cherishman2005 / rtc

webrtc websocket GCC NACK FEC
MIT License
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[ffmpeg][sdp @ 0x3b83ec0] max delay reached. need to consume packet #68

Open cherishman2005 opened 7 months ago

cherishman2005 commented 7 months ago
[sdp @ 0x3b83ec0] max delay reached. need to consume packet
[sdp @ 0x3b83ec0] RTP: missed 4015 packets
cherishman2005 commented 7 months ago

采用以下命令进行拉流转发,运行了几个小时,几乎没有出现丢包现象

ffmpeg -rtsp_transport tcp -i rtsp://192.168.1.168/0 -vcodec copy -f rtsp rtsp://192.168.1.28/11

原因:这是rtsp协议默认使用udp导致的问题,所以rtsp强制使用tcp方式可以一定程度避免丢包。

如果是拉rtsp转rtmp,命令是(没测试)

ffmpeg -rtsp_transport tcp -i rtsp://admin:password@192.168.1.11:554 -vcodec copy -f flv -an rtmp://localhost/live/tests