I am implementing a RTSP client and use http://m.youtube.com <http://m.youtube.com/> to verify my program.
This RTSP server will output burst data in the beginning. It’s about 3-second data fed within the first second. When sending PAUSE and then PLAY command, the sending speed from server in the beginning is much higher than stream bitrate. If PAUSE and PLAY are sent again and again, the burst data is more and more. That is, queued RTP packets are too many to play with correct presentation time. Lastly, the buffer overflows. How to avoid the problem? Enlarging the buffer size seems not a good solution since we have no idea how many PAUSE and PLAY commands will be sent. If you have a suggestion about buffer size or other solution, please kindly advise.
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