパケットで遅延差分がしきい値を超えたと判別する。ここで、Th3はあらかじめ定められたしきい値である。そして、遅延差分判別部255は、遅延差分がしきい値をこえた直後のパケットの番号Nをパケット送受信部250に出力する。
 パケット送受信部250は、遅延差分判別部255からパケット番号Nを入力し、遅延差分がしきい値をこえた直後のパケットの番号N、第N番目のパケットのデータサイズ、(N−1)番目のパケットのデータサイズ、おのおののパケットの受信時刻、送信時刻を、返信パケットのペイロードに格納した上で、図4のeNodeB装置194を経由してパケット転送制御装置190Aに向け送出する。
 ここで、しきい値Th3はあらかじめ定めておいても良いし、τ(n)の一連の数値を見た上でその都度決定してもよい。
 また、判別法としては、他の方法を用いることも出来る。たとえば、T(n)をしきい値と比較し、しきい値をこえた時点のnをNとするようにしてもよい。
 図6において、レート設定部254は、パケット送受信部250から、削減後の音声コーデックのレートを入力し、音声コーデック253のエンコーダに当該レートを設定する。ここでは、レート設定部254は、SDPまたは、RTPペイロードヘッダのCMRフィールドに設定した6.7kbpsの数値を読み取り、音声コーデック253に6.7kbpsを設定するものとする。なお、ここでは、前述のように、音声コーデック253としてはAMRを用いる。
 以上で本発明の第2の実施形態の構成の説明を終えるが、種々の変形が可能である。
 例えば、図5の帯域計測部205における帯域計測アルゴリズムには、別のアルゴリズムを用いることも出来る。上記実施形態では、帯域計測部205は携帯端末からの応答信号をもとに帯域を算出したが、帯域計測部205は携帯端末との間の遅延量を計測し、これをもとに帯域を算出することもできる。
 なお、帯域計測部205はパケット転送部から帯域計測用のプローブパケットを送出することにより帯域計測するのではなく、プローブパケットを用いずに、携帯端末からの応答信号を用いて帯域推定する方法を用いることも出来る。この場合は、図6における遅延差分判別部255は不要となる。
 なお、本第2の実施の形態では、帯域計測部205と輻輳検出部210とレート制御部230をパケット転送制御装置190Aに内蔵させたが、それらの少なくとも一つを外付け装置とすることもできる。
 尚、パケット転送制御装置190Aは、コンピュータによって実行されるプログラムによって実現され得る。すなわち、パケット転送制御装置190Aは、パケット転送制御プロセッサ(図示せず)と記憶装置(図示せず)とから構成されてよい。記憶装置は、パケット転送制御プログラムを記憶する。この場合、パケット転送制御プロセッサは、記憶装置に記憶されたパケット転送制御プログラムに従って上述したパケット転送制御動作を行う。
 次に、本発明の第2の実施形態の効果について説明する。
 本発明の第2の実施形態によれば、LTE/EPCネットワークの帯域幅が、時間的なトラヒック量の変動に依存して時間的に変動した場合に、従来のQoCパラメータによる制御方式では回避が難しかった輻輳状態を回避することが可能となる。その結果、優先的なユーザに対しては、端末で、音が途切れる、画面がフリーズする、といったQoE(Quality of Experience)の劣化を回避することができるという効果がある。さらに、今後、LTE/EPCシステムのパケット通信路を用いて高音質VoIPや高解像度TV電話などのサービスがして開始されても、端末側のQoEの劣化を回避することができるという効果がある。
 以上、実施の形態を参照して本願発明を説明したが、本願発明は上記実施の形態に限定されるものではない。本願発明の構成や詳細には、本願発明のスコープ内で当業者が理解し得る様々な変更をすることができる。
 上記の実施形態の一部又は全部は、以下の付記のようにも記載されうるが、以下には限られない。
(付記1)
 ネットワークを介して端末を接続し、前記端末からの要求によりメディアデータを格納したパケットを転送するパケット転送制御装置であって、
 前記ネットワークの輻輳を検出する輻輳検出部と、
 前記輻輳検出部が前記ネットワークの輻輳を検出した場合に、前記メディアデータに対しレートを削減する要求を相手端末および自端末の少なくとも一方に通知する転送制御部と、
を備えることを特徴とするパケット転送制御装置。
(付記2)
 前記輻輳検出部は、上り方向のパケットから輻輳情報を抽出して、前記ネットワークの輻輳を検出する、ことを特徴とする付記1に記載のパケット転送制御装置。
(付記3)
 前記輻輳検出部は、前記上り方向のパケットのIPヘッダ部のECNフィールドをチェックすることにより、前記輻輳情報を抽出する、ことを特徴とする付記2に記載のパケット転送制御装置。
(付記4)
 前記パケット転送制御装置から送出したパケットに対する前記端末からの返信応答信号に基づいて、前記ネットワークの帯域を算出値として算出する帯域計測部を更に備え、
 前記輻輳検出部は、前記算出値に基づいて前記ネットワークの輻輳を検出する、ことを特徴とする付記1に記載のパケット転送制御装置。
(付記5)
 前記帯域計測部は、セッション接続時又は予め定められた時間間隔毎に、前記ネットワークの帯域を算出する、ことを特徴とする付記4に記載のパケット転送制御装置。
(付記6)
 前記メディアデータが、音声データ、映像データ、及びオーディオの少なくとも一つからなる、付記1乃至5のいずれか1つに記載のパケット転送制御装置。
(付記7)
 付記1乃至6のいずれか1つに記載のパケット転送制御装置と、前記ネットワークを介して前記パケット転送制御装置に接続された前記端末と、を含む通信システム。
(付記8)
 前記輻輳検出部が前記ネットワークの輻輳を検出した場合に、前記端末に対するQoSパラメータを変更して、変更したQoSパラメータと変更フラグとを生成するPCRF装置を更に含み、
 前記パケット転送制御装置は、前記変更したQoSパラメータと前記変更フラグとに応答して、レートを削減する指示を作成するレート制御部を更に備え、
 前記転送制御部は、前記レートを削減する指示に応答して、前記メディアデータに対し前記レートを削減する要求を相手端末および自端末の少なくとも一方に通知する、付記7に記載の通信システム。
(付記9)
 前記QoSパラメータは、QCI、ARP、MBR及びGBRの少なくとも一つである、付記8に記載の通信システム。
(付記10)
 ネットワークを介して端末を接続し、前記端末からの要求によりメディアデータを格納したパケットを転送するパケット転送制御装置におけるパケット転送制御方法であって、
 前記ネットワークの輻輳を検出する輻輳検出ステップと、
 前記ネットワークの輻輳を検出した場合に、前記メディアデータに対しレートを削減する要求を相手端末および自端末の少なくとも一方に通知する転送制御ステップと、
を含むパケット転送制御方法。
(付記11)
 前記輻輳検出ステップは、上り方向のパケットから輻輳情報を抽出して、前記ネットワークの輻輳を検出する、付記10に記載のパケット転送制御方法。
(付記12)
 前記輻輳検出ステップは、前記上り方向のパケットのIPヘッダ部のECNフィールドをチェックすることにより、前記輻輳情報を抽出する、付記11に記載のパケット転送制御方法。
(付記13)
 前記パケット転送制御装置から送出したパケットに対する前記端末からの返信応答信号に基づいて、前記ネットワークの帯域を算出値として算出する帯域計測ステップを更に備え、
