Not able to open rtsp stream with gstreamer version 1.18.0.
Getting error "No supported authentication protocol was found" ./gst-launch-1.0 rtspsrc location=rtsp://admin:admin@192.168.1.20:8551/PSIA/Streaming/channels/2?videoCodecType=H.264 ! rtph264depay ! avdec_h264 ! videoconvert ! xvimagesink 0:00:00.023943631 29448 0x13e7290 DEBUG rtspsrc gstrtspsrc.c:9318:gst_rtspsrc_uri_set_uri:<rtspsrc0> parsing URI 0:00:00.023991492 29448 0x13e7290 DEBUG rtspsrc gstrtspsrc.c:9325:gst_rtspsrc_uri_set_uri:<rtspsrc0> configuring URI 0:00:00.024010038 29448 0x13e7290 DEBUG rtspsrc gstrtspsrc.c:9341:gst_rtspsrc_uri_set_uri:<rtspsrc0> set uri: rtsp://admin:admin@192.168.1.20:8551/PSIA/Streaming/channels/2?videoCodecType=H.264 0:00:00.024025219 29448 0x13e7290 DEBUG rtspsrc gstrtspsrc.c:9343:gst_rtspsrc_uri_set_uri:<rtspsrc0> request uri is: rtsp://192.168.1.20:8551/PSIA/Streaming/channels/2?videoCodecType=H.264 Setting pipeline to PAUSED ... 0:00:00.037557763 29448 0x13e7290 DEBUG rtspsrc gstrtspsrc.c:9070:gst_rtspsrc_start:<rtspsrc0> starting 0:00:00.037745036 29448 0x13e7290 DEBUG rtspsrc gstrtspsrc.c:6026:gst_rtspsrc_loop_send_cmd:<rtspsrc0> sending cmd OPEN 0:00:00.037761472 29448 0x13e7290 DEBUG rtspsrc gstrtspsrc.c:6064:gst_rtspsrc_loop_send_cmd:<rtspsrc0> not interrupting busy cmd unknown Pipeline is live and does not need PREROLL ... 0:00:00.037947783 29448 0x13c2a30 DEBUG rtspsrc gstrtspsrc.c:9017:gst_rtspsrc_thread:<rtspsrc0> got command OPEN 0:00:00.037978989 29448 0x13c2a30 DEBUG rtspsrc gstrtspsrc.c:5183:gst_rtspsrc_connection_flush:<rtspsrc0> set flushing 0 0:00:00.038001607 29448 0x13c2a30 DEBUG rtspsrc gstrtspsrc.c:5049:gst_rtsp_conninfo_connect:<rtspsrc0> creating connection (rtsp://admin:admin@192.168.1.20:8551/PSIA/Streaming/channels/2?videoCodecType=H.264)... Progress: (open) Opening Stream Pipeline is PREROLLED ... Prerolled, waiting for progress to finish... 0:00:00.038321214 29448 0x13c2a30 DEBUG rtspsrc gstrtspsrc.c:5060:gst_rtsp_conninfo_connect:<rtspsrc0> sanitized uri rtsp://192.168.1.20:8551/PSIA/Streaming/channels/2?videoCodecType=H.264 0:00:00.038368796 29448 0x13c2a30 DEBUG rtspsrc gstrtspsrc.c:5094:gst_rtsp_conninfo_connect:<rtspsrc0> connecting (rtsp://admin:admin@192.168.1.20:8551/PSIA/Streaming/channels/2?videoCodecType=H.264)... Progress: (connect) Connecting to rtsp://admin:admin@192.168.1.20:8551/PSIA/Streaming/channels/2?videoCodecType=H.264 0:00:00.048837085 29448 0x13c2a30 DEBUG rtspsrc gstrtspsrc.c:7872:gst_rtspsrc_retrieve_sdp:<rtspsrc0> create options... (async) 0:00:00.048873252 29448 0x13c2a30 DEBUG rtspsrc gstrtspsrc.c:7881:gst_rtspsrc_retrieve_sdp:<rtspsrc0> send options... 