WO2014087764A1 - Terminal and communication system - Google Patents

Terminal and communication system Download PDF

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Publication number
WO2014087764A1
WO2014087764A1 PCT/JP2013/079330 JP2013079330W WO2014087764A1 WO 2014087764 A1 WO2014087764 A1 WO 2014087764A1 JP 2013079330 W JP2013079330 W JP 2013079330W WO 2014087764 A1 WO2014087764 A1 WO 2014087764A1
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WIPO (PCT)
Prior art keywords
network
packet
rate
congestion
terminal
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PCT/JP2013/079330
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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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Priority to JP2012263956 priority Critical
Priority to JP2012-263956 priority
Application filed by 日本電気株式会社 filed Critical 日本電気株式会社
Publication of WO2014087764A1 publication Critical patent/WO2014087764A1/en

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    • HELECTRICITY
    • H04ELECTRIC COMMUNICATION TECHNIQUE
    • H04WWIRELESS COMMUNICATION NETWORKS
    • H04W28/00Network traffic or resource management
    • H04W28/02Traffic management, e.g. flow control or congestion control
    • H04W28/0289Congestion control
    • HELECTRICITY
    • H04ELECTRIC COMMUNICATION TECHNIQUE
    • H04LTRANSMISSION OF DIGITAL INFORMATION, e.g. TELEGRAPHIC COMMUNICATION
    • H04L47/00Traffic regulation in packet switching networks
    • H04L47/10Flow control or congestion control
    • H04L47/26Explicit feedback to the source, e.g. choke packet
    • H04L47/263Source rate modification after feedback
    • HELECTRICITY
    • H04ELECTRIC COMMUNICATION TECHNIQUE
    • H04WWIRELESS COMMUNICATION NETWORKS
    • H04W28/00Network traffic or resource management
    • H04W28/16Central resource management; Negotiation of resources or communication parameters, e.g. negotiating bandwidth or QoS [Quality of Service]
    • H04W28/18Negotiating wireless communication parameters
    • H04W28/22Negotiating communication rate

Abstract

In order to prevent any deterioration in quality of experience (QoE), this terminal, which is connected to a network and transmits and receives media-data-containing packets over said network, is provided with the following: a congestion detection unit that detects network congestion on the basis of received downlink packets; and a packet transmission/reception unit that, if the congestion detection unit has detected network congestion, uses the reverse direction of the network to send an opposing terminal a request to change the rate of the abovementioned media data.

Description

Terminal and communication system

The present invention relates to a terminal such as a portable terminal connected to a network, and more particularly to a terminal used in a communication system that performs congestion control of video, audio, and audio communication using packets.

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.

JP 2004-320452 A ([0051] to [0057])

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.
Note that Patent Document 1 merely discloses a packet transfer control device that selects one or more IP flows having a hop count value lower than a threshold value when the backbone line is congested. That is, Patent Document 1 does not recognize the above-described problem relating to the deterioration of QoE, and does not disclose any specific configuration of a terminal such as a portable terminal.
An object of the present invention is to provide a terminal capable of avoiding QoE degradation.

One aspect of the present invention is a terminal that is connected to a network and transmits / receives a packet storing media data via the network, and detects congestion of the network based on a received downstream packet. And a packet transmission / reception unit that, when the congestion detection unit detects congestion of the network, notifies a counterpart terminal of a request to change the rate of the media data using the reverse direction of the network. It is characterized by providing.

According to the present invention, it is possible to avoid a congestion state even when the traffic changes significantly compared to the statistical value.

FIG. 1 is a block diagram showing a connection configuration of a communication system to which the present invention is applied.
FIG. 2 is a block diagram showing the configuration of the mobile terminal according to the first embodiment of the present invention used in the communication system shown in FIG.
FIG. 3 is a block diagram showing a configuration of a mobile terminal according to the second embodiment of the present invention used in the communication system shown in FIG.