 前記輻輳検出ステップは、前記算出値に基づいて前記ネットワークの輻輳を検出する、付記10に記載のパケット転送制御方法。
(付記14)
 前記帯域計測ステップは、セッション接続時又は予め定められた時間間隔毎に、前記ネットワークの帯域を算出する、付記13に記載のパケット転送制御方法。
(付記15)
 前記メディアデータが、音声データ、映像データ、及びオーディオの少なくとも一つからなる、付記10乃至14のいずれか1つに記載のパケット転送制御方法。
(付記16)
 ネットワークを介して端末に接続されたコンピュータに、前記端末からの要求によりメディアデータを格納したパケットを転送させるパケット転送制御プログラムを記録したコンピュータ読み取り可能な記録媒体であって、前記パケット転送制御プログラムは、前記コンピュータに、
 前記ネットワークの輻輳を検出する輻輳検出手順と、
 前記ネットワークの輻輳を検出した場合に、前記メディアデータに対しレートを削減する要求を相手端末および自端末の少なくとも一方に通知する転送制御手順と、
を実行させるコンピュータ読み取り可能な記録媒体。
Hereinafter, embodiments and operations of the present invention will be described in detail with reference to the drawings.
[First Embodiment]
FIG. 1 is a block diagram showing a configuration of a communication system according to the first embodiment of the present invention. Here, a configuration in which the mobile LTE / EPC packet network 150 is used as the network is shown.
Further, in the communication system of FIG. 1, a packet transfer control device 190 (to be described later) shows a configuration using P-GW (Packet data network Gateway) or S-GW (Serving Gateway) or both. The mobile terminal is assumed to be a so-called Galapagos mobile phone, a smartphone, or a tablet.
The communication system of FIG. 1 shows an example in which user A communicates with a partner user (not shown).
In the communication system of FIG. 1, a user A uses a portable terminal 170 </ b> _ <b> 1 via a mobile network 150 and an IMS (IP Multimedia Subsystem) network 130 via a counterpart network (not shown). (Not shown) and VoIP (Voice Over IP) voice communication. A TV phone may be used in place of the voice communication (voice call), but in FIG.
As an example of congestion detection, FIG. 1 shows a configuration for detecting congestion by receiving congestion information by ECN (Explicit Congestion Notification) for an upstream packet sent from a terminal. Here, when the outdoor LTE radio base station apparatus (eNodeB apparatus) 194 detects a congestion state in the wireless network, the ECN field of the IP (Internet protocol) header portion of the uplink packet from the eNodeB apparatus to the packet transfer control apparatus 190 It is assumed that the packet transfer control device is notified of the congestion state by setting a predetermined value for the packet transfer control device.
In the communication system of FIG. 1, when the mobile terminal 170_1 sends out the IP address and RTP (real-time transport protocol) port number of the destination terminal from the mobile terminal 170_1 as a connection request signal for a voice call, the connection request signal is , A SIP (Session Initiation Protocol) server 110 disposed in an IMS (IP Multimedia Subsystem) network 130, and a PCRF (Policy and Charging Rules Function) 191 at least via the eNodeB device 194 and the packet transfer control device 190, respectively. Forwarded to one side. Further, the mobile terminal 170_1 adds at least one parameter of parameters such as voice call traffic, desired QoS class, MBR (Maximum Bit Rate), GBR (Guaranteed Bit Rate) to the connection request signal, and the parameter is added. The connection request signal thus made can be notified to at least one of the SIP server 110 and the PCRF device 191 via the packet transfer control device 190.
