VoLTE-SRVCC基本原理与系统结构

IMS2G/3GVoLTE网络架构LTEeNodeBSGWHLR/HSS

IMS-HSSHLRLTE-HSSPGWMMEPCRFGERAN/UTRANMSSSGSNMGWP-CSCFA-BGWI-CSCFS-CSCFTASUEMRFCMRFPIMS-MGWMGCFBGCFMRFPVoLTE网元列表网元名称网元所属域网元功能归属网元eNodeBLTE无线支持VoLTE专用承载，支持SRVCC功能eNodeBMMEEPC核心网支持SRVCC功能MMES/P-GWEPC核心网支持VoLTE专用承载S/P-GWPCRFEPC核心网支持Rx接口用于建立VoLTE专用承载PCRFMSS/eMSSR4核心网支持SRVCC功能MSSMGCF互通关口局设备支持IMS/R4话音互通MGCFIM-MGW互通关口局设备支持IMS/R4话音互通IM-MGWI-SBCIMS核心网媒体域用于IMS互通本次测试暂不需要BCFIMS核心网控制域控制BGF用于IMS媒体面P-CSCFBGFIMS核心网媒体域用于IMS媒体面连接，与BCF共同承担SBC功能BGWP-CSCFIMS核心网控制域接入侧IMS核心网控制网元P-CSCFI/S-CSCFIMS核心网控制域用户归属域IMS核心网控制网元I/S-CSCFBGCFIMS核心网控制域IMS用户呼叫其他网络时的路由功能I/S-CSCFDRAIMS核心网控制域提供Diameter接口(如Cx/Dx/Sh/Rx)路由功能DRAENUMDNSIMS核心网控制域将E.164号码翻译为IMS用户的IMPUENUMDNSVoLTE网元列表网元名称网元所属域网元功能归属网元TASIMS核心网业务域用于支持VoLTE话音以及补充业务TASMRFCIMS核心网业务域VoLTE话音服务器中录音通知与混音控制功能TASMRFPIMS核心网业务域VoLTE话音服务器中录音通知与混音媒体功能BGWIP-SM-GWIMS核心网业务域提供VoLTE与CS的短信互通TASIM-SSF(可选)IMS核心网业务域提供VoLTE与CS的智能网互通TASIMS-HSSBEIMS核心网数据域用于存储IMS有关用户数据

HSS-BEIMS-HSSFEIMS核心网数据域用于提供HSS到CSCF或者业务服务器的接口HSS-FESCC-ASIMS核心网业务域提供SRVCC锚定功能的业务服务器TASATCFIMS核心网控制域提供eSRVCC控制面功能P-CSCFATGWIMS核心网数据域提供eSRVCC媒体面功能BGWCGIMS核心网支持域提供IMS计费数据整合功能开通网关(可选)IMS核心网支持域提供HSS/TAS统一开通功能OSSIMS核心网支持域提供IMS网管Needtobeadded3GPPVoLTENetworkInterfaceInterfBetweenProtocolFunctionGmUE↔P-CSCFSIPAspartofitisover-the-air,signalingcompressionisneededMwCSCF↔CSCFSIPUsedforsessionsetup,maintenanceandteardownMrS-CSCF↔MRFCSIPS-CSCFrequestformediaresourcesfromMRFCduringsessioncontrolMxS-CSCF↔ALGSIPDestinationnetworkrequiresIPversioninter-workingMiS-CSCF↔BGCFSIPCSbreakout:

