[Live-devel] Problem with rtcp

[Andrea Ricciardi]

Hi everybody,
I developed an embedded application that creates one MP4 stream + one PCM stream from a camera and a microphone. I need to send them over the network using RTP. My application writes the 2 streams to 2 separate linux fifos, /fifo_video and /fifo_audio, so I simply modified the testMPEG4VideoStreamer and testWAVAudioStreamer to read from the fifos instead of the "test.*" files.
Then I use vlc with an sdp file to receive the streams, but I only get a few random video and audio frames, with long periods with no video or audio.
Moreover, If I modify the sdp file to receive only one stream, either the video or the audio one, everything works perfectly, no frame loss.
So I suspected it could be a problem of audio - video synchronization, and tried to remove the rtpc creation from the test*Streamer applications (commented the RTCPInstance::createNew calls) and this way I can play both audio and video streams together in vlc without any problem (apart from the fact that of course they are 1-2 seconds out of sync).
So the questions are:
1) is my setup ok or do I miss something?
2) Any idea about why the presence of the rtcp messages modify the vlc behavior in such a way that it can't play the video and audio streams?

Thanks in advance,

Andrea
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