WO2017197635A1 - 分组域语音业务调度的方法和装置 - Google Patents

分组域语音业务调度的方法和装置 Download PDF

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Publication number
WO2017197635A1
WO2017197635A1 PCT/CN2016/082786 CN2016082786W WO2017197635A1 WO 2017197635 A1 WO2017197635 A1 WO 2017197635A1 CN 2016082786 W CN2016082786 W CN 2016082786W WO 2017197635 A1 WO2017197635 A1 WO 2017197635A1
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Prior art keywords
value
voice service
service
protocol
scheduling parameter
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PCT/CN2016/082786
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English (en)
French (fr)
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唐欣
杨颜博
李明
魏岳军
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华为技术有限公司
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Priority to JP2018560872A priority Critical patent/JP6807956B2/ja
Priority to PCT/CN2016/082786 priority patent/WO2017197635A1/zh
Priority to KR1020187036389A priority patent/KR102207467B1/ko
Priority to CN201680085030.4A priority patent/CN109076396B/zh
Priority to EP16902020.3A priority patent/EP3448083B1/en
Publication of WO2017197635A1 publication Critical patent/WO2017197635A1/zh
Priority to US16/195,321 priority patent/US11039341B2/en

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    • H—ELECTRICITY
    • H04—ELECTRIC COMMUNICATION TECHNIQUE
    • H04L—TRANSMISSION OF DIGITAL INFORMATION, e.g. TELEGRAPHIC COMMUNICATION
    • H04L65/00—Network arrangements, protocols or services for supporting real-time applications in data packet communication
    • H04L65/80—Responding to QoS
    • 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
    • H04L47/24—Traffic characterised by specific attributes, e.g. priority or QoS
    • H04L47/2416—Real-time traffic
    • H—ELECTRICITY
    • H04—ELECTRIC COMMUNICATION TECHNIQUE
    • H04L—TRANSMISSION OF DIGITAL INFORMATION, e.g. TELEGRAPHIC COMMUNICATION
    • H04L65/00—Network arrangements, protocols or services for supporting real-time applications in data packet communication
    • H04L65/1066—Session management
    • H04L65/1069—Session establishment or de-establishment
    • 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]
    • H—ELECTRICITY
    • H04—ELECTRIC COMMUNICATION TECHNIQUE
    • H04W—WIRELESS COMMUNICATION NETWORKS
    • H04W72/00—Local resource management
    • H04W72/50—Allocation or scheduling criteria for wireless resources
    • H04W72/54—Allocation or scheduling criteria for wireless resources based on quality criteria
    • H—ELECTRICITY
    • H04—ELECTRIC COMMUNICATION TECHNIQUE
    • H04W—WIRELESS COMMUNICATION NETWORKS
    • H04W72/00—Local resource management
    • H04W72/50—Allocation or scheduling criteria for wireless resources
    • H04W72/54—Allocation or scheduling criteria for wireless resources based on quality criteria
    • H04W72/543—Allocation or scheduling criteria for wireless resources based on quality criteria based on requested quality, e.g. QoS
    • H—ELECTRICITY
    • H04—ELECTRIC COMMUNICATION TECHNIQUE
    • H04W—WIRELESS COMMUNICATION NETWORKS
    • H04W80/00—Wireless network protocols or protocol adaptations to wireless operation
    • H04W80/02—Data link layer protocols