0:00:00.048946732 29448 0x13c2a30 DEBUG rtspsrc gstrtspsrc.c:528:default_before_send:<rtspsrc0> default handler 0:00:00.048965438 29448 0x13c2a30 DEBUG rtspsrc gstrtspsrc.c:528:default_before_send:<rtspsrc0> default handler 0:00:00.048976754 29448 0x13c2a30 DEBUG rtspsrc gstrtspsrc.c:6455:gst_rtspsrc_try_send:<rtspsrc0> sending message Progress: (open) Retrieving server options 0:00:00.055753992 29448 0x13c2a30 DEBUG rtspsrc gstrtspsrc.c:6357:gst_rtsp_src_receive_response:<rtspsrc0> received response message 0:00:00.055788635 29448 0x13c2a30 DEBUG rtspsrc gstrtspsrc.c:6376:gst_rtsp_src_receive_response:<rtspsrc0> got response message 200 0:00:00.055810469 29448 0x13c2a30 INFO rtspsrc gstrtspsrc.c:7894:gst_rtspsrc_retrieve_sdp:<rtspsrc0> Now using version: 1.0 0:00:00.055827966 29448 0x13c2a30 DEBUG rtspsrc gstrtspsrc.c:7901:gst_rtspsrc_retrieve_sdp:<rtspsrc0> create describe... 0:00:00.055842932 29448 0x13c2a30 DEBUG rtspsrc gstrtspsrc.c:7918:gst_rtspsrc_retrieve_sdp:<rtspsrc0> send describe... 0:00:00.055899555 29448 0x13c2a30 DEBUG rtspsrc gstrtspsrc.c:528:default_before_send:<rtspsrc0> default handler 0:00:00.055913554 29448 0x13c2a30 DEBUG rtspsrc gstrtspsrc.c:528:default_before_send:<rtspsrc0> default handler 0:00:00.055921643 29448 0x13c2a30 DEBUG rtspsrc gstrtspsrc.c:6455:gst_rtspsrc_try_send:<rtspsrc0> sending message Progress: (open) Retrieving media info 0:00:00.069467384 29448 0x13c2a30 DEBUG rtspsrc gstrtspsrc.c:6357:gst_rtsp_src_receive_response:<rtspsrc0> received response message 0:00:00.069515644 29448 0x13c2a30 DEBUG rtspsrc gstrtspsrc.c:6376:gst_rtsp_src_receive_response:<rtspsrc0> got response message 200 0:00:00.069678144 29448 0x13c2a30 DEBUG rtspsrc gstrtspsrc.c:7976:gst_rtspsrc_retrieve_sdp:<rtspsrc0> parse SDP... 0:00:00.069826253 29448 0x13c2a30 DEBUG rtspsrc gstrtspsrc.c:7591:gst_rtspsrc_parse_range:<rtspsrc0> parsed range npt=0- 0:00:00.069860006 29448 0x13c2a30 DEBUG rtspsrc gstrtspsrc.c:7604:gst_rtspsrc_parse_range:<rtspsrc0> range: type 0, min 0.000000 - type 2, max 0.000000 0:00:00.069873230 29448 0x13c2a30 DEBUG rtspsrc gstrtspsrc.c:7614:gst_rtspsrc_parse_range:<rtspsrc0> range: min 0:00:00.000000000 0:00:00.069885982 29448 0x13c2a30 DEBUG rtspsrc gstrtspsrc.c:7633:gst_rtspsrc_parse_range:<rtspsrc0> range: max 99:99:99.999999999 0:00:00.070012795 29448 0x13c2a30 DEBUG rtspsrc gstrtspsrc.c:2121:gst_rtspsrc_collect_payloads: mapping sdp session level attributes to caps 0:00:00.070041628 29448 0x13c2a30 DEBUG rtspsrc gstrtspsrc.c:2123:gst_rtspsrc_collect_payloads: mapping sdp media level attributes to caps 0:00:00.070057698 29448 0x13c2a30 