Using SRT protocol in Gstreamer

Question

1. Using the following 2 commands I can stream a videotestsrc source over SRT.

gst-launch-1.0 -v videotestsrc ! queue ! x264enc ! queue ! mpegtsmux alignment=7 ! identity silent=false ! queue leaky=downstream ! srtsink uri="srt://:8888" sync=false async=false

gst-launch-1.0 -v srtsrc uri="srt://127.0.0.1:8888" ! identity silent=false ! fakesink async=false

And play it in this way:

gst-play-1.0 srt://127.0.0.1:8888

2. Now I want to stream a rtsp source, and I get it in the following way:

gst-launch-1.0 rtspsrc location=rtsp://localhost:8554/main latency=100 ! queue ! rtph264depay ! h264parse ! avdec_h264 ! videoconvert ! videoscale ! video/x-raw,width=640,height=480 ! srtsink uri="srt://:8888" sync=false async=false

gst-launch-1.0 -v srtsrc uri="srt://127.0.0.1:8888" ! identity silent=false ! fakesink async=false

However, when I when to playback I have this error:

gst-play-1.0 srt://127.0.0.1:8888

Press 'k' to see a list of keyboard shortcuts.
Now playing srt://127.0.0.1:8888
Pipeline is live.
ERROR Could not determine type of stream. for srt://127.0.0.1:8888
ERROR debug information: ../subprojects/gstreamer/plugins/elements/gsttypefindelement.c(999): gst_type_find_element_chain_do_typefinding (): /GstPlayBin:playbin/GstURIDecodeBin:uridecodebin0/GstDecodeBin:decodebin0/GstTypeFindElement:typefind
Reached end of play list.

How can I solve it?