WO2015160223A1 - Procédé et appareil pour permettre un service d'appels dans une situation de surcharge de système de communications mobiles à échange de paquets - Google Patents

Procédé et appareil pour permettre un service d'appels dans une situation de surcharge de système de communications mobiles à échange de paquets Download PDF

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Publication number
WO2015160223A1
WO2015160223A1 PCT/KR2015/003887 KR2015003887W WO2015160223A1 WO 2015160223 A1 WO2015160223 A1 WO 2015160223A1 KR 2015003887 W KR2015003887 W KR 2015003887W WO 2015160223 A1 WO2015160223 A1 WO 2015160223A1
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terminal
bit rate
call service
information
bitrate
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PCT/KR2015/003887
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English (en)
Korean (ko)
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정경훈
정경인
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삼성전자 주식회사
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    • H—ELECTRICITY
    • H04—ELECTRIC COMMUNICATION TECHNIQUE
    • H04W—WIRELESS COMMUNICATION NETWORKS
    • H04W28/00—Network traffic management; Network resource management
    • H04W28/16—Central resource management; Negotiation of resources or communication parameters, e.g. negotiating bandwidth or QoS [Quality of Service]
    • H04W28/18—Negotiating wireless communication parameters
    • H04W28/22—Negotiating communication rate
    • H—ELECTRICITY
    • H04—ELECTRIC COMMUNICATION TECHNIQUE
    • H04L—TRANSMISSION OF DIGITAL INFORMATION, e.g. TELEGRAPHIC COMMUNICATION
    • H04L47/00—Traffic control in data switching networks
    • H04L47/10—Flow control; Congestion control
    • H04L47/26—Flow control; Congestion control using explicit feedback to the source, e.g. choke packets
    • H04L47/263—Rate modification at the source after receiving feedback
    • H—ELECTRICITY
    • H04—ELECTRIC COMMUNICATION TECHNIQUE
    • H04W—WIRELESS COMMUNICATION NETWORKS
    • H04W28/00—Network traffic management; Network resource management
    • H04W28/02—Traffic management, e.g. flow control or congestion control
    • H04W28/10—Flow control between communication endpoints
    • H04W28/12—Flow control between communication endpoints using signalling between network elements

Definitions

  • the present invention relates to a mobile communication system, and more particularly, to a method and apparatus for a packet switched mobile communication system that allows a call service in a network overload situation.
  • LTE Long Term Evolution
  • IP Internet Protocol
  • CSFB circuit switched fallback
  • the CSFB scheme does not provide voice services based on LTE. That is, in the CSFB scheme, data communication is provided using an LTE network, and when a voice service is required, the CSFB scheme is switched to 3G mode (FallBack) to provide a voice service through an existing circuit-based network.
  • 3G mode FeallBack
  • VoLTE Voice over Internet Protocol
  • the present disclosure provides a specific method for adjusting a bit rate to be applied to the terminal in a packet-switched mobile communication system (eg, LTE system).
  • a packet-switched mobile communication system eg, LTE system
  • the present disclosure provides a base station apparatus for supporting a bit rate control of the terminal by periodically transmitting a message including the maximum bit rate information to be applied to the call service and the connection timeout information of the terminal in a packet-switched mobile communication system to provide.
  • the present disclosure provides an originating terminal for performing a bit rate control to be applied to a call service based on the information included in the periodic message in a packet-switched mobile communication system.
  • the present disclosure provides a destination terminal for performing a bit rate control to be applied to a call service on the basis of the information in the session negotiation proposal message including the bit rate to be applied to the call service from the other terminal in the packet-switched mobile communication system to provide.
  • the present disclosure provides a bitrate control method for a call service of an originating terminal in a packet-switched mobile communication system, comprising a periodic message, including a maximum bitrate information that the originating terminal can apply to a call service from a base station.
  • Receiving determining a bitrate to be applied to a call service based on the maximum bitrate information included in the periodic message, and transmitting a session negotiation proposal message including the determined bitrate to another terminal.
  • a control method is included.
