








技术领域technical field
本发明涉及通信技术,尤其涉及通信领域中的自适应多速率分组语音编AMR VoIP码模式调整技术。The invention relates to communication technology, in particular to an adaptive multi-rate packet voice encoding AMR VoIP code mode adjustment technology in the communication field.
背景技术Background technique
传统的电话网是以电路交换方式传输语音,所要求的传输宽带为64kbit/s。且由于电话业务历来都是各国管制最为严格的业务,而各国国际长途电话费存在着严重的不平衡性,国际长途电话业务在很多国家都是垄断经营的,所以,随着因特网Internet的发展,在Internet上实现语音通话成为一种趋势。The traditional telephone network transmits voice in the way of circuit switching, and the required transmission bandwidth is 64kbit/s. And because the telephone service has always been the most strictly regulated business in various countries, and there is a serious imbalance in the international long-distance telephone charges in various countries, the international long-distance telephone service is monopolized in many countries. Therefore, with the development of the Internet, It has become a trend to implement voice calls on the Internet.
随着技术的发展,已实现Internet和已有的公共电话交换网(PSTN,PublicSwitched Telephone Network)结合,使得IP(Internet Protocol)电话从当初的PC(Personal Computer)到PC发展到PC到PC、PC到电话、电话到电话等多种业务形式,以及向IP传输多媒体业务过渡。但不论怎样,IP电话承载网络是Internet,或是遵循TCP/IP协议的专用网或Internet。With the development of technology, the combination of the Internet and the existing PSTN (Public Switched Telephone Network) has been realized, making the IP (Internet Protocol) telephone develop from the original PC (Personal Computer) to PC to PC to PC, PC to telephone, telephone to telephone and other business forms, as well as the transition to IP transmission of multimedia services. But no matter what, the IP telephony bearer network is the Internet, or a dedicated network or the Internet following the TCP/IP protocol.
如在通用移动通信系统(UMTS,Universal Mobile TelecommunicationsSystem)中,UMTS是采用WCDMA(Wideband Code Division Multiple Access,宽带码分多址接入)空中接口技术的第三代移动通信系统。For example, in Universal Mobile Telecommunications System (UMTS, Universal Mobile Telecommunications System), UMTS is a third-generation mobile communication system that adopts WCDMA (Wideband Code Division Multiple Access, Wideband Code Division Multiple Access) air interface technology.
如图1所示,UMTS系统包括RAN(Radio Access Network,无线接入网)和CN(Core Network,核心网)。其中RAN用于处理所有与无线有关的过程,而CN处理UMTS系统内所有的话音呼叫和数据连接,并实现与外部网络的交换和路由功能。CN从逻辑上分为CS(Circuit Switched Domain,电路交换域)和PS(Packet Switched Domain,分组交换域),CS一般包括MSC(MobileSwitching Center,移动交换中心)/VLR(Visitor Location Register,拜访位置寄存器)、GMSC(Gateway Mobile Switching Center,网关移动业务交换中心)、gsmSSF,PS一般包括SGSN(Serving GPRS Support Node,服务GPRS支持节点)、GGSN(Gateway GPRS Support Node,网关GPRS支持节点);CS主要处理有关电话、语音等语音业务,PS则处理有关分组数据业务。UTRAN(UMTSTerritorial Radio Access Network UMTS,陆地无线接入网)、CN与UE(UserEquipment,用户终端)一起构成了整个UMTS系统,UMTS系统连接外部网络,例如:PSTN和互联网。As shown in Figure 1, the UMTS system includes RAN (Radio Access Network, wireless access network) and CN (Core Network, core network). Among them, RAN is used to handle all wireless-related processes, while CN handles all voice calls and data connections in the UMTS system, and implements switching and routing functions with external networks. CN is logically divided into CS (Circuit Switched Domain, Circuit Switched Domain) and PS (Packet Switched Domain, Packet Switched Domain). CS generally includes MSC (Mobile Switching Center, Mobile Switching Center)/VLR (Visitor Location Register, Visitor Location Register) ), GMSC (Gateway Mobile Switching Center, gateway mobile service switching center), gsmSSF, PS generally includes SGSN (Serving GPRS Support Node, serving GPRS support node), GGSN (Gateway GPRS Support Node, gateway GPRS support node); CS mainly handles Regarding voice services such as telephone and voice, PS handles relevant packet data services. UTRAN (UMTS Territorial Radio Access Network UMTS, Terrestrial Radio Access Network), CN and UE (User Equipment, user terminal) together constitute the entire UMTS system, and the UMTS system is connected to external networks, such as PSTN and the Internet.
上述的陆地无线接入网UTRAN,其网络结构框图如图2所示,包含至少一个RNS(Radio Network Subsystem,无线网络子系统),一个RNS由一个RNC(Radio Network Controller,无线网络控制器)和至少一个NodeB(基站)组成,NodeB可覆盖至少一个小区CELL。The above-mentioned terrestrial radio access network UTRAN has a network structure diagram as shown in Figure 2, including at least one RNS (Radio Network Subsystem, radio network subsystem), and one RNS is composed of an RNC (Radio Network Controller, radio network controller) and It consists of at least one NodeB (base station), and the NodeB can cover at least one cell CELL.
