RTP Transmission always lose the last packet

Hi,
I transmit a audio file through rtp by using the JMFStudio(V 2.1.1e), then capture the rtp packets on the sending port, but the last rtp packet is missing every time.
Several developers of my team tried on their own machine and got the same result.
Did anyone try this before or if anybody can tell me how to solve it?
Thanks a lot!

Several thoughts...
If the last "packet" isn't big enough, IE, a full payload worth of data, the underlaying transport mechanism may keep it trapped in the buffer forever. If it sends packets based on a BufferFull event of some sort, then the last packet won't ever be sent because it'll get trapped in the buffer. This is a big problem with TCP sockets, and in that instance, the buffer is flushed automaticlly when a close operation is called on the socket.
I have no idea how you're managing the sending of RTP packets, but that's a possible reason.
Another possible reason would be how you're handling the EndOfMedia event. If you were to, say, stop the entire process down before the final RTP packet got sent out, then you'd also not be getting the last packet sent out. Everything in JMF pretty much spawns a new execution thread, so it's possible the main thread is shutting those other threads down before they're done doing work with calls to whatever.Stop() and whatever.Close()...
Those are just 2 ideas off the top of my head to explain why a last packet may not be sent out...

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