I am trying to work with a developer to troubleshoot echo issues in a mobile-phone SIP application (csipsimple). If I recall from my telecom days, there was sometimes a practice of intentionally feeding back some of the received signal. This was done to give the far-end talker a nice "reverb" sound to his voice, but is a disaster if there is delay in the line. Is there anything like this in pjsip? Again, the issue is that a far-end caller (call him Frank) calls the near end SIP device using pjsip (call her Sue), Frank experiences significant echo of his voice. Sue experiences no echo of her voice. It seems that Frank's echo problems are not reduced even when Sue covers her mic with her finger ("acoustic mute") while Frank talks. This leads me to think that Frank's voice is being resent (digitally, not acousticly) to Frank by Sue's device. Since there is non-trivial delay in the round trip, Frank hears significant echo. Any suggestions? Thanks!