NodeB是WCDMA系统的基站(即无线收发信机),包括无线收发信机和基带处理部件。通过标准的Iub接口和RNC互连,主要完成Uu接口物理层协议的处理,主要功能是扩频、调制、信道编码及解扩、解调、信道解码,还包括基带信号和射频信号的相互转换等功能。NodeB is the base station (that is, the wireless transceiver) of the WCDMA system, including the wireless transceiver and baseband processing components. Through the standard Iub interface and RNC interconnection, it mainly completes the processing of the Uu interface physical layer protocol. The main functions are spread spectrum, modulation, channel coding and despreading, demodulation, channel decoding, and also include the mutual conversion of baseband signals and radio frequency signals. and other functions.
RNC是无线网络控制器,用于控制UTRAN的无线资源,主要完成连接建立和断开、切换、宏分集合并、无线资源管理控制等功能。The RNC is a radio network controller, which is used to control the radio resources of the UTRAN, and mainly completes functions such as connection establishment and disconnection, handover, macro-diversity combination, and radio resource management and control.
在RNC和NodeB之间的Iub接口,一般使用多个AAL2PVC承载UE的数据,这些数据包括UE的CS语音、PS数据。The Iub interface between the RNC and the NodeB generally uses multiple AAL2PVCs to bear the data of the UE, and these data include the CS voice and PS data of the UE.
通过Internet进行语音通信是一个非常复杂的系统工程,所涉及的技术也较多,其中最根本的技术是分组语音(VoIP,Voice over IP)技术。Carrying out voice communication through the Internet is a very complicated system engineering, and involves many technologies, the most fundamental technology of which is packet voice (VoIP, Voice over IP) technology.
VoIP是以IP分组交换网络为传输平台,透过IP网络传输的语音讯号或影像讯号的技术。其基本原理是:通过语音压缩算法对语音数据进行压缩编码处理,然后把这些语音数据按IP等相关协议进行打包,经过IP网络把数据包传输到接收地,再把这些语音数据包串起来,经过解码解压处理后,恢复成原来的语音信号,从而达到由IP网络传送语音的目的。IP电话系统把普通电话的模拟信号转换成计算机可联入因特网传送的IP数据包,同时也将收到的IP数据包转换成声音的模拟电信号。经过IP电话系统的转换及压缩处理,每个普通电话传输速率约占用8~11Kbit/s带宽,因此在与普通电信网同样使用传输速率为64kbit/s的带宽时,IP电话数是原来的5~8倍。VoIP is a technology that uses the IP packet switching network as the transmission platform to transmit voice signals or video signals through the IP network. The basic principle is: compress and code the voice data through the voice compression algorithm, then package the voice data according to IP and other related protocols, transmit the data packets to the receiving place through the IP network, and then string these voice data packets together. After decoding and decompressing, it returns to the original voice signal, so as to achieve the purpose of transmitting voice through the IP network. The IP telephone system converts the analog signal of an ordinary telephone into an IP data packet that can be connected to the Internet by a computer, and at the same time converts the received IP data packet into an analog electrical signal of sound. After the conversion and compression processing of the IP telephone system, the transmission rate of each ordinary telephone occupies about 8-11Kbit/s bandwidth, so when using the same bandwidth as the ordinary telecommunication network with a transmission rate of 64kbit/s, the number of IP telephones is the original 5 to 8 times.
目前在宽带码分多址(WCDMA,Wideband Code Division Multiple Access)系统中,语音采用自适应多速率(AMR,Adaptive Multi-Rate)压缩编码,然后转换为IP数据包在IP网络上进行传输。AMR VoIP编码是一种自适应的编码方法,可以产生8种不同的模式,每一种模式对应于一种速率:12.2、10.2、7.95、7.4、6.7、5.9、5.15和4.75kbit/s,也即不同的模式可以提供不同的语音质量。在块误码率(BLER,Block error rate)小于等于1%的条件下,模式越高,提供的语音质量越高,但是占用的传输信道带宽资源(包括负载资源和Iub资源)也越多。At present, in the Wideband Code Division Multiple Access (WCDMA, Wideband Code Division Multiple Access) system, voice is compressed and encoded by Adaptive Multi-Rate (AMR, Adaptive Multi-Rate), and then converted into IP data packets for transmission on the IP network. AMR VoIP coding is an adaptive coding method that can generate 8 different modes, each mode corresponds to a rate: 12.2, 10.2, 7.95, 7.4, 6.7, 5.9, 5.15 and 4.75kbit/s, also That is, different modes can provide different voice quality. Under the condition that the block error rate (BLER, Block error rate) is less than or equal to 1%, the higher the mode, the higher the voice quality provided, but the more occupied transmission channel bandwidth resources (including load resources and Iub resources).
VoIP业务流程可以如图3所示,在进行无线接入承载(RAB,Radio AccessBearer)建立时,首先由CN向UTRAN发送RAB指配请求消息,请求UTRAN建立RAB。RAB指配建立IMS信令承载,然后主叫用户终端和被叫用户终端通过各自的UTRAN、PS域,根据IMS信令信道进行会话控制协议SIP/SDP会话,主要是协商编码速率等;最后是建立业务承载,核心网CN会指配相应的服务质量(QoS,Quality of Service)参数,无线网络控制器(RNC,Radio NetworkController)根据不同模式的QoS,为VoIP业务分配相应的带宽资源,双方UE进行多媒体业务RTP(Real-time Transport Protocol)/实时业务控制协议RTCP会话。The VoIP service process can be shown in Figure 3. When establishing a radio access bearer (RAB, Radio Access Bearer), first, the CN sends a RAB assignment request message to the UTRAN to request the UTRAN to establish the RAB. RAB assignment establishes IMS signaling bearer, and then the calling user terminal and called user terminal conduct session control protocol SIP/SDP sessions according to the IMS signaling channel through their respective UTRAN and PS domains, mainly to negotiate the coding rate, etc.; finally To establish a service bearer, the core network CN will assign corresponding quality of service (QoS, Quality of Service) parameters, and the radio network controller (RNC, Radio Network Controller) will allocate corresponding bandwidth resources for VoIP services according to different modes of QoS. Conduct multimedia service RTP (Real-time Transport Protocol)/real-time service control protocol RTCP session.
