If the SDP is correct, then you might have specific issues related to your specific deployment case. Snippets from others config files won't help. You really need to investigate and understand your particular issue that you are facing and fix it accordingly.
Regards,
Ovidiu Sas
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
_______________________________________________
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
sr-users@lists.sip-router.org
http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users