On Tue, 15 Mar 2005 22:08:34 -0500, Java Rockx javarockx@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.