[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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