[Freeswitch-users] Manipulating audio streams at sample level

I have a scenario where I'd like to be able to manipulate audio streams coming into FS on a sample level. This is a 3-party SIP call where 2 parties (A and B) are in the same room but using individual handsets. This creates an echo effect for the third party (C) since the audio streams from A and B will arrive at her location slightly out of sync. So at the start of a call I want to examine the audio streams coming from A B as they come into FS to see how out of sync they are, then add an appropriate delay to the one that's ahead and put that delayed audio back into FS. Oivvio