Axis C15 Series - Additional Settings; Calibrate and Run a Remote Speaker Test; Set up Direct SIP (P2 P)

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AXISC15Series

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.

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