On Mon, Aug 31, 2009 at 8:10 AM, Leon Li<[email protected]> wrote: > Ok, I modified the supported codec on the SIP client and the call went > through successfully. However, I got one way audio. UA2(callee) with public > IP can hear, but UA1(caller) with private IP cannot. > > The INVITE msg arrived on UA2 is like with the private IP in SDP. I think > this causes the problem because UA2 (public IP) won't know how to get to > private ip. >
That's right. > How can I debug and check whether OpenSIPs did NAT/mediaproxy part properly? > Add some xlog to your NAT handling part of the script to see if you are correctly calling nat fixing functions. Regards, -- /Saúl http://www.saghul.net | http://www.sipdoc.net _______________________________________________ Users mailing list [email protected] http://lists.opensips.org/cgi-bin/mailman/listinfo/users
