8 Oct
2015
8 Oct
'15
9:03 a.m.
On 30-09-15 13:29, Fred Posner wrote:
Without a version of rtpproxy using the -A flag, you'll need to either
(1) update to a different version of rtpproxy or (2) skip rtpproxy and
have your asterisk handle all the rtp.
I tried rtpproxy v2, with the -A flag in bridge mode ( -A
privateip/publicip ). This doesn't reflect anything in the SIP headers.
The problem is a bit more complex I think, because all INVITEs to
gateways contain the same internal IPs from Asterisk and Kamaialio in
their From and To header. SDP information is correctly being displayed.
But it seems that some UAs disregard what's in the SDP descriptors and
just look at the SIP headers (To/From/Contact).
Can anyone share their config snippets about how they've delt with the
Asterisk behind NAT situation? It would really be appreciated!
Cheers,
Dirk