[reSIProcate] rtp stacks

[Jay Hogg]

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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