[Live-devel] Help: cause of RTP packet loss

[周弘]

Hello,   I can't answer your questions, but I have some suggestions.    First, making sure your stream is good with UDP client and server.Maybe the problem does not come from RTP.    Second, your test program must run in the simplest net environment.    Third, you must know the problem come from which side- receiver or sender or both of them.  
在2007-02-13，xcsmith at rockwellcollins.com 写道：
Hello! I hope you will please help me figure out the reason I am losing RTP packets when recording a live stream. The RTCP of my openRTSP reports packet loss (I'm using ethereal to look at this). My recorded video also shows large chunks of missing data and freezes. When I use VLC to play the recorded video, VLC messages say "buffer underrun" every time the video freezes. So I think that somehow I am losing RTP packets. When I play the video live using VLC as an RTSP client, the video is fine. It seems that about every 30 - 40 seconds, a large number of packets are dropped. I thought this was happening every time the packet reorder threshold is reached, but now I'm not quite so sure of this. I tried increasing the wait threshold to 40 seconds and the RTP socket buffer in MultiFramedRTPSource to 100 MB (which I see is successfully changed). I have varied the wait threshold to 1, 2, 20 and 40 seconds. I also used the default 100ms. I have tried the following things to figure out the cause of lost packets. 1. In ReorderingPacketBuffer::getNextCompletedPacket() print debug statement any time fHeadPacket->rtpSeqNo() != fNextExpectedSeqNo. I got quite a lot of printout from this statement. I printed both the head packet and the expected sequence number. I checked ethereal, and the missing seq no. packets had been sent to me. Later I removed this and just print a single statement any time the "packetLossPreceded" was set to true. 2. Add printf to MultiframedRTPSource::networkReadHandler() for each case in which an RTP packet could be discarded before being stored in fReorderingBuffer. None of my RTP packets seem to be invalidated before being stored. 3. Add printf to MultiframedRTPSource::networkReadHandler() to show if fReorderingBuffer->storePacket fails. No failures reported. 4. Comment out using MPEG2TransportStreamFramer in MediaSession.cpp. It doesn't seem to make much of a difference whether or not MPEG2TransportStreamFramer is used or if the RTPSource goes directly to FileSink. 5. In ReorderingPacketBuffer::storePacket() print debug statement any time we see a late packet. No late packets noted. I also notice that the RTP packets sent to me all have an RTPTime of 0. Ethereal I/O graph shows my data rate to be about 4 Mbps. My host computer is on a private network with 1 other computer and 2 embedded computers which are used as RTSP servers or clients. I can't see how it's possible for network packets to be 30 - 40 seconds late on this private network, but I see that LIVE acts as though this is happening. Can you point me to a part of the code to look at for these RTP packet losses? Any ideas for me? Thank you! Xochitl _______________________________________________ live-devel mailing list live-devel at lists.live555.com http://lists.live555.com/mailman/listinfo/live-devel
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