29 Nov
2007
29 Nov
'07
2:12 p.m.
Hello,
if the call goes through asterisk, it should work without
nathelper/rtpproxy if you set "nat=yes" in asterisk config file.
However, you do not mark the re-INVITE as being for a NATted call, check
openser page of voip-info.org to see some examples.
Cheers,
Daniel
On 11/26/07 11:19, srinivas Antarvedi wrote:
Hello all,
i have users one is on global ip and another behind NAT
am using asterisk as media server
leg 1:
caller : Global ip UAC.
callee: asterisk
leg2:
caller :asterisk
callee: NATed UAC.
sdp of NATed client is handled at openser reply route at first stage
when asterisk re-invites the NATed UAC to bridge the two call-Leg's
the sdp from NATed UAC is not changed ,, even if i call t_on_reply
in the loose route section of the script.. it is still showing privat ip
so finally after 2 or 3 sec's there was an end to the dialog
can anybody have any idea to handle re-invite's 200 ok SDP mangling?
please help me out..
Thanks in advance
regards
srinivas
--
Srinivas Antarvedi
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