如何用GStreamer无转码重新推流RTSP H264流至UDP？

我希望无需转码，通过UDP发送RTSP的H264音视频流，但以下GStreamer管道无法正常工作：

gst-launch-1.0 -v rtspsrc location=rtsp://127.0.0.1:8554/steam  name=demux demux. ! queue ! rtph264depay ! h264parse ! mux. demux. ! queue ! rtpmp4gdepay ! aacparse ! queue ! mux. mpegtsmux name=mux alignment=7 ! udpsink host=192.168.1.1 port=5002

运行日志如下：

...
/GstPipeline:pipeline0/GstRTSPSrc:demux/GstRtpBin:manager/GstRtpSsrcDemux:rtpssrcdemux1.GstPad:rtcp_src_560534233: caps = application/x-rtcp, ssrc=(uint)560534233
/GstPipeline:pipeline0/GstRTSPSrc:demux/GstRtpBin:manager/GstRtpJitterBuffer:rtpjitterbuffer1.GstPad:sink_rtcp: caps = application/x-rtcp, ssrc=(uint)560534233
/GstPipeline:pipeline0/GstRTSPSrc:demux/GstRtpBin:manager/GstRtpSsrcDemux:rtpssrcdemux0.GstPad:rtcp_src_164951109: caps = application/x-rtcp, ssrc=(uint)164951109
/GstPipeline:pipeline0/GstRTSPSrc:demux/GstRtpBin:manager/GstRtpJitterBuffer:rtpjitterbuffer0.GstPad:sink_rtcp: caps = application/x-rtcp, ssrc=(uint)164951109
/GstPipeline:pipeline0/GstRTSPSrc:demux/GstRtpBin:manager/GstRtpJitterBuffer:rtpjitterbuffer0: ts-offset = 119978681
/GstPipeline:pipeline0/GstRTSPSrc:demux/GstRtpBin:manager/GstRtpJitterBuffer:rtpjitterbuffer0: ts-offset = 80018388
/GstPipeline:pipeline0/GstRTSPSrc:demux/GstRtpBin:manager/GstRtpJitterBuffer:rtpjitterbuffer0: ts-offset = 119978931
/GstPipeline:pipeline0/GstRTSPSrc:demux/GstRtpBin:manager/GstRtpJitterBuffer:rtpjitterbuffer0: ts-offset = 79926621
/GstPipeline:pipeline0/GstRTSPSrc:demux/GstRtpBin:manager/GstRtpJitterBuffer:rtpjitterbuffer0: ts-offset = 119935374
0:00:00.0 / 99:99:99.

不过我成功实现了音视频流分开发送，以下两个管道可以正常工作：

gst-launch-1.0 rtspsrc location=rtsp://127.0.0.1:8554/steam ! rtph264depay ! h264parse  ! mpegtsmux alignment=7 ! queue ! udpsink host=192.168.1.1 port=5002 sync=false

gst-launch-1.0 rtspsrc location=rtsp://127.0.0.1:8554/steam ! rtpmp4gdepay ! aacparse  ! mpegtsmux alignment=7 ! queue ! udpsink host=192.168.1.1 port=5002 sync=false

内容的提问来源于stack exchange，提问作者Mike