Definitions

  • Embodiments of the present invention relate to the field of wireless communication technologies, and, more particularly, to a method and apparatus for packet domain voice service scheduling.
  • LTE Long Term Evolution
  • the network side device can adjust parameters such as a modulation mode or a coding rate of the wireless link transmission through an Adaptive Modulation and Coding (AMC) technology to ensure the link. Transmission quality.
  • AMC Adaptive Modulation and Coding
  • the LTE-based voice (Voice Over LTE, VoLTE) service has its own characteristics.
  • the network side device generally generates VoLTE services periodically, for example, periodically generates a fixed-size voice packet every 20 ms, and then the maximum throughput rate of the Media Access Control (MAC) layer when transmitting the VoLTE service is fixed.
  • the adaptive modulation and coding method of the conventional general data service is used to adaptively modulate and encode the VoLTE service, since the maximum throughput of the MAC layer is fixed, the determined final used MCS is also fixed and unchanged. Therefore, the appropriate MCS cannot be selected in time, that is, the purpose of adaptive modulation and coding cannot be achieved, and the transmission quality of the link when the VoLTE service is transmitted cannot be guaranteed.
  • Embodiments of the present invention provide a method and apparatus for scheduling packet domain voice services, which can ensure transmission quality of a link.
  • the first aspect provides a packet domain voice service scheduling method, which includes: acquiring a first scheduling corresponding to each of the at least two protocol sublayers of the L2 voice service. At least one value of the parameter; a condition for determining a quality of service of the voice service; a condition to be satisfied according to a service quality of the voice service, and each of the at least two protocol sublayers of the L2 of the voice service And determining, by the at least one value of the corresponding first scheduling parameter, a value of the service scheduling parameter of the voice service; and scheduling the voice service according to a value of the service scheduling parameter of the voice service.
  • the embodiment of the present invention determines the value of the service scheduling parameter of the voice service according to the scheduling parameter of the L2 different protocol sublayer of the voice service and the condition that the service quality of the voice service needs to be satisfied, and schedules the voice service according to the value of the scheduling parameter of the voice service, so that Can guarantee the transmission quality of the link.
  • the voice service in the embodiment of the present invention may be a VoLTE service.
  • the value of the service scheduling parameter of the VoLTE service may be determined according to the condition that the service quality needs to be met, and the voice service in the VoLTE transmission process is scheduled to ensure Transmission quality of VoLTE services.
  • the at least two protocol sublayers in the embodiment of the present invention may include at least one of the following protocol sublayers: a Radio Link Control (RLC) layer, a Packet Data Convergence Protocol (PDCP) layer, and a MAC layer.
  • RLC Radio Link Control
  • PDCP Packet Data Convergence Protocol
  • MAC MAC
  • the first scheduling parameter in the embodiment of the present invention may be an MCS.
  • the MCS of the RLC layer, the PDCP layer, and the MAC layer may be considered, and then the appropriate MCS is selected to adaptively modulate and encode the transmitted voice service.
  • the first scheduling parameter in the embodiment of the present invention may be the number of retransmissions.
  • the number of retransmissions of the RLC layer, the PDCP layer, and the MAC layer may be considered, and then the appropriate number of retransmissions is selected to adaptively transmit the voice service. pass.
  • the first scheduling parameter in the embodiment of the present invention may be the time of dropping the timer, and may consider the time of the discarding timer of the RLC layer, the PDCP layer, and the MAC layer, and then select the appropriate discarding timer for the transmission time.
  • the voice service performs an adjustment of the adaptive PDCP buffer size.
  • the first scheduling parameter includes any one of the following scheduling parameters: a modulation and coding mode MCS, a retransmission times, and a time of dropping a timer; At least one value of the first scheduling parameter corresponding to each of the at least two protocol sublayers of the at least two protocol sublayers of the L2 includes: according to the other two scheduling parameters except the first scheduling parameter, Obtaining at least one value of a first scheduling parameter of the first protocol sublayer, where the first collaboration sublayer is at least two protocols of the L2 of the voice service.
  • One of the layers configuring or calculating at least one of the at least two protocol sublayers of the L2 of the voice service according to at least one value of the first scheduling parameter of the first protocol sublayer He protocol at least one value of the first scheduling parameter corresponding to the sub-layer.
  • the condition that the service quality of the voice service needs to be satisfied is a voice quality average opinion value of the voice service (Mean Opinion Score, MOS Optimum, wherein the condition that the service quality needs to be satisfied according to the voice service and at least one value of the first scheduling parameter corresponding to at least two protocol sublayers of the L2 of the voice service determine the voice service.
  • a voice quality average opinion value of the voice service Mean Opinion Score, MOS Optimum
  • the value of the service scheduling parameter includes: determining, according to at least one value of the first scheduling parameter of the first protocol sublayer, at least one throughput rate of the corresponding first protocol sublayer; determining that the first protocol sublayer is At least one throughput rate of the other protocol sub-layers corresponding to the one-to-one throughput rate; determining at least one of the at least one of the at least one throughput rate of the first protocol sub-layer and the at least one throughput rate of the other protocol sub-layer Voice quality MOS of the voice service; selecting the first voice quality MOS corresponding to the voice quality MOS of the at least one voice service Service scheduling parameter value as the value of the voice service parameters.
  • the value of the first scheduling parameter corresponding to the voice quality MOS optimal may be selected as the value of the service scheduling parameter of the voice service, so that the voice quality of the voice service transmission process may be ensured.
  • the first scheduling parameter corresponding to the MOS optimal voice quality of the VoLTE service may be selected as the value of the service scheduling parameter to schedule the voice service in the VoLTE transmission process to ensure the voice quality of the service transmission process.
  • the service quality of the voice service needs to be satisfied, that is, the throughput rate of the voice service is the largest, where the And determining, by the at least one value of the first scheduling parameter corresponding to the at least two protocol sub-layers of the voice service, the value of the service scheduling parameter of the voice service, according to the foregoing: At least one value of a first scheduling parameter of a protocol sub-layer determines at least one throughput rate of the corresponding first protocol sub-layer; and determines other protocols that are in one-to-one correspondence with at least one throughput rate of the first protocol sub-layer At least one throughput rate of the layer; determining at least one throughput rate of the corresponding voice service according to at least one throughput rate of the first protocol sublayer and at least one throughput rate of the other protocol sublayer; Selecting, as the service scheduling parameter of the voice service, a value of a first scheduling parameter corresponding to a maximum throughput rate among at least one throughput
  • the condition that the service quality of the voice service needs to be satisfied is that the voice service has the highest transmission rate, and Determining, according to the condition that the service quality of the voice service needs to be satisfied, and at least one value of the first scheduling parameter corresponding to the at least two protocol sublayers of the L2 of the voice service, determining a service scheduling parameter of the voice service.
  • the value includes: determining at least one value of the traffic scheduling parameter, wherein each value of the traffic scheduling parameter is a value from a first scheduling parameter of the first protocol sublayer and a corresponding first of each of the other protocol sublayers a maximum value selected from a value of the scheduling parameter; selecting a maximum value from at least one value of the traffic scheduling parameter as a value of a traffic scheduling parameter of the voice service.
  • the service quality of the voice service needs to be satisfied that the error rate of the voice service is the smallest, where the Determining the service quality of the voice service and the at least one value of the first scheduling parameter corresponding to the L2 at least two protocol sub-layers of the voice service, determining a value of the service scheduling parameter of the voice service, including: determining a service scheduling At least one value of the parameter, wherein each value of the traffic scheduling parameter is a value from a first scheduling parameter of the first protocol sublayer and a value of a first scheduling parameter of each of the other respective protocol sublayers a minimum value selected; a minimum value is selected from at least one value of the service scheduling parameter as a value of a traffic scheduling parameter of the voice service.
  • the first scheduling parameter is an MCS
  • the other three parameters, except the first scheduling parameter are used according to the foregoing three scheduling parameters.
  • the at least one value of the first scheduling parameter of the first protocol sublayer is obtained according to the threshold range of the number of retransmissions.
  • the TTIB switch may be bound according to the number of retransmissions within the threshold range of the number of retransmissions, the time of multiple discarding timers within the time threshold range of the discarding timer, and the transmission time interval.
  • the open/close state determines the number of fragments of the RLC layer.
  • the product of the state value of the TTIB switch and the number of retransmissions is denoted as A, and each number of fragments may be a discard timer.
  • the time is divided by A, or each fragment number can be obtained by dividing the time of the discard timer by the number of decimals that A retains a certain number of digits.
  • the fragment size of the RLC layer may be determined according to the data size of the PDCP layer buffer, the number of fragments of the RLC layer, and the header data size of the RLC layer.
  • the data size of the PDCP layer buffer divided by the number of fragments of the RLC layer is recorded as B, and the fragment size of the RLC layer may be obtained by adding the header data size of the RLC layer.
  • the first scheduling parameter includes any one of the following scheduling parameters: an MCS, a retransmission times, a discarding timer, and a PDCP buffer.
  • the data size the number of fragments of the RLC layer.
  • the first protocol sublayer is a radio link control RLC layer
  • the other protocol sublayer includes a packet data convergence protocol PDCP layer.
  • media access control MAC layer is a radio link control RLC layer
  • the first scheduling parameter in the embodiment of the present invention may be any parameter when the VoLTE service is transmitted.
  • the corresponding parameter values of the three layers of the RLC layer, the PDCP layer, and the MAC layer are estimated according to the method of the embodiment of the present invention, and the VoLTE service is transmitted by selecting an appropriate parameter, so that the link can be ensured. Transmission quality.
  • the first scheduling parameter of the PDCP layer may be obtained according to the first scheduling parameter corresponding to the RLC layer.
  • the throughput rate of the RLC layer can be obtained from the MCS of the RLC layer
  • the throughput rate of the PDCP layer can be obtained according to the throughput rate of the RLC layer and the transmission efficiency factor.
  • the throughput rate of the PDCP layer can be considered as a linear function of the MCS of the PDCP layer, so the MCS of the PDCP layer can be obtained from the throughput rate of the PDCP layer.
  • the MCS of the PDCP layer can be obtained according to the MCS of the RLC layer.
  • the number of retransmissions of the PDCP layer and the time of the discarding timer can be configured as independent parameters.
  • the PDCP layer can independently configure different retransmission times and discard timers.
  • the PDCP layer does not consider retransmission, and the number of retransmissions of the PDCP layer can be considered as 1.
  • the first scheduling parameter of the MAC layer may be estimated by using a Channel Quality Indicator (CQI).
  • CQI Channel Quality Indicator
  • the network side device may obtain the MCS of the MAC layer according to the estimation result of the channel quality.
  • the number of retransmissions of the MAC layer and the time of the discarding timer can be configured as independent parameters.
  • the number of retransmissions of the MAC layer may be the number of retransmissions of the Hybrid Automatic Repeat reQuest (HARQ) of the MAC layer.
  • HARQ Hybrid Automatic Repeat reQuest
  • the MAC layer can independently configure different retransmission times and discard timers.
  • the MAC layer does not actively discard the data packet, and the MAC layer discarding timer can be considered to be infinite.
  • the second aspect provides an apparatus for scheduling packet domain voice service, comprising: an acquiring unit, configured to acquire at least one of a first scheduling parameter corresponding to each of the at least two protocol sublayers of the L2 voice service, respectively. a first determining unit, configured to determine a condition that the service quality of the voice service needs to be met, and a second determining unit, configured to: meet a condition that the service quality of the voice service needs to be determined according to the first determining unit, and The at least one value of the first scheduling parameter corresponding to each of the at least two protocol sublayers of the at least two protocol sublayers of the L2 of the voice service that is obtained by the acquiring unit determines a value of a service scheduling parameter of the voice service, and a scheduling unit, And the method is configured to schedule the voice service according to a value of a service scheduling parameter of the voice service determined by the second determining unit.
  • the first scheduling parameter includes any one of the following scheduling parameters: a modulation and coding mode MCS, a retransmission times, and a time of dropping the timer;
  • the obtaining unit is configured to obtain a value in the threshold range corresponding to the first protocol sub-layer according to the other two scheduling parameters except the first scheduling parameter, to obtain the first protocol.
  • Calculating at least one value of a first scheduling parameter of the layer and configuring or calculating, according to at least one value of the first scheduling parameter of the first protocol sublayer, another protocol sublayer in at least two protocol sublayers of L2 of the voice service
  • the condition that the service quality of the voice service needs to be satisfied is that the voice quality average opinion value MOS of the voice service is optimal
  • the second determining unit is configured to determine, according to the at least one value of the first scheduling parameter of the first protocol sublayer, the at least one throughput rate of the corresponding first protocol sublayer, and determine the first protocol sublayer At least one throughput rate of the other protocol sub-layers corresponding to the at least one throughput rate, determining at least one of the at least one of the at least one throughput rate of the first protocol sub-layer and the at least one throughput rate of the other protocol sub-layer
  • the voice quality MOS of the voice service, and the value of the first scheduling parameter corresponding to the optimal voice quality MOS is selected from the voice quality MOS of the at least one voice service as the value of the service scheduling parameter of the voice service.
  • the condition that the service quality of the voice service needs to be satisfied is that the throughput rate of the voice service is the largest, and the second determining unit is specific. Determining, according to at least one value of the first scheduling parameter of the first protocol sublayer, determining at least one throughput rate of the corresponding first protocol sublayer, and determining at least one throughput rate of the first protocol sublayer At least one throughput rate of a corresponding other protocol sublayer, determining at least one throughput rate of the corresponding voice service according to at least one throughput rate of the first protocol sublayer and at least one throughput rate of the other protocol sublayer And selecting, according to at least one throughput rate of the voice service, a value of a first scheduling parameter corresponding to a maximum throughput rate as a value of a service scheduling parameter of the voice service.
  • the condition that the service quality of the voice service needs to be satisfied is that the transmission rate of the voice service is the largest, and the second determining unit is specific.
  • At least one value for determining a traffic scheduling parameter wherein each value of the traffic scheduling parameter is a value from a first scheduling parameter of the first protocol sublayer and a corresponding first scheduling parameter of another respective protocol sublayer. The selected maximum value of one value, and selecting a maximum value from at least one value of the service scheduling parameter as a value of a traffic scheduling parameter of the voice service.
  • the service quality of the voice service needs to be satisfied that the error rate of the voice service is the smallest
  • the second determining unit is Specifically, it is used to determine at least one value of a service scheduling parameter, where each value of the service scheduling parameter is a value from a first scheduling parameter of the first protocol sublayer and a first scheduling parameter of a corresponding other protocol sublayer A minimum value selected from a value, and a minimum value is selected from at least one value of the traffic scheduling parameter as a value of a traffic scheduling parameter of the voice service.
  • the first scheduling parameter is an MCS
  • the acquiring unit is specifically configured to use a threshold according to the number of retransmissions The plurality of retransmission times in the range, the time of the plurality of discarding timers in the threshold range of the time of the discarding timer, and the opening and closing state of the transmission time interval binding TTIB switch to determine at least the first protocol sublayer a number of fragments, where the number of fragments of the first protocol sublayer corresponds to the number of retransmissions and the time of one discarding timer, according to the maximum cacheable data size of the other protocol sublayers.
  • a slice size determines at least one MCS of the first protocol sublayer.
  • the first scheduling parameter includes any one of the following scheduling parameters: an MCS, a number of retransmissions, a time of dropping a timer, a data size of a PDCP buffer, and a number of fragments of the RLC layer.
  • the first protocol sublayer is a radio link control RLC layer
  • the other protocol sublayer includes a packet data convergence protocol PDCP layer.
  • media access control MAC layer is a radio link control RLC layer
  • the respective operations of the corresponding units and/or devices of the apparatus for scheduling packet voice service scheduling provided by the foregoing second aspect may refer to the respective steps of the method in the first aspect, and are not repeated here.
  • the beneficial technical effects brought by the various technical solutions of the second aspect can be referred to the technical effects in the method of the first aspect, and are not repeated.
  • FIG. 1 is a schematic flowchart of a method for packet domain voice service scheduling according to an embodiment of the present invention.
  • FIG. 2 is a block diagram of an apparatus for packet domain voice service scheduling in accordance with an embodiment of the present invention.
  • FIG. 3 is a block diagram of an apparatus for packet domain voice service scheduling according to another embodiment of the present invention.
  • the access system of the LTE radio access protocol is divided into three layers: layer 1 (L1) is the physical layer (Physical Layer, PHY), layer 2 (L2) is the MAC layer, the RLC layer and the PDCP layer, and the layer 3 ( L3) is a Radio Resource Control (RRC).
  • L1 is the physical layer (Physical Layer, PHY)
  • L2 is the MAC layer
  • RLC Radio Resource Control
  • L3 is a Radio Resource Control (RRC).
  • RRC Radio Resource Control
  • the physical layer is the lowest layer of the wireless access system. It uses the transport channel as the interface to provide services to the upper layer.
  • the PDCP layer is located above the RLC layer and is the top sublayer of L2.
  • the PDCP sublayer can be The packet data carrying the network layer on the air interface, such as the Internet Protocol (IP) data stream, can also handle the radio resource management (RRC) message of the control plane.
  • the PDCP sublayer can process the packet data carried on the interface, compress and encrypt the packet data, and then deliver it to the RLC sublayer.
  • the RLC sublayer is located above the MAC sublayer and is part of L2.
  • the RLC sublayer can provide segmentation and retransmission services for user and control data.
  • a higher-level Protocol Data Unit (PDU) packet of different lengths is segmented (transmitting) recombination (receiver) into a smaller RLC load unit.
  • the MAC layer defines the way in which data frames are transmitted over the medium.
  • FIG. 1 is a schematic flowchart of a method for packet domain voice service scheduling parameters according to an embodiment of the present invention.
  • the method of FIG. 1 may be performed by a network side device, for example, by an evolved base station (Evolved Node B, e-NodeB).
  • Evolved Node B evolved Node B
  • e-NodeB evolved base station
  • the embodiment of the present invention determines the value of the service scheduling parameter of the voice service according to the scheduling parameter of the L2 different protocol sublayer of the voice service and the condition that the service quality of the voice service needs to be satisfied, and schedules the voice service according to the value of the scheduling parameter of the voice service, so that Can guarantee the transmission quality of the link.
  • the embodiments of the present invention may be used in an LTE system.
  • the voice service may be a VoLTE service
  • each protocol sublayer of the L2 may include an RLC layer, a PDCP layer, and a MAC layer.
  • the first scheduling parameter in the embodiment of the present invention may be a parameter involved in a voice service transmission process.
  • the first scheduling parameter in the embodiment of the present invention may be any one of the following parameters: the MCS, the number of retransmissions, the time of the discarding timer, the data size of the PDCP buffer, and the number of fragments of the RLC layer.
  • the first scheduling parameter of each protocol sublayer of the voice service is one of the first scheduling parameters, that is, the first scheduling parameter of each protocol sublayer is the same parameter, but each protocol sublayer The value of the first scheduling parameter can be different.
  • the first scheduling parameter may be an MCS, and in this case, according to the service quality of the voice service.
  • the quantity needs to be satisfied and the value corresponding to the MCS of each L2 protocol sublayer, and the appropriate MCS is selected to adaptively modulate and encode the transmitted voice service.
  • the first scheduling parameter may be the number of retransmissions.
  • the number of retransmission times may be selected according to the condition that the quality of the voice service needs to be met and the number of retransmission times of the L2 protocol sublayers.
  • the voice service is adaptively retransmitted.
  • the first scheduling parameter may be the time for dropping the timer.
  • the appropriate discard timing may be selected according to the condition that the quality of the voice service needs to be met and the time corresponding to the discard timer of the L2 protocol sublayer.
  • the time of the device adjusts the adaptive PDCP buffer size for the transmitted voice traffic.
  • the first scheduling parameters of the other protocol sub-layers may respectively take corresponding values, and the value of the first scheduling parameter of the corresponding transmitted voice service may be determined.
  • the first scheduling parameter of the first protocol sub-layer takes a value within a certain range, multiple values of the first scheduling parameter of the corresponding transmission voice service may be obtained.
  • one of the multiple values of the first scheduling parameter of the transmitted voice service may be selected as the service scheduling parameter for determining the used voice service according to the condition that the service quality of the voice service needs to be satisfied, so as to schedule the voice service.
  • the first protocol sublayer may be one of at least two protocol sublayers of L2 of the voice service.
  • the condition that the service quality of the voice service needs to be satisfied in the embodiment of the present invention may be that the voice quality satisfies certain conditions, for example, the voice quality is optimal.
  • the voice quality can be reflected by the service quality MOS of the voice service, the throughput of the voice service, the transmission rate of the voice service, and the error rate of the voice service. Therefore, in the embodiment of the present invention, the service quality of the voice service in the step 102 needs to be satisfied, that is, the service quality MOS of the voice service is optimal, the throughput of the voice service is the largest, the transmission rate of the voice service is the largest, and the error rate of the voice service is the smallest.
  • it may be a combination of at least two of the above conditions.
  • the service quality of the voice service needs to be satisfied.
  • the maximum value can be selected from the multiple values of the first scheduling parameter of the voice service as the value of the service scheduling parameter of the voice service.