RTP协议是基于用户终端数据报协议(UDP,User Datagram Protocol)/IP提供的承载,如图4所示,为WDMA中承载语音流的IP/UDP/RTP报文结构示意图,总共有60个字节(IPv6)或40个字节(IPv4)的头部,其中IP头部40个字节(IPv6)或20个字节(IPv4),UDP头部8个和RTP头部12个字节。The RTP protocol is based on the bearer provided by the User Datagram Protocol (UDP, User Datagram Protocol)/IP. As shown in Figure 4, it is a structural diagram of the IP/UDP/RTP message carrying the voice stream in WDMA, with a total of 60 characters Section (IPv6) or 40-byte (IPv4) header, where the IP header is 40 bytes (IPv6) or 20 bytes (IPv4), the UDP header is 8 and the RTP header is 12 bytes.
在图4中,对应3G PS终端,RTP PAYLOAD格式为:payload header|tableof contents|speech data,其中payload header为对端UE发给本端UE的编码模式请求CMR,现有技术中,本端只能选择小于等于CMR的模式;speech data为传送的AMR语音帧,现有技术中对于3G PS终端,每一个RTP包只封装一个语音帧;table of contents主要包含该语音帧的模式信息及帧质量指示信息。In Figure 4, corresponding to a 3G PS terminal, the RTP PAYLOAD format is: payload header|tableof contents|speech data, where the payload header is the encoding mode request CMR sent by the peer UE to the local UE. In the prior art, the local terminal only A mode less than or equal to CMR can be selected; speech data is the transmitted AMR voice frame. For 3G PS terminals in the prior art, each RTP packet only encapsulates one voice frame; table of contents mainly includes the mode information and frame quality of the voice frame Instructions.
在现有的数据传输过程中,RNC对语音业务的RTP包是进行透传的,即仅把RTP/UDP/IP包(或是压缩后的RTP/UDP/IP包)作为一个数据包在空口进行传输,并不要求当前的模式信息。但在通讯过程中,由于传输环境和位置等改变而可能会导致信道质量下降,或当系统负载、Iub资源接近饱和,这时候如果仍然采用高模式传输势必对语音质量造成影响,而且还会加重负载和Iub资源的拥塞,造成部分用户终端的业务受损,影响QoS。In the existing data transmission process, the RNC transparently transmits the RTP packet of the voice service, that is, only the RTP/UDP/IP packet (or compressed RTP/UDP/IP packet) is used as a data packet on the air interface For transmission, current mode information is not required. However, during the communication process, the channel quality may decrease due to changes in the transmission environment and location, or when the system load and Iub resources are close to saturation, if the high-mode transmission is still used at this time, the voice quality will be affected, and it will be aggravated The load and the congestion of the Iub resource cause service damage of some user terminals and affect QoS.
发明内容Contents of the invention
有鉴于此,本发明的实施方式提供一种AMR VoIP模式调整方法及基站控制器,使得当资源接近饱和时,用户终端仍保持一定的语音质量或最大限度地延缓负载、基站控制器与基站的接口资源拥塞。In view of this, the embodiments of the present invention provide an AMR VoIP mode adjustment method and a base station controller, so that when the resource is close to saturation, the user terminal still maintains a certain voice quality or delays the load, the base station controller and the base station to the greatest extent. Interface resources are congested.
一种AMR VoIP模式调整方法,其中,包括:获取当前发射功率、基于功率的负载、基于基站控制器与基站的接口负载;确定当前发射功率大于等于预设的发射功率门限、基于功率的负载大于等于预设的基于功率的负载门限、基站控制器与基站的接口负载大于等于预设的接口负载门限的至少之一满足时,基站控制器通过降低RTP数据包中的编码模式请求CMR模式以降低AMR VoIP的编码模式。一种基站控制器,其中,该基站控制器包括AMR VoIP模式存储单元、AMR VoIP模式选择单元、AMR VoIP模式调整单元、判断单元,AMR VoIP模式存储单元在进行SIP/SDP会话时,存储双方用户终端支持的AMR VoIP编码模式集,在数据传输过程中,判断单元判断当前发射功率大于等于预设的发射功率门限、基于功率的负载大于等于预设的基于功率的负载门限、基站控制器与基站的接口负载大于等于预设的接口负载门限的至少之一满足时,通知AMR VoIP模式选择单元从AMR VoIP模式存储单元选择小于当前所用AMR模式的第一AMR模式,AMR VoIP模式调整单元将RTP数据包中的CMR模式降低为第一AMR模式。A method for adjusting an AMR VoIP mode, including: obtaining current transmission power, power-based load, and interface load based on a base station controller and a base station; determining that the current transmission power is greater than or equal to a preset transmission power threshold, and that the power-based load is greater than or equal to When at least one of the preset power-based load threshold and the interface load between the base station controller and the base station is greater than or equal to the preset interface load threshold is met, the base station controller requests the CMR mode by reducing the encoding mode in the RTP data packet to reduce Coding mode for AMR VoIP. A base station controller, wherein the base station controller includes an AMR VoIP mode storage unit, an AMR VoIP mode selection unit, an AMR VoIP mode adjustment unit, and a judging unit, and the AMR VoIP mode storage unit stores both users when performing a SIP/SDP session. The AMR VoIP encoding mode set supported by the terminal, in the process of data transmission, the judging unit judges that the current transmit power is greater than or equal to the preset transmit power threshold, the power-based load is greater than or equal to the preset power-based load threshold, the base station controller and the base station When at least one of the preset interface load thresholds is satisfied, the AMR VoIP mode selection unit is notified to select the first AMR mode less than the currently used AMR mode from the AMR VoIP mode storage unit, and the AMR VoIP mode adjustment unit converts the RTP data The CMR mode in the package is reduced to the first AMR mode.