The SIP server 110 receives a connection request signal for a voice call and sends a connection request signal to a partner terminal (not shown) via a partner network (not shown). When the SIP server 110 receives the Ack signal from the counterpart terminal, the SIP server 110 sends the Ack signal to the portable terminal 170_1 via the packet transfer control device 190 and the eNodeB device 194. When this is received by the portable terminal 170_1, control signals for voice call are exchanged. Here, not only the IP address and RTP port number of the mobile terminal 170_1 but also at least one of the parameters of voice call traffic, desired QoS class, MBR (Maximum Bit Rate), GBR (Guaranteed Bit Rate) from the counterpart terminal. Can also be sent out. These parameters can be transmitted not only to the SIP server 110 but also to the PCRF device 191.
The PCRF device 191 inputs the voice call traffic, the IP address of the portable terminal 170_1, and the port number from the packet transfer control device 190 for at least one of the upstream and downstream directions. If necessary, the PCRF device 191 also inputs parameters such as a desired QoS class, MBR (Maximum Bit Rate), GBR (Guaranteed Bit Rate), etc. from the packet transfer control device 190 as QoS information.
Next, the PCRF device 191 generates a QoS parameter for QoS control. The QoS parameter for QoS control is at least one of QCI (Quality Class Identifier) which is a value for identifying a QoS class, ARP (Allocation and Retention Priority) indicating the priority of resource reservation and retention, MBR, and GBR. is there. Here, the MBR and the GBR are used as they are when received from the packet transfer control device 190, and are generated by the PCRF device 191 when there is no reception.
The PCRF device 191 generates at least one of these four types of QoS parameters for each of the uplink direction and the downlink direction, and sends the generated parameter to the packet transfer control device 190. For the mobile terminal 170_1, since the traffic is a voice call, specifically, the QoS parameter values are, for example, QCI = 1 (Conversational Voice), ARP = 2, GBR = 12, for both uplink and downlink. .2 kbps and MBR = 22.8 kbps are set. Here, as an example, assuming that an AMR-NB (Adaptive Multi-Rate Narrowband) audio codec is used in a mobile terminal, the above parameter values are used. As another configuration, an AMR-WB (Adaptive Multi-Rate Wideband) audio codec can be used. In this case, the value of GBR can be changed.
Next, the configuration of the packet transfer control device 190 will be described with reference to FIG. As shown in FIG. 2, the packet transfer control device 190 includes a packet transfer unit 176, a transfer control unit 188, a congestion detection unit 200, a control unit 211, and a rate control unit 230.
In FIG. 2, the control unit 211 relays a control signal from the mobile terminal 170_1 to the SIP server 110, and relays a control signal and an Ack signal from the SIP server 110 to the mobile terminal 170_1. In addition, the control unit 211 inputs at least one of four types of QoS parameters of QCI, ARP, MBR, and GBR for each traffic data from the PCRF device 191.
Here, in the first embodiment, since two types of traffic, that is, voice call traffic and Web content download data traffic, are generated, the control unit 211 controls the uplink and downlink directions of the voice call traffic. At least one of the four types of QoS parameters and at least one of the four types of QoS parameters for the downlink direction of the download data traffic are input from the PCRF device 191. The control unit 211 sends these QoS parameters to the rate control unit 230.
The congestion detection unit 200 checks the ECN (Explicit Congestion Notification) field of the IP header for the upstream packet sent from the eNodeB device 194 via the packet transfer unit 176. When the ECN field has a predetermined value, the congestion detection unit 200 recognizes that the downlink direction from the eNodeB device 194 to the portable terminal is in a congestion state, and controls downlink congestion detection information indicating the recognition result. Output to the unit 211 and the rate control unit 230. The control unit 211 sends the congestion detection information to the PCRF device 191 in FIG.
When the congestion detection information is input from the control unit 211 in FIG. 2, the PCRF device 191 in FIG. 1 checks the QoS parameter for the portable terminal 170_1 currently connected to the eNodeB device 194.
As described above, the QoS parameters of the portable terminal 170_1 are QCI = 1 (Conversational Voice), ARP = 2, GBR = 12.2 kbps, and MBR = 22.8 kbps for both uplink and downlink.
On the other hand, since the downlink direction is congestion detection as the congestion information, the PCRF device 191 determines that the QoS parameter for the portable terminal 170_1 is to be changed based on this information, and sets the change flag = 1. . Here, as an example, the mobile terminal 170_1 is changed to QCI = 1 (conversational voice), ARP = 2, GBR = 6.7 kbps, MBR = 12.2 kbps, and the PCRF device 191 performs QoS related to the mobile terminal 170_1. The parameter and the change flag are sent to the control unit 211 of the packet transfer control device 190.
Returning to FIG. 2, the rate control unit 230 inputs congestion detection information from the congestion detection unit 200, and inputs a QoS parameter and a change flag from the control unit 211.
Here, the configuration of the rate control unit 230 will be described with reference to FIG. The rate control unit 230 includes a congestion identification unit 231, a flag identification unit 232, a QoS parameter identification unit 233, and a rate change setting unit 234.
In FIG. 3, the congestion identification unit 231 receives the congestion detection information, identifies that congestion has occurred in the downstream direction, and sends the identification result to the rate change setting unit 234.
The flag identifying unit 232 inputs a change flag related to the QoS parameter for the mobile terminal 170_1, recognizes that there is a change in the QoS parameter for the mobile terminal 170_1, and instructs the rate change setting unit 234 to change the rate setting for the mobile terminal 170_1. To send.
The QoS parameter identification unit 233 inputs the QoS parameter for the mobile terminal 170_1, and recognizes that the GBR and MBR are changed (reduced) among the QoS parameters. Then, the QoS parameter identification unit 233 sends these QoS parameters to the rate change setting unit 234.
