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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