On Tue, 15 Mar 2005 22:08:34 -0500, Java Rockx <javarockx(a)gmail.com> wrote:
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?
I'm working on reinvite with ser and mediaproxy, but on an older ser
(pre 0.8.14) version.
What I've seen is, is that this is a bug/feature in the mediaproxy
itself where a new INVITE will be ignored on an existing call. I've
fixed this on our development proxy a long time ago and can't remember
how much I had to fix.
When I'm done I will port it to the current source (if necessary) with
also some other stuff I'm working on.
--
Andreas Sikkema