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VoIP for IPv6
ThisdocumentdescribesVoIPinIPv6(VoIPv6),afeaturethataddsIPv6capabilitytoexistingVoIPfeatures.
Thisfeatureaddsdual-stack(IPv4andIPv6)supportonvoicegatewaysandmediaterminationpoints(MTPs),
IPv6supportforSessionInitiationProtocol(SIP)trunks,andsupportforSkinnyClientControlProtocol
(SCCP)-controlledanalogvoicegateways.Inaddition,theSessionBorderController(SBC)functionality
ofconnectingaSIPIPv4orH.323IPv4networktoaSIPIPv6networkisimplementedonaCiscoUBEto
facilitatemigrationfromVoIPv4toVoIPv6.
• FindingFeatureInformation, page 1
• PrerequisitesforVoIPforIPv6, page 1
• RestrictionsforImplementingVoIPforIPv6, page 2
• InformationAboutVoIPforIPv6, page 3
• HowtoConfigureVoIPforIPv6, page 9
• ConfigurationExamplesforVoIPoverIPv6, page 36
• TroubleshootingTipsforVoIPforIPv6, page 36
• VerifyingandTroubleshootingTips, page 37
• FeatureInformationforVoIPforIPv6, page 53
Finding Feature Information
Yoursoftwarereleasemaynotsupportallthefeaturesdocumentedinthismodule.Forthelatestcaveatsand
featureinformation,seeBugSearchToolandthereleasenotesforyourplatformandsoftwarerelease.To
findinformationaboutthefeaturesdocumentedinthismodule,andtoseealistofthereleasesinwhicheach
featureissupported,seethefeatureinformationtableattheendofthismodule.
UseCiscoFeatureNavigatortofindinformationaboutplatformsupportandCiscosoftwareimagesupport.
ToaccessCiscoFeatureNavigator,gotowww.cisco.com/go/cfn.AnaccountonCisco.comisnotrequired.
Prerequisites for VoIP for IPv6
•CiscoExpressForwardingforIPv6mustbeenabled.
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Restrictions for Implementing VoIP for IPv6
•Virtualroutingandforwarding(VRF)isnotsupportedinIPv6calls.
CiscoUnifiedBorderElement
•CiscoIOSRelease12.4(22)ToralaterreleasemustbeinstalledandrunningonyourCiscoUBE.
CiscoUnifiedBorderElement(Enterprise)
•CiscoIOSXERelease3.3SoralaterreleasemustbeinstalledandrunningonyourCiscoASR1000
SeriesRouter.
Restrictions for Implementing VoIP for IPv6
ThefollowingaretherestrictionsforCiscoUBEfeatures:
Media Flow–Through
•VideocallflowswithAlternativeNetworkAddressTypes(ANAT)arenotsupported.
•WebExcallflowwithANATarenotsupported(CiscoUBEdoesnotsupportANATonVideoand
Applicationmediatypes).
SDP Pass-Through
•SupportsonlyEarlyOffer(EO)–EarlyOffer(EO)andDelayedOffer(DO)–DelayedOffer(DO)call
flows.
•DelayedOffer–EarlyOffercallflowfallsbacktoDelayedOffer–DelayedOffercallflow.
•SupplementaryservicesarenotsupportedonSDPPass-Through.
•TranscodingandDTMFinterworkingarenotsupported.
Note TheaboveSDPPass–ThroughrestrictionsareapplicableforbothIPv4andIPv6.
•SDPPass–Throughdoesnotsupportthedual-stackfunctionality.
•ANATcallflowsdoesnotsupportIPv4-to-IPv6andIPv6-to-IPv4Mediainterworking.
UDP Checksum
•CEFandprocessoptionsarenotsupportedonASR1000seriesrouters.
•NoneoptionispartiallysupportedonISR–G2.
Media Anti–Trombone
•MediaAnti–TromboneisnotenablediftheinitialcallbeforetromboningisinFlow–Around(FA)mode.
•MediaAnti–Trombonesupportsonlysymmetricmediaaddresstypeinterworking(IPv4-IPv4orIPv6-IPv6
media)withorwithoutANAT.
