Some SIP providers and server developers thinks that it is possible to guess if a user can send/receive RTP traffic direct in his IP if the SIP Signaling network is public IP.
WRONG!
The correct way to do this is using a connectivity check algorithm (STUN, ECHO, etc...) based on the UDP Channel that you may use to send/receive the streaming.
SIP Providers, stop doing guesses about User Agents NAT description based on the signaling channel.
This kills the interoperability of your SIP Services.
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