Hereinafter, embodiments and operations of the present invention will be described in detail with reference to the drawings.
FIG. 1 is a block diagram showing a configuration of a communication system to which the present invention is applied. 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 to connect to a partner terminal (not shown) via a mobile network 150 and an IMS (IP Multimedia Subsystem) network 130 via a partner network (not shown). (Not shown) and VoIP (Voice Over IP) voice communication.
Note that the same configuration can be adopted for a videophone that exchanges video and audio with a partner terminal, but the description in that case is omitted here.
In the communication system of FIG. 1, as an example of congestion detection, a configuration in which the mobile terminal 170 detects congestion by receiving congestion information by ECN (Explicit Connection Notification) for downlink packets received from the mobile network 150 is shown. . Here, when the outdoor LTE radio base station apparatus (eNodeB apparatus) 194 detects a congestion state in the wireless network, the ECN of the IP (Internet protocol) header portion of the downlink packet that the eNodeB apparatus 194 sends to the mobile terminal 170 It is assumed that the mobile terminal 170 is in a congested state by setting a CE (Congestion Experience) bit in the field.
In the communication system of FIG. 1, when the mobile terminal 170 sends out the IP address and RTP (real-time transport protocol) port number of the destination terminal as a voice call connection request, the connection request is sent to the eNodeB device 194 and the packet. The data is transferred to at least one of a SIP (Session Initiation Protocol) server 110 and a PCRF (Policy and Charging Rules Function) 191 that are arranged in an IMS (IP Multimedia Subsystem) network 130 via the transfer control device 190. Further, the mobile terminal 170 adds at least one parameter such as voice call traffic, desired QoS class, MBR (Maximum Bit Rate), GBR (Guaranteed Bit Rate) to the connection request, and the packet transfer control device 190. It is also possible to notify at least one of the SIP server 110 and the PCRF device 191 via the route.
The SIP server 110 receives a connection request signal for a voice call and sends a connection request 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 transmits the Ack signal to the mobile terminal 170 via the packet transfer control device 190 and the eNodeB device 194. When the mobile terminal 170 receives this Ack signal, control signals for voice call are exchanged. Here, not only the IP address and RTP port number of the mobile terminal 170 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 be transmitted in addition to the Ack signal. These parameters can be transmitted not only to the SIP server 110 but also to the PCRF apparatus 191.
The PCRF device 191 inputs the voice call traffic, the IP address and port number of the mobile terminal 170 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 QoS parameters to the packet transfer control device 190. For the mobile terminal 170, since the traffic is a voice call, the values of the QoS parameters are specifically, 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, the above-described parameter values are used on the assumption that the mobile terminal 170 uses an AMR-NB (Adaptive Multi-Rate Narrowband) audio codec. For details of the AMR-NB audio codec, for example, the 3GPP TS26.090 standard can be referred to, and the description thereof is omitted here.
As another audio codec, an AMR-WB (Adaptive Multi-Rate Wide band) audio codec can also be used. In this case, the value of GBR can be changed. For details of the AMR-WB audio codec, for example, the 3GPP TS26.190 standard can be referred to, and the description thereof is omitted here.
The packet transfer control device 190 relays the control signal from the mobile terminal 170 to the SIP server 110 and relays the control signal and the Ack signal from the SIP server 110 to the mobile terminal 170. The packet transfer control device 190 inputs at least one of four types of QoS parameters, QCI, ARP, MBR, and GBR, for each traffic data from the PCRF device 191. That is, the packet transfer control device 190 receives at least one of the four types of QoS parameters for the uplink direction and the downlink direction of the voice call traffic, and at least one of the four types of QoS parameters for the downlink direction of the download data traffic. , Input from the RCRF device 191, and performs uplink and downlink packet transfer control according to the set value of the QoS parameter.
[First Embodiment]
Next, the configuration of the mobile terminal 170 according to the first embodiment of the present invention will be described with reference to FIG. FIG. 2 is a block diagram illustrating a configuration of the mobile terminal 170. Here, the counterpart terminal also has the same configuration as that shown in FIG.
As shown in FIG. 2, the portable terminal 170 includes a packet receiver 250, a packet transmitter 251, a voice decoder 253, a rate setting unit 254, a congestion detector 255, and a voice encoder 256. .
In FIG. 2, the packet receiving unit 250 first receives a downlink packet transmitted from the eNodeB apparatus 194 in FIG. Then, the packet receiving unit 250 extracts information on the IP header portion, information on the payload header portion, and payload data from the received packet. The packet receiving unit 250 sends the information of the IP header part to the congestion detection unit 255, sends the information of the payload header part to the rate setting unit 254, and sends the payload data to the audio decoder 253. Here, it is assumed that an RTP / UDP (user datagram protocol) / IP packet is used as the protocol of the received packet.
The congestion detection unit 255 inputs the information of the IP header portion of the downstream packet, and checks the ECN (Explicit Connection Notification) field of the IP header portion. When the CE bit is set in the ECN field, the congestion detection unit 255 indicates that the downlink network from the eNodeB device 194 to the portable terminal 170 or the downlink network from the packet transfer control device 190 to the portable terminal 170 is congested. It detects that there is, and sends down congestion detection information to the rate setting unit 254.
When the downstream congestion detection information is input from the congestion detection unit 255, the rate setting unit 254 sets the changed rate in order to change the rate of the speech encoder of the counterpart terminal. Specifically, when the rate setting before the rate change is 12.2 kbps, the rate setting unit 254 changes the rate to 6.7 kbps after detecting congestion. Then, in order to request the changed rate from the counterpart terminal using an uplink packet that is the reverse direction, the rate setting unit 254 sets the rate after the change in the CMR (Codec Mode Request) field of the payload header of the uplink packet. To the packet transmission unit 251.
Further, the rate setting unit 254 inputs the payload header information extracted from the downstream packet in the packet receiving unit 250 from the packet receiving unit 250, and checks the CMR field in the payload header information. If the rate specified in the CMR field has been changed, a request for rate change has been received from the counterpart terminal to the voice encoder of the mobile terminal 170 via a downstream packet. The rate setting unit 254 changes the rate to the changed value designated in the CMR field, and sends the changed rate to the speech encoder 256.
