[OpenSER-Users] Handling re-INVITES --hellp required
Daniel-Constantin Mierla
daniel at voice-system.ro
Thu Nov 29 13:12:29 CET 2007
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
> ------------------------------------------------------------------------
>
> _______________________________________________
> Users mailing list
> Users at lists.openser.org
> http://lists.openser.org/cgi-bin/mailman/listinfo/users
>
More information about the Users
mailing list