[Libav-user] RTP missed or jitter buffer full

[wolverin]

Hi, I'm trying to modify ffserver.c of the old version to use rtps server.
 
If I send an rtsp/rtp stream via  av_write_frame(pFmtCtxOut, ppktout) to rtsp-simple-server and watch it in VLC, then there are no errors, I install pts/dts myself in my source.
 
But if I send x264 encoded packets via udp protocol to a connection implemented so as not to use temporary files.
 
    AVDictionary * pDicStm = NULL;
    av_dict_set(&pDicStm, "rtsp_flags", "listen", 0);
    av_dict_set(&pDicStm, "protocol_whitelist", "tcp,udp,rtp", 0);
    av_dict_set(&pDicStm, "rtsp_transport", "udp", 0);
    av_dict_set(&pDicStm, "allowed_media_types", "video", 0);
    av_dict_set(&pDicStm, "max_delay", "5000000", 0);
    avformat_open_input(&pFmtCtxStm, rtsp_url, NULL, &pDicStm);
 
Then when reading from such a connection, I see errors
 
[rtsp @ ] max delay reached. need to consume packet
[rtsp @ ] RTP: missed 17 packets
[rtsp @ ] max delay reached. need to consume packet
[rtsp @ ] RTP: missed 10 packets
[rtsp @ ] jitter buffer full
[rtsp @ ] RTP: missed 50 packets
[rtsp @ ] max delay reached. need to consume packet
[rtsp @ ] RTP: missed 33 packets
[rtsp @ ] max delay reached. need to consume packet
[rtsp @ ] RTP: missed 1 packets
 
Increasing max_delay only increases the number of messages jitter buffer full
 
Is it possible to defeat packet loss somehow?
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