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Information About VoIP for IPv6
•DoesnotprovidesupportforIPv4-IPv6interworkingcaseswithorwithoutANATbecauseCiscoUBE
cannotoperateinFAmodeposttromboning.
Information About VoIP for IPv6
SIP Features Supported on IPv6
TheSessionInitiationProtocol(SIP)isanalternativeprotocoldevelopedbytheInternetEngineeringTask
Force(IETF)formultimediaconferencingoverIP.
TheCiscoSIPfunctionalityenablesCiscoaccessplatformstosignalthesetupofvoiceandmultimediacalls
overIPnetworks.SIPfeaturesalsoprovideadvantagesinthefollowingareas:
•Protocolextensibility
•Systemscalability
•Systemscalability
•Personalmobilityservices
•Interoperabilitywithdifferentvendors
ASIPUserAgent(UA)operatesinoneofthefollowingthreemodes:
•IPv4-only:CommunicationwithonlyIPv6UAisunavailable.
•IPv6-only:CommunicationwithonlyIPv4UAisunavailable.
•Dual-stack:CommunicationwithonlyIPv4,onlyIPv6anddual-stackUAsareavailable.
Dual-stackSIPUAsuseAlternativeNetworkAddressTransport(ANAT)groupingsemantics:
•IncludesbothIPv4andIPv6addressesintheSessionDescriptionProtocol(SDP).
•Isautomaticallyenabledindual-stackmode(canbedisabledifrequired).
•RequiresmediatobeboundtoaninterfacethathavebothIPv4andIPv6addresses.
•DescribedinRFC4091andRFC4092(RFC5888describesgeneralSDPgroupingframework).
SIPUAsuse“sdp-anat”optiontagintheRequiredandSupportedSIPheaderfields:
•EarlyOffer(EO)INVITEusingANATsemanticsplaces“sdp-anat”intheRequireheader.
•DelayedOffer(DO)INVITEplaces“sdp-anat”intheSupportedheader.
SIPSignalingandMediaAddressSelection:
•SourceaddressforSIPsignalingisselectedbasedonthedestinationsignalingaddresstypeconfigured
inthesession-targetoftheoutbounddial-peer:
◦Ifsignalingbindisconfigured,sourceSIPsignalingaddressischosenfromtheboundinterface.
◦Ifsignalingbindisnotconfigured,sourceSIPsignalingaddressischosenbasedonthebestaddress
intheUAtoreachthedestinationsignalingaddress.
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SIP Voice Gateways in VoIPv6
SDPmayormaynotuseANATsemantics:
•WhenANATisused,mediaaddressesinSDParechosenfromtheinterfacemediathatisconfigured.
WhenANATisnotused,mediaaddressesinSDParechosenfromtheinterfacemediathatisconfigured
ORbasedonthebestaddresstoreachthedestinationsignalingaddress(whennomediabindis
configured).
SIP Voice Gateways in VoIPv6
SessionInitiationProtocol(SIP)isasimple,ASCII-basedprotocolthatusesrequestsandresponsestoestablish
communicationamongthevariouscomponentsinthenetworkandtoultimatelyestablishaconferencebetween
twoormoreendpoints.
InadditiontothealreadyexistingfeaturesthataresupportedonIPv4andIPv6,theSIPVoiceGateways
supportthefollowingfeatures:
•History–Info:TheSIPHistory–infoHeaderSupportfeatureprovidessupportforthehistory-infoheader
inSIPINVITEmessagesonly.TheSIPgatewaygenerateshistoryinformationintheINVITEmessage
forallforwardedandtransferredcalls.Thehistory-infoheaderrecordsthecallordialoghistory.The
receivingapplicationusesthehistory-infoheaderinformationtodeterminehowandwhythecallhas
reachedit.
Formoreinformation,refertothe“SIPHistoryINFO”sectioninthe CiscoUnifiedBorderElement
(Enterprise)SIPSupportConfigurationGuide.
•Handling181/183Responseswith/withoutSDP:TheHandling181/183Responseswith/withoutSDP
featureprovidessupportforSIP181(CallisBeingForwarded)andSIP183(SessionProgress)messages
eithergloballyoronaspecificdial-peer.Also,youcancontrolwhenthespecifiedSIPmessageisdropped
basedoneithertheabsenceorpresenceofSDPinformation.
