Call failed - incompatible SDP Error in Webrtc call

– 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?


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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