I've observed that FreeSWITCH will initially transmit RTP to the remote party's private IP address until it has received at least 10 packets, at which time it will start sending to the remote party's NATed address.  Is there some way to suspend the transmission of RTP altogether until at least one packet has been received from the remote party?  It would be ideal if this could be controlled on a per call and/or per SIP profile basis.

Thanks,

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