Formoreinformation,referto“SIP–Enhanced180ProvisionalResponseHandling”sectioninthe Cisco
UnifiedBorderElementConfigurationGuide.
•LimitingtheRateofIncomingSIPCallsperDial-Peer(CallSpike):Thecallrate-limitingfeature
forincomingSIPcallsstartsworkingafteraswitchoverinaSIPcall.Therate–limitingisdonefornew
callsreceivedonthenewActive.TheIOStimersthattrackthecallratelimitsrunsonactiveandstandby
modeanddoesnotrequireanycheckpoint.However,somestatisticsforcallsrejectedrequirestobe
checkedfortheshowcommandstobeconsistentbeforeandaftertheswitchover.
•PPI/PAI/PrivacyandRPIDPassing:ForincomingSIPrequestsorresponsemessages,whenthePAI
orPPIprivacyheaderisset,theSIPgatewaybuildsthePAIorPPIheaderintothecommonSIPstack,
therebyprovidingsupporttohandlethecalldatapresentinthePAIorPPIheader.ForoutgoingSIP
requestsorresponsemessages,whenthePAIorPPIprivacyheaderisset,privacyinformationissent
usingthePAIorPPIheader.
Formoreinformation,refertothe“SupportforPAIDPPIDPrivacyPCPIDandPAURIHeaderson
CiscoUBE”sectionintheCiscoUnifiedBorderElementSIPSupportConfigurationGuide.
•SIPVMWIforFXSphones:SIPprovidesvisiblemessagewaitingindication(VMWI)onFXSphones.
Thisfeatureprovidesuserswiththeoptiontoenableonemessagewaitingindication(MWI):audible,
visible,orboth.TheVMWImechanismusesSIPSubscribeorNotifytogetMWIupdatesfromavirtual
machine(VM)system,andthenforwardsupdatestotheFXSphoneontheport.
Formoreinformation,refertothe“ConfiguringSIPMWIFeatures”sectioninthe SIPConfiguration
Guide.
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•SIPSessiontimer(RFC4028):ThisfeatureallowsforaperiodicrefreshofSIPsessionsthrougha
re-INVITEorUPDATErequest.Therefreshallowsbothuseragentsandproxiestodeterminewhether
theSIPsessionisstillactive.Twoheaderfieldscanbedefined:Session-Expires,whichconveysthe
lifetimeofthesession,andMin-SE,whichconveystheminimumallowedvalueforthesessiontimer.
Formoreinformation,refertothe“SIPSessionTimerSupport”sectioninthe CiscoUnifiedBorder
ElementSIPSupportConfigurationGuide.
•SIPMediaInactivityDetection:TheSIPMediaInactivityDetectionTimerfeatureenablesCisco
gatewaystomonitoranddisconnectVoIPcallsifnoReal-TimeControlProtocol(RTCP)packetsare
receivedwithinaconfigurabletimeperiod.
Formoreinformation,refertotheSIPMediaInactivityTimersection.
TheSIPVoiceGatewaysfeatureissupportedforanalogendpointsthatareconnectedtoForeignExchange
Station(FXS)portsoraCiscoVG224AnalogPhoneGatewayandcontrolledbyaCiscocall-controlsystem,
suchasaCiscoUnifiedCommunicationsManager(CiscoUnifiedCM)oraCiscoUnifiedCommunications
ManagerExpress(CiscoUnifiedCME).
FormoreinformationonSIPGatewayfeaturesandinformationaboutconfiguringtheSIPvoicegatewayfor
VoIPv6,seetheConfiguringVoIPforIPv6.
VoIPv6 Support on Cisco UBE
CiscoUBEinVoIPv6addsIPv6capabilitytoVoIPfeatures.Thisfeatureaddsdual-stacksupportonvoice
gateways,IPv6supportforSIPtrunks,supportforSCCP-controlledanalogvoicegateways,supportfor
real-timecontrolprotocol(RTCP)pass-through,andsupportforT.38faxoverIPv6.
