Actually, it should work without any NAT traversal done in Asterisk, if Asterisk communicates never direct with the phones, but only via Kamailio and rtpproxy. In this case, Asterisk can use private IP addresses. All the near-end NAT traversal can be done in Kamailio.
regards Klaus
On 21.01.2014 14:06, meres wrote:
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.15@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