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
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