Formoreinformationonthesefeatures,refertothefollowing:
•“ConfiguringCiscoIOSGateways”sectionintheDeployingIPv6inUnifiedCommunicationsNetworks
withCiscoUnifiedCommunicationsManager
•“Trunks”sectioninDeployingIPv6inUnifiedCommunicationsNetworkswithCiscoUnified
CommunicationsManager
•“SCCP-controlledanalogvoicegateways”sectionintheSCCPControlledAnalog(FXS)Portswith
SupplementaryFeaturesinCiscoIOSGateways
•“RTCPPass-Through”sectionin CiscoUBERTCPVoicePass-ThroughforIPv6
•“T.38faxoverIPv6”sectionin Fax,Modem,andTextSupportoverIPConfigurationGuide
SupporthasbeenaddedforaudiocallsinmediaFlow–Through(FT)andFlow–Around(FA)modes,High
Density(HD)transcoding,LocalTranscodingInterface(LTI),alongwithVoiceClassCodec(VCC)support,
supportforHold/Resume,REFER,re-INVITE,302basedservices,andsupportformediaanti-trombonehave
beenaddedtoCiscoUBE.
CiscoUBEbeingasignalingproxyprocessesallsignalingmessagesforsettingupmediachannels.This
enablesCiscoUBEtoaffecttheflowofmediapacketsusingthemediaflow-throughandthemediaflow-around
modes.
•MediaFTandMediaFAmodessupportthefollowingcallflows:
◦EO–to–EO
◦DO–to–DO
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◦DO–to–EO
•MediaFlow-Through(FT):Inamediaflow–throughmode,betweentwoendpoints,bothsignaling
andmediaflowsthroughtheIP-to-IPGateway(IPIPGW).TheIPIPGWperformsbothsignalingand
mediainterworkingbetweenH.323/SIPIPv4andSIPIPv6networks.
Figure 1: H.323/SIP IPv4 – SIP IPv6 interworking in media flow-through mode
•MediaFlow-Around(FA):Mediaflow–aroundprovidestheabilitytohaveaSIPvideocallwhereby
signalingpassesthroughCiscoUBEandmediapassdirectlybetweenendpointsbypassingtheCisco
UBE.
Figure 2: H.323/SIP IPv4 - SIP IPv6 interworking in media flow-around mode
•AssistedRTCP(RTCPKeepalive):AssistedReal-timeTransportControlProtocol(RTCP)enables
CiscoUBEtogenerateRTCPkeepalivereportsonbehalfofendpoints;however,endpoints,suchas
secondgenerationCiscoIPphones(7940/7960)andNortelMediaGateways(MG1000T)donotgenerate
anyRTCPkeepalivereports.AssistedRTCPsenablecustomerstouseCiscoUBEtointeroperatebetween
endpointsandcallcontrolagents,suchasMicrosoftOCS/LyncsothatRTCPreportsaregeneratedto
indicatesessionlivelinessduringperiodsofprolongedsilence,suchascallholdorcallonmute.
TheassistedRTCPfeaturehelpsCiscoUBEtogeneratestandardRTCPkeepalivereportsonbehalfof
endpoints.RTCPreportsdeterminethelivelinessofamediasessionduringprolongedperiodsofsilence,
suchasacallonholdoracallonmute.
•SDPPass–Through:SDPisconfiguredtopassthroughtransparentlyattheCiscoUBE,sothatboth
theremoteendscannegotiatemediaindependentlyoftheCiscoUBE.
SDPpass-throughisaddressedintwomodes:
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•Flow-through—CiscoUBEplaysnoroleinthemedianegotiation,itblindlyterminatesand
re-originatestheRTPpacketsirrespectiveofthecontenttypenegotiatedbyboththeends.This
supportsaddresshidingandNATtraversal.
•Flow-around—CiscoUBEneitherplaysapartinmedianegotiation,nordoesitterminateand
re-originatemedia.Medianegotiationandmediaexchangeiscompletelyend-to-end.
Formoreinformation,refertothe“ConfigurablePass-throughofSIPINVITEParameters”sectionin
theCiscoUnifiedBorderElementSIPSupportConfigurationGuide.
