Dear All,
I need to make all my rtp traffic through OpenSips to pass through rtp
proxy...I have the following route:
if(!cr_route("default", "0", "$rU", "$rU", "call_id")){
sl_send_reply("403", "Not allowed");
} else {
# In cas of failure, re-route the request
t_on_failure("1");
force_rtp_proxy();
t_relay();
}
The call is working fine but with no audio...How can i fix this issue in
order to have 2 way audio through rtpproxy?
Regards
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