Problem with RTPExport output video files

Hi, I have a problem with RTPExport output video files. One side streams H263/RTP(AVTransmit2.java) and other write this steam to a file by RTPExport.java. When network conditions are ideal, output video file has same fps and same number of frames like original file. Problem occures, when theres packet lost in network, then output file has different fps,and also has less frames like original video(because it didnt write missing frames to file, and thats why it get shorter). Pls how can I achieve output file that will have the same fps like original one? How to write to file an identical copy of what I can see while receiveing video with AVReceive2.java? Its there a way to modifi rtpexport or avreceiver to do this? Thanks a lot!

Trubka wrote:
When network conditions are ideal, output video file has same fps and same number of frames like original file. Problem occures, when theres packet lost in network, then output file has different fps,and also has less frames like original video(because it didnt write missing frames to file, and thats why it get shorter). Okay, first off, the second file is smaller on purpose. RTP intentionally drops packets that are old/out of order in order to make sure real-time video stays as close to real time as it can. This is by design, so there's really nothing that can be done about it.
How to write to file an identical copy of what I can see while receiveing video with AVReceive2.java? Technically speaking, what you're getting in the RTPExport is exactly what you got on the receiving end. Any frames that are dropped during transmission will not be seen by the receiver, nor saved by the receiver.
Pls how can I achieve output file that will have the same fps like original one? I'm not 100% sure that you can, but, you can give the following idea a try. I make no guarentees that it'll work, but it should work for you...
[http://java.sun.com/javase/technologies/desktop/media/jmf/2.1.1/solutions/RTPConnector.html]
That example is an example of a "custom transport layer" for RTP connections. Essentially, it's some code that's handed the RTP packets on the transmission end, and it's expected to deliver those RTP packets on the other end. And it doesn't care about how they get from A to B, only that they do.
If you were to replace the UDP socket in that example with a TCP socket, you would be guarenteed not to drop packets due to network reasons. Every RTP packet you were handed by the transmitter, you would then hand to the receiver. There is no guarentee that none of the packets would be cast away as "old" by the RTP protocol itself, but there's also no guarentee any of them would be. It's a crap-shoot at best, but it's certainly worth a try.

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