ffmpeg rtp流错误: RTP:丢弃收到的旧数据包太迟

Question

我通过ffmpeg开始视频传输，如下所示：

ffmpeg -f video4linux2 -i /dev/video0 -vcodec libx264 -preset ultrafast -crf 20 -tune zerolatency -s 800x600 -r 25 -b:v 0.9M -sdp_file video.sdp -f rtp rtp://192.168.10.24:5010    

我是这样复制的：

ffplay -protocol_whitelist file,udp,rtp video.sdp    

一切都运行得很好。然后我中断传输，几秒钟后我恢复。ffplay不会立即开始重现，但会出现错误：

....
[sdp @ 0x6ebf80] RTP: dropping old packet received too lateB f=1/1
    Last message repeated 14 times
[sdp @ 0x6ebf80] RTP: dropping old packet received too lateB f=1/1
    Last message repeated 33 times
[sdp @ 0x6ebf80] RTP: dropping old packet received too lateB f=1/1
    Last message repeated 41 times
[sdp @ 0x6ebf80] RTP: dropping old packet received too lateB f=1/1
    Last message repeated 49 times
[sdp @ 0x6ebf80] RTP: dropping old packet received too lateB f=1/1
    Last message repeated 33 times
[sdp @ 0x6ebf80] RTP: dropping old packet received too lateB f=1/1
    Last message repeated 27 times
[sdp @ 0x6ebf80] RTP: dropping old packet received too lateB f=1/1
    Last message repeated 14 times
[sdp @ 0x6ebf80] RTP: dropping old packet received too lateB f=1/1
    Last message repeated 48 times
[sdp @ 0x6ebf80] RTP: dropping old packet received too lateB f=1/1
    Last message repeated 34 times
......    

一段时间后，播放恢复，但它太长了。有没有一种方法可以消除或最小化这种性质的错误的发生，当传入的流被挂起时，可以选择或其他什么吗？阅读手册ffmpeg没有任何有价值的关于这没有纳瑞尔....(

Answer 1

好的，我也遇到了同样的问题，并且花了一个小时的时间。深入研究了ffmpeg代码；在libavformat/rtpdec.c中

    if ((s->seq == 0 && !s->queue) || s->queue_size <= 1) {
    /* First packet, or no reordering */
    return rtp_parse_packet_internal(s, pkt, buf, len);
} else {
    uint16_t seq = AV_RB16(buf + 2);
    int16_t diff = seq - s->seq;
    if (diff < 0) {
        /* Packet older than the previously emitted one, drop */
        av_log(s->ic, AV_LOG_WARNING,
               "RTP: dropping old packet received too late\n");
        return -1;
    } else if (diff <= 1) {

它提到了no reordering。因此，我跟进了queue_size，在libavformat/rtsp.c中谈到了reordering_queue_size，并在文档https://ffmpeg.org/ffmpeg-protocols.html#rtsp中展示了

-reorder_queue_size

    Set number of packets to buffer for handling of reordered packets.

它不是显式的，但是如果您添加-reorder_queue_size 0，它就解决了这个问题，因为它会停止对数据包进行排序。我验证并解决了我的问题。我可以打断消息来源。2分钟后重新启动，它就能正常工作了

主要是因为作为udp和实时我不需要任何命令，它应该只播放通过

希望这也能解决你的问题