  • the present disclosure provides a bitrate control method for a call service of a called terminal in a packet-switched mobile communication system, the periodic message including the maximum bitrate information that the called terminal can apply to the call service from the base station
  • a control method including determining a bit rate to be applied to a call service based on the bit rate to be applied to, and transmitting a session negotiation response message including the determined bit rate to the other terminal.
  • the present disclosure provides an originating terminal for controlling a bit rate for a call service in a packet-switched mobile communication system, wherein the originating terminal includes, from a base station, maximum bit rate information that the calling terminal can apply to a call service. And a transceiver for receiving a periodic message and transmitting a session negotiation proposal message including the determined bit rate to another terminal, and a controller for determining a bit rate to be applied to a call service based on the information included in the periodic message. It provides a calling terminal.
  • the present disclosure provides a destination terminal for controlling a bit rate for a call service in a packet-switched mobile communication system, wherein the destination terminal includes, from a base station, the maximum bit rate information that the destination terminal can apply to the call service;
  • a transceiver for receiving a periodic message, receiving a session negotiation proposal message including a bit rate to be applied to a call service from another terminal, and transmitting a session negotiation response message including the determined bit rate to the other terminal; It provides a called terminal comprising a control unit for determining the bit rate to be applied to the call service based on the information included in the periodic message and the session negotiation proposal message.
  • the present disclosure also provides a base station apparatus supporting bit rate control for a call service of a terminal in a packet switched mobile communication system, wherein the base station is connected to the terminal and the maximum bit rate information that the terminal can apply to a call service.
  • a base station apparatus including a control unit configured to configure a message including time-out information and to periodically transmit the configured message to at least one terminal is provided.
  • the terminal may directly adjust the bit rate of the voice or video call in a situation where the network is overloaded.
  • GSM Global System for Mobile communication
  • FIG. 2 is an exemplary diagram of an operation of adjusting call quality and network capacity in a WCDMA mobile communication system which is an example of a third generation communication system;
  • FIG. 3 is an exemplary diagram of an operation of adjusting call quality and network capacity in an LTE communication system that is an example of a fourth generation communication system;
  • FIG. 4 is a diagram illustrating a packet-switched mobile communication network to which a service-specific access control operation is applied;
  • FIG. 5 is a diagram illustrating a service-specific access control method
  • FIG. 6 is a diagram illustrating a communication bit rate control method of a base station and a terminal according to an embodiment of the present disclosure
  • FIG. 7 is a diagram illustrating session negotiation proposal operation of a call originating terminal according to an embodiment of the present disclosure
  • FIG. 8 illustrates a session negotiation response operation of a called terminal according to an embodiment of the present disclosure
  • FIG. 9 is an exemplary configuration diagram of a base station apparatus according to an embodiment of the present disclosure.
  • FIG. 10 is an exemplary configuration diagram of a terminal device according to an embodiment of the present disclosure.
  • Generations of mobile communication technologies can be classified according to data transmission speeds.
  • the transmission speed determines the transmission target (voice, text, video, multimedia, etc.).
  • the division of households is determined by the International Telecommunication Union (ITU) under the United Nations.
  • the first generation (1G) refers to the era of analog communication where only voice calls can be made.
  • the analog mobile communication system uses analog frequency modulation (FM) for voice transmission and frequency shift keying (FSK) for signal transmission.
  • FM analog frequency modulation
  • FSK frequency shift keying
  • the second generation (2G) refers to digital mobile phones, and can transmit data such as text messages and e-mails in addition to voice calls.
  • the second generation mobile communication supports the transmission of still images due to the low data transmission speed, but does not provide the transmission of moving images.
  • the third generation (3G) was defined by the ITU as a service that provides data rates of 144kbps to 2Mbps and video, and the CDMA (Code Division Multiple Access) 2000-1x EV DO, WCDMA (Wideband Code Division Multiple Access), etc. Belongs to the generation.
  • users can enjoy multimedia data such as text, voice, and moving pictures at a speed and quality similar to those of the current wired Internet.
  • LTE Long Term Evolution
  • ITU adopted LTE-advanced and WiMax-evolution as international standards for 4G mobile communication.