The audio decoder 253 inputs the payload data from the packet receiving unit 250, operates the audio decoder, inputs the audio compression-encoded bitstream included in the payload data, decodes it, and reproduces it by decoding Output audio signals. Here, as described above, an AMR-NB decoder is used as the audio decoder.
The voice encoder 256 inputs the changed rate from the rate setting unit 254, and inputs the rate based on the changed rate when the rate is changed, or based on the previous rate when the rate is not changed. A bit stream obtained by compressing and encoding the audio signal is sent to the packet transmission unit 251. Here, as described above, an AMR-NB encoder is used as the speech encoder.
The packet transmission unit 251 stores the compressed and encoded bit stream input from the audio encoder 256 in the payload portion of the transmission packet. Further, in order to request the changed rate from the counterpart terminal, the packet transmission unit 251 sets the changed rate value input from the rate setting unit 254 in the CMR field of the payload header portion of the transmission packet. The packet is sent to the eNodeB device 194. Here, as described above, the RTP / UDP / IP packet is used as the protocol of the transmission 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, the configuration in which the packet CMR field is used as a request to change the rate to the counterpart terminal has been described. However, as another configuration, an RTCP (RTP Control Protocol) packet or an RTCP-APP (APPLICATION SPECIFIC) is used. ) Packets can be used to describe rate values, rate change values, or CMR values. Furthermore, SIP (Session Initiation Protocol) or SDP (Session Description Protocol) can also be used as a request for changing the rate to the partner terminal. When using SDP, it can also be included in the parameters of the mode set, for example.
As the audio codec, in addition to AMR-NB, AMR-WB and other audio codecs operating at a plurality of bit rates can be used.
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, congestion detection uses ECN information, but other information can also be used.
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).
In addition, an IP network such as NGN (Next Generation Network) can be used instead of the 3G network. Further, instead of the eNodeB device 194, a W-LAN (Wireless Local Area Network) access point may be used.
The portable terminal 170 can be realized by a program executed by a computer. That is, the mobile terminal 170 may be configured by a packet transmission / reception control processor (not shown) and a storage device (not shown). The storage device stores a packet transmission / reception control program. In this case, the packet transmission / reception control processor performs the above-described packet transmission / reception control operation according to the packet transmission / reception 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 network congestion is detected at a terminal, a request to change the rate of at least one of video, audio, and audio is sent to the opposite terminal in the reverse direction of the network. Can be notified using. As a result, it is possible to avoid a congestion state, and there is an effect that it is possible to avoid deterioration of QoE (Quality of Experience) such as the sound being interrupted and the screen being frozen at the terminal. Furthermore, even if services such as high sound quality VoIP and high resolution TV phone are started using the packet communication path of the LTE / EPC system in the future, there is an effect that deterioration of QoE on the terminal side can be avoided.
[Second Embodiment]
Next, the configuration of a mobile terminal 170A according to the second embodiment of the present invention will be described with reference to FIG.
FIG. 3 shows a configuration in which portable terminal 170A estimates a network bandwidth and calculates a rate to be changed based on the estimated value when congestion is detected. In FIG. 3, the constituent elements having the same numbers as those in FIG. 2 perform the same operations as in FIG.
The mobile terminal 170A has the same configuration as the mobile terminal 170 shown in FIG. 2 except that a band estimation unit 257 is added and the operation of the rate setting unit is changed as will be described later. Therefore, the reference numeral 258 is attached to the rate setting unit.
When the output signal from the congestion detection unit 255 indicates that the downstream network congestion has been detected, the bandwidth estimation unit 257 is the nth immediately after the congestion detection in the packet reception unit 250 according to the following equation (1). For a packet received after the time, the delay time T (n) of the packet is measured.
T (n) = R (n) −S (n) (1)
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 bandwidth estimation unit 257 measures delay values for a plurality of subsequent consecutive packets, smooths them in the time direction, and then obtains the bandwidth W of the downstream network from the following equation (2). Is estimated.
W = D / ST (2)
Here, W is the estimated bandwidth of the network, D is the time smoothed value of the received data size, and ST is the time smoothed value of the delay time.
Then, the bandwidth estimation unit 257 sends the estimated bandwidth W to the rate setting unit 258.
When the downlink congestion detection information is input from the congestion detection unit 255, the rate setting unit 258 sets the changed rate in order to change the rate of the speech encoder of the counterpart terminal. Specifically, the rate setting unit 258 receives the bandwidth estimation value W of the downstream network from the bandwidth estimation unit 257, and the following equation (3) is selected from a plurality of rates supported by the voice codec. Select a rate that satisfies, and set this to the new rate.
B (i) <W (3)
Here, B (i) is the i-th rate among the N types of rates supported by the audio codec, and 1 <i <N. Here, N is 8 for the AMR-NB audio codec and 9 for the AMR-WB audio codec.
For example, when there are two types, 6.7 kbps and 4.75 kbps, as the rates satisfying Expression (3), the rate setting unit 258 selects the higher bit rate. Accordingly, here, the rate setting unit 258 selects 6.7 kbps.
Then, in order to request the changed rate from the counterpart terminal using an uplink packet that is the reverse direction, the rate setting unit 258 sets the rate after the change in the CMR (Codec Mode Request) field of the payload header of the uplink packet. To the packet transmission unit 251.
Further, the rate setting unit 258 inputs the payload header information extracted from the downlink packet in the packet receiving unit 250 from the packet receiving unit 250, and checks the CMR field in the payload header information. If the rate specified in the CMR field has been changed, a rate change request has been made to the voice encoder of the portable terminal 170A via the downstream packet from the counterpart terminal. The unit 258 changes the rate to the changed value specified in the CMR field, and sends the changed rate to the speech encoder 256.
This is the end of the description of the configuration of the second exemplary embodiment of the present invention, but various modifications are possible.