S-CSCFtowardaCSdestinationnetworkMjBGCF↔MGCFSIPCSbreakoutMkBGCF↔BGCFSIPInter-PLMNIMSinter-workingISCAS↔S-CSCFSIPForthecontroltransfertoaASfromS-CSCFCxI/S-CSCF↔HSSDiameterI/S-CSCFobtainsandupdatesuserandserviceprofiledatafromHSSDxI/S-CSCF↔SLFDiameterI/S-CSCFaskSLFtolocatetheHSSinmulti-HSSnetworkShAS↔HSSDiameterAStoobtainandupdateuserandserviceprofiledatafromHSSRfAS/CSCF/MGCF↔CDFDiameterForcreatingofflinechargingrecordsRoAS/CSCF/MGCF↔OCSDiameterForon-linechargingandsession/servicecontrolRxP-CSCF↔PCRFDiameterP-CSCFrequestresourcereservationGxPCRF↔P-GWDiameterPolicyandChargingruletoPaCoMpMRFC↔MRFPH.248ForcontrolmediaresourcesMnMGCF↔IMS-MGWH.248ForcontrolmediaresourcesMbUE↔IMS-MGW/MRFPRTPCarryIPpacketsofmediastreamUtUE↔ASXCAPForUEapp.Clienttomanageservice-relateddatawithASSrMRFC↔ASXCAPMTRFobtaininfoaboutmediafromASVoLTEbearerconceptVoicetraffic-QCI1SIPSignaling-QCI5Videotraffic–QCI2EPCeNBVoLTErequiresownIMSAPN:SIPsignalingalwaysusesaQCI=5bearerwithnon-GBRVoicetrafficalwaysusesadedicatedQCI=1bearerwithGBROtherIMSmedia-ifpresent,e.g.forRCS-useQCI=8/9mediawithnonGBRInternettrafficforanother(internet)APNUMDRBUMDRBAMDRBIMSAPNBesteffortdata-QCI9AMDRBInternetAPNotherIMSmedia-QCI8/9AMDRBQoSaspectsofE2EVoLTECallFlow(IR.92)ThomasHammannPhase5:UserDataandGBR/NoneGBRschedulingE-UTRANS-GWS1-UEPSDefaultBearerforIMSAPN(QCI5,e-RAB=5,EBI=5)EPSBearerforDefaultBearerforDataAPN(QCI9,e-RAB=6EBI=6)P-GWS5/S8RadioAccessBearerismappedtoGTPTunnel.TFTstoselecte-RABinULTFTstoselectBearer(EBI)inDLSIPTrafficNoneGBRBearerforSIPSignallingPriority9,PacketDelayBudget300msec,PacketErrorLossRate10-6

DataTrafficNoneGBRBearerforDataPriority1,PacketDelayBudget100msec,PacketErrorLossRate10-6EPSDedicatedBearer(QCI1,e-RAB=7,EBI=7)VoIPDataGBRBearerforVoIPDataPriority2,PacketDelayBudget100msec,PacketErrorLossRate10-22ePreconditionUlfSpangenbergerPrecondition:asetofconstraintsthatmustbefulfilledbeforetheuserisalerted(rfc3312)newSDPattributescurr(currentstatus),des(desiredstatus),conf(confirmationstatus)Offer/AnswermessageflowInitialINVITE(SDP)183SessionProgress(SDP)UPDATE(SDP)200OK(SDP)RABsetupARABsetupB180RingingA->BDelayringinguntilcurr=desforAABB->AInformmewhencurr=confA->Bcurr=conf=desIMSInviteProcedure–RxInterfaceUlfSpangenberger2eP-CSCFAS-CSCFAS-CSCFBP-CSCFBPCRFPCRFINVITE(SDP) IPaddress-A,port-A,orderedlistofsupportedcodecs,[bandwidth]180RINGING200OK(SDP) IPaddress-B,port-B,orderedlistofnegociatedcodecs,[bandwidth]ACKSAE-GWSAE-GWSetupdedicatedbearerwithQCI=1AccessAuthorization,AAR/AFlowdescriptor:UL/DLforRTP/RTCPBandwidth:UL/DLforRTP/RTCPCodeclistCallrouting(IMPU)UE-AUE-BPCRFforVoLTE2eElisabethKozielPGWPCRFCreateDedicatedBearerReqPCCRulesProvisioning(PCCRules,EventTriggers)GxRAACreateDedicatedBearerRespServicesP-CSCFSessionandmediainfoPGWdecidesbasedontheQCI/ARPpairwhetheranexistingbearercanbeusedorwhetheranewdedicatedbearerisneededforVoIP12345DefaultBearerforsignalingcreated0Gx_CCR-I,QCI=5,ARP=2,SessionId=1Rx_AAR:MediaComponenDescriptionGx_RAR:Charging-Rule-Install(Flow,Qos),SessionId=1RxAAAIR.92basedQoSmodel:IMSAPNisusedfortheIMStraffic,otherAPNfordatatrafficE2EVoLTECallFlow(IR.92)2ePhase1Attach/PrimaryDefaultBearerwithIMSAPN(QCI=5)Phase2SIPRegisterw/primaryDefaultBearer/2ndPDNconnectionforDataAPNPhase3InitiateVoIPcallusingPrimarydefaultbearerPhase4DedicatedBearerforVoIPDataPhase5UserDataandGBR/Non-GBRsheduleingPhase6ReleaseofVoIPcallanddedicatedbearerLTE/EPCPointofViewPhase1SIPRegisterPhase2SIPInvitePhase3SIPReleaseIMSPointofViewOption1CombinedEPS/IMSIcombinedEPS/IMSIattachUEbehaviourforIMSPSVoicepreferredwithCSVoiceassecondaryTS23.221Option2noncombinedEPS/IMSIattachXVoLTE呼叫流程VoLTEHandoverProcedures

