[OpenSIPS-Users] RTPProxy No Audio on Outbound Calls
Mark Farmer
farmorg at gmail.com
Wed Feb 13 09:30:44 EST 2019
Hello everyone, all help gratefully received, I've been slogging away at
this for ages!
I have OpenSIPS 2.4.4 & RTPProxy behind 1:1 NAT's (different hosts).
RTPProxy runs so:
/usr/local/bin/rtpproxy -s unix:/var/run/rtpproxy/rtpproxy.sock -u rtpproxy
rtpproxy -p /var/run/rtpproxy/rtpproxy.pid -s udp:10.96.16.58 7722 -l
10.96.0.58 10.98.0.58 -A ext.ip.addr.ess 10.98.0.58 -d DBUG LOG_LOCAL0 -m
10000 -M 20000
OpenSIPS is sitting between my provider & an Asterisk server which has
phones registered.
When I make calls 'Provider -> OpenSIPS/RTPProxy -> Asterisk -> Phone' all
is good, 2 way audio.
But when the call flows in the opposite direction, I get no audio since SDP
is the same as the 1st call.
How do I get it to reverse the rtpproxy_offer/answer flags?
These are the bits that handles it all:
route[RTPPROXY] {
if (is_method("BYE|CANCEL")) {
rtpproxy_unforce();
}
if (is_method("INVITE")) {
rtpproxy_offer("corwfie");
}
}
onreply_route[DROUTING] {
if (is_method("BYE|CANCEL")) {
sip_trace("tid","d");
rtpproxy_unforce();
}
if ($rs=~"(2[0-9][0-9])") {
rtpproxy_answer("corwfei");
}
}
--
Mark Farmer
farmorg at gmail.com
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