  • the service quality of the voice service needs to be satisfied, when the error rate of the voice service is the smallest, the smallest value may be selected from the multiple values of the first scheduling parameter of the voice service as the value of the service scheduling parameter of the voice service.
  • the throughput rate of the voice service or the voice quality MOS of the voice service may be determined by at least three parameters of the MCS, the number of retransmissions, and the time of the discarding timer.
  • the three parameters of the MCS, the number of retransmissions, and the time of the discard timer any two parameters When taking the corresponding value, the throughput or voice quality MOS can be considered as a function of the third parameter. Therefore, the value of the third parameter corresponding to the throughput rate or the voice quality MOS maximum can be found as the service scheduling parameter when actually transmitting the voice service.
  • the number of retransmissions and the time of the discarding timer may be corresponding values within a certain range.
  • the throughput of the voice service or the voice quality MOS may be regarded as MCS.
  • the function can use the MCS corresponding to the maximum throughput or the voice quality MOS as the service scheduling parameter of the voice service, and adaptively modulate and encode the voice service.
  • the specific implementation manners of the embodiments of the present invention are exemplified by taking the time of the first scheduling parameter as the MCS, the number of retransmissions, or the discarding timer as an example.
  • the voice service is a VoLTE service
  • at least two protocol sublayers of the L2 include an RLC layer, a PDCP layer, and a MAC layer
  • the first protocol sublayer is an RLC layer as an example.
  • the MCS is taken as an example for the first scheduling parameter as an example.
  • the network side device may obtain the value of the MCS of the RLC layer, the value of the MCS of the PDCP layer, and the value of the MCS of the MAC layer, and determine the actual transmission VoLTE service according to the value of the MCS of the layer 3 and the condition that the service quality of the voice service needs to be met.
  • the value of the final MCS is adopted, and finally the VoLTE service is scheduled according to the value of the final MCS.
  • the system needs to define different data transmission MCS formats.
  • the MCS format corresponds to various modulation orders and coding rates.
  • the system can select different MCS schemes for adaptive modulation and coding according to channel conditions. In order to adapt to the impact of channel changes, the transmission quality of the link is guaranteed in real time.
  • the number of retransmissions and the time of discarding the timer can be configured through the system or specify the respective range of values.
  • the specific mode of obtaining the value of the MCS of the RLC layer by the network side device is as follows.
  • the network side device can obtain the maximum bufferable data size of the PDCP layer by using the discard timer of the PDCP layer.
  • the time of the discarding timer can be configured by the network side device and sent to the user equipment.
  • the data size of the maximum bufferable (Buffer) of the PDCP layer can be used to calculate the fragment size of the RLC layer, and further determine the value of the MCS of the RLC layer.
  • the network side device may also estimate the size of the buffered data of the current PDCP layer by using a Buffer Status Report (BSR) reported by the User Equipment (UE), an MCS before the current time, and a VoLTE coding rate.
  • BSR Buffer Status Report
  • UE User Equipment
  • the RLC fragment size and the value of the MCS are in one-to-one correspondence. That is, by using the above parameters (PDCP Buffer status, number of transmissions, TTIB switch, etc.), the fragment size of the corresponding RLC layer can be determined, thereby determining the value of the MCS of the RLC layer.
  • At least one RB may be scheduled, and the number of RBs is not fixed.
  • the RB number and the MCS can jointly determine the slice size of the RLC layer.
  • the fragment size of the RLC layer can be obtained according to the above parameters (PDCP Buffer status, number of transmissions, TTIB switch, etc.) and the number of RBs, and then the value of the MCS of the RLC layer can be obtained from the fragment size and the number of RBs of the RLC layer.
  • the network side device can obtain the value of the MCS of the PDCP layer by determining the throughput rate of the RLC layer according to the number of RLC layer fragments and the fragment size.
  • the throughput rate of the RLC layer the fragment size of the RLC layer * the number of fragments of the RLC layer.
  • the throughput rate of the PDCP layer RLC layer throughput rate * transmission efficiency factor, and the transmission efficiency factor can be determined by the number of fragments of the RLC layer and the head overhead of the RLC layer.
  • the transmission efficiency factor the fragment size of the RLC layer / [(the fragment size of the RLC layer + the head overhead size of the RLC layer) * the number of fragments].
  • the throughput rate of the PDCP layer can be regarded as a linear function of the MCS of the PDCP layer, and the value of the MCS of the PDCP layer can be obtained according to the throughput rate of the PDCP layer.
  • the network device may obtain the value of the MCS of the MAC layer by using the following manner: the network side device may estimate the channel quality, and determine the value of the MCS of the MAC layer according to the estimation result of the channel quality and the channel quality threshold corresponding to each MCS.
  • the channel quality here may be a signal to noise ratio or a bit error rate.
  • the network side device can configure the maximum number of retransmissions and the maximum time for dropping the timer.
  • the number of retransmissions is different in the range of less than or equal to the maximum number of retransmissions, and the time of discarding the timer takes different values within a range less than or equal to the longest time, so that the RLC layer can be obtained according to the method for calculating the value of the MCS.
  • each MCS of the RLC layer may be selected from the value of one MCS of the RLC layer, and the value of one MCS of the corresponding PDCP layer and the value of one MCS of the MAC layer, for example, from One of the values of the MCS of the above three layers is selected as the value of one MCS of the corresponding voice service.
  • the condition that the service quality of the voice service needs to be satisfied may be that the service quality MOS of the voice service is optimal.
  • the voice quality MOS of the VoLTE service can be calculated by the MCS of the RLC layer, the MCS of the PDCP layer, and the MCS of the MAC layer, and the value of the MCS corresponding to the MOS optimal is selected as the value of the final MCS, and the VoLTE service is scheduled.
  • the voice quality of the VoLTE service can be guaranteed.
  • the voice quality MOS of the VoLTE service can be obtained by estimating the throughput rate of the RLC layer, the throughput rate of the PDCP layer, and the throughput rate of the MAC layer according to the MCS of the RLC layer, the MCS of the PDCP layer, and the MCS of the MAC layer, respectively.
  • the throughput rate and the MCS of this layer can be considered a linear relationship.
  • the minimum value is selected from the throughput rate of the RLC layer, the throughput rate of the PDCP layer, and the throughput rate of the MAC layer as the actual throughput rate at the time of VoLTE traffic transmission.
  • the MOS is determined based on the actual throughput rate.
  • MOS Const-a*max[1-actual throughput rate/(voice rate*DTX ratio), 0], where Const represents the maximum MOS score for the speech coding mode and speech rate, usually through offline Obtained after training a large number of sequences.
  • the throughput of the PDCP layer can be obtained according to the following manner: determining the transmission efficiency factor according to the number of fragments of the RLC layer and the header data overhead of the RLC layer, and multiplying the throughput rate of the RLC layer by the transmission efficiency factor to obtain the throughput of the PDCP layer. rate.
  • the throughput of the MAC layer can be estimated by estimating the channel quality, and the estimation result of the channel quality is obtained, and the MCS of the MAC layer is determined according to the estimation result and the threshold value of the channel quality corresponding to different MCSs.
  • the condition that the service quality of the voice service needs to be satisfied may be that the throughput of the voice service is the largest.
  • the actual throughput rate of the VoLTE service can be calculated by using the MCS of the RLC layer, the MCS of the PDCP layer, and the MCS of the MAC layer.
  • the MCS corresponding to the actual throughput rate can be selected as the final MCS value, and the VoLTE service is scheduled. In this way, the actual throughput rate of the voice transmission service can be maximized, thereby making the VoLTE service have better voice quality.
  • the maximum value may be selected from the values of the plurality of MCSs of the voice service determined by the value of the MCS of the three layers.
  • the value of the MCS as the traffic scheduling parameter of the final voice service.
  • the service quality of the voice service needs to be met when the error rate of the voice service is the smallest.
  • the smallest value is selected from the plurality of MCSs of the voice service determined by the value of the MCS of the above three layers as the value of the traffic scheduling parameter MCS of the voice service.
  • Different modulation modes have different characteristics. Low-order modulation adds more redundancy, resulting in lower actual efficiency, but can ensure higher reliability.
  • the final MCS takes the maximum or minimum of the MCS in the three layers, which is determined by the actual demand. For example, in order to ensure that the error rate of the voice service transmitted between the network side device and the user equipment is small, a smaller MCS may be selected. To ensure that the transmission rate of voice traffic between the network side device and the user equipment is the largest, a larger MCS can be selected.
  • the values of the three MCSs of the RLC layer, the PDCP layer, and the MAC layer of the L2 may determine the value of one MCS of the voice service.
  • the value of the service scheduling parameter MCS of the finally determined voice service may be the value of one MCS selected from the values of the plurality of MCSs of the voice service determined according to the above method, and each layer uses the value of the finally selected MCS. As a modulation and coding method for this layer.
  • the first scheduling parameter is used as an example for retransmission times as an example.
  • HARQ is a retransmission method of the MAC layer combining forward error correction coding (FEC) and automatic repeat request (ARQ).
  • FEC forward error correction coding
  • ARQ automatic repeat request
  • the key words of HARQ are storage, request retransmission, and combined demodulation.
  • the receiver saves the received data in the case of decoding failure, and requests the sender to retransmit the data, and the receiver combines the retransmitted data with the previously received data and decodes it. There is a certain diversity gain in it, which reduces the number of retransmissions and thus reduces the delay.
  • HARQ can efficiently compensate for the error caused by link adaptation, can improve the data transmission rate, and can reduce the data transmission delay.
  • the network side device can obtain the value range of the retransmission times of the RLC layer, the value range of the retransmission times of the PDCP layer of the packet data convergence protocol, and the value range of the retransmission times of the MAC layer, and according to the three layers and one-to-one correspondence
  • the value of the number of retransmissions of the voice service determines a value of the number of retransmissions of the voice service, and further determines the multiple values of the number of retransmissions of the voice service according to the value range of the number of retransmissions of each layer, and according to the service quality of the voice service.
  • the condition to be satisfied is determined by determining a value from the plurality of values of the number of retransmissions as the number of retransmissions of the final voice service used for actually transmitting the VoLTE service, and finally scheduling the VoLTE service according to the number of retransmissions of the final voice service.
  • the system can select different retransmission times according to channel conditions to adapt to the impact of channel changes. In this way, the adaptive retransmission adjustment of the VoLTE service transmission can be performed to ensure the transmission quality of the link in real time, the bit error rate can be reduced, and the data transmission rate can be improved.
  • the specific manner in which the network device obtains the number of retransmissions of the RLC layer is as follows.
  • the manner in which the network side device obtains the data size of the maximum bufferable (Buffer) of the PDCP layer and the size of the buffered data of the PDCP layer is the same as that in the first embodiment. To avoid repetition, details are not described herein again.
  • the network side device can configure the maximum MCS value and the maximum time of the discard timer.
  • the value of the MCS of the RLC layer may be in a range of values less than or equal to the maximum MCS, and the time of dropping the timer may be in a range less than or equal to the longest time, and may be based on the MCS and the drop timer.
  • the time determines the number of retransmissions corresponding to it. For example, when the number of resource blocks (RBs) is fixed, the value of the MCS of the RLC layer is in one-to-one correspondence with the fragment size of the RLC layer. When the value of the MCS of the RLC layer is given, the slice size of the RLC layer can be obtained.
  • the number of fragments corresponding to the number of fragments can be obtained.
  • the number of retransmissions is obtained according to the estimation method of the number of fragments in the first embodiment.
  • the number of retransmissions of the RLC layer, the PDCP layer, and the MAC layer in the embodiment of the present invention may be independently configured.
  • the value of the number of retransmissions may be set according to the delay.
  • multiple throughput rates or multiple voice quality MOSs for transmitting VoLTE services may be obtained according to the value of the number of retransmissions, and the weight that maximizes the throughput rate or the voice quality MOS is selected.
  • the value of the number of transmissions is taken as the value of the number of retransmissions of the final voice service.
  • the throughput rate of the VoLTE service may be calculated by the number of retransmissions of the RLC layer, the PDCP layer, and the MAC layer, and the number of retransmissions corresponding to the maximum throughput rate is determined to be the actual transmission VoLTE service. The number of retransmissions.
  • the voice quality MOS of the VoLTE service may be calculated by the number of retransmissions of the RLC layer, the PDCP layer, and the MAC layer, and the number of retransmissions corresponding to the voice quality is determined to be the actual transmission VoLTE. The number of retransmissions of the service.
  • the values of the three retransmission times of the RLC layer, the PDCP layer, and the MAC layer of the L2 may determine the value of one retransmission of the voice service. .
  • the value of the number of retransmissions of the finally determined voice service may be a value of one retransmission number selected from the values of the number of retransmission times of the voice service determined according to the above method, and each layer is selected by the final selection. The value of the number of retransmissions is retransmitted.
  • the condition for satisfying the service quality of the voice service is a voice service.
  • the value of the maximum number of retransmissions may be selected from the values of the number of retransmission times of the voice service determined in the range of the number of retransmissions of the RLC layer, the PDCP layer, and the MAC layer. The value of the number of retransmissions of the scheduled voice service that is ultimately used. In this way, multiple retransmissions of voice data can ensure the transmission quality of the link.
  • the value of the number of retransmissions of the RLC layer, the PDCP layer, and the MAC layer may be determined.
  • the value of the minimum number of retransmissions is selected as the value of the number of retransmissions of the scheduled used voice service. This can increase the transfer rate.
  • the time when the first scheduling parameter is the discarding timer is taken as an example for exemplary description.
  • the network measuring device can obtain the value range of the discarding timer of the RLC layer, the value range of the discarding timer of the PDCP layer, and the value range of the discarding timer of the MAC layer, and according to the third layer.
  • the time of the one-to-one corresponding three discarding timers determines a value of the time of the discarding timer of the voice service, and the time range of the discarding timer of the voice service can be determined according to the value range of the time of the discarding timer of each layer.
  • a value is determined from a plurality of values of the time of the discarding timer as a value of the time of the discarding timer of the final voice service used for actually transmitting the VoLTE service, and finally according to The value of the time of the final discard timer is scheduled for the VoLTE service.
  • the system can select different discard timers according to channel conditions to adapt to the effects of channel changes. In this way, the size of the data to be buffered of the VoLTE service can be adaptively adjusted to ensure the transmission quality of the link in real time.
  • the time when the network device acquires the discard timer of the RLC layer is as follows.
  • the manner in which the network side device obtains the maximum storable data size of the PDCP layer and the size of the cached data of the PDCP layer is the same as that in the first embodiment. To avoid repetition, details are not described herein again.
  • the network side device may configure the value of the maximum MCS and the value of the maximum number of retransmissions.
  • the value of the MCS of the RLC layer is less than or equal to the value of the maximum MCS, and the number of retransmissions is less than or equal to the maximum number of retransmissions
  • the value of the MCS and the number of retransmissions may be determined according to the value of the MCS and the number of retransmissions.
  • the value of the time of the corresponding drop timer For example, since the value of the MCS of the RLC layer is in one-to-one correspondence with the slice size of the RLC layer.
  • the slice size of the RLC layer can be obtained.
  • the relationship between the size of the slice and the number of slices can be obtained from the size of the number of slices.
  • the time of discarding the timer is obtained according to the estimation method of the number of fragments in the first embodiment.
  • the values of the time of the drop timer of the RLC layer, the PDCP layer, and the MAC layer layer 3 in the embodiment of the present invention may be independently configured.
  • the value of the time for discarding the timer may be set according to the delay.
  • the throughput or voice quality MOS of the transmitted VoLTE service may be obtained according to the time of the discard timer, and the discard rate is maximized or the voice quality MOS is optimally discarded.
  • the value of the time of the timer is taken as the value of the time of the final discard timer.
  • the throughput rate of the VoLTE service may be calculated by the time of the drop timer of the RLC layer, the PDCP layer, and the MAC layer, and the value of the time of the drop timer corresponding to the maximum throughput rate is determined. The value of the time of the discard timer for actually transmitting the VoLTE service.
  • the voice quality MOS of the VoLTE service may be calculated by the time of the drop timer of the RLC layer, the PDCP layer, and the MAC layer layer 3, and the time of the discard timer corresponding to the voice quality MOS optimal may be selected.
  • the value of the value is determined as the value of the time at which the discard timer of the VoLTE service is actually transmitted.
  • the voice quality parameter is the time of the discarding timer
  • the value of the time of selecting the maximum or minimum discarding timer from the number of retransmissions in the RLC layer, the PDCP layer, and the MAC layer may be used as the final transmitted voice.
  • the value of the time of the service discard timer which in turn ensures the transmission quality of the link.
  • the foregoing three embodiments respectively use the first scheduling parameter as the MCS, the number of retransmissions, and the time of the discarding timer as an example.
  • the first scheduling parameter in the embodiment of the present invention may also be the size of the PDCP layer cache data.
  • Other parameters such as the number of fragments of the RLC layer.
  • Other parameters can be obtained according to the corresponding parameters of the RLC layer, the PDCP layer and the MAC layer, and the parameters used in the actual transmission of the voice service are adjusted, and then the parameters of the VoLTE service are adaptively adjusted to ensure the transmission quality of the link. .
  • the apparatus 10 for packet domain voice service scheduling of FIG. 2 may include an obtaining unit 11, a first determining unit 12, and a Two determining unit 13 and adjusting unit 14.
  • the obtaining unit 11 is configured to acquire at least one value of the first scheduling parameter corresponding to each of the at least two protocol sublayers of the L2 of the voice service.
  • the first determining unit 12 is configured to determine a condition that the quality of service of the voice service needs to be met.
  • the second determining unit 13 is configured to: the condition that the service quality of the voice service that is determined by the first determining unit is to be satisfied, and the first scheduling that is respectively corresponding to each of the at least two protocol sublayers of the L2 of the voice service acquired by the acquiring unit At least one value of the parameter determines the value of the traffic scheduling parameter of the voice service.
  • the adjusting unit 14 is configured to schedule the voice service according to the value of the service scheduling parameter of the voice service determined by the second determining unit.
  • the embodiment of the present invention determines the value of the service scheduling parameter of the voice service according to the scheduling parameter of the L2 different protocol sublayer of the voice service and the condition that the service quality of the voice service needs to be satisfied, and schedules the voice service according to the value of the scheduling parameter of the voice service, so that Can guarantee the transmission quality of the link.
  • the apparatus for packet domain voice service scheduling may correspond to the method of packet domain voice service scheduling in FIG. 1 of the embodiment of the present invention, and each unit/module and other operations and/or functions described above in the apparatus.
  • the corresponding processes performed by the method shown in FIG. 1 are respectively omitted for brevity.
  • the apparatus 20 includes a processor 21, a memory 22, and a bus system 23, the processor 21 and the memory 22 being coupled by a bus system 23 for storing instructions for executing instructions stored by the memory 22.
  • the apparatus 20 is caused to perform the various steps in the flow of the method of FIG.
  • the method disclosed in the foregoing embodiment of the present invention may be applied to the processor 21 or implemented by the processor 21.
  • each step of the above method may be completed by an integrated logic circuit of hardware in the processor 21 or an instruction in the form of software.
  • the processor 21 can be a general purpose processor, a digital signal processor, an application specific integrated circuit, a field programmable gate array or other programmable logic device, a discrete gate or transistor logic device, a discrete hardware component, and can be implemented or executed in an embodiment of the invention.
  • a general purpose processor can be a microprocessor or any conventional processor or the like.
  • the steps of the method disclosed in the embodiments of the present invention may be directly implemented as a hardware processor, or may be performed by a combination of hardware and software modules in the processor.
  • the software module can be located in a conventional storage medium such as random access memory, flash memory, read only memory, programmable read only memory or electrically erasable programmable memory, registers, and the like.
  • the storage medium is located The memory 22, the processor 21 reads the information in the memory 22, and completes the steps of the above method in combination with its hardware.
  • the processor 21 is configured to obtain at least one value of the first scheduling parameter corresponding to each of the at least two protocol sublayers of the L2 of the voice service, and determine a condition that the service quality of the voice service needs to be satisfied, according to the voice.
  • the service quality of the service needs to be satisfied, and at least one value of the first scheduling parameter corresponding to each of the at least two protocol sublayers of the L2 of the voice service determines the value of the service scheduling parameter of the voice service, and finally according to the voice service.
  • the value of the traffic scheduling parameters is scheduled for voice traffic.
  • the embodiment of the present invention determines the value of the service scheduling parameter of the voice service according to the scheduling parameter of the L2 different protocol sublayer of the voice service and the condition that the service quality of the voice service needs to be satisfied, and schedules the voice service according to the value of the scheduling parameter of the voice service, so that Can guarantee the transmission quality of the link.
  • the apparatus for packet domain voice service scheduling may correspond to the method of packet domain voice service scheduling in FIG. 1 of the embodiment of the present invention, and each unit/device and other operations and/or functions described above in the apparatus.
  • the corresponding processes performed by the method shown in FIG. 1 are respectively omitted for brevity.
  • RAM random access memory
  • ROM read only memory
  • EEPROM electrically programmable ROM
  • EEPly erasable programmable ROM registers
  • hard disk removable disk
  • CD-ROM computer-readable media

