Axis M4215-V - Page 41

AXISM4215-VDomeCamera
Thewebinterface
EnableSIP:CheckthisoptiontomakeitpossibletoinitiateandreceiveSIPcalls.
Allowincomingcalls:CheckthisoptiontoallowincomingcallsfromotherSIPdevices.
Callhandling
•Callingtimeout:Setthemaximumdurationofanattemptedcallifnooneanswers.
•Incomingcallduration:Setthemaximumtimeanincomingcallcanlast(max10min).
•Endcallsafter:Setthemaximumtimethatacallcanlast(max60minutes).SelectInnitecalldurationifyoudon’t
wanttolimitthelengthofacall.
Ports
Aportnumbermustbebetween1024and65535.
•SIPport:ThenetworkportusedforSIPcommunication.Thesignalingtrafcthroughthisportisnon-encrypted.
Thedefaultportnumberis5060.Enteradifferentportnumberifrequired.
•TLSport:ThenetworkportusedforencryptedSIPcommunication.Thesignalingtrafcthroughthisportisencrypted
withTransportLayerSecurity(TLS).Thedefaultportnumberis5061.Enteradifferentportnumberifrequired.
•RTPstartport:ThenetworkportusedfortherstRTPmediastreaminaSIPcall.Thedefaultstartportnumberis
4000.SomerewallsblockRTPtrafconcertainportnumbers.
NATtraversal
UseNAT(NetworkAddressTranslation)traversalwhenthedeviceislocatedonanprivatenetwork(LAN)andyouwantto
makeitavailablefromoutsideofthatnetwork.
Note
ForNATtraversaltowork,theroutermustsupportit.TheroutermustalsosupportUPnP®.
EachNATtraversalprotocolcanbeusedseparatelyorindifferentcombinationsdependingonthenetworkenvironment.
•ICE:TheICE(InteractiveConnectivityEstablishment)protocolincreasesthechancesofndingthemostefcient
pathtosuccessfulcommunicationbetweenpeerdevices.IfyoualsoenableSTUNandTURN,youimprovetheICE
protocol’schances.
•STUN:STUN(SessionTraversalUtilitiesforNAT)isaclient-servernetworkprotocolthatletsthedevicedetermineif
itislocatedbehindaNATorrewall,andifsoobtainthemappedpublicIPaddressandportnumberallocatedfor
connectionstoremotehosts.EntertheSTUNserveraddress,forexample,anIPaddress.
•TURN:TURN(TraversalUsingRelaysaroundNAT)isaprotocolthatletsadevicebehindaNATrouterorrewallreceive
incomingdatafromotherhostsoverTCPorUDP.EntertheTURNserveraddressandthelogininformation.
Audioandvideo
•Audiocodecpriority:SelectatleastoneaudiocodecwiththedesiredaudioqualityforSIPcalls.Drag-and-dropto
changethepriority.
Note
Theselectedcodecsmustmatchthecallrecipientcodec,sincetherecipientcodecisdecisivewhenacallismade.
•Audiodirection:Selectallowedaudiodirections.
•H.264packetizationmode:Selectwhichpacketizationmodetouse.
-Auto:(Recommended)Thedevicedecideswhichpacketizationmodetouse.
-None:Nopacketizationmodeisset.Thismodeisofteninterpretedasmode0.
-0:Non-interleavedmode.
-1:SingleNALunitmode.
•Videodirection:Selectallowedvideodirections.
Additional
•UDP-to-TCPswitching:SelecttoallowcallstoswitchtransportprotocolsfromUDP(UserDatagramProtocol)toTCP
(TransmissionControlProtocol)temporarily.Thereasonforswitchingistoavoidfragmentation,andtheswitchcan
takeplaceifarequestiswithin200bytesofthemaximumtransmissionunit(MTU)orlargerthan1300bytes.
•Allowviarewrite:SelecttosendthelocalIPaddressinsteadoftherouter'spublicIPaddress.
•Allowcontactrewrite:SelecttosendthelocalIPaddressinsteadoftherouter'spublicIPaddress.
•Registerwithserverevery:SethowoftenyouwantthedevicetoregisterwiththeSIPserverfortheexisting
SIPaccounts.
•DTMFpayloadtype:ChangesthedefaultpayloadtypeforDTMF.
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