Hello ppl,
This post is to do with the variables 'nat' or 'canreinvite' for sip
entities.
Idealy users, wont be static, they could be roaming all over the globe.
So, setting someone as behind NAT, and disabling canreinvite, etc.,
restricts the roaming capabilities of a user.
Is there any way for Asterisk to auto detect, if a user is behind NAT,
also, if two users are behind the same NAT, help in having a
peer-to-peer rtp flow between the two users in the call??
cheerz
- Ben
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