The rate change setting unit 234 creates an instruction to reduce the rate based on the voice communication based on the input values from the congestion control unit 231, the flag identification unit 232, and the QoS parameter identification unit 233. Specifically, for the uplink direction, the rate change setting unit 234 gives an instruction to change the rate of the audio codec transmitted from the mobile terminal 170_1 to a rate equal to or less than the rate of the uplink GBR by using a downlink signal to the mobile terminal 170_1. It sends it to the transfer control unit 188 (FIG. 2) for notification. Here, as an example, the rate after reduction is set to 6.7 kbps.
Furthermore, it is necessary to reduce the rate of the voice codec transmitted from the counterpart terminal to the portable terminal 170_1, and the rate change setting unit 234 sends an instruction to change to a rate equal to or lower than the downlink GBR to the uplink to the counterpart terminal. The data is sent to the transfer control unit 188 (FIG. 2) for notification by a signal. Here, as an example, the rate after reduction is set to 6.7 kbps.
In this way, the rate change setting unit 234 outputs these uplink and downlink rate reduction requests to the transfer control unit 188 in FIG.
The transfer control unit 188 in FIG. 2 sends an instruction signal for controlling transfer to the packet transfer unit 176 in accordance with the changed GBR and MBR for the downstream packet to the mobile terminal 170_1. Further, the transfer control unit 188 instructs the mobile terminal 170_1 to use a downstream packet so as to reduce the audio codec transmission rate to 6.7 kbps. For example, when SIP / SDP (Session Description Protocol) is used, the “b =” field of the SDP portion is set to 6.7 kbps. Alternatively, 6.7 kbps can be set in the CMR (Codec Mode Request) field in the RTP payload format header of the RTP packet.
On the other hand, the transfer control unit 188 sends to the packet transfer unit 176 an instruction signal for controlling transfer in accordance with the changed GBR and MBR for the uplink packet. Furthermore, the transfer control unit 188 gives an instruction to the partner terminal using the uplink packet so as to reduce the transmission rate of the voice codec to 6.7 kbps. For example, when SIP / SDP is used, the “b =” field of the SDP portion is set to 6.7 kbps. Alternatively, 6.7 kbps can be set in the CMR field in the RTP payload format header of the RTP packet.
The packet transfer unit 176 in FIG. 2 inputs an instruction signal from the transfer control unit 188, and controls transfer to the portable terminal 170_1 according to the changed GBR = 6.7 kbps and MBR = 12.2 kbps. Further, the packet transfer unit 176 instructs the mobile terminal 170_1 to use the downstream packet so as to reduce the audio codec transmission rate to 6.7 kbps. For example, when SIP / SDP is used, the “b =” field of the SDP portion is set to 6.7 kbps. Alternatively, 6.7 kbps can be set in the CMR field in the RTP payload format header of the RTP packet.
Further, the packet transfer unit 176 transfers the packet according to GBR = 6.7 kbps and MBR = 12.2 kbps for the uplink packet. Further, the packet transfer unit 176 gives an instruction to the partner terminal using the uplink packet so as to reduce the transmission rate of the voice codec to 6.7 kbps. For example, when SIP / SDP is used, the “b =” field of the SDP portion is set to 6.7 kbps. Alternatively, 6.7 kbps can be set in the CMR field in the RTP payload format header of the RTP packet.
Although the description of the configuration of the first embodiment of the present invention has been completed above, various modifications are possible.
In the first embodiment, a case where a voice call is made has been described. However, for example, a TV phone can be handled with the same configuration. It can also be applied to audio signals.
In the first embodiment, ECN information is used to detect congestion, but other information can also be used.
Although the congestion detection unit 200 and the rate control unit 230 are built in the packet transfer control device 190, they can be external devices.
Further, the function of the PCRF device 191 can be built in the packet transfer device 190.
The mobile network 150 may be a 3G network, and the packet transfer control device 190 may be an SGSN (Serving GPRS Support Node) or a GGSN (Gateway GPRS Support Node).
The packet transfer control device 190 can be realized by a program executed by a computer. That is, the packet transfer control device 190 may be composed of a packet transfer control processor (not shown) and a storage device (not shown). The storage device stores a packet transfer control program. In this case, the packet transfer control processor performs the above-described packet transfer control operation according to the packet transfer control program stored in the storage device.
Next, effects of the first exemplary embodiment of the present invention will be described.
According to the first embodiment of the present invention, when the bandwidth of the LTE / EPC network temporally varies depending on the temporal traffic variation, the conventional QoS parameter control method avoids this. It becomes possible to avoid the congested state that was difficult. As a result, there is an effect that it is possible to avoid QoE (Quality of Experience) degradation such as sound being interrupted or screen being frozen at a terminal for a preferential user. Furthermore, even if a service such as a high-quality VoIP or a high-resolution TV phone is started in the future using the packet communication path of the LTE / EPC system, it is possible to avoid deterioration of QoE on the terminal side. .
[Second Embodiment]
Next, the configuration of a communication system according to the second embodiment of the present invention will be described with reference to FIG. As will be described later, the illustrated communication system is different only in the packet transfer control device, and the other configuration is the same as the configuration of the communication system according to the first embodiment shown in FIG. Accordingly, reference numeral 190A is assigned to the packet transfer control device.
FIG. 5 is a block diagram showing a configuration of the packet transfer control device 190A. In FIG. 5, the constituent elements having the same numbers as those in FIG. 2 perform the same operations as those in FIG.
The packet transfer control device 190A is the same as the packet transfer control device 190 shown in FIG. 2 except that the bandwidth measuring unit 205 is added and the operation of the congestion detection unit is changed as will be described later. It has a configuration. Accordingly, reference numeral 210 is assigned to the congestion detection unit.
In FIG. 5, the bandwidth measuring unit 205 calculates at least one bandwidth of the network to which the mobile terminal 170_1 is connected at the time of session connection or at predetermined time intervals.