Abstract

本发明实施例提供了一种分组域语音业务调度的方法和装置。该方法包括获取语音业务的L2至少两个协议子层中每个协议子层分别对应的第一调度参数的至少一个值，确定语音业务的业务质量需要满足的条件，根据语音业务的业务质量需要满足的条件以及语音业务的L2至少两个协议子层中每个协议子层分别对应的第一调度参数的至少一个值确定所述语音业务的业务调度参数的值，根据语音业务的业务调度参数的值调度语音业务。这样在满足一定条件下适时调整业务调度参数，可以保证链路的传输质量。

Description

分组域语音业务调度的方法和装置 技术领域
本发明实施例涉及无线通信技术领域，并且更具体地，涉及分组域语音业务调度的方法和装置。
背景技术
随着技术的发展，长期演进(Long Term Evolution，LTE)提供的高带宽为移动带宽带来了蓬勃发展的机会。无线链路数据传输时如果整个传输过程中的参数固定，整个链路传输过程的实际情况可能不同，这样无法确保链路的传输质量。
现有技术中，针对LTE普通的数据业务，网络侧设备可以通过自适应调制与编码(Adaptive Modulation and Coding，AMC)技术调整无线链路传输的调制方式或编码速率等参数，以确保链路的传输质量。
同LTE普通的数据业务相比，基于LTE的语音(Voice Over LTE，VoLTE)业务有自身的特点。网络侧设备一般周期性地产生VoLTE业务，例如，每20ms周期性产生一个固定大小的语音包，那么，传输VoLTE业务时媒体接入控制(Media Access Control，MAC)层的最大吞吐率为定值。此时，如果使用传统的普通数据业务的自适应调制与编码的方法对VoLTE业务进行自适应调制与编码，由于MAC层最大吞吐率为定值，使得确定的最终使用的MCS也是固定没有改变的，这样便不能适时选择合适的MCS，即达不到自适应调制与编码的目的，无法保证传输VoLTE业务时链路的传输质量。
另外，现有技术中也没有针对传输VoLTE业务过程中其他参数，例如，重传次数和丢弃计时器的时间等参数，进行自适应调整的方法。这样，很难保证传输VoLTE业务时链路的传输质量。
发明内容
本发明实施例提供一种分组域语音业务调度的方法和装置，能够保证链路的传输质量。
第一方面，提供了一种分组域语音业务调度的方法，其特征在于，包括：获取语音业务的L2至少两个协议子层中每个协议子层分别对应的第一调度 参数的至少一个值；确定所述语音业务的业务质量需要满足的条件；根据所述语音业务的业务质量需要满足的条件以及所述语音业务的L2至少两个协议子层中每个协议子层分别对应的第一调度参数的至少一个值确定所述语音业务的业务调度参数的值；根据所述语音业务的业务调度参数的值调度所述语音业务。
本发明实施例根据语音业务的L2不同协议子层的调度参数和语音业务的业务质量需要满足的条件确定语音业务的业务调度参数的值，并根据语音业务的调度参数的值调度语音业务，这样能够保证链路的传输质量。
具体地，本发明实施例中的语音业务可以为VoLTE业务，这时可以根据业务质量需要满足的条件确定VoLTE业务的业务调度参数的值，并对VoLTE传输过程中的语音业务进行调度，以保证VoLTE业务的传输质量。
本发明实施例中的至少两个协议子层可以包括下列至少一个协议子层：无线链路控制(Radio Link Control，RLC)层、分组数据会聚协议(Packet Data Convergence Protocol，PDCP)层、MAC层。
本发明实施例中的第一调度参数可以为MCS，此时，可以通过考虑RLC层、PDCP层和MAC层三层的MCS，然后选择合适的MCS对传输的语音业务进行自适应调制与编码。
本发明实施例中的第一调度参数可以为重传次数，可以通过考虑RLC层、PDCP层和MAC层三层的重传次数，然后选择合适的重传次数对传输的语音业务进行自适应重传。
本发明实施例中的第一调度参数可以为丢弃计时器的时间，可以通过考虑RLC层、PDCP层和MAC层三层的丢弃计时器的时间，然后选择合适的丢弃计时器的时间对传输的语音业务进行自适应PDCP缓存大小的调整。
结合第一方面，在第一方面的一种实现方式中，所述第一调度参数包括以下任一种调度参数：调制编码方式MCS、重传次数和丢弃计时器的时间；所述获取语音业务的L2至少两个协议子层中每个协议子层分别对应的第一调度参数的至少一个值包括：根据上述三种调度参数中除所述第一调度参数外的另两种调度参数在第一协议子层分别对应的阈值范围内取值，得到所述第一协议子层的第一调度参数的至少一个值，所述第一协作子层为所述语音业务的L2至少两个协议子层中的一个；配置或根据所述第一协议子层的第一调度参数的至少一个值计算所述语音业务的L2至少两个协议子层中的其 他协议子层分别对应的第一调度参数的至少一个值。
结合第一方面及其上述实现方式，在第一方面的另一种实现方式中，所述语音业务的业务质量需要满足的条件为所述语音业务的语音质量平均意见值(Mean Opinion Score，MOS)最优，其中，所述根据所述语音业务的业务质量需要满足的条件以及所述语音业务的L2至少两个协议子层分别对应的第一调度参数的至少一个值确定所述语音业务的业务调度参数的值包括：根据所述第一协议子层的第一调度参数的至少一个值确定对应的所述第一协议子层的至少一个吞吐率；确定与所述第一协议子层的至少一个吞吐率一一对应的其他协议子层的至少一个吞吐率；根据所述第一协议子层的至少一个吞吐率和所述其他协议子层的至少一个吞吐率确定对应的至少一个所述语音业务的语音质量MOS；从所述至少一个所述语音业务的语音质量MOS中选择最优的语音质量MOS对应的第一调度参数的值作为所述语音业务的业务调度参数的值。
本发明的一个实施例中，可以选择语音质量MOS最优时对应的第一调度参数的值作为语音业务的业务调度参数的值，这样可以保证语音业务传输过程的语音质量。例如，对于VoLTE业务，可以选择使得VoLTE业务的语音质量的MOS最优时对应的第一调度参数作为业务调度参数的值对VoLTE传输过程中的语音业务进行调度，以保证业务传输过程的语音质量。
结合第一方面及其上述实现方式，在第一方面的另一种实现方式中，所述语音业务的业务质量需要满足的条件为所述语音业务的吞吐率最大，其中，所述根据所述语音业务的业务质量需要满足的条件以及所述语音业务的L2至少两个协议子层分别对应的第一调度参数的至少一个值确定所述语音业务的业务调度参数的值包括：根据所述第一协议子层的第一调度参数的至少一个值确定对应的所述第一协议子层的至少一个吞吐率；确定与所述第一协议子层的至少一个吞吐率一一对应的其他协议子层的至少一个吞吐率；根据所述第一协议子层的至少一个吞吐率和所述其他协议子层的至少一个吞吐率确定对应的所述语音业务的至少一个吞吐率；从所述所述语音业务的至少一个吞吐率中选择最大的吞吐率对应的第一调度参数的值作为所述语音业务的业务调度参数的值。
结合第一方面及其上述实现方式，在第一方面的另一种实现方式中，所述语音业务的业务质量需要满足的条件为所述语音业务的传输速率最大，其 中，所述根据所述语音业务的业务质量需要满足的条件以及所述语音业务的L2至少两个协议子层分别对应的第一调度参数的至少一个值确定所述语音业务的业务调度参数的值包括：确定业务调度参数的至少一个值，其中，业务调度参数的每个值为从所述第一协议子层的第一调度参数的一个值和对应的其他每个协议子层的第一调度参数的一个值中选择的最大值；从所述业务调度参数的至少一个值中选择最大值作为所述语音业务的业务调度参数的值。
结合第一方面及其上述实现方式，在第一方面的另一种实现方式中，所述语音业务的业务质量需要满足的条件为所述语音业务的误码率最小，其中，所述根据所述语音业务的业务质量需要满足的条件以及所述语音业务的L2至少两个协议子层分别对应的第一调度参数的至少一个值确定所述语音业务的业务调度参数的值包括：确定业务调度参数的至少一个值，其中，业务调度参数的每个值为从所述第一协议子层的第一调度参数的一个值和对应的其他每个协议子层的第一调度参数的一个值中选择的最小值；从所述业务调度参数的至少一个值中选择最小值作为所述语音业务的业务调度参数的值。
结合第一方面及其上述实现方式，在第一方面的另一种实现方式中，所述第一调度参数为MCS，所述根据上述三种调度参数中除所述第一调度参数外的另两种调度参数在第一协议子层分别对应的阈值范围内取值，得到所述第一协议子层的第一调度参数的至少一个值包括：根据在所述重传次数的阈值范围内的多个重传次数、所述丢弃计时器的时间的阈值范围内的多个丢弃计时器的时间、传输时间间隔绑定TTIB开关的开闭状态确定所述第一协议子层的至少一个分片数，其中，所述第一协议子层的每个分片数与一个重传次数和一个丢弃计时器的时间相对应；根据所述其他协议子层最大可缓存的数据大小、所述第一协议子层的至少一个分片数和所述第一协议子层的头数据大小确定所述第一协议子层的至少一个分片大小；根据所述第一协议子层的至少一个分片大小确定所述第一协议子层的至少一个MCS。
在本发明的一个实施例中，可以根据在重传次数的阈值范围内的多个重传次数、丢弃计时器的时间阈值范围内的多个丢弃计时器的时间、传输时间间隔绑定TTIB开关的开闭状态确定RLC层的多个分片数。例如，将TTIB开关的状态值和重传次数的乘积记为A，每个分片数可以是对丢弃计时器的 时间除以A得到，或者，每个分片数可以是对丢弃计时器的时间除以A保留一定位数的小数四舍五入得到。
在本发明的一个实施例中，可以根据PDCP层缓存的数据大小、RLC层的分片数和RLC层的头数据大小确定RLC层的分片大小。例如，将PDCP层缓存的数据大小除以RLC层的多个分片数记为B，RLC层的分片大小可以是B加上RLC层的头数据大小得到。
结合第一方面及其上述实现方式，在第一方面的另一种实现方式中，所述第一调度参数包括以下任一种调度参数：MCS、重传次数、丢弃计时器的时间、PDCP缓存的数据大小、所述RLC层的分片数。
结合第一方面及其上述实现方式，在第一方面的另一种实现方式中，所述第一协议子层为无线链路控制RLC层，所述其他协议子层包括分组数据汇聚协议PDCP层和媒体接入控制MAC层。
本发明实施例中的第一调度参数可以为传输VoLTE业务时的任一参数。在本发明的一个实施例中，可以根据本发明实施例的方法，估计RLC层、PDCP层和MAC层这三层的对应参数值，并选择合适的参数传输VoLTE业务，这样能够确保链路的传输质量。
本发明实施例的调整语音业务的方法应用于服务质量(Quality Of Service，QoS)等级标识(QoS Class Identifier，QCI)1业务或QCI2业务中时，可以使得保证链路传输质量的效果更明显。本发明实施例中，PDCP层的第一调度参数可以根据RLC层对应的第一调度参数得到。例如，可以根据RLC层的MCS得到RLC层的吞吐率，根据RLC层的吞吐率和传输效率因子得到PDCP层的吞吐率。PDCP层的吞吐率可以认为是PDCP层的MCS的线性函数，所以，由PDCP层的吞吐率可以得到PDCP层的MCS。由此，如果第一调度参数为MCS，可以根据RLC层的MCS得到PDCP层的MCS。
本发明实施例中PDCP层的重传次数和丢弃计时器的时间都可以作为独立的参数进行配置。
PDCP层可以独立配置不同的重传次数和丢弃计时器的时间。另外，一般地，PDCP层不考虑重传，此时可以认为PDCP层的重传次数为1。
本发明实施例中MAC层的第一调度参数可以对信道质量(Channel Quality Indicator，CQI)进行估计得到。例如，如果第一调度参数为MCS，网络侧设备可以根据信道质量的估计结果得到MAC层的MCS。
本发明实施例中MAC层的重传次数和丢弃计时器的时间都可以作为独立的参数进行配置。MAC层的重传次数可以是MAC层的混合自动重传请求(Hybrid Automatic Repeat reQuest，HARQ)的重传次数。
MAC层可以独立配置不同的重传次数和丢弃计时器的时间。另外，一般地，MAC层不会主动丢弃数据包，可以认为MAC层丢弃计时器的时间为无限大。
第二方面，提供了一种分组域语音业务调度的装置，包括：获取单元，用于获取语音业务的L2至少两个协议子层中每个协议子层分别对应的第一调度参数的至少一个值；第一确定单元，用于确定所述语音业务的业务质量需要满足的条件；第二确定单元，用于根据所述第一确定单元确定的所述语音业务的业务质量需要满足的条件以及所述获取单元获取的所述语音业务的L2至少两个协议子层中每个协议子层分别对应的第一调度参数的至少一个值确定所述语音业务的业务调度参数的值；调度单元，用于根据所述第二确定单元确定的所述语音业务的业务调度参数的值调度所述语音业务。
结合第二方面，在第二方面的一种实现方式中，所述第一调度参数包括以下任一种调度参数：调制编码方式MCS、重传次数和丢弃计时器的时间；
所述获取单元具体用于根据上述三种调度参数中除所述第一调度参数外的另两种调度参数在第一协议子层分别对应的阈值范围内取值，得到所述第一协议子层的第一调度参数的至少一个值，并配置或根据所述第一协议子层的第一调度参数的至少一个值计算所述语音业务的L2至少两个协议子层中的其他协议子层分别对应的第一调度参数的至少一个值，其中，所述第一协作子层为所述语音业务的L2至少两个协议子层中的一个。
结合第二方面及其上述实现方式，在第二方面的另一种实现方式中，所述语音业务的业务质量需要满足的条件为所述语音业务的语音质量平均意见值MOS最优，所述第二确定单元具体用于根据所述第一协议子层的第一调度参数的至少一个值确定对应的所述第一协议子层的至少一个吞吐率，确定与所述第一协议子层的至少一个吞吐率一一对应的其他协议子层的至少一个吞吐率，根据所述第一协议子层的至少一个吞吐率和所述其他协议子层的至少一个吞吐率确定对应的至少一个所述语音业务的语音质量MOS，并从所述至少一个所述语音业务的语音质量MOS中选择最优的语音质量MOS对应的第一调度参数的值作为所述语音业务的业务调度参数的值。
结合第二方面及其上述实现方式，在第二方面的另一种实现方式中，所述语音业务的业务质量需要满足的条件为所述语音业务的吞吐率最大，所述第二确定单元具体用于根据所述第一协议子层的第一调度参数的至少一个值确定对应的所述第一协议子层的至少一个吞吐率，确定与所述第一协议子层的至少一个吞吐率一一对应的其他协议子层的至少一个吞吐率，根据所述第一协议子层的至少一个吞吐率和所述其他协议子层的至少一个吞吐率确定对应的所述语音业务的至少一个吞吐率，从所述语音业务的至少一个吞吐率中选择最大的吞吐率对应的第一调度参数的值作为所述语音业务的业务调度参数的值。
