I am using FreeSWITCH as a B2BUA to handle WebRTC calls and link them to a PBX that does not have WebRTC support. When the PBX calls a WebRTC user, FreeSWITCH answers the initial call, then initiates the call to the WebRTC user. This causes the first 2-4 seconds of the audio to be lost.

Is there a way to have the first call wait until the second call is answered?

Thank you,

Dan