The bandwidth measuring unit 205 uses the transmission packet from the packet transfer unit 176 and the return packet from the mobile terminal 170 to determine the upstream bandwidth and the downstream bandwidth for the network connected to the mobile terminal 170. At least one is measured. In the present embodiment, a configuration in the case of measuring both upstream and downstream bands will be described.
First, the bandwidth measuring unit 205 instructs the packet transfer unit 176 to send a plurality of specific packets (probe packets) at a predetermined timing. In accordance with the instruction, the packet transfer unit 176 sends a plurality of specific packets to the portable terminal 170_1 at a timing when the instruction is received, and continuously transmits in a predetermined order in time.
Here, the plurality of packets indicates two or more packets. The sending order is a predetermined order. For example, packets are sent in order from a packet having a small data size to a packet having a large data size. Further, the time interval between the packet and the next packet is a predetermined time interval. Here, for example, RTP / UDP (user datagram protocol) / IP is used as the protocol.
Next, the packet transfer unit 176 receives a reply packet from the portable terminal 170_1 as a result of sending the plurality of packets.
Here, the reply packet includes, for example, the packet number received by the portable terminal 170_1 below the threshold with respect to the differential delay, the data size of the packet, the time when the packet was transmitted from the server, and the packet. Include information such as the time received by the mobile terminal.
The packet transfer unit 176 obtains, from the return signal received from the mobile terminal 170_1, the packet number that the mobile terminal 170_1 has received below the delay difference threshold, the transmission time from the server, and the reception time information at the mobile terminal. These are extracted and output to the bandwidth measuring unit 205.
The bandwidth measuring unit 205 receives the above information from the packet transfer unit 176 and calculates the network bandwidth.
The bandwidth measuring unit 205 receives the packet number, the packet data size, the transmission time of the packet from the server, and the reception time of the packet at the portable terminal at the portable terminal 170_1 below the delay difference threshold. The downstream band W_d is calculated according to the following equation (1).
D (N) / W_d = R (N) -R (N-1) (1)
In the above equation (1), D (N) represents the data size of the Nth packet transmitted from the packet transfer unit 176. Here, N and D (N) indicate the number and data size of a packet that can be received by the first portable terminal 170_1 below the delay difference threshold value, respectively. R (N) is the reception time at the portable terminal 170_1 of the Nth packet sent from the server, and R (N-1) is the first portable of the (N-1) th packet sent from the server. The reception times at terminal 170_1 are respectively shown.
Next, the bandwidth measuring unit 205 smoothes the downstream bandwidth measurement value W_d temporally according to the following equation (2) to calculate B (n) _d.
B (n) _d = (l−β) * B (n−1) _d + βW_d (2)
Here, B (n) _d indicates a downstream band measurement value after smoothing at the nth time, and β is a constant in a range of 0 <β <1. Note that the time direction smoothing according to the equation (2) may not be performed if unnecessary.
Next, the mobile terminal 170_1 sends a plurality of specific packets to the packet transfer control device 190A in a predetermined order at any timing after the connection request or when the reply signal is transmitted. Thus, the upstream band is measured in the packet transfer control device 190A. Here, the plurality of packets indicates two or more packets. The sending order is a predetermined order. For example, packets are sent in order from a packet having a small data size to a packet having a large data size. Further, the time interval between the packet and the next packet is a predetermined time interval. Here, for example, RTP / UDP / IP is used as the protocol.
The bandwidth measuring unit 205 receives a plurality of packets sent from the portable terminal 170_1 from the packet transfer unit 176, and receives the packet number that is received below the threshold of the delay difference, the data size of the packet, the mobile phone of the packet Information on the transmission time from the terminal and the reception time of the packet at the packet transfer unit 176 is input, and the upstream bandwidth W_u is calculated according to the following equation (3).
D (M) / W_u = P (M) -P (M-1) (3)
Next, the bandwidth measuring unit 205 smoothes the bandwidth measurement value W_u in the upstream direction temporally according to the following equation (4) to calculate B (n) _u.
B (n) _u = (l−β) * B (n−1) _u + βW_u (4)
The packet transfer unit 176 sends information necessary for network bandwidth measurement to the portable terminal 170_1 only at the time of session connection or at predetermined time intervals starting from the time of session connection and the time of session connection. A response signal is received from the mobile terminal.
The bandwidth measuring unit 205 calculates B (n) _d and B (n) _u, for example, at predetermined time intervals, and sends them to the congestion detection unit 210.
The congestion detection unit 210 performs the following determination to detect congestion.
1) Downlink bandwidth calculation B (n) _d ≧ If GBR + α in the downlink direction of a voice call by the mobile terminal 170_1, the congestion detection unit 210 does not change any of the downlink QoS parameters for the mobile terminal 170_1. This is transmitted to the PCRF device 191 in FIG. 4 via the control unit 211. Here, α is a margin, and a predetermined value is used.
2) When B (n) _d <GBR + α in the downlink direction of a voice call, the congestion detection unit 210 detects congestion and notifies the PCRF device 191 of the congestion information.
3) Concerning the uplink direction, the congestion detection unit 210 determines the same as in 1) and 2) above, and notifies the PCRP device 191 whether or not congestion has been detected.
Returning to FIG. 4, when the congestion detection information is input from the congestion detection unit 210 of FIG. 5, the PCRP apparatus 191 checks at least one of the user profile information and the QoS parameter for the portable terminal 170_1.