  • the present disclosure has been described using a VoLTE system as an example, the present invention can be applied to other wireless communication systems that can adjust the bit rate without any addition or subtraction.
  • the voice bit rate of all terminals connected to the base stations of the various cells controlled by the base station controller is controlled in real time to collectively control call quality and Network capacity can be adjusted accordingly.
  • the base station or the base station controller may adjust the number of more terminals to a lower quality by lowering the voice bit rate of the terminal.
  • the base station or the base station controller may use a flexible network operation strategy such as maintaining a voice bit rate of the terminal in the call and lowering the voice bit rate of the newly accepted call to allow more calls.
  • the base station or base station controller lowers the voice bit rate of the terminal to reduce the signal-to-noise ratio of each bit rate at a limited radio frequency (RF) output of the terminal.
  • RF radio frequency
  • SNR signal to noise ratio
  • a circuit-switched mobile communication network such as a 2nd and 3rd generation communication system
  • compression and restoration of voice call are performed in both a wireless section of an originating terminal and a wireless section of a terminating terminal.
  • the centralized bitrate management by the base station or the base station controller has an advantage in that it can quickly respond to a sudden change in call demand.
  • FIG. 1 illustrates an operation of adjusting call quality and network capacity in a Global System for Mobile communication (GSM) mobile communication system, which is an example of second generation communication.
  • GSM Global System for Mobile communication
  • a mobile station (MS) 100 a mobile station (MS) 100, a base transceiver station (BTS) 102, a base station controller (BSC) 104, and a telephone network are connected to the MS.
  • MS Mobile switching centers
  • PSTN public switched telephone networks
  • MS 100 may transmit voice data to public switched telephone network (PSTN) 108 through base station 102, base station controller 104, and mobile switching center 106, and from public switched telephone network 108 Voice data may also be received.
  • PSTN public switched telephone network
  • the MS 100 may encode voice data and transmit the encoded voice data to the base station 102, and the encoded data may be decoded by the base station 102 and then transferred to the public switched telephone network 108 ( 110).
  • the MS 100 may receive the encoded voice data transmitted from the public switched telephone network 108 through the base station 102, decode it, and restore the voice signal.
  • the base station controller 104 may adjust the transmission rate to control the call quality of the MS 100 in consideration of the network capacity (115). For example, in a time division multiple access (TDMA) based GSM system, the base station controller 104 may assign a full rate (FR) channel or an HR (half rate) channel to the MS 100. You can also control call quality by assigning.
  • the FR channel is a channel having a transmission rate of 22.8 kbps
  • the HR is a channel having a transmission rate of 11.4 Kbps.
  • the base station 102 adapts the bit rate of the voice to the channel situation within a range allowed by the channel allocated to each MS 100.
  • the control operation that is, codec mode control (CMC) or codec mode request (CMR) may be exchanged (120).
  • bit rate control may be performed in a CDMA2000 system.
  • a code division multiple access (CDMA) based CDMA2000 system a voice codec such as enhanced variable rate codec (EVRC) is used, and the base station controller 104 is connected to each MS 100 connected to an affiliated base station 102.
  • Call quality and network capacity may be indirectly adjusted by transmitting 'RATE_REDUC', which is a parameter indicating an upward average bitrate, to the MS 100 in order to adjust the voice average bitrate.
  • FIG. 2 illustrates an operation of adjusting call quality and network capacity in a WCDMA mobile communication system, which is an example of a third generation communication system.
  • a user equipment (UE) 200 In the 3rd generation mobile communication system, a user equipment (UE) 200, a base station (NodeB) 202, a radio network controller (RNC) 204, and a telephone network are connected to the UE 200.
  • Mobile switching center (MSC) 206 and public switched telephone networks (PSTN) 208 that provide circuit switching services.
  • the UE 200 may transmit voice data to a public switched telephone network (PSTN) 208 via a base station 202, a wireless network controller 204, and a mobile switching center 206, and the public switched telephone network 208. Voice data may also be received from.
