Hello, I have a problem with some calls where the rtp stream from the other side changes the rtp ptime multiple times. I make a call with the sdp line "a=ptime:20" but the initial rtp comes with 40ms ptime. FreeSWITCH detects it and changes the ptime to 40ms and all works ok. Later in the conversation, without any re-invite, the rtp stream changed to 20ms. At this moment, FreeSWITCH does not detect the change and the audio sended by FreeSWITCH to the other legs is choppy. I think that the problem is on the other side, but the other side is a big operator and always this is a problem to solve issues. I tried to set "rtp_media_autofix_timing", but if I set this property, the first part of the conversation is discarded (ptime at 40ms, silence in my phone) and later I can listen the audio (when the ptime=20ms). Is there any solution for this? Best regards -- Jose Fco. Irles Durá