•UDPChecksumforIPv6:UserDatagramProtocol(UDP)checksumsprovidedataintegrityfor
addressingdifferentfunctionsatthesourceanddestinationofthedatagram,whenaUDPpacketoriginates
fromanIPv6node.
•IPTollFraud:TheIPTollFraudfeaturechecksthesourceIPaddressofthecallsetupbeforerouting
thecall.IfthesourceIPaddressdoesnotmatchanexplicitentryintheconfigurationasatrustedVoIP
source,thecallisrejected.
Formoreinformation,refertothe“ConfiguringTollFraudPrevention”sectioninthe CiscoUnified
CommunicationsManagerExpressSystemAdministratorGuide.
•RTPPortRange:ProvidesthecapabilitywheretheportrangeismanagedperIPaddressrange.This
featuressolvestheproblemoflimitednumberofrtpportsformorethan4000calls.Itenablescombination
ofanIPaddressandaportasauniqueidentificationforeachcall.
•Hold/Resume:CiscoUBEsupportssupplementaryservicessuchasCallHoldandResume.Anactive
callcanbeputinheldstateandlaterthecallcanberesumed.
Formoreinformation,refertothe“ConfiguringCallHold/ResumeforSharedLinesforAnalogPorts”
sectioninSupplementaryServicesFeaturesforFXSPortsonCiscoIOSVoiceGatewaysConfiguration
Guide.
•CallTransfer(re-INVITE,REFER):Calltransferisusedforconferencecalling,wherecallscan
transitionsmoothlybetweenmultiplepoint-to-pointlinksandIPlevelmulticasting.
Formoreinformation,refertothe“ConfigurablePass-throughofSIPINVITEParameters”sectionin
theCiscoUnifiedBorderElementSIPSupportConfigurationGuide.
•CallForward(302based):SIPprovidesamechanismforforwardingorredirectingincomingcalls.A
UniversalAccessServers(UAS)canredirectanincomingINVITEbyrespondingwitha302message
(movedtemporarily).
•Consumptionof302atstacklevelissupportedforEO-EO,DO-DOandDO-EOcallsforall
combinationofIPv4/IPv6/ANAT.
•Consumptionof302atstacklevelissupportedforbothFTandFAcalls.
Formoreinformation,refertothe“ConfiguringCallTransferandForwarding”sectioninCiscoUnified
CommunicationsManagerExpressSystemAdministratorGuide.
•MediaAntitrombone:AntitromboningisamediasignalingserviceinSIPentitytoovercomethemedia
loops.MediaTrombonesaremedialoopsinaSIPentityduetocalltransferorcallforward.Medialoops
inCiscoUBEarenotdetectedbecauseCiscoUBElooksatbothcalltypesasindividualcallsandnot
callsrelatedtoeachother.
Antitromboneservicehastobeenabledonlywhennomediainterworkingisrequiredinbothlegs.Media
antitromboneissupportedonlywhentheinitialcallisinIPv4toIPv4orIPv6toIPv6modeonly.
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Formoreinformation,refertothe“ConfiguringMediaAntitrombone”sectionintheCiscoUnified
BorderElementProtocol-IndependentFeaturesandSetupConfigurationGuide.
•RE-INVITEConsumption:TheRe-INVITE/UPDATEconsumptionfeaturehelpstoavoid
interoperabilityissuesbyconsumingthemid-callRe-INVITEs/UPDATEsfromCiscoUBE.AsCisco
UBEblocksRE-INVITE/mid-callUPDATE,remoteparticipantisnotmadeawareoftheSDPchanges,
suchasCallHold,CallResume,andCalltransfer.
Formoreinformation,refertothe“CiscoUBEMid-callRe-INVITE/UPDATEConsumption”section
intheCiscoUnifiedBorderElementProtocol-IndependentFeaturesandSetupConfigurationGuide.
•AddressHiding:TheaddresshidingfeatureensuresthattheCiscoUBEistheonlypointofsignaling
andmediaentry/exitinallscenarios.Whenyouconfigureaddress-hiding,signalingandmediapeer
addressesarealsohiddenfromtheendpoints,especiallyforsupplementaryserviceswhentheCisco
UBEpassesREFER/3xxmessagesfromonelegtotheother.