  • PSTN public switched telephone network
  • the UE 200 encodes voice data and transmits the encoded voice data to the mobile switching center 206, and the coded data is decoded at the mobile switching center 206 and then transferred to the public switched telephone network 208. It may be 210.
  • the UE 200 may receive the encoded voice data transmitted from the public switched telephone network 208 through the mobile switching center 206, decode, and restore the encoded voice data.
  • the wireless network controller 204 may adjust the transmission rate to control the call quality of the UE 200 in consideration of network capacity (220).
  • a voice codec such as AMR, AMR-WB (wideband) may be used, and the wireless network controller 204 may be configured to perform the operation of each UE 200 connected to an affiliated base station 202.
  • the wireless network controller 204 adjusts the bit rate of the voice to fit the channel situation within a range allowed by the channel allocated to each UE 200, that is, codec mode control (CMC: codec mode). control) command can be sent.
  • CMC codec mode control
  • FIG. 3 illustrates an operation of adjusting call quality and network capacity in an LTE communication system, which is an example of a fourth generation communication system.
  • a user equipment (UE) 300 a user equipment (UE) 300, a base station (eNodeB) 302, a serving gateway (S-GW) 304, a packet data network gateway (P-GW) 306, and an IMS IP multimedia subsystem (IMS) 308 may be included.
  • UE user equipment
  • eNodeB base station
  • S-GW serving gateway
  • P-GW packet data network gateway
  • IMS IMS IP multimedia subsystem
  • a packet-switched mobile communication network compression of a speech signal using a specific codec is performed at the calling terminal 300, and the compressed speech signal is delivered to the called terminal through the IMS network 308, and then the compressed at the called terminal.
  • the restored voice signal is restored (310). That is, since the compressed voice signal passes through the outgoing terminal side wireless section and the destination terminal side wireless section, it is not possible to adjust the voice bit rate in consideration of the channel condition of a specific wireless section. Therefore, in a packet switched (PS) mobile communication network (eg, an LTE system), the base station eNodeB cannot directly adjust the voice bit rate of each UE.
  • PS packet switched
  • the voice bitrate is controlled between terminals by using a codec mode request (CMR) command of a real-time transport protocol (RTP) payload header or a real-time transport control protocol application defined (RTCP-APP) message. It may be requested directly (310).
  • CMR codec mode request
  • RTP real-time transport protocol
  • RTCP-APP real-time transport control protocol application defined
  • the base station 302 can indirectly control the call quality by assigning a radio resource to a packet (for example, a voice packet) to be transmitted as soon as possible through scheduling, ECN function, or SIB2 transmission procedure ( 320).
  • a radio resource for example, a voice packet
  • the base station 302 may warn the UE 300 in advance of an overload of the network or a deterioration of the connection state with the UE 300 through an explicit congestion notification (ECN) function.
  • ECN is a function that the base station 302 signals the overload to the UE 300 and, for example, marking in the 2-bit area of the IP packet header transmitted by the base station 302 to the UE 300 (ie, ECN-CE). It can be implemented through.
  • ECN-CE is a function that the base station 302 signals the overload to the UE 300 and, for example, marking in the 2-bit area of the IP packet header transmitted by the base station 302 to the UE 300 (ie, ECN-CE). It can be implemented through.
  • the terminal 300 may start a procedure for avoiding degradation of a call quality such as lowering a voice bit rate.
  • the base station 302 probabilistically permits the application of a voice or video call of the terminal 300 in an overload situation in which call requests increase rapidly through a system information block (SIB) type 2 periodically broadcasted or Deferred control, that is, service specific access control (SSAC) may be performed.
  • SIB system information block
  • SSAC service specific access control
  • FIG. 4 is a diagram illustrating a service-specific access control operation in a packet-switched mobile communication network.
  • a packet switched mobile communication network such as LTE may include one or more base stations 400 and 420 connected to the IMS 440 and one or more terminals 410 and 415 serviced by the base station 400.
  • the one or more base stations may include a digital unit (DU) 420, which is a base station responsible for signal processing in a plurality of base stations, and a small base station remote radio head (RRH) 400 that extends the coverage of the DU 420. Can be.
