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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