与现有技术相比,上述的技术方案,首先获取当前发射功率、基于功率的负载、基于基站控制器与基站的接口负载信息,然后当确定发射功率大于等于门限、基于功率的负载大于等于门限、基站控制器与基站的接口负载大于等于门限的至少之一满足时,基站控制器降低AMR VoIP编码模式,所以使得当信道质量下降或系统负载、基站控制器与基站的接口负载资源接近饱和时,用户终端仍保持一定的语音质量或最大限度地延缓负载、基站控制器与基站的接口资源拥塞。Compared with the prior art, the above technical solution first obtains the current transmission power, power-based load, and load information based on the interface between the base station controller and the base station, and then when it is determined that the transmission power is greater than or equal to the threshold and the power-based load is greater than or equal to the threshold 1. When at least one of the interface load between the base station controller and the base station is greater than or equal to the threshold, the base station controller reduces the AMR VoIP coding mode, so that when the channel quality drops or the system load, the interface load resource between the base station controller and the base station is close to saturation , the user terminal still maintains a certain voice quality or delays the load to the greatest extent, and the resource congestion of the interface between the base station controller and the base station.
附图说明Description of drawings
图1为现有技术之UMTS系统的网络结构框图。FIG. 1 is a block diagram of a network structure of a UMTS system in the prior art.
图2为现有技术之UTRAN的网络结构框图。FIG. 2 is a block diagram of the network structure of the UTRAN in the prior art.
图3为现有技术之VoIP业务流程建立示意图。FIG. 3 is a schematic diagram of establishing a VoIP service flow in the prior art.
图4为现有技术之WDMA中承载语音流的IP/UDP/RTP报文结构示意图。FIG. 4 is a schematic diagram of the structure of IP/UDP/RTP packets carrying voice streams in WDMA in the prior art.
图5为本发明较佳第一实施方式之AMR VoIP编码模式调整过程框图。Fig. 5 is a block diagram of the AMR VoIP coding mode adjustment process of the preferred first embodiment of the present invention.
图6为本发明较佳实施方式之A事件上报示意图。FIG. 6 is a schematic diagram of event A reporting in a preferred embodiment of the present invention.
图7为本发明较佳实施方式之流量B事件上报示意图。FIG. 7 is a schematic diagram of reporting traffic B events in a preferred embodiment of the present invention.
图8为本发明较佳第二实施方式之AMR VoIP编码模式调整过程框图。Fig. 8 is a block diagram of the AMR VoIP coding mode adjustment process of the preferred second embodiment of the present invention.
图9为本发明较佳实施方式之AMR模式动态调整系统结构框图。FIG. 9 is a structural block diagram of an AMR mode dynamic adjustment system in a preferred embodiment of the present invention.
具体实施方式Detailed ways
为使本发明的目的、技术方案和优点更加清楚明白,以下结合具体实施方式及附图,对本发明作进一步详细的说明。In order to make the object, technical solution and advantages of the present invention clearer, the present invention will be further described in detail below in conjunction with specific implementation methods and accompanying drawings.
本发明实施例的技术方案可以用于多种通信系统,如可以用于码分多址接入系统(CDMA,Code Division Multiple Access)、宽带码分多址接入系统(WCDMA,Wideband Code Division Multiple Access)、全球移动通信系统(GSM,Global System for Mobile communications)、通用分组无线业务(GPRS,General Packet Radio Service)等通信系统。在各个通信系统中,数据发送可分为上行发送和下行发送,上行是指发送方是用户终端,接收方为基站;下行发送是指发送方是基站,接收方为用户终端。The technical scheme of the embodiment of the present invention can be used in various communication systems, as can be used in code division multiple access system (CDMA, Code Division Multiple Access), wideband code division multiple access system (WCDMA, Wideband Code Division Multiple Access system) Access), Global System for Mobile communications (GSM, Global System for Mobile communications), General Packet Radio Service (GPRS, General Packet Radio Service) and other communication systems. In various communication systems, data transmission can be divided into uplink transmission and downlink transmission. Uplink means that the sender is a user terminal and the receiver is a base station; downlink means that the sender is a base station and the receiver is a user terminal.
但为更方便说明本发明之技术方案,下述仅以WCDMA系统的上行、下行的传输过程为例进行说明。However, in order to illustrate the technical solution of the present invention more conveniently, the following only takes the uplink and downlink transmission processes of the WCDMA system as an example for illustration.
在WCDMA系统中,RNC配置能反应系统资源拥塞的一些测量门限(其它通信系统如CDMA、GSM,由基站控制器配置),如UE和NodeB的发射功率门限,预设A、B事件的迟滞时间(Hysteresis Time)和A、B事件门限(Requestedthreshold);如基于功率的负载门限或基于Iub的负载门限(具体为带宽占有率门限)等。In the WCDMA system, RNC configures some measurement thresholds that can reflect system resource congestion (other communication systems such as CDMA and GSM are configured by the base station controller), such as the transmit power threshold of UE and NodeB, and the delay time of preset A and B events (Hysteresis Time) and A, B event threshold (Requested threshold); such as power-based load threshold or Iub-based load threshold (specifically bandwidth occupancy threshold), etc.