  • the packet-switched mobile communication network may further include an S-GW and a P-GW 430 that provide connectivity between the terminal and the IMS network 440.
  • the base station 400 may periodically broadcast information necessary for the terminals 410 and 415 to access the base station 400 through SIB messages of a radio resource control (RRC) protocol.
  • SIB can be defined from Type 2 to 16, each SIB Type is configured to be received by all the terminals in the cell through the appropriate period and transmission method.
  • SSAC which is a function of probabilistically accepting a voice and video call request of a terminal, may use SIB Type2 (ie, SIB2). By receiving the SIB2, the UE may voluntarily defer the voice or video call request for a specified period as in the following procedure.
  • Table 1 is an example of the SIB2 message structure that the terminal 410 receives from the base station 400.
  • terminal A 410 is currently making a voice call through base station A 400, and terminal B 415 is attempting a video or voice call.
  • SIB2 includes information necessary for the terminal to access the cell, in particular, ssac-BarringForMMTEL-Video-r9, which is restriction information for video call, ssac-BarringForMMTEL-Voice--r9, restriction information for voice call, connection restriction At least one of the element ac-BarringFactor, and the connection time-out ac-BarringTime (unit: seconds) may be included.
  • 5 is a diagram illustrating a connection control method according to SSAC.
  • the terminal may periodically receive information necessary for accessing the base station from the base station through an SIB message (eg, an SIB2 message). If the terminal intends to attempt a video call or a voice call, SSAC may proceed as follows.
  • SIB message eg, an SIB2 message
  • step 500 the terminal generates a random number (rand1) greater than or equal to 0 and less than 1. (0 ⁇ rand1 ⁇ 1)
  • step 505 the terminal compares the generated random number (rand1) with the ac-BarringFactor of the information contained in the SIB2 received from the base station.
  • the terminal may immediately try to connect with another terminal.
  • the terminal If the random number rand1 generated by the terminal is greater than or equal to the ac-BarringFactor, the terminal generates another random number rand2 greater than or equal to 0 and less than 1 in step 515. (0 ⁇ rand2 ⁇ 1)
  • step 520 the UE calculates ((0.7 + 0.6 * rand2) * ac-BarringTime) using ac-BarringTime among the information included in the SIB2 received from the base station, and calls for the same time as the calculated value. Defer the connection. The terminal may proceed to step 500 again.
  • Tables 2 and 3 illustrate session negotiation proposal messages sent by an originating terminal of a call in a packet-switched mobile communication network, which is a fourth generation communication system.
  • the terminal A supports the H.264 video codec and has a maximum of 500kbps.
  • RTP real-time transport protocol
  • UDP user datagram protocol
  • IP internet protocol
  • RR RTCP receive report
  • the SDP Offer message which is a session negotiation proposal message transmitted from the terminal A to the terminal B, supports the two voice codecs of the AMR-WB and AMR, and is described relatively earlier in the SDP Offer message than the AMR.
  • AMR-WB supports a total of nine compression schemes between 6.6 and 23.85kbps
  • AMR or AMR-NB (narrow band)
  • AS is an integer value obtained by adding RTP, UDP, and IP header to one frame of compressed voice, and means the maximum bit rate required for a call.
  • B AS value of AMR 12.2kbps is 30.
  • the bit rate of the voice call and the video call may have various bit rates of tens to hundreds of kbps.
  • the SSAC method described with reference to FIG. 5 is suitable for 3G mobile communication, in which the bitrates of the voice call and the video call are fixed at 12.2kbs and 64kbps, respectively, so that bitrate adjustment is unnecessary.
  • the call connection is delayed as described in FIG. 5 to control call quality and network capacity. Rather than controlling the connection in step 520, attempting a call by lowering the bit rate of the call will use resources more efficiently. Accordingly, the present disclosure proposes a flexible access control scheme that enables a terminal to adjust a bit rate of a call in a packet switched mobile communication system.
  • Table 4 illustrates the structure of a SIB2 message according to an embodiment of the present disclosure.