In the second embodiment, the configuration using the packet CMR field as a request for changing the rate to the counterpart terminal is shown. However, as another configuration, an RTCP (RTP Control Protocol) packet or an RTCP-APP (APPLICATION SPECIFIC) is used. ) Packets can be used to describe rate values, rate change values, or CMR values. Furthermore, SIP (Session Initiation Protocol) or SDP (Session Description Protocol) can also be used as a request for changing the rate to the partner terminal. When using SDP, it can also be included in the parameters of the mode set, for example.
As the audio codec, in addition to AMR-NB, AMR-WB and other audio codecs operating at a plurality of bit rates can be used.
Also, other methods can be used for the band estimation method and the rate changing method.
In the second embodiment, a case where a voice call is performed 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 second embodiment, ECN information is used to detect congestion, but other information can also be used.
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).
In addition, an IP network such as NGN (Next Generation Network) can be used instead of the 3G network. Further, instead of the eNodeB device 194, a W-LAN (Wireless Local Area Network) access point may be used.
The portable terminal 170A can be realized by a program executed by a computer. That is, the mobile terminal 170A may be composed of a packet transmission / reception control processor (not shown) and a storage device (not shown). The storage device stores a packet transmission / reception control program. In this case, the packet transmission / reception control processor performs the above-described packet transmission / reception control operation according to the packet transmission / reception 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 network congestion is detected at a terminal, a request to change the rate for at least one of video, audio, and audio is sent to the opposite terminal in the reverse direction of the network. Can be notified using. As a result, it is possible to avoid a congestion state, and there is an effect that it is possible to avoid deterioration of QoE (Quality of Experience) such as the sound being interrupted and the screen being frozen at the terminal. Furthermore, even if services such as high sound quality VoIP and high resolution TV phone are started using the packet communication path of the LTE / EPC system in the future, there is an effect that deterioration of QoE on the terminal side can be avoided.
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 terminal connected to a network for transmitting and receiving packets storing media data via the network;
A congestion detector that detects congestion of the network based on the received downstream packet;
A packet transmission / reception unit for notifying a request for changing the rate of the media data to the counterpart terminal using the reverse direction of the network when the congestion detection unit detects congestion of the network;
A terminal comprising:
(Appendix 2)
The terminal according to appendix 1, wherein the congestion detection unit extracts congestion information from the downlink packet and detects congestion of the network.
(Appendix 3)
The congestion detection unit extracts the congestion information by checking an ECN (Explicit Connection Notification) field of an IP (Internet Protocol) header portion of the downstream packet, wherein the congestion information is extracted. Terminal.
(Appendix 4)
The request for changing the rate includes SIP (Session Initiation Protocol) / SDP (Session Description Protocol), CMR (Codec Mode Protocol), RTCP (RTP Control Protocol), and RTCP-APP (RTP Protocol Control). The terminal according to any one of appendices 1 to 3, wherein one of the two is used.
(Appendix 5)
The packet transmitter / receiver
When the congestion detection unit detects congestion of the network, a bandwidth estimation unit that estimates a bandwidth of the network based on the received downstream packet;
A rate setting unit that calculates a rate to be changed based on the estimated network bandwidth;
Based on the calculated rate, a packet transmitter that sends a request to change the rate to the counterpart terminal;
The terminal according to any one of supplementary notes 1 to 4, further comprising:
(Appendix 6)
The terminal according to any one of appendices 1 to 5, wherein the media data includes at least one of video data, audio data, and audio data.
(Appendix 7)
A communication system including the terminal according to any one of supplementary notes 1 to 6 and a packet transfer control device connected to the terminal via the network.
(Appendix 8)
A packet transmission / reception control method in a terminal connected to a network and transmitting / receiving a packet storing media data via the network,
A congestion detection step of detecting congestion of the network based on the received downstream packet;
A packet transmission / reception step for notifying a request for changing the rate of the media data to the counterpart terminal using the reverse direction of the network when congestion of the network is detected;
Packet transmission / reception control method including
(Appendix 9)
9. The packet transmission / reception control method according to appendix 8, wherein the congestion detection step extracts congestion information from the downstream packet and detects congestion of the network.
(Appendix 10)
The packet transmission / reception control method according to claim 9, wherein the congestion detection step extracts the congestion information by checking an ECN (Explicit Connection Notification) field of an IP (Internet Protocol) header portion of the downlink packet. .
(Appendix 11)
The request for changing the rate includes SIP (Session Initiation Protocol) / SDP (Session Description Protocol), CMR (Codec Mode Protocol), RTCP (RTP Control Protocol), and RTCP-APP (RTP Protocol Control). The packet transmission / reception control method according to any one of appendices 8 to 10, characterized in that any one of them is used.
(Appendix 12)
The packet transmission / reception step includes:
A bandwidth estimation step of estimating the bandwidth of the network based on the received downstream packet when congestion of the network is detected;
A rate calculating step for calculating a rate to be changed based on the estimated network bandwidth;
A sending step of sending a request to change the rate to the counterpart terminal based on the calculated rate;
The packet transmission / reception control method according to any one of appendices 8 to 11 including:
(Appendix 13)
The packet transmission / reception control method according to any one of appendices 8 to 12, wherein the media data includes at least one of video data, audio data, and audio data.
(Appendix 14)
A computer-readable recording medium that records a packet transmission / reception control program that causes a computer that is a terminal connected to a network to transmit and receive packets storing media data via the network, the packet transmission / reception control program comprising: On the computer,
A congestion detection procedure for detecting congestion of the network based on the received downstream packet;
A packet transmission / reception procedure for notifying a counterpart terminal of a request to change the rate of the media data using the reverse direction of the network when congestion of the network is detected;
Recording medium that executes