结合第二方面及其上述实现方式，在第二方面的另一种实现方式中，所述语音业务的业务质量需要满足的条件为所述语音业务的传输速率最大，所述第二确定单元具体用于确定业务调度参数的至少一个值，其中，业务调度参数的每个值为从所述第一协议子层的第一调度参数的一个值和对应的其他各个协议子层的第一调度参数的一个值中选择的最大值，并从所述业务调度参数的至少一个值中选择最大值作为所述语音业务的业务调度参数的值。
结合第二方面及其上述实现方式，在第二方面的另一种实现方式中，所述语音业务的业务质量需要满足的条件为所述语音业务的误码率最小，所述第二确定单元具体用于确定业务调度参数的至少一个值，其中，业务调度参数的每个值为从所述第一协议子层的第一调度参数的一个值和对应的其他协议子层的第一调度参数的一个值中选择的最小值，并从所述业务调度参数的至少一个值中选择最小值作为所述语音业务的业务调度参数的值。
结合第二方面及其上述实现方式，在第二方面的另一种实现方式中，其特征在于所述第一调度参数为MCS，所述获取单元具体用于根据在所述重传次数的阈值范围内的多个重传次数、所述丢弃计时器的时间的阈值范围内的多个丢弃计时器的时间、传输时间间隔绑定TTIB开关的开闭状态确定所述第一协议子层的至少一个分片数，其中，所述第一协议子层的每个分片数与一个重传次数和一个丢弃计时器的时间相对应，根据所述其他协议子层最大可缓存的数据大小、所述第一协议子层的至少一个分片数和所述第一协议子层的头数据大小确定所述第一协议子层的至少一个分片大小，并根据所述第一协议子层的至少一个分片大小确定所述第一协议子层的至少一个MCS。
结合第二方面及其上述实现方式，在第二方面的另一种实现方式中，所 述第一调度参数包括以下任一种调度参数：MCS、重传次数、丢弃计时器的时间、PDCP缓存的数据大小、所述RLC层的分片数。
结合第二方面及其上述实现方式，在第二方面的另一种实现方式中，所述第一协议子层为无线链路控制RLC层，所述其他协议子层包括分组数据汇聚协议PDCP层和媒体接入控制MAC层。
上述第二方面提供的分组域语音业务调度的装置的相应单元和/或器件的各个操作可以参照第一方面中的方法的各个步骤，在此不再重复。所述第二方面的各个技术方案带来的有益技术效果，可以参照第一方面方法中的技术效果，在不再重复。
附图说明
为了更清楚地说明本发明实施例的技术方案，下面将对本发明实施例中所需要使用的附图作简单地介绍，显而易见地，下面所描述的附图仅仅是本发明的一些实施例，对于本领域普通技术人员来讲，在不付出创造性劳动的前提下，还可以根据这些附图获得其他的附图。
图1是本发明一个实施例的分组域语音业务调度的方法的示意性流程图。
图2是本发明一个实施例的分组域语音业务调度的装置的框图。
图3是本发明另一实施例的分组域语音业务调度的装置的框图。
具体实施方式
下面将结合本发明实施例中的附图，对本发明实施例中的技术方案进行清楚、完整地描述，显然，所描述的实施例是本发明的一部分实施例，而不是全部实施例。基于本发明中的实施例，本领域普通技术人员在没有做出创造性劳动的前提下所获得的所有其他实施例，都应属于本发明保护的范围。
首先，简单介绍一下LTE无线接入协议的接入系统的分层结构。LTE无线接入协议的接入系统分为三层：层一(layer 1，L1)为物理层(Physical Layer，PHY)，层二(L2)为MAC层、RLC层和PDCP层，层三(L3)为无线资源控制层(Radio Resource Control，RRC)。
物理层是无线接入系统最底层，它以传输信道为接口，向上层提供服务。PDCP层位于RLC层之上，是L2的最上面的一个子层。PDCP子层可以处 理空中接口上承载网络层的分组数据，例如网络互联协议(Internet Protocol，IP)数据流，也可以处理控制面的无线资源管理(RRC)消息。PDCP子层可以处理接口上承载的分组数据，并对分组数据进行压缩和加密，然后递交到RLC子层。RLC子层位于MAC子层之上，属于L2的一部分。RLC子层可以为用户和控制数据提供分段(segement)和重传业务。例如，将长度不同的高层协议数据单元(Protocol Data Unit，PDU)分组进行分段(发送端)重组(接收端)为较小的RLC负荷单元。MAC层定义了数据帧在介质上进行传输的传输方式。
图1是本发明一个实施例的分组域语音业务调度参数的方法的示意性流程图。图1的方法可以由网络侧设备执行，例如，可以由演进型基站(Evolved Node B，e-NodeB)执行。
101，获取语音业务的L2至少两个协议子层中每个协议子层分别对应的第一调度参数的至少一个值。
102，确定语音业务的业务质量需要满足的条件。
103，根据语音业务的业务质量需要满足的条件以及语音业务的L2至少两个协议子层中每个协议子层分别对应的第一调度参数的至少一个值确定语音业务的业务调度参数的值。
104，根据语音业务的业务调度参数的值调度语音业务。
本发明实施例根据语音业务的L2不同协议子层的调度参数和语音业务的业务质量需要满足的条件确定语音业务的业务调度参数的值，并根据语音业务的调度参数的值调度语音业务，这样能够保证链路的传输质量。
本发明实施例可以用于LTE系统，此时，语音业务可以为VoLTE业务，L2的各个协议子层可以包括RLC层、PDCP层、MAC层。
本发明实施例中的第一调度参数可以是语音业务传输过程中的涉及到的参数。例如，本发明实施例中的第一调度参数可以为下列参数中的任意一个：MCS、重传次数、丢弃计时器的时间、PDCP缓存的数据大小、RLC层的分片数。
本发明实施例中，语音业务的各个协议子层的第一调度参数同时为上述第一调度参数中的一种，即各个协议子层的第一调度参数为同一参数，但各个协议子层的第一调度参数的值可以不同。
例如，第一调度参数可以为MCS，此时，可以根据语音业务的业务质 量需要满足的条件和L2各协议子层的MCS对应的值，选择合适的MCS对传输的语音业务进行自适应调制与编码。
又如，第一调度参数可以为重传次数，此时，可以根据语音业务的业务质量需要满足的条件和L2各协议子层的重传次数对应的值，选择合适的重传次数对传输的语音业务进行自适应重传。
再如，第一调度参数可以为丢弃计时器的时间，此时，可以根据语音业务的业务质量需要满足的条件和L2各协议子层的丢弃计时器的时间对应的值，选择合适的丢弃计时器的时间对传输的语音业务进行自适应PDCP缓存大小的调整。
当第一协议子层的第一调度参数取一个值时，其他协议子层的第一调度参数分别可以对应的取相应值，此时可以确定对应的传输语音业务的第一调度参数的值。当第一协议子层的第一调度参数在一定范围内取值时，可以得到对应的传输语音业务的第一调度参数的多个值。最后，可以根据语音业务的业务质量需要满足的条件从传输语音业务的第一调度参数的多个值中选择一个作为确定使用的语音业务的业务调度参数，以对语音业务进行调度。这里，第一协议子层可以为语音业务的L2至少两个协议子层中的一个。
本发明实施例中语音业务的业务质量需要满足的条件可以是语音质量满足一定条件，例如，语音质量最优。语音质量可以通过语音业务的业务质量MOS、语音业务的吞吐量、语音业务的传输速率、语音业务的误码率等来体现。所以，本发明实施例中步骤102语音业务的业务质量需要满足的条件可以为语音业务的业务质量MOS最优、语音业务的吞吐率最大、语音业务的传输速率最大、语音业务的误码率最小等，另外，还可以是上述至少两个条件的综合。
语音业务的业务质量需要满足的条件为语音业务的传输速率最大时，可以从语音业务的第一调度参数的多个值中选择最大的值作为语音业务的业务调度参数的值。语音业务的业务质量需要满足的条件为语音业务的误码率最小时，可以从语音业务的第一调度参数的多个值中选择最小的值作为语音业务的业务调度参数的值。
本发明一个实施例中，传输语音业务的吞吐率或语音业务的语音质量MOS可以认为至少是MCS、重传次数和丢弃计时器的时间三个参量共同决定的。当MCS、重传次数和丢弃计时器的时间三个参量中，任意两个参量 取相应的值时，吞吐率或语音质量MOS可以认为是第三个参量的函数。所以，可以找到吞吐率或语音质量MOS最大时对应的第三个参量的值作为实际传输语音业务时的业务调度参数。例如，对传输的语音业务进行自适应调整与编码时，可以将重传次数和丢弃计时器的时间在一定范围内取相应的值，此时语音业务的吞吐量或语音质量MOS可以认为是MCS的函数，可以将吞吐率最大或语音质量MOS最优时对应的MCS作为语音业务的业务调度参数，对语音业务进行自适应调制与编码。
下面结合具体实施例分别以第一调度参数为MCS、重传次数或丢弃计时器的时间为例，举例说明本发明实施例的具体实现方式。并且，具体实施例中以语音业务为VoLTE业务，L2的至少两个协议子层包括RLC层、PDCP层和MAC层，第一协议子层为RLC层为例进行示例性说明。
下面的实施例一以第一调度参数为MCS为例进行示例性说明。
网络侧设备可以获取RLC层的MCS的值、PDCP层的MCS的值和MAC层的MCS的值，并根据该三层的MCS的值和语音业务的业务质量需要满足的条件确定实际传输VoLTE业务采用的最终的MCS的值，最后根据最终的MCS的值调度VoLTE业务。这种AMC方式，系统需要定义不同的数据传输MCS格式，MCS格式对应于各种调制阶数和编码速率，当信道条件变化时，系统可以根据信道条件选择不同的MCS方案进行自适应调制与编码以适应信道变化带来的影响，从而实时保证链路的传输质量。
第一调度参数为MCS时，重传次数和丢弃计时器的时间可以通过系统配置或指定各自的取值范围。
网络侧设备获取RLC层的MCS的值具体方式如下。
网络侧设备可以通过PDCP层的丢弃计时器的时间(Discard Timer)获取PDCP层最大可缓存(Buffer)的数据大小，丢弃计时器的时间可以由网络侧设备配置并下发给用户设备。PDCP层最大可缓存(Buffer)的数据大小可以用来计算RLC层的分片大小，进一步确定RLC层的MCS的值。
网络侧设备还可以通过用户设备(User Equipment，UE)上报的缓存状态报告(Buffer Status Report，BSR)、当前时刻之前的MCS、VoLTE编码速率估计当前PDCP层已缓存数据的大小。
RLC层的分片数可以根据下列方式估计：根据PDCP层的丢弃计时器的时间、重传次数、TTIB开关估计RLC层的分片数。例如，如果PDCP层 丢弃计时器的时间为100ms，重传次数为8次，TTIB开关打开，那么当重传进程全部被VoLTE业务占满时对应的RLC分片数＝round(100/(4*8))。
RLC层的分片大小可以根据下列方式估计：RLC分片大小＝PDCP最大可缓存的数据大小/RLC分片数+RLC头数据的大小。
当资源块(Resource Block，RB)数固定时，RLC分片大小和MCS的值是一一对应的。即通过上述这些参数(PDCP Buffer状态、传输次数、TTIB开关等)可以确定对应的RLC层的分片大小，进而确定RLC层的MCS的值。
本发明实施例中可以调度至少一个RB，RB数不固定。此时，RB数和MCS可以共同决定RLC层的分片大小。此时，根据上述参数(PDCP Buffer状态、传输次数、TTIB开关等)和RB数可以得到RLC层的分片大小，然后由RLC层的分片大小和RB数可以得到RLC层的MCS的值。
网络侧设备可以通过下列方式获取PDCP层的MCS的值：根据RLC层分片数和分片大小确定RLC层的吞吐率。例如，RLC层的吞吐率＝RLC层的分片大小*RLC层的分片数。PDCP层的吞吐率＝RLC层吞吐率*传输效率因子，传输效率因子可以由RLC层的分片数和RLC层的头开销大小决定。例如，传输效率因子＝RLC层的分片大小/[(RLC层的分片大小+RLC层的头开销大小)*分片数]。另外，PDCP层的吞吐率可以视为PDCP层的MCS的线性函数，可以根据PDCP层的吞吐率得到PDCP层的MCS的值。
网络设备可以通过下列方式获取MAC层的MCS的值：网络侧设备可以对信道质量进行估计，并根据信道质量的估计结果和各个MCS对应的信道质量门限值确定MAC层的MCS的值。这里的信道质量可以为信噪比或误码率等。
网络侧设备可以配置最大重传次数和丢弃计时器的最长时间。将重传次数在小于或者等于最大重传次数范围内取不同值，丢弃计时器的时间在小于或者等于最长时间的范围内取不同值，这样根据上述计算MCS的值的方法可以得到RLC层的多个MCS的值。
本发明实施例中，RLC层的每个MCS可以是RLC层的一个MCS的值、及其分别对应的PDCP层的一个MCS的值和MAC层的一个MCS的值中选择出来的，例如，从上述三层的MCS的值中选择一个最大值作为对应的语音业务的一个MCS的值。
本发明一个实施例中，语音业务的业务质量需要满足的条件可以为语音业务的业务质量MOS最优。此时，可以通过RLC层的MCS、PDCP层的MCS和MAC层的MCS计算VoLTE业务的语音质量MOS，选择MOS最优对应的MCS的值作为最终的MCS的值，对VoLTE业务进行调度，这样可以保证VoLTE业务的语音质量。
VoLTE业务的语音质量MOS可以通过下列方式得到：根据RLC层的MCS、PDCP层的MCS和MAC层的MCS分别估计RLC层的吞吐率、PDCP层的吞吐率和MAC层的吞吐率，每层的吞吐率和该层的MCS可以视为线性关系。从RLC层的吞吐率、PDCP层的吞吐率和MAC层的吞吐率中选择最小值作为VoLTE业务传输时的实际吞吐率。最后根据实际吞吐率确定MOS。例如，MOS＝Const-a*max[1-实际吞吐率/(语音速率*DTX比例)，0]，其中Const表示该语音编码方式和语音速率下的最大MOS分值，通常可以通过离线状态下对大量序列进行训练后获取得到。
其中，PDCP层的吞吐率可以根据下列方式得到：根据RLC层的分片数和RLC层的头数据开销确定传输效率因子，将RLC层的吞吐率和传输效率因子相乘可以得到PDCP层的吞吐率。传输效率因子可以通过下列方式得到：传输效率因子＝RLC层的分片大小/[(RLC层的分片大小+RLC层的头开销大小)*分片数]。