  • Table 4 shows that ssac-MaxVoiceBitRate, ssac-MaxVideoBitRate, and ssac-BarringTime information have been added.
  • the SIB Type 2 message includes three fields, in particular, ssac-MaxVoiceBitRate, ssac-MaxVideoBitRate, and ssac-BarringTime, and the type of the fields is integer and the unit is kbps.
  • the terminal When the ssac-MaxVoiceBitRate or ssac-MaxVideoBitRate is received, the terminal performs call connection when the bit rates of the video codec and the audio codec to be used for the call service through the packet-switched mobile communication network are less than or equal to ssac-MaxVoiceBitRate and ssac-MaxVideoBitRate (kbps), respectively. Try. Otherwise, the terminal attempts to connect the call by lowering the bitrates of the video codec and the voice codec to ssac-MaxVoiceBitRate and ssac-MaxVideoBitRate (kbps), respectively, or attempts to connect a call having a bitrate exceeded ssac-. Can be limited (delayed) by BarringTime (seconds).
  • the time limit for the call connection may be calculated by combining ssac-BarringTime and a value randomly generated by the terminal.
  • the present embodiment may not be applied simultaneously when ssac-BarringForMMTEL-Voice-r9 or ssac-BarringForMMTEL-Video-r9 is used.
  • SSAC which is an existing technology for restricting network access
  • Some terminals may allow a normal quality of service.
  • FIG. 6 is a diagram illustrating a communication bit rate control method of a base station and a terminal according to an embodiment of the present disclosure.
  • User terminal A 410 and user terminal B 415 may receive the SIB2 message from the base station 400.
  • the base stations periodically transmitting the SIB2 message may also be different from each other.
  • only a plurality of base stations are represented as one base station 400 for convenience of description.
  • the user terminal A 410 (that is, the originating terminal) attempting the call may transmit an SDP Offer message, which is a session negotiation proposal message, to the user terminal B 415 (that is, the called terminal) through the base station 400 (610). , 615).
  • SDP Offer message which is a session negotiation proposal message
  • the user terminal B 415 may transmit a session negotiation response message SDP Answer message to the user terminal A 410 through the base station 400 (620 and 625).
  • a call session is established between the terminals 410 and 415 by providing and receiving an SDP offer message and an SDP answer message, and a call service is provided.
  • FIG. 7 is a diagram illustrating a session negotiation proposal operation of a call originating terminal according to an embodiment of the present disclosure.
  • the calling terminal may receive the SIB2 message broadcast from the base station.
  • the SIB2 message may include information about an allowable maximum bit rate described in Table 4 above.
  • the SIB2 message may include at least one information of ssac-MaxVoiceBitRate, which is the maximum allowable voice bitrate, ssac-MaxVideoBitRate, which is the maximum allowable video bitrate, and ssac-BarringTime, which is the time limit.
  • step 705 it is determined whether the allowable maximum bit rate is greater than the bit rate to be used by the terminal.
  • an SDP offer message including a codec of the bit rate to be used by the terminal may be transmitted to the counterpart terminal.
  • the terminal transmits an SDP offer message for suggesting a bit rate less than the allowed maximum bit rate to the counterpart terminal, or for a predetermined time limit.
  • the timeout may be the ssac-BarringTime, or may be calculated by combining the ssac-BarringTime and a value randomly generated by the terminal.
  • FIG. 8 is a diagram illustrating a session negotiation response operation of a called terminal according to an embodiment of the present disclosure.
  • the called terminal may receive the SIB2 message broadcast from the base station.
  • the SIB2 message may include information about an allowable maximum bit rate described in Table 4 above.
  • the SIB2 message may include at least one information of ssac-MaxVoiceBitRate, which is the maximum allowable voice bitrate, ssac-MaxVideoBitRate, which is the maximum allowable video bitrate, and ssac-BarringTime, which is the time limit.
  • step 805 the called terminal receives an SDP offer message, which is a session negotiation proposal message, from the calling terminal.
  • step 810 the called terminal compares the allowed maximum bit rate with the bit rate of the codec included in the SDP offer message.