Formoreinformation,refertothe“ConfiguringAddressHiding”sectionintheSIP-to-SIPConnections
onaCiscoUnifiedBorderElement.
•HeaderPassing:HeaderPassthroughenablesheaderpassingforSIPINVITE,SUBSCRIBEand
NOTIFYmessages;disablingheaderpassingaffectsonlyincomingINVITEmessages.Enablingheader
passingresultsinaslightincreaseinmemoryandCPUutilization.
Formoreinformation,refertothe“SIP-to-SIPConnectionsonaCiscoUnifiedBorderElement”section
intheSIP-to-SIPConnectionsonCiscoUnifiedBorderElement.
•Refer–ToPassing:TheRefer-toPassingfeatureisenabledwhenyouconfigurerefer-to-passingin
ReferPassthroughmodeandthesupplementaryserviceSIPReferisalreadyconfigured.Thisenables
thereceivedrefer-toheaderinReferPassthroughmodetomovetotheoutboundlegwithoutany
modification.However,whenrefer-to-passingisconfiguredinReferConsumptionmodewithout
configuringthesupplementary-serviceSIPRefer,thereceivedRefer-toURIisusedintherequest-URI
ofthetriggeredinvite.
Formoreinformation,refertothe“ConfiguringSupportforDynamicREFERHandlingonCiscoUBE”
sectionintheCiscoUnifiedBorderElementSIPConfigurationGuide.
•ErrorPass-through:TheSIPerrormessagepassthroughfeatureallowsareceivederrorresponsefrom
oneSIPlegtopasstransparentlyovertoanotherSIPleg.ThisfunctionalitywillpassSIPerrorresponses
thatarenotyetsupportedontheCiscoUBEorwillpreservetheQ.850causecodeacrosstwosipcall-legs.
Formoreinformation,refertothe“ConfiguringSIPErrorMessagePassthrough”sectionintheCisco
UnifiedBorderElementSIPSupportConfigurationGuide.
•SIPUPDATEInterworking:TheSIPUPDATEfeatureallowsaclienttoupdateparametersofasession
(suchas,asetofmediastreamsandtheircodecs)buthasnoimpactonthestateofadialog.UPDATE
withSDPwillsupportSDPPassthrough,mediaflowaroundandmediaflowthrough.UPDATEwith
SDPsupportforSIPtoSIPcallflowsissupportedinthefollowingscenarios:
•EarlyDialogSIPtoSIPmediachanges.
•MidDialogSIPtoSIPmediachanges.
Formoreinformation,refertothe“SIPUPDATEMessageperRFC3311”sectionintheCiscoUnified
BorderElementSIPSupportConfigurationGuide.
•SIPOPTIONSPing:TheOPTIONSpingmechanismmonitorsthestatusofaremoteSessionInitiation
Protocol(SIP)server,proxyorendpoints.CiscoUBEmonitorstheseendpointsperiodically.
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Formoreinformation,refertothe“CiscoUBEOut-of-dialogOPTIONSPingforSpecifiedSIPServers
orEndpoints”sectionintheConfigurationofSIPTrunkingforPSTNAccess(SIP-to-SIP)Configuration
Guide.
•ConfigurableErrorResponseCodeinOPTIONSPing:CiscoUBEprovidesanoptiontoconfigure
theerrorresponsecodewhenadialpeerisbusiedoutbecauseofanOut-of-DialogOPTIONSping
failure.
Formoreinformation,refertothe“ConfiguringanErrorResponseCodeuponanOut-of-DialogOPTIONS
PingFailure”sectioninthe CiscoUnifiedBorderElementSIPSupportConfigurationGuide.
•SIPProfiles:SIPprofilescreateasetofprovisioningpropertiesthatyoucanapplytoSIPtrunk.
•DynamicPayloadTypeInterworking(DTMFandCodecPackets):TheDynamicPayloadType
InterworkingforDTMFandCodecPacketsforSIP-to-SIPCallsfeatureprovidesdynamicpayloadtype
interworkingfordualtonemultifrequency(DTMF)andcodecpacketsforSessionInitiationProtocol
(SIP)toSIPcalls.TheCiscoUBEinterworksbetweendifferentdynamicpayloadtypevaluesacross
thecalllegsforthesamecodec.Also,CiscoUBEsupportsanypayloadtypevalueforaudio,video,
namedsignalingevents(NSEs),andnamedtelephoneevents(NTEs)inthedynamicpayloadtyperange
96to127.
