I have been trying to find documentation or examples of how to bridge two calls and then transfer the media to the new endpoint. Basically, the server places a call to a SIP URI and the call is established with endpoint A. Then the server places a call to a second endpoint B. After the call is established with endpoint B, the server performs a reinvite to transfer the RTP media stream to establish the RTP connection between endpoint A and endpoint B. I understand what needs to happen from a SIP standpoint, but I have not found the information on how to perform that operation with the Sofia-SIP library. Can someone please point me towards an example that shows how to accomplish this task. Thank you.
Jonathan
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