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...@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


_______________________________________________
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

Reply via email to