SRVCCHandoverforVoicefromLTEto2G/3GCSIMS/VoIP/VoLTE

DomainCScoverageLTE/EPSDomain2G/3GCS&PSDomainS6aDUE

(A-Party)Gr/S6dUE

(A-Party)LTE

coverageGmCxISCSv/

SGsGxRxMwSingle

Radio

VoiceCall

Continuity

(SR-VCC)eNBSAE-GWMMEPCRFCSCF

(ofA-Party)HSS/

HLRMSC-SMGWTASBSC/RNCSGSNGGSNOtherOperatorsLTE

CellCSCF

(ofB-Party)SCC-ASUE

(B-Party)S1-US1-MMES11VoicepathwhentheA-PartyisinLTEVoicepathwhentheA-PartyisinCSSRVCCrelatedsignallingSignallinginterfacesMAPSIPInvite(STN-SR,SDP=MGW)0.Userplanebefore

SR-VCCHandover6.UserplaneafterSR-VCChandover4.TheSCCASinitiatesa

SIPRe-InvitetotheB-Party

toswitchtheUserPlane

fromPDN-GWtotheMGW5.TriggeredbytheSIPRe-Invite,theB-partyswitchestheUserPlanetotheMGW2.MMEtriggers“PStoCSHand-overRequest”toMSC-SincludingTargetCellID,STN-SR,etc.1.eNbinitiates

SR-VCCHandover3.TheMSC-SreservesCSradioresources

andinitiatesaSIPInvitetowardstheIMS(SCC-AS)usingtheSTN-SR.STN-SRSessionTransferNumberforSRVCC(staticconfiguredinHSS)->SCC-ASIMS/VoIP/VoLTE

DomainCScoverageLTE/EPSDomain2G/3GCS&PSDomainS6aDUE

(A-Party)Gr/S6dUE

(A-Party)LTE

coverageGmCxISCSv/

SGsGxRxMwSingle

Radio

VoiceCall

Continuity

(SR-VCC)SheNBSAE-GWMMEPCRFCSCF

(ofA-Party)HSS/

HLRMSC-SMGWTASBSC/RNCSGSNGGSNOtherOperatorsLTE

CellCSCF

(ofB-Party)SCC-ASUE

(B-Party)S1-US1-MMES11VoicepathwhentheA-PartyisinLTEVoicepathwhentheA-PartyisinCSSRVCCrelatedsignallingSignallinginterfacesMAPATGWATCFMGWP-GW0.Userplaneisbefore

