[OpenSIPS-Users] One way audio - media port changed (opensips / mediaproxy)

[Magnus Burman]

None of my users are behind NAT, they're all on public IPs (I control their
connection).

Sorry if it's a stupid question, but what do you mean with "the SDP is not
modified by mediaproxy"?

On line 276 in the re-invite (Opensips --> UA) the port used is different:
m=audio 40518 RTP/AVP 18 8 0 101'
The original invite (line 73) reads: m=audio 58928 RTP/AVP 18 8 0 101'

In the statistics dumped at the end, port says 58928.

This is my config in regards to mediaproxy. I included the loose_route above
as well, as this is all that comes before it in the route:

    ##Loose_route packets
    if (loose_route()) {
        # mark routing logic in request
        append_hf("P-hint: rr-enforced\r\n");
        if(method=="BYE") {
                setflag(1);
                $avp(s:can_uri) = $ru;
        }
        route(1);
    };

    if (method==INVITE && !has_totag()) {
        if(avp_db_query("select 1 from subscriber_debug where username =
'$fU'", "$avp(s:debug)")) {
            # No media proxy if in debug
        } else {
            engage_media_proxy();
        }
    }


2010/2/3 Iñaki Baz Castillo <ibc at aliax.net>

> El Miércoles, 3 de Febrero de 2010, Magnus Burman escribió:
> > Nice little utility, saves alot of time on typing. :-)
> >
> > Here's a pastbin with the correct format (ngrep-sip b) of the same call:
> > http://pastebin.ca/1776903
>
> As you can see, the SDP in not modified by mediaproxy module for the
> re-INVITE
> and the 200 response for the re-INVITE. This means that you get
> one-way-audio
> as the caller is behind NAT.
>
> You should inspect why you are not calling mediaproxy functions when
> handling
> in-dialog INVITE and its responses.
>
>
> --
> Iñaki Baz Castillo <ibc at aliax.net>
>
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