DESCRIPTION OF SYMBOLS 110 SIP server 130 IMS network 150 Mobile network 170, 170A Portable terminal 190 Packet transfer control device 191 PCRF device 194 eNodeB device 250 Packet reception unit 251 Packet transmission unit 253 Speech decoder 254 Rate setting unit 255 Congestion detection unit 256 Speech encoder 257 Band estimation Part 258 Rate Setting Part This application claims priority based on Japanese Patent Application No. 2012-263958 filed on December 3, 2012, the entire disclosure of which is incorporated herein.

Claims (10)

  1. A terminal connected to a network for transmitting and receiving packets storing media data via the network;
    A congestion detector that detects congestion of the network based on the received downstream packet;
    A packet transmission / reception unit for notifying a request for changing the rate of the media data to the counterpart terminal using the reverse direction of the network when the congestion detection unit detects congestion of the network;
    A terminal comprising:
  2. The terminal according to claim 1, wherein the congestion detection unit extracts congestion information from the downlink packet and detects congestion of the network.
  3. The congestion detection unit extracts the congestion information by checking an ECN (Explicit Connection Notification) field of an IP (Internet Protocol) header portion of the downlink packet. Terminal.
  4. The request to change the rate is SIP (Session Initiation Protocol) / SDP (Session Description Protocol), CMR (Codec Mode Protocol), RTCP (RTP Control Protocol), and RTCP-APP (RTPProc) The terminal according to claim 1, wherein one of the terminals is used.
  5. The packet transmitter / receiver
    When the congestion detection unit detects congestion of the network, a bandwidth estimation unit that estimates a bandwidth of the network based on the received downstream packet;
    A rate setting unit that calculates a rate to be changed based on the estimated network bandwidth;
    Based on the calculated rate, a packet transmitter that sends a request to change the rate to the counterpart terminal;
    The terminal according to any one of claims 1 to 4, further comprising:
  6. The terminal according to any one of claims 1 to 5, wherein the media data includes at least one of video data, audio data, and audio data.
  7. A communication system including the terminal according to any one of claims 1 to 6 and a packet transfer control device connected to the terminal via the network.
  8. A packet transmission / reception control method in a terminal connected to a network and transmitting / receiving a packet storing media data via the network,
    A congestion detection step of detecting congestion of the network based on the received downstream packet;
    A packet transmission / reception step for notifying a partner terminal of a request to change the rate of the media data using the reverse direction of the network when congestion of the network is detected;
    Packet transmission / reception control method including
  9. The request to change the rate is SIP (Session Initiation Protocol) / SDP (Session Description Protocol), CMR (Codec Mode Protocol), RTCP (RTP Control Protocol), and RTCP-APP (RTPProc) The packet transmission / reception control method according to claim 8, wherein one of them is used.
  10. The packet transmission / reception step includes:
    Estimating the bandwidth of the network based on the received downstream packet when congestion of the network is detected;
    Calculating a rate to be changed based on the estimated network bandwidth;
    Sending a request to change the rate to the counterpart terminal based on the calculated rate;
    The packet transmission / reception control method according to claim 8 or 9, comprising:
PCT/JP2013/079330 2012-12-03 2013-10-23 Terminal and communication system WO2014087764A1 (en)

