[reSIProcate] rtp stacks
I am desperately looking for an rtp stack to accompany resiprocate in a
process to design a voip testing application.
The Application Flow:
client --- INVITE -> server
server --- OK -> client
client --- ACK -> server
client starts rtp to server
server starts rtp to client
client --- BYE (all transports) -> server
Basically I'm getting the rtcp reports from the rtp for some testing and
analysis. The issue I'm hitting among most rtp stacks is that they don't
autodestruct on bye and/or don't implement timeout interfaces. When two
applications are streaming rtp at each other from files or such, there is no
clear way to see when the streams have ended but with a timeout so it seems.
The SIP code is done and 100% working atm.
So what I'm looking for is an rtp stack that does multiplexing on two
sockets (rtp, rtcp) for all rtp sessions. Then my SipHandler would register
new RtpSessions with it. Any suggestions of one, or even a different
approach?
Dan