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VoIPTutorialLi,Xiaomeng2010.06.10ContentsVoIPBasics1VoIPArchitecture2VoIPStandard&Protocols3VoIPEvolution4VoIPTerminologyVoIP:VoiceOverIPPSTN:PublicswitchedtelephonenetworkATMAsynchronoustransfermode,acellswitchedcommunicationstechnologySS7:SignalingSystem7SG:SignalinggatewayMG:MediagatewaySBC:SessionBorderControllerDTMF:DualtonemultiplefrequencyPCM:PulseCodeModulationADPCM:AdaptiveDifferentialPulseCodeModulationCS-ACELP:ConjugateStructureAlgebraic-CodeExcitedLinearPredictionAMR:AdaptiveMulti-RateIETF:InternetEngineeringTaskForceITU:InternationalTelecommunicationsUnionH.323:AnITU-Tstandardprotocolsuiteforreal-timecommunicationsoverapacketnetwork.H.225:AnITU-Tcallsignalingprotocol(partoftheH.323suite).H.235:AnITU-Tsecurityprotocol(partoftheH.323suite).H.245:AnITU-Tcapabilityexchangeprotocol(partoftheH.323suite).SIP:SessionInitialProtocolRTP:Real-timeTransportProtocolRTCP:Real-timeCotrolProtocolRTSPReal-timestreamingprotocolSDPSessiondescriptionprotocolVoIPConceptsVoIP:TransmissionofvoiceoverInternetHowVoIPworks–Continuouslysampleaudio–Encodeeachsampletodigitalstream–SenddigitizedstreamacrossInternetinpackets–Receiveandconvertpacketstodigitalstream–DecodethestreambacktoanalogforplaybackVoIPAdvantages&Disadvantages–Advantages•Muchlowercost(multiplexing,geographicinsensitive)•Portability:callatanywhere•Extensivecallfeatures•Convergedvoice,video,dataoverIPnetwork–Disadvantages:•Continuousserviceduringapoweroutage•Emergencycalls•OperabilityVoIPFunctionalitiesSignaling:theprocessofestablishingandterminatingacall•Registration,Callinitiation,Callteardown•Bearercontrol(codecnegotiation)Media(Voice)Streaming–Codec:encodevoiceintobitstream/decodethestreamtovoice•G711/PCM:mu-law,A-law,64kbps•G.723:5.3,6.4kbps•G.726/ADPCM:16,24,32,and40kbps•G.729/CS-ACELP:8kbps•AMR:Variableratefrom4.75~12.2kbps–RTPStreaming:PacketizingstreamandtransmittingoverIPnetwork•RTPoverUDPinsteadofTCP:Why?VoIPComponentsServers–SessionController–MediaGatewayController•ManaginginteractionwithPBX–ApplicationServerGateways–MediaGateway:transcodeaudiobetweenIPnetworkandPSTN–SignalingGateway:translatessignalingoperations–ResidentialGatewayEnd-pointdevices–H323Phone–SIPPhone–MGCPEndpointOperabilityChallengesVoiceQuality–Latency•VoIPtypicallytoleratesdelaysupto150msbeforethequalityofthecalldegrades.–Jitter•Instantaneousbufferusecausesdelayvariationinthesamevoicestream.–Packetloss•Lossofpacketsseverelydegradesthevoiceapplication.–Bandwidth•BandwidthsharingbetweenvoiceanddatamayworsethevoiceReliability–Networkstabilitywillmakeithardtoguaranteethe99.999%reliabilityScalabilitySecurityFeaturesInteroperabilitySwitchovercostCallFeaturesTraditionalCallFeatures–CLASS5CallFeatures•CallerIDIndication•CallWaiting•CallForwarding•3-WayCalling•CallBaringIn-NetworkCallingCurrentandFutureFeatures–Centrex(NGN/IMS)–MultimediaMessaging–Presence–PoC(PushtoTalkoverCellular)–…ContentsVoIPBasics1VoIPArchitecture2VoIPSignaling&Media3VoIPEvolution4ITUVoIPModelIETFVoIPModelTheWholeVoIPArchitectureContentsVoIPBasics1VoIPArchitecture2VoIPSignaling&Media3VoIPEvolution4VoIPProtocolStackVoIPSignalingProtocolsH.323–ITUstandard,ISDN-based,distributedtopology–90%+ofallServiceProviderVoIPnetworks–ThecurrentinterconnectforCallManagertoServiceProviders–UsefulforvideoapplicationsSIP–IETFRFC3261withmanyextensiveRFCs–DistributedCall-Control–UsedformorethanVoIP…SIMPLE:InstantMessaging/Presence–SessioncontrolprotocolinNGN/IMSH.248/Megaco–ITU-TH.248/IETFRFC3525–CentralizedCall-ControlArchitecture–UsedbetweenCall-Agents(MGC)&Gateways(MG)–MediacontrolprotocolinNGN/IMSMGCP–IETFRFC3435–CentralizedCall-ControlArchitecture–UsedbetweenCall-Agents(MGC)&Gateways(MG)H.323OverviewAnITUrecommendationapplicableto“Packet-basedmultimediacommunicationssystems”.H.323definesadistributedarchitectureforcreatingmultimediaapplications,includingVoIPH.323consistsofasetofprotocolsworkingtogethertohandleallaspectsofcommunication,including:–Transmissionofadigitalaudiophonecall–Signalingtosetupandmanagephonecall–Allowstransmissionofvideoanddatawhileaphonecallisinprogress–Sendsbinarymessage–Incorporatesprotocolsforsecurity–UsesaspecialhardwareMultipointControlUnitforconferencingcalls–Definesserversforaddressresolution,authentication,accounting,features,etcOlderandmoreestablishedprotocolH.323ComponentsH.323ScopeDeploymentofH.323NetworkBasicH.323CallGatekeeperAGatekeeperBRRQ/RCFARQRRQ/RCFLRQIPNetworkPhoneAGatewayAGatewayBH.225(Q.931)SetupH.225(Q.931)AlertandConnectH.245RTPACFLCFVVARQACFPhoneBH.323SignalingMGCP,H.248/Megaco-ArchitectureDeployingMGCP/H.248/MegacoNetworksSIPSIP:SessionInitiationProtocol–IETFRFC3261,ReplacesRFC2543–SIPisanapplication-layercontrol(signaling)protocolforcreating,modifyingandterminatingsessionswithoneormoreparticipants.”–EvolvingwithseriesofRFCs•RFC3262:PRACK•RFC3265:SUBSCRIBE/NOTIFY•RFC3311:UPDATE•RFC3312:Preconditions&resourcereservation•RFC3329:Securitymechanism•RFC3515:REFER•RFC4028:Sessionrefreshment•……Canbeusedforvoice,video,instantmessaging,gamin
本文标题:VoIP基础教程
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