Hi,
I can not get this working:
Asterisk on public IP.
Two SIP phones behind NAT - in the same LAN.
I works perfectly (two way sound) when each peer (friend) can not
reinvite - audio stream goes through Asterisk.
The problem pops up when I define canreinvite=yes on each peer
definision so I suppose to stream audio directly between phones (in the
same local LAN).
Right after called party answers, Asterisk sends new INVITE's to each
phone pointing in DSP messages that audio should be sent to ip:port of
each phone. So, phone A is sending RTP to gateways public ip ad port of
phone B - all this fails, response from gateway is 'Destination
unreachable (Port unreachable)?
Why is it so?
In 'no reinvite' scenario Asterisk communicates whit each phone without
any problems so why phone can not send rtp to another port and Asterisk can?
Is it possible to get this working at all?
Lukasz.