As described above, the QoS parameters for the portable terminal 170_1 are QCI = 1 (Conversational Voice), ARP = 2, GBR = 12.2 kbps, MBR = 22.8 kbps in both uplink and downlink, but input congestion detection information. Therefore, for example, the PCRP device 191 changes GBR = 6.7 kbps and MBR = 12.2 kbps in both uplink and downlink, and sends the changed QoS parameter to the packet transfer control device 190A. Specifically, the PCRP apparatus 191 sends the changed QoS parameter to the rate control unit 230 via the control unit 211 of FIG.
FIG. 6 is a block diagram illustrating a configuration of the mobile terminal 170_1. The portable terminal 170_1 includes a packet transmission / reception unit 250, an audio codec 253, a rate setting unit 254, and a delay difference determination unit 255.
In FIG. 6, the packet transmitting / receiving unit 250 generates a reply packet for the received probe packet and sends the generated reply packet to the network. Here, the reply packet is generated as follows, for example.
The packet transmission / reception unit 250 receives each of the plurality of probe packets transmitted from the packet transfer unit 176 in FIG. 5 and outputs the received packet to the delay difference determination unit 255.
The delay difference determination unit 255 measures the delay time T (n) for each packet by the following equation (5).
T (n) = R (n) -S (n) (5)
Here, T (n), R (n), and S (n) indicate the delay time of the nth packet, the reception time of the nth packet, and the transmission time of the nth packet, respectively. .
Further, the delay difference determination unit 255 calculates the delay difference τ (n) between the packets by the following equation (6).
τ (n) = T (n) −T (n−1) (6)
Here, τ (n) represents the delay difference of the nth packet.
Next, the delay difference determination unit 255 uses τ (n) to determine whether or not the delay difference exceeds a predetermined threshold value. If τ (n) ≧ Th3, the delay difference determination unit 255 determines that the delay difference has exceeded the threshold value in the nth packet. Here, Th3 is a predetermined threshold value. Then, the delay difference determining unit 255 outputs the packet number N immediately after the delay difference exceeds the threshold value to the packet transmitting / receiving unit 250.
The packet transmission / reception unit 250 receives the packet number N from the delay difference determination unit 255, the packet number N immediately after the delay difference exceeds the threshold, the data size of the Nth packet, the (N−1) th The data size of each packet, the reception time and transmission time of each packet are stored in the payload of the reply packet, and then transmitted to the packet transfer control device 190A via the eNodeB device 194 in FIG.
Here, the threshold value Th3 may be determined in advance, or may be determined each time after looking at a series of values of τ (n).
Also, other methods can be used as the discrimination method. For example, T (n) may be compared with a threshold value, and n when the threshold value is exceeded may be set to N.
In FIG. 6, the rate setting unit 254 receives the reduced audio codec rate from the packet transmission / reception unit 250 and sets the rate in the encoder of the audio codec 253. Here, it is assumed that the rate setting unit 254 reads the numerical value of 6.7 kbps set in the CMR field of the SDP or RTP payload header, and sets 6.7 kbps in the audio codec 253. Here, as described above, AMR is used as the audio codec 253.
This is the end of the description of the configuration of the second exemplary embodiment of the present invention, but various modifications are possible.
For example, another algorithm can be used as the band measurement algorithm in the band measurement unit 205 of FIG. In the above embodiment, the bandwidth measuring unit 205 calculates the bandwidth based on the response signal from the mobile terminal, but the bandwidth measuring unit 205 measures the amount of delay with the mobile terminal and based on this, It can also be calculated.
The bandwidth measuring unit 205 does not measure the bandwidth by sending a probe packet for bandwidth measurement from the packet transfer unit, but uses a response signal from the mobile terminal without using the probe packet. It can also be used. In this case, the delay difference determination unit 255 in FIG. 6 is unnecessary.
In the second embodiment, the bandwidth measurement unit 205, the congestion detection unit 210, and the rate control unit 230 are built in the packet transfer control device 190A, but at least one of them may be an external device. it can.
The packet transfer control device 190A can be realized by a program executed by a computer. That is, the packet transfer control device 190A may be composed of a packet transfer control processor (not shown) and a storage device (not shown). The storage device stores a packet transfer control program. In this case, the packet transfer control processor performs the above-described packet transfer control operation according to the packet transfer control program stored in the storage device.
Next, effects of the second exemplary embodiment of the present invention will be described.
According to the second embodiment of the present invention, when the bandwidth of the LTE / EPC network temporally varies depending on the temporal traffic variation, the conventional control method using the QoS parameters can be avoided. It becomes possible to avoid the congested state that was difficult. As a result, there is an effect that it is possible to avoid QoE (Quality of Experience) degradation such as sound being interrupted or screen being frozen at a terminal for a preferential user. Furthermore, even if a service such as a high-quality VoIP or a high-resolution TV phone is started in the future using the packet communication path of the LTE / EPC system, it is possible to avoid deterioration of QoE on the terminal side. .
Although the present invention has been described with reference to the embodiments, the present invention is not limited to the above embodiments. Various changes that can be understood by those skilled in the art can be made to the configuration and details of the present invention within the scope of the present invention.
A part or all of the above-described embodiment can be described as in the following supplementary notes, but is not limited thereto.
(Appendix 1)
A packet transfer control device for connecting a terminal via a network and transferring a packet storing media data in response to a request from the terminal,
A congestion detection unit for detecting congestion of the network;
When the congestion detection unit detects congestion of the network, a transfer control unit that notifies a request to reduce the rate of the media data to at least one of the partner terminal and the own terminal;
A packet transfer control device comprising:
(Appendix 2)
The packet transfer control device according to appendix 1, wherein the congestion detection unit extracts congestion information from an upstream packet and detects congestion of the network.