不同的信道质量值映射不同的AMR VoIP编码模式,代表当前用户终端在信道上可接收数据的能力,NodeB在调度用户终端时将考察每个用户终端的信道质量值,信道质量较低的用户终端被分配到的信道资源就较少,AMR VoIP编码模式也较低。Different channel quality values map different AMR VoIP coding modes, which represent the ability of the current user terminal to receive data on the channel. NodeB will examine the channel quality value of each user terminal when scheduling user terminals. User terminals with lower channel quality The allocated channel resources are less, and the AMR VoIP coding mode is also lower.
如图5所示,为本发明较佳第一实施方式的一种上行AMR模式动态调整过程,主要如下所述。As shown in FIG. 5 , it is a dynamic adjustment process of an uplink AMR mode in a preferred first embodiment of the present invention, which is mainly described as follows.
步骤111,NodeB测量当前信道质量和UE的发射功率,RNC测量当前Iub负载;
RNC向NodeB发出测量控制命令,NodeB根据测量控制命令测量当前信道质量信息,可以通过A、B事件上报给RNC,其中,A事件反映当前信道质量好,B事件反映当前信道质量差。The RNC sends a measurement control command to the NodeB, and the NodeB measures the current channel quality information according to the measurement control command, which can be reported to the RNC through A and B events, where the A event reflects the current channel quality is good, and the B event reflects the current channel quality is poor.
A、B事件可以采用滞后的A、B事件上报的方法,参见图6,图7所示。The A and B events can be reported by lagged A and B events, as shown in Fig. 6 and Fig. 7 .
在图6中,信道质量高于预先设定的A事件门限值(可以是信道质量的上限门限值)的时间达到预先设定的迟滞时间,此时,NodeB进行A事件报告,说明此时信道质量比较好,图6中示出两次A事件报告示例。In Fig. 6, the time when the channel quality is higher than the preset A event threshold value (which may be the upper limit threshold value of the channel quality) reaches the preset delay time, and at this time, NodeB reports the A event, indicating that When the channel quality is relatively good, Figure 6 shows two examples of event A reports.
在图7中,信道质量低于预先设定的B事件门限值(可以是信道质量的下限门限值)的时间达到预先设定的迟滞时间,此时,NodeB进行B事件报告,说明此时信道质量比较差,图7中示出了两次B事件报告示例。In Fig. 7, the time when the channel quality is lower than the preset B event threshold value (which may be the lower limit threshold value of the channel quality) reaches the preset delay time, at this time, NodeB reports the B event, illustrating that When the channel quality is relatively poor, Fig. 7 shows two examples of B event reports.
RNC给NodeB配置测量周期,NodeB根据测量控制命令周期性地向RNC上报UE的发射功率,当RNC确定UE发射功率大于等于预设的发射功率门限时,说明系统资源负载紧张。或当RNC确定基于功率的负载资源大于等于门限时,说明系统资源负载紧张。The RNC configures the measurement period for the NodeB, and the NodeB periodically reports the transmit power of the UE to the RNC according to the measurement control command. When the RNC determines that the transmit power of the UE is greater than or equal to the preset transmit power threshold, it indicates that the system resource load is tight. Or when the RNC determines that the power-based load resource is greater than or equal to the threshold, it indicates that the system resource load is tight.
RNC根据所配置的带宽占有率门限监控Iub接口负载使用情况,一旦发现带宽占有率大于等于带宽占有率门限,也就是Iub负载大于等于预设的Iub负载门限时,即告警,说明Iub资源紧张。RNC monitors the Iub interface load usage situation according to the configured bandwidth occupancy threshold, once it is found that the bandwidth occupancy is greater than or equal to the bandwidth occupancy threshold, that is, when the Iub load is greater than or equal to the preset Iub load threshold, it will give an alarm, indicating that the Iub resources are in short supply.
步骤112,确定UE发射功率大于等于预设的发射功率门限、基于功率的负载资源大于等于门限、基于Iub的负载资源大于等于门限的至少之一满足时,RNC降低AMR VoIP编码模式;
本较佳实施方式中,在IMS信令承载建立后,主被叫双方UE进行SIP/SDP会话,RNC通过解SIP/SDP包获得双方UE支持的AMR VoIP编码模式集,则在数据传输过程中,RNC周期性地根据当前UE的发射功率、基于功率的负载资源、基于Iub的负载资源至少之一进行AMR VoIP模式动态调整。In this preferred embodiment, after the IMS signaling bearer is set up, both UEs of the calling party and the called party carry out a SIP/SDP session, and the RNC obtains the AMR VoIP coding mode set supported by the UEs of both parties by deciphering the SIP/SDP packet. , the RNC periodically performs dynamic adjustment of the AMR VoIP mode according to at least one of the transmit power of the current UE, power-based load resources, and Iub-based load resources.
