On 21.01.2014 13:24, John Smith wrote:
I might be making wrong assumptions regarding this traffic flow. Is that
correct?
That depends on your policy. It is up to you to define how RTP should be
routed. There are basically 2 choices:
a) RTP from clients is handled by rtpproxy:
phone1 <-nat-> rtpproxy <--> Asterisk <--> rtpproxy <-nat-> phone2
In this case, only the private IP of Kamailio and rtpproxy (can be the
same IP address) must be mapped to a public IP address.
b) RTP directly to Asterisk:
phone1 <-nat-> Asterisk <-nat-> phone2
In this case, the private IPs of Kamailio and Asterisk must be mapped to
a public IP address.
When using version a) you have to make sure to set the proper IP address
in the SDP. For example, SDPs in messages from Kamailio to the phone
must contains the PUBLIC IP of rtpproxy in the c=... line. SDPs in
messages from Kamailio to Asterisk must contain the PRIVATE IP of
rtpproxy in the c=... line.
regards
Klaus
_______________________________________________
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