MAC层的吞吐率可以通过对信道质量进行估计，得到信道质量的估计结果，并根据估计结果和不同MCS对应的信道质量的门限值确定MAC层的MCS。
本发明一个实施例中，语音业务的业务质量需要满足的条件可以为语音业务的吞吐率最大。此时，可以通过RLC层的MCS、PDCP层的MCS和MAC层的MCS计算传输VoLTE业务的实际吞吐率，可以选择实际吞吐率最大对应的MCS作为最终的MCS的值，对VoLTE业务进行调度，这样可以使得传输语音业务的实际吞吐率最大，进而使得VoLTE业务具有较好的语音质量。
作为本发明的一个实施例，语音业务的业务质量需要满足的条件为语音业务的传输速率最大时，可以从上述三层的MCS的值确定的语音业务的多个MCS的值中选择最大的值作为最终的语音业务的业务调度参数MCS的值。语音业务的业务质量需要满足的条件为语音业务的误码率最小时，可以 从上述三层的MCS的值确定的语音业务的多个MCS中选择最小的值作为语音业务的业务调度参数MCS的值。不同的调制方式有不同的特征，低阶调制增加了较多的冗余导致实际效率较低，但能够保证较高的可靠性，高阶调试具有较高的效率但可靠性差，对信道条件提出了较高的要求，只有在信道很好的条件下才能获得较高的增益。最终的MCS取三层中MCS的最大值还是最小值由实际的需求决定。例如，为了保证网络侧设备和用户设备之间传输语音业务的误码率小，可以选择较小的MCS。为了保证网路侧设备和用户设备之间传输语音业务的传输速率最大，可以选择较大的MCS。
在本发明的一个实施例中，如果第一调度参数为MCS，L2的RLC层、PDCP层和MAC层三层的三个MCS的值可以确定语音业务的一个MCS的值。最终确定的语音业务的业务调度参数MCS的值可以是从按照上述方法确定的语音业务的多个MCS的值中选择出的一个MCS的值，每一层都以该最终选择出的MCS的值作为该层的调制编码方式。
下面的实施例二以第一调度参数为重传次数为例进行示例性说明。
HARQ是一种将前向纠错编码(FEC)和自动重传请求(ARQ)相结合的MAC层的重传方式。HARQ的关键词是存储、请求重传、合并解调。接收方在解码失败的情况下，保存接收到的数据，并要求发送方重传数据，接收方将重传的数据和先前接收到的数据进行合并后再解码。这里面就有一定的分集增益，减少了重传次数，进而减少了时延。HARQ可以高效地补偿由于采用链路适配所带来的误码，能够提高数据传输速率，可以减小数据传输时延。
网络侧设备可以获取RLC层的重传次数的取值范围、分组数据汇聚协议PDCP层的重传次数的取值范围和MAC层的重传次数的取值范围，并根据该三层一一对应的三个重传次数的值确定语音业务的重传次数的一个值，进而可以根据各层重传次数的取值范围确定语音业务的重传次数的多个值，并根据语音业务的业务质量需要满足的条件，从重传次数的多个值中确定一个值作为实际传输VoLTE业务采用的最终的语音业务的重传次数，最后根据最终的语音业务的重传次数进行VoLTE业务的调度。当信道条件变化时，系统可以根据信道条件选择不同的重传次数以适应信道变化带来的影响。这种方式可以对VoLTE业务的传输进行自适应重传的调整，以实时保证链路的传输质量，可以减小误码率，能够提高数据的传输速率。
网络设备获取RLC层的重传次数具体方式如下。网络侧设备获取PDCP层最大可缓存(Buffer)的数据大小和PDCP层已缓存数据的大小的方式与实施例一中的方式相同，为避免重复，在此不再详细赘述。
当第一调度参数为重传次数时，网络侧设备可以配置最大的MCS的值和丢弃计时器的最长时间。RLC层的MCS的值可以在小于或者等于最大的MCS的值的范围内取值，丢弃计时器的时间可以在小于或者等于最长时间的范围内取值，这时可以根据MCS和丢弃计时器的时间确定与之对应的重传次数。例如，由于当资源块(Resource Block，RB)数固定时，RLC层的MCS的值与RLC层的分片大小一一对应。当给定RLC层的MCS的值时，可以得到RLC层的分片大小。根据实施例一中分片大小和分片数的关系，可以由分片数的大小得到与之对应的分片数。最后根据实施例一中分片数的估计方式得到重传次数。具体各个参数之间的关系式参照实施例一中的描述，为避免重复，在此不再详细赘述。
本发明实施例中的RLC层、PDCP层和MAC层三层的重传次数可以独立配置。配置重传次数时，可以根据时延等要求进行，进而配置时重传次数的值会有一定的取值范围。在三层的重传次数配置为不同值时，可以根据重传次数的值得到传输VoLTE业务的多个吞吐率或多个语音质量MOS，并选择使得吞吐率最大或语音质量MOS最优的重传次数的值作为最终语音业务的重传次数的值。
在本发明的一个实施例中，可以由RLC层、PDCP层和MAC层三层的重传次数计算VoLTE业务的吞吐率，并选择吞吐率最大时对应的重传次数确定为实际传输VoLTE业务的重传次数。
在本发明的一个实施例中，可以由RLC层、PDCP层和MAC层三层的重传次数计算VoLTE业务的语音质量MOS，并选择语音质量最优时对应的重传次数确定为实际传输VoLTE业务的重传次数。
本发明的一个实施例中，如果第一调度参数为的重传次数，L2的RLC层、PDCP层和MAC层三层的三个重传次数的值可以确定语音业务的一个重传次数的值。最终确定的语音业务的重传次数的值可以是从按照上述方法确定的语音业务的多个重传次数的值中选择出的一个重传次数的值，每一层都以该最终选择出的重传次数的值进行重传。
在本发明的一个实施例中，语音业务的业务质量满足的条件为语音业务 的传输速率最大时，可以从RLC层、PDCP层和MAC层三层的重传次数的取值范围内确定的上述语音业务的多个重传次数的值中选择最大的重传次数的值作为最终使用的调度语音业务的重传次数的值。这样进行语音数据的多次重传可以保证链路的传输质量。
在本发明的一个实施例中，语音业务的业务质量满足的条件为语音业务的误码率最小时，可以从RLC层、PDCP层和MAC层三层的重传次数的取值范围内确定的上述语音业务的多个重传次数的值中选择最小的重传次数的值作为最终使用的调度语音业务的重传次数的值。这样可以提高传输速率。
下面的实施例三以第一调度参数为丢弃计时器的时间为例进行示例性说明。
网络测设备可以获取RLC层的丢弃计时器的时间的取值范围、PDCP层的丢弃计时器的时间的取值范围和MAC层的丢弃计时器的时间的取值范围、，并根据该三层一一对应的三个丢弃计时器的时间确定语音业务的丢弃计时器的时间的一个值，进而可以根据各层丢弃计时器的时间的取值范围确定语音业务的丢弃计时器的时间的多个值，并根据语音业务的业务质量需要满足的条件，从丢弃计时器的时间的多个值中确定一个值作为实际传输VoLTE业务采用的最终的语音业务的丢弃计时器的时间的值，最后根据最终的丢弃计时器的时间的值进行VoLTE业务的调度。当信道条件变化时，系统可以根据信道条件选择不同的丢弃计时器的时间以适应信道变化带来的影响。这种方式可以对VoLTE业务的待缓存数据大小进行自适应调整，以实时保证链路的传输质量。
网络设备获取RLC层的丢弃计时器的时间具体方式如下。网络侧设备获取PDCP层最大可缓存的数据大小和PDCP层已缓存数据的大小的方式与实施例一中的方式相同，为避免重复，在此不再详细赘述。
当第一调度参数为丢弃计时器的时间时，网络侧设备可以配置最大的MCS的值和最大重传次数的值。当RLC层的MCS小于或者等于最大的MCS的值的范围内取值，重传次数在小于或者等于最大重传次数的范围内取值时，可以根据MCS的值和重传次数的值确定与之对应的丢弃计时器的时间的值。例如，由于RLC层的MCS的值与RLC层的分片大小一一对应。当给定RLC层的MCS的值时，可以得到RLC层的分片大小。根据实施例一 中分片大小和分片数的关系，可以由分片数的大小得到与之对应的分片数。最后根据实施例一中分片数的估计方式得到丢弃计时器的时间。具体各个参数之间的关系式参照实施例一中的描述，为避免重复，在此不再详细赘述。
本发明实施例中的RLC层、PDCP层和MAC层三层的丢弃计时器的时间的值可以独立配置。配置丢弃计时器的时间的值时，可以根据时延等要求进行，进而配置时丢弃计时器的时间的值会有一定的取值范围。在三层的丢弃计时器的时间配置为不同值时，可以根据丢弃计时器的时间得到不同的传输VoLTE业务的吞吐率或语音质量MOS，并选择使得吞吐率最大或语音质量MOS最优的丢弃计时器的时间的值作为最终的丢弃计时器的时间的值。
在本发明的一个实施例中，可以由RLC层、PDCP层和MAC层三层的丢弃计时器的时间计算VoLTE业务的吞吐率，并选择吞吐率最大时对应的丢弃计时器的时间的值确定为实际传输VoLTE业务的丢弃计时器的时间的值。
在本发明的一个实施例中，可以由RLC层、PDCP层和MAC层三层的丢弃计时器的时间计算VoLTE业务的语音质量MOS，并选择语音质量MOS最优时对应的丢弃计时器的时间的值确定为实际传输VoLTE业务的丢弃计时器的时间的值。
当语音质量参数为丢弃计时器的时间时，可以是从RLC层、PDCP层和MAC层三层中的重传次数中直接选择最大或最小的丢弃计时器的时间的值作为最终使用的传输语音业务的丢弃计时器的时间的值，进而可以保证链路的传输质量。
上述三个实施例分别以第一调度参数为MCS、重传次数和丢弃计时器的时间为例进行示例性说明，本发明实施例中的第一调度参数还可以为PDCP层缓存数据的大小、RLC层的分片数等其他参数。其他参数都可以根据RLC层、PDCP层和MAC层三层的对应参数得到实际传输语音业务所采用的该参数，进而对VoLTE业务的传输进行该参数的自适应调整，以保证链路的传输质量。
本发明实施例的调整语音业务的方法应用于QCI1业务或QCI2业务中时，可以使得保证链路传输质量的效果更明显。
图2是本发明一个实施例的分组域语音业务调度的装置的框图。图2的分组域语音业务调度的装置10可以包括获取单元11、第一确定单元12、第 二确定单元13和调整单元14。
获取单元11用于获取语音业务的L2至少两个协议子层中每个协议子层分别对应的第一调度参数的至少一个值。
第一确定单元12用于确定语音业务的业务质量需要满足的条件。
第二确定单元13用于根据第一确定单元确定的语音业务的业务质量需要满足的条件以及获取单元获取的语音业务的L2至少两个协议子层中每个协议子层分别对应的第一调度参数的至少一个值确定语音业务的业务调度参数的值。
调整单元14用于根据第二确定单元确定的语音业务的业务调度参数的值调度语音业务。
本发明实施例根据语音业务的L2不同协议子层的调度参数和语音业务的业务质量需要满足的条件确定语音业务的业务调度参数的值，并根据语音业务的调度参数的值调度语音业务，这样能够保证链路的传输质量。
根据本发明实施例的分组域语音业务调度的装置可对应于本发明实施例图1中的分组域语音业务调度的方法，并且，该装置中的各个单元/模块和上述其他操作和/或功能分别为了实现图1所示方法所执行的相应流程，为了简洁，在此不再赘述。
图3是本发明另一实施例的分组域语音业务调度的装置的框图。该装置20包括处理器21、存储器22和总线系统23，该处理器21和该存储器22通过总线系统23相连，该存储器22用于存储指令，该处理器21用于执行该存储器22存储的指令，使得该装置20执行图1方法的流程中的各个步骤。
上述本发明实施例揭示的方法可以应用于处理器21中，或者由处理器21实现。在实现过程中，上述方法的各步骤可以通过处理器21中的硬件的集成逻辑电路或者软件形式的指令完成。处理器21可以是通用处理器、数字信号处理器、专用集成电路、现场可编程门阵列或者其他可编程逻辑器件、分立门或者晶体管逻辑器件、分立硬件组件，可以实现或者执行本发明实施例中的公开的各方法、步骤及逻辑框图。通用处理器可以是微处理器或者任何常规的处理器等。结合本发明实施例所公开的方法的步骤可以直接体现为硬件处理器执行完成，或者用处理器中的硬件及软件模块组合执行完成。软件模块可以位于随机存储器，闪存、只读存储器，可编程只读存储器或者电可擦写可编程存储器、寄存器等本领域成熟的存储介质中。该存储介质位于 存储器22，处理器21读取存储器22中的信息，结合其硬件完成上述方法的步骤。
具体地，处理器21用于获取语音业务的L2至少两个协议子层中每个协议子层分别对应的第一调度参数的至少一个值，确定语音业务的业务质量需要满足的条件，根据语音业务的业务质量需要满足的条件以及语音业务的L2至少两个协议子层中每个协议子层分别对应的第一调度参数的至少一个值确定语音业务的业务调度参数的值，最后根据语音业务的业务调度参数的值调度语音业务。
本发明实施例根据语音业务的L2不同协议子层的调度参数和语音业务的业务质量需要满足的条件确定语音业务的业务调度参数的值，并根据语音业务的调度参数的值调度语音业务，这样能够保证链路的传输质量。
根据本发明实施例的分组域语音业务调度的装置可对应于本发明实施例图1中的分组域语音业务调度的方法，并且，该装置中的各个单元/器件和上述其他操作和/或功能分别为了实现图1所示方法所执行的相应流程，为了简洁，在此不再赘述。
应理解，说明书通篇中提到的“一个实施例”或“一实施例”意味着与实施例有关的特定特征、结构或特性包括在本发明的至少一个实施例中。因此，在整个说明书各处出现的“在一个实施例中”或“在一实施例中”未必一定指相同的实施例。此外，这些特定的特征、结构或特性可以任意适合的方式结合在一个或多个实施例中。
本领域普通技术人员可以意识到，结合本文中所公开的实施例中描述的各方法步骤和单元，能够以电子硬件、计算机软件或者二者的结合来实现，为了清楚地说明硬件和软件的可互换性，在上述说明中已经按照功能一般性地描述了各实施例的步骤及组成。这些功能究竟以硬件还是软件方式来执行，取决于技术方案的特定应用和设计约束条件。本领域普通技术人员可以对每个特定的应用来使用不同方法来实现所描述的功能，但是这种实现不应认为超出本发明的范围。
结合本文中所公开的实施例描述的方法或步骤可以用硬件、处理器执行的软件程序，或者二者的结合来实施。软件程序可以置于随机存储器(RAM)、内存、只读存储器(ROM)、电可编程ROM、电可擦除可编程ROM、寄存器、硬盘、可移动磁盘、CD-ROM、或技术领域内所公知的任意其他形式的 存储介质中。
尽管通过参考附图并结合优选实施例的方式对本发明进行了详细描述，但本发明并不限于此。在不脱离本发明的精神和实质的前提下，本领域普通技术人员可以对本发明的实施例进行各种等效的修改或替换，而这些修改或替换都应在本发明的涵盖范围内。

