<div>Hi:</div> <div> </div> <div> Thank you for answering my question!<BR> <BR> I checked the packet loss by using the "-Q" option to "openRTSP", and the statistics are below:<BR> <BR> begin_QOS_statistics<BR> server_availability 100<BR> stream_availability 100<BR> subsession video/H264<BR> num_packets_received 6153<BR> num_packets_lost 941<BR> elapsed_measurement_time 3.003171<BR> kBytes_received_total 2793.262000<BR> measurement_sampling_interval_ms
1000<BR> kbits_per_second_min 6510.232637<BR> kbits_per_second_ave 7440.833705<BR> kbits_per_second_max 8846.432576<BR> packet_loss_percentage_min 1.653747<BR> packet_loss_percentage_ave 13.264731<BR> packet_loss_percentage_max 18.069058<BR> inter_packet_gap_ms_min 0.038000<BR> inter_packet_gap_ms_ave 0.486910<BR> inter_packet_gap_ms_max 111.935000<BR> end_QOS_statistics</div> <div> <BR> <BR>1. On the server, I checked that all datas are correctly write to the socket,<BR> but on the client, less data can be read from
that socket. What causes data loss? <BR> <BR> The frame rate of my bytestream is fixed(0.02s), so I set durationTime as 20ms, and send one frame every 20ms.<BR> <BR> On the server, I just added my own H264VideoFileServerMediaSession and xH264VideoStreamFramer.<BR> On the client, I just added one sentence in MediaSession.cpp's MediaSession::lookupPayloadFormat():<BR> <BR> case 96: {temp = "H264"; freq = 90000; nCh = 1; break;}<BR> <BR> <BR>2. The num of bytes received on the client is not constant, it changes every time. It's so puzzling!<BR> <BR> My network is Ok, I used livemediaServer to transport a MPEG2/TS file to
openRTSP,<BR> and found no packet loss.</div> <div> </div> <div><BR>3.Increasing Socket send & receive buffer size from 50*1024 Bytes to 1000*1024 Bytes makes no difference.</div> <div> </div> <div> </div> <div>4.When I increase the duration from 20ms to 40ms, less packets lost. Why?</div><p> 
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