Hi John, rtpproxy is not enough if you are using asterisk in your environment. You have to check that asterisk is configured to work with NAT, otherwise you will experience audio problems. Are the asterisk RTP ports enabled/forwarded on your firewall?
Regards, Kostas On Jan 21, 2014, at 2:24 PM, John Smith <jsmith...@mail.com> wrote: > Hi Fred, > > I have followed your HOWTO and the scenario remains exactly the same. > > I see traffic from Phone1 IP to Kamailio private IP, from Kamailio private IP > to Asterisk IP, and back directly to Phone2 public IP. > > I might be making wrong assumptions regarding this traffic flow. Is that > correct? > > Thank you > > _______________________________________________ > 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 _______________________________________________ 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