在数据传输过程中,当RNC确定UE的发射功率大于等于预设的发射功率门限时;或当RNC确定基于功率的负载资源大于等于门限时;或RNC确定基于Iub的负载资源大于等于门限;或进一步在数据传输过程中,RNC将NodeB上报的UE发射功率转换成专用物理数据信道(DPDCH,Dedicated Physical DataChannel)平均发射功率,然后和发射功率门限值进行比较,判断发射功率大于等于发射功率门限值;只要上述满足上述至少之一,就执行下述操作:During data transmission, when the RNC determines that the transmit power of the UE is greater than or equal to the preset transmit power threshold; or when the RNC determines that the power-based load resource is greater than or equal to the threshold; or the RNC determines that the Iub-based load resource is greater than or equal to the threshold; or Further in the data transmission process, the RNC converts the UE transmit power reported by the NodeB into the average transmit power of a dedicated physical data channel (DPDCH, Dedicated Physical DataChannel), and then compares it with the transmit power threshold value to determine that the transmit power is greater than or equal to the transmit power gate Limits; as long as at least one of the above is met, do the following:
降低RTP数据包中的CMR模式,该降低的CMR模式从UE所支持的CMR模式集中选择,CMR模式的降低可以是逐级降低,也可以根据所测量的结果越级降低,也可以直接降低为UE所支持模式集中的最小模式。Reduce the CMR mode in the RTP data packet. The reduced CMR mode is selected from the set of CMR modes supported by the UE. The reduction of the CMR mode can be reduced step by step, or it can be reduced step by step according to the measured results, or it can be directly reduced to UE The smallest pattern in the set of supported patterns.
上述描述的AMR VoIP模式动态调整过程是在上行传输过程的场景下,在下行传输过程的场景下,稍有点不同,具体如下所述。The dynamic adjustment process of the AMR VoIP mode described above is slightly different in the scenario of the uplink transmission process and in the scenario of the downlink transmission process, as described below.
如图8所示,为本发明较佳第二实施方式的一种下行AMR模式动态调整过程,主要如下所述。As shown in FIG. 8 , it is a dynamic adjustment process of the downlink AMR mode in the preferred second embodiment of the present invention, which is mainly described as follows.
步骤121,UE反馈当前信道质量、NodeB发射功率,RNC测量当前Iub负载;
NodeB周期性地广播导频信号,UE测量导频获得当前信道质量信息,并向NodeB反馈信道质量指示(CQI,Channel Quality Indication)。当前信道质量较好时,Node上报CQI A事件;当前信道质量较差时,上报CQI B事件,CQI A事件和CQI B事件的上报方式可以采用类似上述较佳第一实施方式中的A、B事件的上报方式。The NodeB broadcasts the pilot signal periodically, and the UE measures the pilot to obtain current channel quality information, and feeds back a Channel Quality Indication (CQI, Channel Quality Indication) to the NodeB. When the current channel quality is good, the Node reports the CQI A event; when the current channel quality is poor, it reports the CQI B event, and the reporting methods of the CQI A event and the CQI B event can be similar to A and B in the above-mentioned preferred first implementation mode How the event is reported.
UE在接收数据同时检测NodeB发射功率,并向NodeB反馈,RNC给NodeB配置周期,NodeB周期性地上报NodeB发射功率。当RNC确定NodeB发射功率大于等于预设的发射功率门限时,说明系统资源负载紧张。或当RNC确定基于功率的负载资源大于等于门限时,说明系统资源负载紧张。The UE detects the NodeB transmit power while receiving data, and feeds back to the NodeB. The RNC configures a period for the NodeB, and the NodeB periodically reports the NodeB transmit power. When the RNC determines that the NodeB transmit power is greater than or equal to the preset transmit power threshold, it indicates that the system resource load is tight. Or when the RNC determines that the power-based load resource is greater than or equal to the threshold, it indicates that the system resource load is tight.
RNC根据所配置的带宽占有率门限监控Iub接口负载使用情况,一旦发现带宽占有率大于等于带宽占有率门限,也就是Iub负载大于等于预设的Iub负载门限时,即告警,说明Iub资源紧张。RNC monitors the Iub interface load usage situation according to the configured bandwidth occupancy threshold, once it is found that the bandwidth occupancy is greater than or equal to the bandwidth occupancy threshold, that is, when the Iub load is greater than or equal to the preset Iub load threshold, it will give an alarm, indicating that the Iub resources are in short supply.
步骤122,确定NodeB发射功率大于等于预设的发射功率门限、基于功率的负载资源大于等于门限、基于Iub的负载资源大于等于门限的至少之一满足时,RNC降低AMR VoIP编码模式;
本较佳实施方式中,在IMS信令承载建立后,主被叫双方UE进行SIP/SDP会话,RNC通过解SIP/SDP包获得双方UE支持的AMR VoIP编码模式集,则在数据传输过程中,RNC周期性地根据当UE发射功率、基于功率的负载资源、基于Iub的负载资源至少之一进行AMR VoIP模式动态调整。In this preferred embodiment, after the IMS signaling bearer is set up, both UEs of the calling party and the called party carry out a SIP/SDP session, and the RNC obtains the AMR VoIP coding mode set supported by the UEs of both parties by deciphering the SIP/SDP packet. , the RNC periodically performs dynamic adjustment of the AMR VoIP mode according to at least one of UE transmit power, power-based load resources, and Iub-based load resources.
在数据传输过程中,当RNC确定NodeB的发射功率大于等于预设的发射功率门限时;或当RNC确定基于功率的负载资源大于等于门限时;或RNC确定基于Iub的负载资源大于等于门限;或进一步在数据传输过程中,RNC将NodeB上报的NodeB发射功率转换成专用物理数据信道DPDCH平均发射功率,然后和发射功率门限值进行比较,判断发射功率大于等于发射功率门限值;只要上述满足上述至少之一,就执行下述操作:During data transmission, when the RNC determines that the transmit power of the NodeB is greater than or equal to the preset transmit power threshold; or when the RNC determines that the load resource based on power is greater than or equal to the threshold; or the RNC determines that the load resource based on Iub is greater than or equal to the threshold; or Further in the data transmission process, the RNC converts the NodeB transmission power reported by the NodeB into the average transmission power of the dedicated physical data channel DPDCH, and then compares it with the transmission power threshold value to determine that the transmission power is greater than or equal to the transmission power threshold value; as long as the above conditions are satisfied At least one of the above, do the following:
降低RTP数据包中的CMR模式,该降低的CMR模式从UE所支持的CMR模式集中选择,CMR模式的降低可以是逐级降低,也可以根据所测量的结果越级降低,也可以直接降低为UE所支持模式集中的最小模式。Reduce the CMR mode in the RTP data packet. The reduced CMR mode is selected from the set of CMR modes supported by the UE. The reduction of the CMR mode can be reduced step by step, or it can be reduced step by step according to the measured results, or it can be directly reduced to UE The smallest pattern in the set of supported patterns.