Claims (18)

  1. 一种分组域语音业务调度的方法，其特征在于，包括：
    获取语音业务的L2至少两个协议子层中每个协议子层分别对应的第一调度参数的至少一个值；
    确定所述语音业务的业务质量需要满足的条件；
    根据所述语音业务的业务质量需要满足的条件以及所述语音业务的L2至少两个协议子层中每个协议子层分别对应的第一调度参数的至少一个值确定所述语音业务的业务调度参数的值；
    根据所述语音业务的业务调度参数的值调度所述语音业务。
  2. 如权利要求1所述的方法，其特征在于，所述第一调度参数包括以下任一种调度参数：调制编码方式MCS、重传次数和丢弃计时器的时间；
    所述获取语音业务的L2至少两个协议子层中每个协议子层分别对应的第一调度参数的至少一个值包括：
    根据上述三种调度参数中除所述第一调度参数外的另两种调度参数在第一协议子层分别对应的阈值范围内取值，得到所述第一协议子层的第一调度参数的至少一个值，所述第一协作子层为所述语音业务的L2至少两个协议子层中的一个；
    配置或根据所述第一协议子层的第一调度参数的至少一个值计算所述语音业务的L2至少两个协议子层中的其他协议子层分别对应的第一调度参数的至少一个值。
  3. 如权利要求2所述的方法，其特征在于，所述语音业务的业务质量需要满足的条件为所述语音业务的语音质量平均意见值MOS最优，
    其中，
    所述根据所述语音业务的业务质量需要满足的条件以及所述语音业务的L2至少两个协议子层分别对应的第一调度参数的至少一个值确定所述语音业务的业务调度参数的值包括：
    根据所述第一协议子层的第一调度参数的至少一个值确定对应的所述第一协议子层的至少一个吞吐率；
    确定与所述第一协议子层的至少一个吞吐率一一对应的其他协议子层的至少一个吞吐率；
    根据所述第一协议子层的至少一个吞吐率和所述其他协议子层的至少一个吞吐率确定对应的至少一个所述语音业务的语音质量MOS；
    从所述至少一个所述语音业务的语音质量MOS中选择最优的语音质量MOS对应的第一调度参数的值作为所述语音业务的业务调度参数的值。
  4. 如权利要求2所述的方法，其特征在于，所述语音业务的业务质量需要满足的条件为所述语音业务的吞吐率最大，
    其中，
    所述根据所述语音业务的业务质量需要满足的条件以及所述语音业务的L2至少两个协议子层分别对应的第一调度参数的至少一个值确定所述语音业务的业务调度参数的值包括：
    根据所述第一协议子层的第一调度参数的至少一个值确定对应的所述第一协议子层的至少一个吞吐率；
    确定与所述第一协议子层的至少一个吞吐率一一对应的其他协议子层的至少一个吞吐率；
    根据所述第一协议子层的至少一个吞吐率和所述其他协议子层的至少一个吞吐率确定对应的所述语音业务的至少一个吞吐率；
    从所述所述语音业务的至少一个吞吐率中选择最大的吞吐率对应的第一调度参数的值作为所述语音业务的业务调度参数的值。
  5. 如权利要求2所述的方法，其特征在于，所述语音业务的业务质量需要满足的条件为所述语音业务的传输速率最大，
    其中，
    所述根据所述语音业务的业务质量需要满足的条件以及所述语音业务的L2至少两个协议子层分别对应的第一调度参数的至少一个值确定所述语音业务的业务调度参数的值包括：
    确定业务调度参数的至少一个值，其中，业务调度参数的每个值为从所述第一协议子层的第一调度参数的一个值和对应的其他每个协议子层的第一调度参数的一个值中选择的最大值；
    从所述业务调度参数的至少一个值中选择最大值作为所述语音业务的业务调度参数的值。
  6. 如权利要求2所述的方法，其特征在于，所述语音业务的业务质量需要满足的条件为所述语音业务的误码率最小，
    其中，
    所述根据所述语音业务的业务质量需要满足的条件以及所述语音业务的L2至少两个协议子层分别对应的第一调度参数的至少一个值确定所述语音业务的业务调度参数的值包括：
    确定业务调度参数的至少一个值，其中，业务调度参数的每个值为从所述第一协议子层的第一调度参数的一个值和对应的其他每个协议子层的第一调度参数的一个值中选择的最小值；
    从所述业务调度参数的至少一个值中选择最小值作为所述语音业务的业务调度参数的值。
  7. 如权利要求2-6中任一项所述的方法，其特征在于，所述第一调度参数为MCS，
    所述根据上述三种调度参数中除所述第一调度参数外的另两种调度参数在第一协议子层分别对应的阈值范围内取值，得到所述第一协议子层的第一调度参数的至少一个值包括：
    根据在所述重传次数的阈值范围内的多个重传次数、所述丢弃计时器的时间的阈值范围内的多个丢弃计时器的时间、传输时间间隔绑定TTIB开关的开闭状态确定所述第一协议子层的至少一个分片数，其中，所述第一协议子层的每个分片数与一个重传次数和一个丢弃计时器的时间相对应；
    根据所述其他协议子层最大可缓存的数据大小、所述第一协议子层的至少一个分片数和所述第一协议子层的头数据大小确定所述第一协议子层的至少一个分片大小；
    根据所述第一协议子层的至少一个分片大小确定所述第一协议子层的至少一个MCS。
  8. 如权利要求2-7中任一项所述的方法，其特征在于，所述第一协议子层为无线链路控制RLC层，所述其他协议子层包括分组数据汇聚协议PDCP层和媒体接入控制MAC层。
  9. 如权利要求8所述的方法，其特征在于，所述第一调度参数包括以下任一种调度参数：MCS、重传次数、丢弃计时器的时间、PDCP缓存的数据大小、所述RLC层的分片数。
  10. 一种分组域语音业务调度的装置，其特征在于，包括：
    获取单元，用于获取语音业务的L2至少两个协议子层中每个协议子层 分别对应的第一调度参数的至少一个值；
    第一确定单元，用于确定所述语音业务的业务质量需要满足的条件；
    第二确定单元，用于根据所述第一确定单元确定的所述语音业务的业务质量需要满足的条件以及所述获取单元获取的所述语音业务的L2至少两个协议子层中每个协议子层分别对应的第一调度参数的至少一个值确定所述语音业务的业务调度参数的值；
    调度单元，用于根据所述第二确定单元确定的所述语音业务的业务调度参数的值调度所述语音业务。
  11. 如权利要求10所述的装置，其特征在于，所述第一调度参数包括以下任一种调度参数：调制编码方式MCS、重传次数和丢弃计时器的时间；
    所述获取单元具体用于根据上述三种调度参数中除所述第一调度参数外的另两种调度参数在第一协议子层分别对应的阈值范围内取值，得到所述第一协议子层的第一调度参数的至少一个值，并配置或根据所述第一协议子层的第一调度参数的至少一个值计算所述语音业务的L2至少两个协议子层中的其他协议子层分别对应的第一调度参数的至少一个值，其中，所述第一协作子层为所述语音业务的L2至少两个协议子层中的一个。
  12. 如权利要求11所述的装置，其特征在于，所述语音业务的业务质量需要满足的条件为所述语音业务的语音质量平均意见值MOS最优，所述第二确定单元具体用于根据所述第一协议子层的第一调度参数的至少一个值确定对应的所述第一协议子层的至少一个吞吐率，确定与所述第一协议子层的至少一个吞吐率一一对应的其他协议子层的至少一个吞吐率，根据所述第一协议子层的至少一个吞吐率和所述其他协议子层的至少一个吞吐率确定对应的至少一个所述语音业务的语音质量MOS，并从所述至少一个所述语音业务的语音质量MOS中选择最优的语音质量MOS对应的第一调度参数的值作为所述语音业务的业务调度参数的值。
  13. 如权利要求11所述的装置，其特征在于，所述语音业务的业务质量需要满足的条件为所述语音业务的吞吐率最大，所述第二确定单元具体用于根据所述第一协议子层的第一调度参数的至少一个值确定对应的所述第一协议子层的至少一个吞吐率，确定与所述第一协议子层的至少一个吞吐率一一对应的其他协议子层的至少一个吞吐率，根据所述第一协议子层的至少一个吞吐率和所述其他协议子层的至少一个吞吐率确定对应的所述语音业 务的至少一个吞吐率，从所述语音业务的至少一个吞吐率中选择最大的吞吐率对应的第一调度参数的值作为所述语音业务的业务调度参数的值。
  14. 如权利要求11所述的装置，其特征在于，所述语音业务的业务质量需要满足的条件为所述语音业务的传输速率最大，所述第二确定单元具体用于确定业务调度参数的至少一个值，其中，业务调度参数的每个值为从所述第一协议子层的第一调度参数的一个值和对应的其他各个协议子层的第一调度参数的一个值中选择的最大值，并从所述业务调度参数的至少一个值中选择最大值作为所述语音业务的业务调度参数的值。
  15. 如权利要求11所述的装置，其特征在于，所述语音业务的业务质量需要满足的条件为所述语音业务的误码率最小，所述第二确定单元具体用于确定业务调度参数的至少一个值，其中，业务调度参数的每个值为从所述第一协议子层的第一调度参数的一个值和对应的其他协议子层的第一调度参数的一个值中选择的最小值，并从所述业务调度参数的至少一个值中选择最小值作为所述语音业务的业务调度参数的值。
  16. 如权利要求11-15中任一项所述的装置，其特征在于所述第一调度参数为MCS，所述获取单元具体用于根据在所述重传次数的阈值范围内的多个重传次数、所述丢弃计时器的时间的阈值范围内的多个丢弃计时器的时间、传输时间间隔绑定TTIB开关的开闭状态确定所述第一协议子层的至少一个分片数，其中，所述第一协议子层的每个分片数与一个重传次数和一个丢弃计时器的时间相对应，根据所述其他协议子层最大可缓存的数据大小、所述第一协议子层的至少一个分片数和所述第一协议子层的头数据大小确定所述第一协议子层的至少一个分片大小，并根据所述第一协议子层的至少一个分片大小确定所述第一协议子层的至少一个MCS。
  17. 如权利要求11-16中任一项所述的装置，其特征在于，所述第一协议子层为无线链路控制RLC层，所述其他协议子层包括分组数据汇聚协议PDCP层和媒体接入控制MAC层。
  18. 如权利要求17所述的装置，其特征在于，所述第一调度参数包括以下任一种调度参数：MCS、重传次数、丢弃计时器的时间、PDCP缓存的数据大小、所述RLC层的分片数。
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Families Citing this family (2)

