[Live-devel] Why H264 annexB bytestream transported incompletely?

[Felix]

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

       
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