Formoreinformation,refertothe“DynamicPayloadTypeInterworkingforDTMFandCodecPackets
forSIP-to-SIPCalls”sectionintheCiscoUnifiedBorderElement(Enterprise)Protocol-Independent
FeaturesandSetupConfigurationGuide.
•AudioTranscodingusingLocalTranscodingInterface(LTI):LocalTranscodingInterface(LTI)is
aninterfacecreatedtoremovetherequirementofSCCPclientforCiscoUBEtranscoding.
Forinformation,refertoCiscoUnifiedBorderElement9.0LocalTranscodingInterface(LTI).
•VoiceClassCodec(VCC)withorwithoutTranscoding:TheVoiceClassCodecfeaturesupports
basicandallRe-Invitebasedsupplementaryserviceslikecall-hold/resume,callforward,calltransfer,
whereifanymid-callcodecchanges,CiscoUBEinserts/removes/modifiesthetranscoderasneeded.
SupportfornegotiationofanAudioCodeconeachlegofaSIP–SIPcallontheCiscoUBEfeature
supportsnegotiationofanaudiocodecusingtheVoiceClassCodec(VCC)infrastructureonCiscoUBE.
VCCsupportsSIP-SIPcallsonCiscoUBEandallowsmid-callcodecchangeforsupplementaryservices.
How to Configure VoIP for IPv6
Configuring VoIP for IPv6
SIPisasimple,ASCII-basedprotocolthatusesrequestsandresponsestoestablishcommunicationamong
thevariouscomponentsinthenetworkandtoultimatelyestablishaconferencebetweentwoormoreendpoints.
UsersinaSIPnetworkareidentifiedbyuniqueSIPaddresses.ASIPaddressissimilartoane-mailaddress
andisintheformatofsip:[email protected].
Thegatewaycanbeeitheradomain(withorwithoutahostname)oraspecificInternetIPv4orIPv6address.
ASIPtrunkcanoperateinoneofthreemodes:SIPtrunkinIPv4-onlymode,SIPtrunkinIPv6-onlymode,
andSIPtrunkindual-stackmode,whichsupportsbothIPv4andIPv6.
ASIPtrunkusestheAlternativeNetworkAddressTransport(ANAT)mechanismtoexchangemultipleIPv4
andIPv6mediaaddressesfortheendpointsinasession.ANATisautomaticallyenabledonSIPtrunksin
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dual-stackmode.TheANATSessionDescriptionProtocol(SDP)groupingframeworkallowsuseragents
(UAs)toincludebothIPv4andIPv6addressesintheirSDPsessiondescriptions.TheUAisthenabletouse
anyofitsmediaaddressestoestablishamediasessionwitharemoteUA.
ACiscoUnifiedBorderElementcaninteroperatebetweenH.323/SIPIPv4andSIPIPv6networksinmedia
flow-throughmode.Inmediaflow-throughmode,bothsignalingandmediaflowsthroughtheCiscoUnified
BorderElement,andtheCiscoUnifiedBorderElementperformsbothsignalingandmediainteroperation
betweenH.323/SIPIPv4andSIPIPv6networks(seethefigurebelow).
Figure 3: H.323/SIP IPv4--SIP IPv6 Interoperating in Media Flow-Through Mode
Shutting Down or Enabling VoIPv6 Service on Cisco Gateways
SUMMARY STEPS
1. enable
2. configure terminal
3. voiceservicevoip
4. shutdown[forced]
DETAILED STEPS
Command or Action Purpose
Step 1 enable EnablesprivilegedEXECmode.
•Enteryourpasswordifprompted.
Example:
Device> enable
Step 2 configure terminal Entersglobalconfigurationmode.
Example:
Device# configure terminal
Step 3 voiceservicevoip EntersvoiceserviceVoIPconfigurationmode.
Example:
Device(config)# voice service voip
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