  • the UE can transmit an SDP answer message including the codec included in the SDP offer message to the counterpart terminal in step 815.
  • the terminal transmits an SDP answer message including a codec less than or equal to the allowable maximum bit rate to the counterpart terminal.
  • the timeout may be the ssac-BarringTime, or may be calculated by combining the ssac-BarringTime and a value randomly generated by the terminal.
  • Table 5 is an example of an SDP Offer message for a voice call according to an embodiment of the present disclosure.
  • Table 5 illustrates an SDP Offer message in a network low load situation in which ssac-MaxVoiceBitRate is not received through the SIB2 message.
  • the terminal may transmit a Session Initiation Protocol (SIP) Invite message including the SDP offer messages of Table 5 to the counterpart terminal via a packet switched mobile communication system.
  • SIP Session Initiation Protocol
  • Table 6 is another example of an SDP Offer message for a voice call according to an embodiment of the present disclosure.
  • Table 6 exemplifies a case in which ssac-MaxVoiceBitRate is included in the SIB2 message received from the base station and its value is 30.
  • a call connection attempt may not be allowed during ssac-BarringTime (seconds), during which the outgoing The terminal cannot send a SIP Invite message including an SDP offer.
  • Tables 7 and 8 are examples of SDP Answer messages for voice calls according to one embodiment of the disclosure.
  • the called terminal is capable of supporting both AMR-WB and AMR, as shown in Table 7, the called terminal can accept a call request with an AMR having a bit rate of 30 or less limited by SIB2, and the maximum of AMR-WB as shown in Table 8 below.
  • a call connection attempt may not be allowed for ssac-BarringTime (seconds) if the received ssac-MaxVoiceBitRate value is less than the minimum bitrate value of AMR or AMR-WB.
  • the time for which the call connection attempt is not allowed may be calculated by combining the ssac-BarringTime with a value randomly generated by the called terminal.
  • Table 9 is an example of an SDP Offer message for a video call according to an embodiment of the present disclosure.
  • Table 9 illustrates an SDP Offer message in a network low load situation in which ssac-MaxVideoBitRate is not received through the SIB2 message.
  • the originating terminal may transmit a Session Initiation Protocol (SIP) Invite message including the SDP offer message of Table 9 to the counterpart terminal through a packet switched telecommunication system.
  • SIP Session Initiation Protocol
  • an SDP offer message including the corresponding video bitrate may be transmitted.
  • Table 10 is another example of an SDP Offer message for a video call according to an embodiment of the present disclosure.
  • Table 10 exemplifies a case where ssac-MaxVideoBitRate is included in the SIB2 message received from the base station and its value is 400.
  • the call is not allowed for ssac-BarringTime seconds and no SIP Invite message can be sent during this period.
  • the time at which the message is not allowed to be sent may be calculated by combining the ssac-BarringTime with a value randomly generated by the calling terminal.
  • Tables 11 and 12 are examples of SDP offer message and SDP Answer message for video call according to one embodiment of the present disclosure, respectively.
  • FIG. 9 is an exemplary configuration diagram of a base station apparatus according to an embodiment of the present disclosure.
  • the base station apparatus 900 may include a transceiver 902 for transmitting and receiving a signal, data, or message with a terminal, and a controller 904 for controlling the transceiver 902.
  • the controller 904 may be interpreted as performing all operations related to bit rate control of the base station described above in the present disclosure. Specifically, for example, the controller 904 may control the base station apparatus 900 to transmit an SIB2 message including at least one of maximum allowed bitrate and connection attempt timeout information to at least one terminal. have.
  • transceiver 902 and the controller 904 are shown separately, but the transceiver 902 and the controller 904 may be implemented as one component.
  • FIG. 10 is an exemplary configuration diagram of a terminal device according to an embodiment of the present disclosure.
  • the terminal device 1000 may include a transceiver 1002 that transmits and receives a signal, data, or message with a base station and another terminal device, and a controller 1004 that controls the transceiver 1002.