* Cited by examiner, † Cited by third party
Publication number Priority date Publication date Assignee Title
CN111510394B (zh) * 2019-01-31 2022-04-12 华为技术有限公司 一种报文调度方法、相关设备及计算机存储介质
US11304198B2 (en) * 2020-08-19 2022-04-12 Qualcomm Incorporated Packet data convergence protocol traffic monitoring

Citations (4)

* Cited by examiner, † Cited by third party
Publication number Priority date Publication date Assignee Title
WO2006135201A1 (en) * 2005-06-15 2006-12-21 Electronics And Telecommunications Research Institute Wireless communication system with protocol architecture for improving latency
CN101517994A (zh) * 2006-09-28 2009-08-26 高通股份有限公司 用于确定通信系统的服务质量的方法和装置
CN101945440A (zh) * 2010-09-03 2011-01-12 北京天碁科技有限公司 一种数据传输处理方法和系统
CN102752087A (zh) * 2012-07-09 2012-10-24 华中科技大学 一种基于amc-arq跨层的链路自适应方法

Family Cites Families (49)

* Cited by examiner, † Cited by third party
Publication number Priority date Publication date Assignee Title
US6542490B1 (en) * 1999-01-29 2003-04-01 Nortel Networks Limited Data link control proctocol for 3G wireless system
EP1198107B1 (en) * 2000-10-07 2012-11-28 LG Electronics Inc. Method for transmitting data from an RLC layer in a radio communication system
DE50106700D1 (de) * 2000-10-09 2005-08-11 Siemens Ag Verfahren zur übertragung von datenpaketen über eine luftschnittstelle eines mobilfunksystems
DE10107700A1 (de) * 2001-02-19 2002-08-29 Siemens Ag Verfahren und Vorrichtung zum Multiplexen und/oder Demultiplexen sowie entsprechende Computerprogramme und ein entsprechendes Computerprogramm-Erzeugnis
US7200154B1 (en) * 2001-05-23 2007-04-03 Nortel Networks Limited QoS link protocol (QLP)
US7254143B2 (en) * 2001-11-19 2007-08-07 Innovative Sonic Limited Local suspend scheme for wireless communication systems
DE60232023D1 (de) * 2002-02-19 2009-05-28 Alcatel Lucent Telekommunikationsendgerät, verfahren zum senden und empfangen und sendungssystem
FR2843269B1 (fr) * 2002-08-01 2004-10-15 Cit Alcatel Dispositif de prise de controle de ressources dans un reseau de communications, pour l'insertion de trafic
US7164890B2 (en) 2003-11-21 2007-01-16 Telefonaktiebologet Lm Ericsson (Publ) Link adaptation for point-to-multipoint channel
US8111698B2 (en) * 2005-03-31 2012-02-07 Alcatel Lucent Method of performing a layer operation in a communications network
US7843991B2 (en) 2005-04-21 2010-11-30 Qualcomm Incorporated Methods and apparatus for monitoring voice quality on a wireless communication device
US8804626B2 (en) * 2005-05-10 2014-08-12 Ntt Docomo, Inc. Transmission rate control method, mobile station, radio network controller, and radio base station
US8289952B2 (en) 2005-05-25 2012-10-16 Telefonaktiebolaget Lm Ericsson (Publ) Enhanced VoIP media flow quality by adapting speech encoding based on selected modulation and coding scheme (MCS)
US7929410B2 (en) * 2005-06-29 2011-04-19 Interdigital Technology Corporation Protocol engine for processing data in a wireless transmit/receive unit
WO2007130325A2 (en) * 2006-05-01 2007-11-15 Interdigital Technology Corporation Method and apparatus for facilitating lossless handover in 3gpp long term evolution systems
JP4850600B2 (ja) 2006-06-29 2012-01-11 京セラ株式会社 無線端末装置及びチャネル制御方法
WO2008040021A1 (en) 2006-09-28 2008-04-03 Qualcomm Incorporated Methods and apparatus for determining quality of service in a communication system
CN102573086A (zh) * 2007-03-01 2012-07-11 株式会社Ntt都科摩 基站装置和通信控制方法
KR101513033B1 (ko) * 2007-09-18 2015-04-17 엘지전자 주식회사 다중 계층 구조에서 ＱｏＳ를 보장하기 위한 방법
KR101505687B1 (ko) * 2007-11-05 2015-03-31 엘지전자 주식회사 무선통신 시스템에서 전력제어 방법
JP5194844B2 (ja) 2008-01-31 2013-05-08 富士通株式会社 無線通信装置及び通信方法
US8441934B2 (en) * 2008-11-11 2013-05-14 Qualcomm Incorporated Efficient UE QoS/UL packet build in LTE
CN101925088B (zh) 2009-06-12 2015-02-18 株式会社Ntt都科摩 一种基于用户体验质量测量的编码方法、终端和系统
US9668083B2 (en) * 2011-12-22 2017-05-30 Taiwan Semiconductor Manufacturing Co., Ltd. Systems and methods for cooperative applications in communication systems
EP2451215A4 (en) * 2009-07-01 2015-02-25 Fujitsu Ltd Radio communication control and radio communication device
US20120033563A1 (en) * 2010-08-05 2012-02-09 Apple Inc. Packet classification and prioritization using an ip header in a mobile wireless device
US8620372B2 (en) * 2010-09-30 2013-12-31 Apple Inc. Method and apparatus for transport format selection in a mobile wireless device
CN102571265B (zh) * 2011-01-04 2015-11-25 华为技术有限公司 数据传输方法和发送设备
US10075969B2 (en) * 2011-04-25 2018-09-11 Texas Instruments Incorporated Medium access control schedulers for wireless communication
KR101568921B1 (ko) 2011-05-27 2015-11-12 후아웨이 테크놀러지 컴퍼니 리미티드 음성 신호 처리 방법과 장치, 및 액세스 네트워크 시스템
CN102883441B (zh) * 2011-07-15 2015-04-22 华为技术有限公司 一种无线宽带通信方法和装置
CN105357773B (zh) * 2011-07-15 2020-06-02 华为技术有限公司 一种无线宽带通信方法，装置和系统
US20140155043A1 (en) * 2011-12-22 2014-06-05 Cygnus Broadband, Inc. Application quality management in a communication system
US20140153392A1 (en) * 2011-12-22 2014-06-05 Cygnus Broadband, Inc. Application quality management in a cooperative communication system
CN104126281B (zh) * 2012-01-11 2018-06-15 诺基亚通信公司 用于站间载波聚合的辅小区准备的方法和装置
US9246842B2 (en) * 2012-04-27 2016-01-26 Intel Corporation QoE-aware radio access network architecture for http-based video streaming
CN103716885B (zh) * 2012-09-28 2017-09-29 电信科学技术研究院 一种本地承载管理方法及设备
US9420511B2 (en) * 2012-11-01 2016-08-16 Intel Corporation Signaling QoS requirements and UE power preference in LTE-A networks
TWI470974B (zh) * 2013-01-10 2015-01-21 Univ Nat Taiwan 多媒體資料傳輸速率調節方法及網路電話語音資料傳輸速率調節方法
EP2995090A1 (en) * 2013-05-07 2016-03-16 VID SCALE, Inc. Qoe-aware wifi enhancements for video for video applications
US10091707B2 (en) * 2014-03-21 2018-10-02 Lg Electronics Inc. Method and apparatus for indicating D2D related information in wireless communication system
WO2015147605A1 (en) * 2014-03-28 2015-10-01 Lg Electronics Inc. Method and apparatus for performing d2d operation in wireless communication system
WO2016013744A1 (en) * 2014-07-24 2016-01-28 Lg Electronics Inc. Method and apparatus for transmitting uplink data in wireless communication system
US9577793B2 (en) * 2014-09-23 2017-02-21 Intel Corporation Soft buffer management
WO2017062627A1 (en) * 2015-10-06 2017-04-13 Kodiak Networks, Inc. System and method for improved push-to-talk communication performance
US20170118707A1 (en) * 2015-10-21 2017-04-27 Qualcomm Incorporated Techniques for achieving optimal power and mean opinion scores for internet protocol multimedia subsystem-based dual connectivity calling
US10555368B2 (en) * 2015-11-19 2020-02-04 Telefonaktiebolaget Lm Ericsson (Publ) Active period of a discontinuous reception cycle
US10143002B2 (en) * 2016-01-12 2018-11-27 Cisco Technology, Inc. System and method to facilitate centralized radio resource management in a split radio access network environment
US10959240B2 (en) * 2016-04-15 2021-03-23 Qualcomm Incorporated Providing quality-of-service in wireless communications

Patent Citations (4)

* Cited by examiner, † Cited by third party
Publication number Priority date Publication date Assignee Title
WO2006135201A1 (en) * 2005-06-15 2006-12-21 Electronics And Telecommunications Research Institute Wireless communication system with protocol architecture for improving latency
CN101517994A (zh) * 2006-09-28 2009-08-26 高通股份有限公司 用于确定通信系统的服务质量的方法和装置
CN101945440A (zh) * 2010-09-03 2011-01-12 北京天碁科技有限公司 一种数据传输处理方法和系统
CN102752087A (zh) * 2012-07-09 2012-10-24 华中科技大学 一种基于amc-arq跨层的链路自适应方法

Non-Patent Citations (1)

* Cited by examiner, † Cited by third party
Title
See also references of EP3448083A4 *

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