Axis C1310-E - SIP via PBX Server Setup

AXISC1310-ENetworkHornSpeaker
Additionalsettings
-RTPstartport–EntertheportusedfortherstRTPmediastreaminaSIPcall.Thedefaultstartportfor
mediatransportis4000.SomerewallsmightblockRTPtrafconcertainportnumbers.Aportnumbermust
bebetween1024and65535.
5.UnderNATtraversal,selecttheprotocolsyouwanttoenableforNATtraversal.
Note
UseNATtraversalwhenthedeviceisconnectedtothenetworkfrombehindaNATrouterorarewall.Formoreinformation
seeNATtraversalonpage12.
6.UnderAudio,selectatleastoneaudiocodecwiththedesiredaudioqualityforSIPcalls.Drag-and-droptochange
thepriority.
7.UnderAdditional,selectadditionaloptions.
-UDP-to-TCPswitching–SelecttoallowcallstoswitchtransportprotocolsfromUDP(UserDatagramProtocol)
toTCP(TransmissionControlProtocol)temporarily.Thereasonforswitchingistoavoidfragmentation,and
theswitchcantakeplaceifarequestiswithin200bytesofthemaximumtransmissionunit(MTU)orlarger
than1300bytes.
-Allowviarewrite–SelecttosendthelocalIPaddressinsteadoftherouter'spublicIPaddress.
-Allowcontactrewrite–SelecttosendthelocalIPaddressinsteadoftherouter'spublicIPaddress.
-Registerwithserverevery–SethowoftenyouwantthedevicetoregisterwiththeSIPserverfortheexisting
SIPaccounts.
-DTMFpayloadtype–ChangesthedefaultpayloadtypeforDTMF.
8.ClickSave.
SetupSIPthroughaserver(PBX)
UseaPBX-serverwhenthecommunicationshouldbebetweenaninnitenumberofuseragentswithinandoutsidetheIPnetwork.
AdditionalfeaturescouldbeaddedtothesetupdependingonthePBX-provider.TobetterunderstandhowP2Pworks,seePrivate
BranchExchange(PBX)onpage1 1.
Formoreinformationaboutsettingoptions,seeSIPonpage26.
1.RequestthefollowinginformationfromyourPBXprovider:
-UserID
-Domain
-Password
-AuthenticationID
-CallerID
-Registrar
-RTPstartport
2.Toaddanewaccount,gotoSystem>SIP>SIPaccountsandclick+Account.
3.EnterthedetailsyoureceivedfromyourPBXprovider.
4.SelectRegistered.
5.Selectatransportmode.
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