16 Mar
2005
16 Mar
'05
3:46 p.m.
Adrian,
Thanks for the tip. Just to make sure I understand your comment should
I do what is shown below? If I do this won't calls that didn't use
mediaproxy on the original INVITE be sent through the mediproxy on
re-INVITEs or is this incorrect because I didn't call
fix_nated_contact() in the loose_route() section?
route {
#usual sanity checks
if (loose_route()) {
if (method=="INVITE") {
use_media_proxy();
};
r_relay();
break;
};
#ususal other processing like lookup("location")
}
Regards,
Paul
On Wed, 16 Mar 2005 15:39:41 +0100, Adrian Georgescu <ag(a)ag-projects.com> wrote:
Just call the use of media_proxy, if the session
exists it will use it
otherwise not. Simple.
Arian
>>>>>>
Hi All.
I'm using ser-0.9.1.
Is there a way to determine if mediaproxy is in use for an existing
SIP call so that re-INVITE messages can avoid losing audio when one or
the other SIP UAs are NATed?
But nothing has led me to a solution. I cannot just use
lookup("location") and test the nat_flag since that won't always work
on re-INVITEs. A mediaproxy function for something like
is_existing_session() would be awesome to lookup the Call-ID in the
existing mediaproxy sessions.
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