我正在尝试使用GStreamer流式传输RTSP视频。我正在使用的test-launch
示例gst-rtsp-server
。
服务器:
./test-launch "(videotestsrc ! x264enc ! h264parse ! rtph264pay)"
客户:VLC。
当我尝试rtsp://0.0.0.0:8554/test
在VLC中打开URI 时出现错误:
live555 debug: we will now try HTTP tunneling mode
live555 debug: connection error -115
live555 error: Failed to connect with rtsp://127.0.0.1:8554/test
core debug: no access_demux modules matched
core debug: creating access 'rtsp' location='127.0.0.1:8554/test', path='(null)'
core debug: looking for access module matching "rtsp": 25 candidates
core debug: net: connecting to 127.0.0.1 port 8554
core debug: connection succeeded (socket = 28)
access_realrtsp debug: rtsp connected
access_realrtsp warning: only real/helix rtsp servers supported for now
core debug: no access modules matched
core error: open of `rtsp://127.0.0.1:8554/test' failed
我试图用wget简单测试它并得到错误 503: Service Unavailable
gstreamer日志:
0:00:10.219925652 2772 0xcb4ca0 ERROR rtspclient rtsp-client.c:767:find_media: client 0xe140c0: can't prepare media
0:00:10.220158204 2772 0xcb4ca0 ERROR rtspclient rtsp-client.c:2283:handle_describe_request: client 0xe140c0: no media
我做错了什么?
流是来自相机吗?如果是的话,什么型号?看起来您刚刚在流中
—
输入
现在,我只是尝试使用
—
Alexey Markov
videotestsrc
,没有相机。当我运行服务器时,它会打印此URL,所以我认为它是正确的