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Citations (4)

* Cited by examiner, † Cited by third party
Publication number Priority date Publication date Assignee Title
JPH10336626A (en) * 1997-05-30 1998-12-18 Nec Software Ltd Transfer method and transfer device for video data
WO2002025878A1 (en) * 2000-09-22 2002-03-28 Matsushita Electric Industrial Co., Ltd. Data transmitting/receiving method, transmitting device, receiving device, transmitting/receiving system, and program
JP2010533419A (en) * 2007-07-09 2010-10-21 テレフオンアクチーボラゲット エル エム エリクソン(パブル) Adaptive rate control in communication systems.
US20110170410A1 (en) * 2010-01-11 2011-07-14 Research In Motion Limited Explicit congestion notification based rate adaptation using binary marking in communication systems

Patent Citations (4)

* Cited by examiner, † Cited by third party
Publication number Priority date Publication date Assignee Title
JPH10336626A (en) * 1997-05-30 1998-12-18 Nec Software Ltd Transfer method and transfer device for video data
WO2002025878A1 (en) * 2000-09-22 2002-03-28 Matsushita Electric Industrial Co., Ltd. Data transmitting/receiving method, transmitting device, receiving device, transmitting/receiving system, and program
JP2010533419A (en) * 2007-07-09 2010-10-21 テレフオンアクチーボラゲット エル エム エリクソン(パブル) Adaptive rate control in communication systems.
US20110170410A1 (en) * 2010-01-11 2011-07-14 Research In Motion Limited Explicit congestion notification based rate adaptation using binary marking in communication systems

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