Sorry... should have added that OpenSIPS box is acting as mid-registrar On Thu, 7 Jan 2021, 12:12 Mark Allen, <m...@allenclan.co.uk> wrote:
> 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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