Re: [reSIProcate] rtp stacks
Dan,
I've been using Jori's stuff and it handles all the RTP/RTCP stuff
internally.
http://research.edm.luc.ac.be/jori/jrtplib/jrtplib.html
If what I am reading below is correct then you are letting the RTP
handle the termination via
RTCP instead of using the SIP BYE message. From what I've seen and been
reading the number
of implementations that actually include RTCP is small - it is
preferrable to use the SIP BYE
message to set an indication to the RTP stream control to terminate the
data on both sides.
Jay
Dan Weber wrote:
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
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