  • the controller 1004 may be interpreted as performing all operations related to bit rate control of the terminal described above in the present disclosure. Specifically, for example, the controller 1004 may allow the terminal device 1000 to send an SDP offer message or message to the other terminal using at least one of allowed maximum bit rate and connection attempt timeout information included in the SIB2 message. You can control to send SDP answer messages.
  • transceiver 1002 and the controller 1004 are shown separately, but the transceiver 1002 and the controller 1004 may be implemented as one component.
  • FIGS. 1 to 10 an example of a bitrate control method, a configuration diagram of an apparatus, and the like are not intended to limit the scope of the present disclosure. That is, all components, or steps of the operations described in FIGS. 1 to 10 should not be interpreted as essential components for the implementation of the invention, and may include only some of the components within the scope that does not impair the essence of the invention. Can be implemented.
  • the above-described operations can be realized by providing a memory device storing the corresponding program code to an entity, a function, a base station, or any component in the terminal device of the communication system. That is, the controller of an entity, a function, a base station, or a terminal device can execute the above-described operations by reading and executing a program code stored in a memory device by a processor or a central processing unit (CPU).
  • a processor or a central processing unit (CPU).
  • the various components, modules, etc. of an entity, function, base station, or terminal device described herein may be a hardware circuit, for example, a complementary metal oxide semiconductor based logic circuit. And hardware circuitry such as firmware and a combination of software and / or hardware and software embedded in firmware and / or machine readable media.
  • various electrical structures and methods may be implemented using transistors, logic gates, and electrical circuits such as application specific semiconductors.

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  • Engineering & Computer Science (AREA)
  • Computer Networks & Wireless Communication (AREA)
  • Signal Processing (AREA)
  • Quality & Reliability (AREA)
  • Mobile Radio Communication Systems (AREA)

Abstract

L'invention concerne un procédé de régulation du débit binaire pour un service d'appels d'un terminal appelant dans un système de communications mobiles à échange de paquets, comportant les étapes consistant à: recevoir, en provenance d'une station de base, un message périodique comprenant des informations de débit binaire maximal susceptible d'être appliqué au service d'appels par le terminal appelant; déterminer un débit binaire à appliquer au service d'appels en se basant sur les informations de débit binaire maximal figurant dans le message périodique; et envoyer à un autre terminal un message de proposition de négociation de session comprenant le débit binaire déterminé.
PCT/KR2015/003887 2014-04-17 2015-04-17 Procédé et appareil pour permettre un service d'appels dans une situation de surcharge de système de communications mobiles à échange de paquets WO2015160223A1 (fr)

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KR1020140046150A KR102208795B1 (ko) 2014-04-17 2014-04-17 패킷 교환 이동통신 시스템의 과 부하 상황에서 통화 서비스를 허용하는 방법 및 장치
KR10-2014-0046150 2014-04-17

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EP3497965A4 (fr) * 2016-08-11 2020-03-25 Kyocera Corporation Adaptation du débit assistée par ran
WO2024036531A1 (fr) * 2022-08-17 2024-02-22 Nokia Shanghai Bell Co., Ltd. Adaptation de débit binaire de codec

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JP6893975B2 (ja) * 2016-09-30 2021-06-23 京セラ株式会社 モビリティのもとでのｒａｎアシストレート適応
US11611909B2 (en) 2018-09-07 2023-03-21 Apple Inc. Apparatus and method for signaling ran-assisted codec adaptation capabilities in IMS multimedia telephony sessions

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KR20030063474A (ko) * 2000-12-29 2003-07-28 노키아 코포레이션 비트율 결정
US20040102194A1 (en) * 2001-05-25 2004-05-27 Siamak Naghian Handover in cellular communication system
KR100501273B1 (ko) * 2003-02-26 2005-07-18 엘지전자 주식회사 이동통신 단말기의 음성/영상 모드 선택에 따른 화질개선방법
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EP3497965A4 (fr) * 2016-08-11 2020-03-25 Kyocera Corporation Adaptation du débit assistée par ran
WO2024036531A1 (fr) * 2022-08-17 2024-02-22 Nokia Shanghai Bell Co., Ltd. Adaptation de débit binaire de codec

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