Axis C1510 - Additional Settings; Calibrate and Run a Remote Speaker Test; Set up Direct SIP (P2 P)

AXISC15Series
Additionalsettings
Additionalsettings
Calibrateandrunaremotespeakertest
Youcanrunaspeakertesttoverifyfromaremotelocationthataspeakerisworkingasintended.Thespeakerperformsthetestby
playingaseriesoftesttonesthatareregisteredbythebuilt-inmicrophone.Everytimeyourunthetest,theregisteredvaluesare
comparedwiththevaluesthatwereregisteredduringthecalibration.
Note
Thetestmustbecalibratedfromitsmountedpositionattheinstallationsite.Ifthespeakerismovedorifitslocal
surroundingschange,forinstanceifawallisbuiltorremoved,thespeakershouldbere-calibrated
Duringcalibration,itisrecommendedthatsomeoneisphysicallypresentattheinstallationsitetolistentothetesttonesand
ensurethatthetesttonesarenotmufedorblockedbyanyunintendedobstructionsinthespeaker’sacousticpath.
1.Gotothedeviceinterface>Audio>Speakertest.
2.Tocalibratetheaudiodevice,clickCalibrate.
Note
OncetheAxisproductiscalibrated,thespeakertestcanberunatanytime.
3.Torunthespeakertest,clickRunthetest.
Note
Itisalsopossibletorunthecalibrationbypressingthecontrolbuttononthephysicaldevice.SeeProductoverviewonpage
44toidentifythecontrolbutton.
SetupdirectSIP(P2P)
Usepeer-to-peerwhenthecommunicationisbetweenafewuseragentswithinthesameIPnetworkandthereisnoneedforextra
featuresthataPBX-servercouldprovide.TobetterunderstandhowP2Pworks,seePeer-to-peerSIP(P2PSIP)onpage13.
Formoreinformationaboutsettingoptions,seeSIPonpage32.
1.GotoSystem>SIP>SIPsettingsandselectEnableSIP.
2.Toallowthedevicetoreceiveincomingcalls,selectAllowincomingcalls.
3.UnderCallhandling,setthetimeoutanddurationforthecall.
4.UnderPorts,entertheportnumbers.
-SIPport–ThenetworkportusedforSIPcommunication.Thesignalingtrafcthroughthisportisnon-encrypted.
Thedefaultportnumberis5060.Enteradifferentportnumberifrequired.
-TLSport–ThenetworkportusedforencryptedSIPcommunication.Thesignalingtrafcthroughthisportis
encryptedwithTransportLayerSecurity(TLS).Thedefaultportnumberis5061.Enteradifferentportnumber
ifrequired.
-RTPstartport–EntertheportusedfortherstRTPmediastreaminaSIPcall.Thedefaultstartportfor
mediatransportis4000.SomerewallsmightblockRTPtrafconcertainportnumbers.Aportnumbermust
bebetween1024and65535.
5.UnderNATtraversal,selecttheprotocolsyouwanttoenableforNATtraversal.
Note
UseNATtraversalwhenthedeviceisconnectedtothenetworkfrombehindaNATrouterorarewall.Formoreinformation
seeNATtraversalonpage14.
7