本发明另一实施例还公开了一种AMR模式动态调整系统,如图9所示,包括无线网络控制器RNC200和基站NodeB300。其中,RNC200进一步包括AMRVoIP模式存储单元201、AMR VoIP模式选择单元202、AMR VoIP模式调整单元203、发射功率门限设置单元204、Iub负载门限设置单元205、发射功率获取单元206、判断单元208、Iub负载测量单元209、A/B事件检测单元210。NodeB300进一步包括发射功率测量单元301、发射功率反馈单元302、信道质量测量单元303、信道质量获取单元304、信道质量反馈单元305。Another embodiment of the present invention also discloses an AMR mode dynamic adjustment system, as shown in FIG. 9 , including a radio network controller RNC200 and a base station NodeB300. Wherein, RNC200 further includes AMRVoIP
在RNC200中,发射功率门限设置单元204设置NodeB和UE的发射功率门限;Iub负载门限设置单元205设置Iub接口的负载门限,如可以设置Iub带宽占有率门限。In RNC200, transmit power
主被叫双方UE进行SIP/SDP会话时,RNC通过解SIP/SDP包获得双方UE支持的AMR VoIP编码模式集,并将AMR VoIP编码模式存储于AMR VoIP模式存储单元201中。则在数据传输过程中,AMR VoIP模式选择单元202和AMRVoIP模式调整单元203周期性地根据当前UE或NodeB发射功率、基于功率的负载资源、基于Iub的负载资源至少之一进行AMR模式动态调整,该调整主要工作过程如下所述。When both the calling and called UEs conduct a SIP/SDP session, the RNC obtains the AMR VoIP coding mode set supported by both UEs by deciphering the SIP/SDP packet, and stores the AMR VoIP coding mode in the AMR VoIP
为进行AMR模式调整,RNC首先需要获得当前UE或NodeB发射功率、基于功率的负载资源、基于Iub的负载资源等至少之一,根据传输过程是上行还是下行,获得这些信息的方式稍有不同。In order to adjust the AMR mode, the RNC first needs to obtain at least one of the current UE or NodeB transmit power, power-based load resources, and Iub-based load resources. Depending on whether the transmission process is uplink or downlink, the way to obtain this information is slightly different.
上行传输时,获取上述当前UE发射功率、基于功率的负载资源、基于Iub的负载资源的方法可以如下所述。During uplink transmission, the methods for acquiring the above-mentioned current UE transmit power, power-based load resources, and Iub-based load resources may be described as follows.
RNC200向NodeB300发出测量控制命令,信道质量测量单元303根据测量控制命令测量当前信道质量信息,并传给信道质量反馈单元305,信道质量反馈单元305以A、B事件的形式上报给A/B事件检测单元210,其中,A事件反映当前信道质量好,B事件反映当前信道质量差。RNC200 sends a measurement control command to NodeB300, channel
A、B事件可以采用滞后的A、B事件上报的方法,参见图6,图7所示。The A and B events can be reported by lagged A and B events, as shown in Fig. 6 and Fig. 7 .
在图6中,信道质量反馈单元305判断信道质量高于预先设定的A事件门限值(可以是信道质量的上限门限值)的时间达到预先设定的迟滞时间时,信道质量反馈单元305进行A事件报告,报告给A/B事件检测单元210,说明此时信道质量比较好,图6中示出两次A事件报告示例。In FIG. 6, when the channel
在图7中,信道质量反馈单元305判断信道质量低于预先设定的B事件门限值(可以是信道质量的下限门限值)的时间达到预先设定的迟滞时间时,信道质量反馈单元305进行B事件报告,报告给A/B事件检测单元210,说明此时信道质量比较差,图7中示出了两次B事件报告示例。In FIG. 7, when the channel
发射功率测量单元301根据测量控制命令测量当前UE发射功率,并将UE发射功率传给发射功率反馈单元302,发射功率反馈单元302上报UE发射功率给发射功率获取单元206。The transmission
Iub负载测量单元209监控Iub接口负载使用情况,测量Iub接口的带宽占有率,并将所测得的带宽占有率传给判断单元208。The Iub
下行传输时,获取当前NodeB发射功率、基于功率的负载资源、基于Iub的负载资源的方法可以如下所述。During downlink transmission, the methods for obtaining the current NodeB transmit power, power-based load resources, and Iub-based load resources can be described as follows.