(Appendix 3)
The packet transfer control device according to appendix 2, wherein the congestion detection unit extracts the congestion information by checking an ECN field of an IP header portion of the uplink packet.
(Appendix 4)
Based on a reply response signal from the terminal to the packet sent from the packet transfer control device, further comprising a bandwidth measuring unit that calculates the bandwidth of the network as a calculated value,
The packet transfer control device according to appendix 1, wherein the congestion detection unit detects congestion of the network based on the calculated value.
(Appendix 5)
The packet transfer control device according to appendix 4, wherein the bandwidth measuring unit calculates the bandwidth of the network at the time of session connection or at a predetermined time interval.
(Appendix 6)
The packet transfer control device according to any one of appendices 1 to 5, wherein the media data includes at least one of audio data, video data, and audio.
(Appendix 7)
A communication system including the packet transfer control device according to any one of supplementary notes 1 to 6 and the terminal connected to the packet transfer control device via the network.
(Appendix 8)
A PCRF device for changing the QoS parameter for the terminal and generating a changed QoS parameter and a change flag when the congestion detection unit detects congestion of the network;
The packet transfer control device further includes a rate control unit that creates an instruction to reduce a rate in response to the changed QoS parameter and the change flag,
8. The communication system according to appendix 7, wherein the transfer control unit notifies at least one of the partner terminal and the own terminal of a request to reduce the rate for the media data in response to an instruction to reduce the rate.
(Appendix 9)
The communication system according to appendix 8, wherein the QoS parameter is at least one of QCI, ARP, MBR, and GBR.
(Appendix 10)
A packet transfer control method in a packet transfer control device for connecting a terminal via a network and transferring a packet storing media data in response to a request from the terminal,
A congestion detection step of detecting congestion of the network;
A transfer control step of notifying at least one of the partner terminal and the own terminal of a request to reduce the rate of the media data when congestion of the network is detected;
A packet transfer control method including:
(Appendix 11)
11. The packet transfer control method according to appendix 10, wherein the congestion detection step detects congestion of the network by extracting congestion information from an upstream packet.
(Appendix 12)
12. The packet transfer control method according to appendix 11, wherein the congestion detection step extracts the congestion information by checking an ECN field of an IP header part of the uplink packet.
(Appendix 13)
Based on a reply response signal from the terminal to the packet transmitted from the packet transfer control device, further comprising a bandwidth measurement step of calculating the bandwidth of the network as a calculated value,
The packet transfer control method according to appendix 10, wherein the congestion detection step detects congestion of the network based on the calculated value.
(Appendix 14)
14. The packet transfer control method according to appendix 13, wherein the bandwidth measurement step calculates the bandwidth of the network at the time of session connection or at a predetermined time interval.
(Appendix 15)
15. The packet transfer control method according to any one of appendices 10 to 14, wherein the media data includes at least one of audio data, video data, and audio.
(Appendix 16)
A computer-readable recording medium recorded with a packet transfer control program for transferring a packet storing media data in response to a request from the terminal to a computer connected to the terminal via a network, the packet transfer control program comprising: To the computer,
A congestion detection procedure for detecting congestion of the network;
A transfer control procedure for notifying at least one of the partner terminal and the own terminal of a request to reduce the rate of the media data when congestion of the network is detected;
A computer-readable recording medium for executing
 110 SIPサーバ
 130 IMS網
 140 インターネット網
 145 Webサーバ
 150 モバイルネットワーク
 170_1 携帯端末
 176 パケット転送部
 188 転送制御部
 190、190A パケット転送制御装置
 191 PCRF装置
 194 eNodeB装置
 205 帯域計測部
 200、210 輻輳検出部
 211 制御部
 230 レート制御部
 250 パケット送受信部
 253 音声コーデック
 254 レート設定部
 255 遅延差分判別部
 この出願は、2012年11月29日に出願された日本特許出願第2012−260636号を基礎とする優先権を主張し、その開示の全てをここに取り込む。
DESCRIPTION OF SYMBOLS 110 SIP server 130 IMS network 140 Internet network 145 Web server 150 Mobile network 170_1 Portable terminal 176 Packet transfer part 188 Transfer control part 190, 190A Packet transfer control apparatus 191 PCRF apparatus 194 eNodeB apparatus 205 Bandwidth measurement part 200, 210 Congestion detection part 211 Control unit 230 Rate control unit 250 Packet transmission / reception unit 253 Audio codec 254 Rate setting unit 255 Delay difference determination unit This application is based on Japanese Patent Application No. 2012-260636 filed on November 29, 2012 And the entire disclosure is incorporated herein.

Claims (10)

  1.  ネットワークを介して端末を接続し、前記端末からの要求によりメディアデータを格納したパケットを転送するパケット転送制御装置であって、
     前記ネットワークの輻輳を検出する輻輳検出部と、
     前記輻輳検出部が前記ネットワークの輻輳を検出した場合に、前記メディアデータに対しレートを削減する要求を相手端末および自端末の少なくとも一方に通知する転送制御部と、
    を備えることを特徴とするパケット転送制御装置。
    A packet transfer control device for connecting a terminal via a network and transferring a packet storing media data in response to a request from the terminal,
    A congestion detection unit for detecting congestion of the network;
    When the congestion detection unit detects congestion of the network, a transfer control unit that notifies a request to reduce the rate of the media data to at least one of the partner terminal and the own terminal;
    A packet transfer control device comprising:
  2.  前記輻輳検出部は、上り方向のパケットから輻輳情報を抽出して、前記ネットワークの輻輳を検出する、ことを特徴とする請求項1に記載のパケット転送制御装置。 2. The packet transfer control device according to claim 1, wherein the congestion detection unit extracts congestion information from an upstream packet and detects congestion of the network.