SR-VCCHandover2.MMEtriggers“PStoCSHand-overRequest”toMSC-SincludingTargetCellID,STN-SR,etc.1.eNbinitiates

SR-VCCHandover6.UserplaneafterSR-VCChandover3.TheMSC-SreservesCSradioresources

andinitiatesaSIPInvitetowardstheATCFusingtheSTN-SR.5.SCC-ASisinformedfortheSRVCCHOwithoutthebearerchangeindication.SincethereisnoSDPupdate,no

SIPRe-InviteissenttotheB-Party.4.ATCFconfiguresATGWwithSDP=MGWwhichswitchesthebearerfromPDN-GW(LTE)totheMGW(CS)VoLTEHandoverProcedures

eSRVCCHandoverforVoicefromLTEto2G/3GCSSTN-SRSessionTransferNumberforSRVCC(updatedduringRegistration)->ATCFAMR简介全称AdaptiveMulti-Rate，自适应多速率编码，主要用于移动设备的音频，压缩比比较大，但相对其他的压缩格式质量比较差，由于多用于人声，通话，效果还是很不错的。1.AMR:又称为AMR-NB，相对于下面的WB而言，语音带宽范围：300－3400Hz，8KHz抽样2.AMR-WB:AMRWideBand，语音带宽范围：50－7000Hz16KHz抽样“AMR-WB”全称为“AdaptiveMulti-rate-Wideband”，即“自适应多速率宽带编码”，采样频率为16kHz，是一种同时被国际标准化组织ITU-T和3GPP采用的宽带语音编码标准，也称为G722.2标准。AMR-WB提供语音带宽范围达到50～7000Hz，用户可主观感受到话音比以前更加自然、舒适和易于分辨。AMR-NBFrameTypeModeIndicationModeRequestFramecontent(AMRmode,comfortnoise,orother)000AMR4,75kbit/s111AMR5,15kbit/s222AMR5,90kbit/s333AMR6,70kbit/s(PDC-EFR)444AMR7,40kbit/s(TDMA-EFR)555AMR7,95kbit/s666AMR10,2kbit/s777AMR12,2kbit/s(GSM-EFR)8--AMRSID9--GSM-EFRSID10--TDMA-EFRSID11--PDC-EFRSID12-14--Forfutureuse15--NoData(Notransmission/Noreception)AMR

一共有16种编码方式，

0-7对应8种不同的编码方式，

8-15

用于噪音或者保留用。AMR-WBFrameTypeIndexModeIndicationModeRequestFramecontent(AMR-WBmode,comfortnoise,orother)000AMR-WB6.60kbit/s111AMR-WB8.85kbit/s222AMR-WB12.65kbit/s333AMR-WB14.25kbit/s444AMR-WB15.85kbit/s555AMR-WB18.25kbit/s666AMR-WB19.85kbit/s777AMR-WB23.05kbit/s888AMR-WB23.85kbit/s9--AMR-WBSID(ComfortNoiseFrame)10-13--Forfutureuse14--speechlost15--NoData(Notransmission/Noreception)

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时延和抖动时延：端到端包传递的时间ITUG.114规范建议，在传输语音流量时,单向语音包端到端延迟要低于150ms(对于国际长途呼叫,特别是卫星传输时,可接受的单向延迟为300ms。如果超过300ms则通话的质量会变的让人不能忍受。过多的包延迟可以引起通话声音不清晰、不连贯或破碎。经验：大多数用户察觉不到小于100毫秒的延迟，当延迟在100毫秒和300毫秒之间时，说话者可以察觉到对方回复的轻微停顿。这种停顿可以影响到通话双方的交流。超过300毫秒，延迟就会很明显，用户开始互相等待对方的回复，通话过程变成类似对讲机式的模式。而且较长的时延也会将回声问题的影响放大。时延的产生因素编码的处理：模拟形式的