Call failed - incompatible SDP Error in Webrtc call

FreePBX Endpoints
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– I have fresh install freepbx server.
–For testing webrtc call using online website - tryit-jssip
–I have configured pjsip two extensions only (no trunk config done)
–Now testing call between those two extensions but from above website cann’t make call
–Getting error - Call failed - incompatible SDP
–when make call using normal sip in zoiper - call works fine.
–The whole purpose of this test is to build a new softphone app like Zoiper on webrtc protocol.

Please guide if we need to do any configurations after fresh installation of freepbx in order to make call webrtc success?

Thanks

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Asterisk is rejecting the call due to a lack of compatible codecs or other SDP parameter in the received INVITE request. You need to look at the logs to get more detail and arrive at a root cause. It could be caused by a missing STUN configuration. Do you have ws transport enabled on PJSIP settings?

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