  3.  前記輻輳検出部は、前記上り方向のパケットのIPヘッダ部のECNフィールドをチェックすることにより、前記輻輳情報を抽出する、ことを特徴とする請求項2に記載のパケット転送制御装置。 3. The packet transfer control device according to claim 2, wherein the congestion detection unit extracts the congestion information by checking an ECN field of an IP header portion of the uplink packet.
  4.  前記パケット転送制御装置から送出したパケットに対する前記端末からの返信応答信号に基づいて、前記ネットワークの帯域を算出値として算出する帯域計測部を更に備え、
     前記輻輳検出部は、前記算出値に基づいて前記ネットワークの輻輳を検出する、ことを特徴とする請求項1に記載のパケット転送制御装置。
    Based on a reply response signal from the terminal to the packet sent from the packet transfer control device, further comprising a bandwidth measuring unit that calculates the bandwidth of the network as a calculated value,
    The packet transfer control device according to claim 1, wherein the congestion detection unit detects congestion of the network based on the calculated value.
  5.  前記帯域計測部は、セッション接続時又は予め定められた時間間隔毎に、前記ネットワークの帯域を算出する、ことを特徴とする請求項4に記載のパケット転送制御装置。 The packet transfer control device according to claim 4, wherein the bandwidth measuring unit calculates the bandwidth of the network at the time of session connection or at predetermined time intervals.
  6.  前記メディアデータが、音声データ、映像データ、及びオーディオの少なくとも一つからなる、請求項1乃至5のいずれか1つに記載のパケット転送制御装置。 The packet transfer control device according to any one of claims 1 to 5, wherein the media data includes at least one of audio data, video data, and audio.
  7.  請求項1乃至6のいずれか1つに記載のパケット転送制御装置と、前記ネットワークを介して前記パケット転送制御装置に接続された前記端末と、を含む通信システム。 A communication system including the packet transfer control device according to any one of claims 1 to 6 and the terminal connected to the packet transfer control device via the network.
  8.  前記輻輳検出部が前記ネットワークの輻輳を検出した場合に、前記端末に対するQoSパラメータを変更して、変更したQoSパラメータと変更フラグとを生成するPCRF装置を更に含み、
     前記パケット転送制御装置は、前記変更したQoSパラメータと前記変更フラグとに応答して、レートを削減する指示を作成するレート制御部を更に備え、
     前記転送制御部は、前記レートを削減する指示に応答して、前記メディアデータに対し前記レートを削減する要求を相手端末および自端末の少なくとも一方に通知する、請求項7に記載の通信システム。
    A PCRF device for changing the QoS parameter for the terminal and generating a changed QoS parameter and a change flag when the congestion detection unit detects congestion of the network;
    The packet transfer control device further includes a rate control unit that creates an instruction to reduce a rate in response to the changed QoS parameter and the change flag,
    The communication system according to claim 7, wherein the transfer control unit notifies a request for reducing the rate of the media data to at least one of a partner terminal and a local terminal in response to an instruction to reduce the rate.
  9.  前記QoSパラメータは、QCI、ARP、MBR及びGBRの少なくとも一つである、請求項8に記載の通信システム。 The communication system according to claim 8, wherein the QoS parameter is at least one of QCI, ARP, MBR, and GBR.
  10.  ネットワークを介して端末を接続し、前記端末からの要求によりメディアデータを格納したパケットを転送するパケット転送制御装置におけるパケット転送制御方法であって、
     前記ネットワークの輻輳を検出する輻輳検出ステップと、
     前記ネットワークの輻輳を検出した場合に、前記メディアデータに対しレートを削減する要求を相手端末および自端末の少なくとも一方に通知する転送制御ステップと、
    を含むパケット転送制御方法。
    A packet transfer control method in a packet transfer control device for connecting a terminal via a network and transferring a packet storing media data in response to a request from the terminal,
    A congestion detection step of detecting congestion of the network;
    A transfer control step of notifying at least one of the partner terminal and the own terminal of a request to reduce the rate of the media data when congestion of the network is detected;
    A packet transfer control method including:
PCT/JP2013/078399 2012-11-29 2013-10-11 Packet transfer control device and communications system WO2014083961A1 (en)

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Citations (4)

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Publication number Priority date Publication date Assignee Title
JPH10247944A (en) * 1997-03-05 1998-09-14 Kokusai Denshin Denwa Co Ltd <Kdd> Relay controller and its method
JP2006121166A (en) * 2004-10-19 2006-05-11 Ntt Comware Corp Voice quality control system and method, band control device, and computer program
WO2012077701A1 (en) * 2010-12-07 2012-06-14 日本電気株式会社 Gateway device and voice communication method
JP2012222380A (en) * 2011-04-04 2012-11-12 Nec Corp Access network system, gateway device, and network quality ensuring method

Patent Citations (5)

* Cited by examiner, † Cited by third party
Publication number Priority date Publication date Assignee Title
JPH10247944A (en) * 1997-03-05 1998-09-14 Kokusai Denshin Denwa Co Ltd <Kdd> Relay controller and its method
US6052734A (en) * 1997-03-05 2000-04-18 Kokusai Denshin Denwa Kabushiki Kaisha Method and apparatus for dynamic data rate control over a packet-switched network
JP2006121166A (en) * 2004-10-19 2006-05-11 Ntt Comware Corp Voice quality control system and method, band control device, and computer program
WO2012077701A1 (en) * 2010-12-07 2012-06-14 日本電気株式会社 Gateway device and voice communication method
JP2012222380A (en) * 2011-04-04 2012-11-12 Nec Corp Access network system, gateway device, and network quality ensuring method

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