[OpenSIPS-Users] RTPProxy No Audio on Outbound Calls

[John Quick]

Mark,

You can detect if the INVITE came from your Asterisk by testing the $si
pseudo-variable.
That will allow you to identify the direction of the call. I usually set a
flag for this purpose. For example:
If ($si == "my.ast.er.isk")
    setflag(DIR_OUT);

At the point where you engage the rtpproxy, you will then be able to reverse
the internal/external parameters for the function call depending on the
direction of the call

If (isflagset(DIR_OUT)) {
    rtpproxy_offer("corfei");
} else {
    rtpproxy_offer("corfie");
}

The same flag should still be valid in the onreply handler where you can do
something similar. [Not sure if I have ie/ei the right way round in my
example].

That said, I'm not sure this topology is a good one to be using.
I would generally try to avoid having the media proxy behind NAT and also
using it in bridging mode - it makes life too complicated.

P.S. Looks like you sorted out the problems with the call to do_routing().

John Quick
Smartvox Limited