Hi, Ross, thanks for quick response. In general, I don't need RTSP for receiving audio. My ipcam has RTSP server with two audio/video subsessions and needs some interface to output sound, just to play all incoming data (in current implementation).
So I started from this topic http://lists.live555.com/pipermail/live-devel/ 2012-September/015923.html and made working example. Now I need TCP. I didn't find examples with pure RTP-over-TCP and figured out that I should get it from RTSPClient implementation. Answers for your questions: 1. Server is Android application and it doesn't use live555 library, it sends RTP packets, not RTSP. RTP packet: 4 bytes (RFC 2326, section 10.12) + 12 bytes (header) + 160 byte (payload, G711) 2. Client uses live555 library, receives and decodes RTP using subclass of MediaSink with SimpleRTPSource. _______________________________________________ live-devel mailing list live-devel@lists.live555.com http://lists.live555.com/mailman/listinfo/live-devel
