[OpenSIPS-Users] Mediaproxy configuration

[Mark Allen]

I wonder if anyone can help me with this? I am trying to configure
Mediaproxy to handle RTP traffic coming from outside our local network.
Here's the setup:

    UAC ---> IPA ---> IPB ---> Mediaproxy / OpenSIPS ---> Asterisk

IPA (a public IP address 4x.xxx.xxx.xxx) maps ports ports 5060 and 10000 to
65535 to IPB (local IP address 192.168.xxx.xxx). IPB is actually a Virtual
IP managed by keepalived.
UAC is MizuDroid app running on my Android phone connected to my home
network (NATed) with a public IP of 5.xxx.xxx.xxx. Everything else relates
to our office network.
Mediaproxy Dispatcher and Relay are both running on the same (OpenSIPS)
system

SIP conversation between UAC and Asterisk via OpenSIPS looks to be working
fine. Endpoints connect, exchange data, and hangup. The problem is with SDP
addressing (NAT problem) causing no audio either way, which is what I want
Mediaproxy to handle.

In opensips.cfg I'm passing control for calls arriving at IPA to
Mediaproxy...

    if (is_method("INVITE")) {
        if (!has_totag()) {
            if ($fd == "4x.xxx.xxx.xxx") {
                xlog("Passing control to Mediaproxy...");
                engage_media_proxy();
            }
        }
    }

In /etc/mediaproxy/config.ini all settings are defaults except for setting
dispatcher as IPB...

    dispatchers = 192.168.xxx.xxx

...and I've tried it with and without advertised_ip set to IPA...

    advertised_ip = 4x.xxx.xxx.xxx


I can see that Mediaproxy is taking control of calls as instructed and
making changes to SDP but it's not solving my audio problems. What am I
doing wrong????
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