问题描述
我需要实现和 RTSP 服务器，它将连接到其他 RTSP 流并将其传输到客户端，因此只是一个虚拟的 rtsp 转发器（端口转发或其他 NAT 解决方案在这里不是一个选项）。我尝试使用 gstreamer rtsp 服务器并将下一个启动字符串提供给它：
rtspsrc location=rtsp://127.0.0.1/axis-media/media.amp ! rtph264depay ! rtph264pay name=pay0 pt=96
使用 GST_DEBUG=3 运行我的服务器会得到一个结果：
0:00:11.702615217 8269 0x7fe5ec005d40 WARN rtspsrc gstrtspsrc.c:5423:gst_rtspsrc_reconnect:<rtspsrc0> warning: Could not receive any UDP packets for 5.0000 seconds,maybe your firewall is blocking it. retrying using a tcp connection.
0:00:11.702773239 8269 0x7fe5f800de30 WARN rtspmedia rtsp-media.c:2735:default_handle_message: 0x7fe5f80301c0: got warning Could not read from resource. (gstrtspsrc.c(5423): gst_rtspsrc_reconnect
(): /GstPipeline:media-pipeline/GstBin:bin0/GstRTSPSrc:rtspsrc0:
Could not receive any UDP packets for 5.0000 seconds,maybe your firewall is blocking it. retrying using a tcp connection.)
client (127.0.0.1) connected
connected sessions: 0
clients connected: 2
0:00:11.891762200 8269 0x7fe5ec005d40 FIXME rtpjitterbuffer gstrtpjitterbuffer.c:1535:gst_jitter_buffer_sink_parse_caps:<rtpjitterbuffer0> Unsupported timestamp reference clock
0:00:11.891774847 8269 0x7fe5ec005d40 FIXME rtpjitterbuffer gstrtpjitterbuffer.c:1543:gst_jitter_buffer_sink_parse_caps:<rtpjitterbuffer0> Unsupported media clock
0:00:11.910516807 8269 0x562e7e5fa680 FIXME rtspmedia rtsp-media.c:3835:gst_rtsp_media_suspend: suspend for dynamic pipelines needs fixing
0:00:11.910893371 8269 0x7fe5ec005d40 WARN rtspsrc gstrtspsrc.c:5915:gst_rtsp_src_receive_response:<rtspsrc0> receive interrupted
0:00:11.910903309 8269 0x7fe5ec005d40 WARN rtspsrc gstrtspsrc.c:8242:gst_rtspsrc_pause:<rtspsrc0> PAUSE interrupted
0:00:11.911159840 8269 0x7fe5ec005d40 WARN rtspsrc gstrtspsrc.c:5995:gst_rtspsrc_try_send:<rtspsrc0> send interrupted
0:00:11.911166353 8269 0x7fe5ec005d40 WARN rtspsrc gstrtspsrc.c:7669:gst_rtspsrc_close:<rtspsrc0> TEARDOWN interrupted
client (127.0.0.1) closed
0:00:12.052945700 8269 0x7fe5f00171e0 FIXME default gstutils.c:3981:gst_pad_create_stream_id_internal:<fakesrc1:src> Creating random stream-id,consider implementing a deterministic way of creating a stream-id
connected sessions: 0
clients connected: 1
connected sessions: 0
clients connected: 1
0:00:17.170566970 8269 0x7fe5f00170a0 WARN rtspsrc gstrtspsrc.c:5423:gst_rtspsrc_reconnect:<rtspsrc1> warning: Could not receive any UDP packets for 5.0000 seconds,maybe your firewall is blocking it. retrying using a tcp connection.
0:00:17.170649535 8269 0x7fe5f00172d0 WARN rtspmedia rtsp-media.c:2735:default_handle_message: 0x7fe5f8030390: got warning Could not read from resource. (gstrtspsrc.c(5423): gst_rtspsrc_reconnect
(): /GstPipeline:media-pipeline/GstBin:bin1/GstRTSPSrc:rtspsrc1:
Could not receive any UDP packets for 5.0000 seconds,maybe your firewall is blocking it. retrying using a tcp connection.)
0:00:17.425226083 8269 0x7fe5f00170a0 FIXME rtpjitterbuffer gstrtpjitterbuffer.c:1535:gst_jitter_buffer_sink_parse_caps:<rtpjitterbuffer1> Unsupported timestamp reference clock
0:00:17.425254552 8269 0x7fe5f00170a0 FIXME rtpjitterbuffer gstrtpjitterbuffer.c:1543:gst_jitter_buffer_sink_parse_caps:<rtpjitterbuffer1> Unsupported media clock
0:00:17.434697010 8269 0x562e7e5fa680 FIXME rtspmedia rtsp-media.c:3835:gst_rtsp_media_suspend: suspend for dynamic pipelines needs fixing
0:00:17.435554904 8269 0x7fe5f00170a0 WARN rtspsrc gstrtspsrc.c:5915:gst_rtsp_src_receive_response:<rtspsrc1> receive interrupted
0:00:17.435578828 8269 0x7fe5f00170a0 WARN rtspsrc gstrtspsrc.c:8242:gst_rtspsrc_pause:<rtspsrc1> PAUSE interrupted
0:00:17.436368348 8269 0x7fe5f00170a0 WARN rtspsrc gstrtspsrc.c:5995:gst_rtspsrc_try_send:<rtspsrc1> send interrupted
0:00:17.436393638 8269 0x7fe5f00170a0 WARN rtspsrc gstrtspsrc.c:7669:gst_rtspsrc_close:<rtspsrc1> TEARDOWN interrupted
0:00:17.488454868 8269 0x562e7e5fa680 ERROR rtspclient rtsp-client.c:1707:handle_play_request: client 0x7fe5fc009250: media not found
0:00:17.489046867 8269 0x562e7e5fa680 ERROR rtspclient rtsp-client.c:1259:handle_teardown_request: client 0x7fe5fc009250: no media for uri
client (127.0.0.1) closed
client (127.0.0.1) connected
client (127.0.0.1) closed
0:00:17.722597026 8269 0x7fe5f00170f0 FIXME default gstutils.c:3981:gst_pad_create_stream_id_internal:<fakesrc2:src> Creating random stream-id,consider implementing a deterministic way of creating a stream-id
任何建议我做错了什么？
解决方法
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