NodeB200周期性地广播导频信号,UE测量导频获得当前信道质量信息,并向NodeB200的信道质量获取单元304反馈信道质量指示(CQI,ChannelQuality Indication)。信道质量获取单元304将CQI传给信道质量反馈单元305,信道质量反馈单元305可以通过CQI A、CQIB事件上报给A/B事件检测单元210,其中,CQI A事件反映当前信道质量好,CQI B事件反映当前信道质量差。当前信道质量较好,CQI A事件和CQI B事件的上报方式可以采用上述类似A、B事件的上报方式。The NodeB200 broadcasts the pilot signal periodically, and the UE measures the pilot signal to obtain current channel quality information, and feeds back a channel quality indication (CQI, ChannelQuality Indication) to the channel
UE在接收数据同时检测NodeB发射功率,并向发射功率反馈单元302反馈,RNC200给发射功率反馈单元302配置反馈周期,发射功率反馈单元302周期性地上报NodeB发射功率给发射功率获取单元206。The UE detects the NodeB transmission power while receiving data, and feeds back to the transmission
Iub负载测量单元209监控Iub接口负载使用情况,测量Iub接口的带宽占有率,并将所测得的带宽占有率传给判断单元208。The Iub
不论是上行传输过程,还是下行传输过程,均根据当前发射功率、基于功率的负载资源、基于Iub的负载资源至少之一进行AMR模式调整,主要如下所述。Regardless of the uplink transmission process or the downlink transmission process, the AMR mode adjustment is performed according to at least one of the current transmit power, power-based load resources, and Iub-based load resources, mainly as follows.
发射功率获取单元206获取UE或NodeB发射功率后,将UE或NodeB发射功率传给判断单元208,判断单元208将该UE发射功率与发射功率门限设置单元204所设置的UE发射功率门限进行比较,或判断单元208将该NodeB发射功率与发射功率门限设置单元204所设置的NodeB发射功率门限进行比较。After the transmission
如果UE或NodeB大于等于相对应的发射功率门限,则判断单元208通知AMR VoIP模式选择单元202。If the UE or NodeB is greater than or equal to the corresponding transmit power threshold, the
AMR VoIP模式选择单元202根据所述通知从AMR VoIP模式存储单元201选择相应的AMR VoIP模式,选择方式可以是逐级降低,或越级降低,或也可以直接降低为AMR VoIP模式存储单元201所存储模式集中的最小模式。AMRVoIP模式调整单元203根据AMR VoIP模式选择单元202所选择的AMR VoIP模式,将RTP数据包中的CMR模式降低到所选择的AMR VoIP模式。The AMR VoIP
或进一步,发射功率获取单元206获取UE或NodeB发射功率后,将UE或NodeB发射功率传给判断单元208。判断单元208将该上报的UE或NodeB发射功率转换成专用物理数据信道平均发射功率,然后和发射功率门限设置单元204所设置的相应发射功率门限进行比较,即判断单元208将该UE平均发射功率与发射功率门限设置单元204所设置的UE发射功率门限进行比较,或判断单元208将该NodeB平均发射功率与发射功率门限设置单元204所设置的NodeB发射功率门限进行比较。Or further, the transmit
如果所述UE或NodeB平均发射功率大于等于相应发射功率门限,则判断单元208通知AMR VoIP模式选择单元202。If the average transmit power of the UE or NodeB is greater than or equal to the corresponding transmit power threshold, the
AMR VoIP模式选择单元202根据所述通知从AMR VoIP模式存储单元201选择相应的AMR VoIP模式,选择方式可以是逐级降低,或越级降低,或也可以直接降低为AMR VoIP模式存储单元201所存储模式集中的最小模式。AMRVoIP模式调整单元203根据AMR VoIP模式选择单元202所选择的AMR VoIP模式,将RTP数据包中的CMR模式降低至所选择的AMR VoIP模式。The AMR VoIP
判断单元208接收Iub负载测量单元209所测量的Iub接口的带宽占有率,判断单元208将该带宽占有率与Iub负载门限设置单元205所设置的带宽占有率门限进行比较,一旦发现该带宽占有率大于等于带宽占有率门限,也就是Iub负载大于等于预设的Iub负载门限时,即告警,判断单元208通知AMR VoIP模式选择单元202。Judging
AMR VoIP模式选择单元202根据所述通知从AMR VoIP模式存储单元201选择相应的AMR VoIP模式,选择方式可以是逐级降低,或越级降低,或也可以直接降低为AMR VoIP模式存储单元201所存储模式集中的最小模式。AMRVoIP模式调整单元203根据AMR VoIP模式选择单元202所选择的AMR VoIP模式,将RTP数据包中的CMR模式降低至所选择的AMR VoIP模式。The AMR VoIP
但上述仅为本发明的较佳实施方式,并非用于限定本发明的保护范围,任何熟悉本技术领域的技术人员应当认识到,凡在本发明的精神和原则范围之内,所做的任何修饰、等效替换、改进等,均应包含在本发明的权利保护范围之内。However, the above is only a preferred embodiment of the present invention, and is not intended to limit the protection scope of the present invention. Any person familiar with the technical field should recognize that within the scope of the spirit and principles of the present invention, any Modifications, equivalent replacements, improvements, etc., should all be included within the protection scope of the present invention.
| Application Number | Priority Date | Filing Date | Title |
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| CNB2007100732777ACN100493223C (en) | 2007-02-10 | 2007-02-10 | Adaptive Multi-Rate Packet Speech Coding Mode Adjustment Method and Base Station Controller |
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| CNB2007100732777ACN100493223C (en) | 2007-02-10 | 2007-02-10 | Adaptive Multi-Rate Packet Speech Coding Mode Adjustment Method and Base Station Controller |
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| CN100493223Ctrue CN100493223C (en) | 2009-05-27 |
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| CNB2007100732777AActiveCN100493223C (en) | 2007-02-10 | 2007-02-10 | Adaptive Multi-Rate Packet Speech Coding Mode Adjustment Method and Base Station Controller |
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