DEBUG rtspsrc gstrtspsrc.c:2141:gst_rtspsrc_collect_payloads:<rtspsrc0> looking at 0 pt: 96 0:00:00.070119584 29448 0x13c2a30 DEBUG rtspsrc gstrtspsrc.c:2280:gst_rtspsrc_create_stream:<rtspsrc0> stream 0, (0x7f136c038c20) 0:00:00.070131043 29448 0x13c2a30 DEBUG rtspsrc gstrtspsrc.c:2281:gst_rtspsrc_create_stream:<rtspsrc0> port: 0 0:00:00.070142575 29448 0x13c2a30 DEBUG rtspsrc gstrtspsrc.c:2282:gst_rtspsrc_create_stream:<rtspsrc0> container: 0 0:00:00.070153780 29448 0x13c2a30 DEBUG rtspsrc gstrtspsrc.c:2283:gst_rtspsrc_create_stream:<rtspsrc0> control: track1 0:00:00.070170582 29448 0x13c2a30 DEBUG rtspsrc gstrtspsrc.c:2338:gst_rtspsrc_create_stream:<rtspsrc0> setup: rtsp://192.168.1.20:8551/PSIA/Streaming/channels/2/track1?videoCodecType=H.264 0:00:00.070198809 29448 0x13c2a30 DEBUG rtspsrc gstrtspsrc.c:507:default_select_stream:<rtspsrc0> default handler 0:00:00.070212684 29448 0x13c2a30 DEBUG rtspsrc gstrtspsrc.c:518:select_stream_accum: accum 1 0:00:00.070225674 29448 0x13c2a30 DEBUG rtspsrc gstrtspsrc.c:507:default_select_stream:<rtspsrc0> default handler 0:00:00.070242390 29448 0x13c2a30 DEBUG rtspsrc gstrtspsrc.c:7311:gst_rtspsrc_setup_streams_start:<rtspsrc0> doing setup of stream 0x7f136c038c20 with rtsp://192.168.1.20:8551/PSIA/Streaming/channels/2/track1?videoCodecType=H.264 0:00:00.070257888 29448 0x13c2a30 DEBUG rtspsrc gstrtspsrc.c:7326:gst_rtspsrc_setup_streams_start:<rtspsrc0> protocols = 0x7, protocol mask = 0x1 0:00:00.070269728 29448 0x13c2a30 DEBUG rtspsrc gstrtspsrc.c:6790:gst_rtspsrc_create_transports_string:<rtspsrc0> got transports (NULL) 0:00:00.070282453 29448 0x13c2a30 DEBUG rtspsrc gstrtspsrc.c:6820:gst_rtspsrc_create_transports_string:<rtspsrc0> adding UDP unicast 0:00:00.070295149 29448 0x13c2a30 DEBUG rtspsrc gstrtspsrc.c:6847:gst_rtspsrc_create_transports_string:<rtspsrc0> prepared transports RTP/AVP;unicast;client_port=%%u1-%%u2 0:00:00.070306202 29448 0x13c2a30 DEBUG rtspsrc gstrtspsrc.c:7341:gst_rtspsrc_setup_streams_start:<rtspsrc0> replace ports in RTP/AVP;unicast;client_port=%%u1-%%u2 0:00:00.071970268 29448 0x13c2a30 DEBUG rtspsrc gstrtspsrc.c:2537:gst_rtspsrc_alloc_udp_ports:<rtspsrc0> got RTP port 39939 0:00:00.071991632 29448 0x13c2a30 DEBUG rtspsrc gstrtspsrc.c:2545:gst_rtspsrc_alloc_udp_ports:<rtspsrc0> RTP port not even 0:00:00.072002264 29448 0x13c2a30 DEBUG rtspsrc gstrtspsrc.c:2547:gst_rtspsrc_alloc_udp_ports:<rtspsrc0> free RTP udpsrc 0:00:00.072081086 29448 0x13c2a30 DEBUG rtspsrc gstrtspsrc.c:2552:gst_rtspsrc_alloc_udp_ports:<rtspsrc0> retry 1 0:00:00.072302081 29448 0x13c2a30 DEBUG rtspsrc gstrtspsrc.c:2537:gst_rtspsrc_alloc_udp_ports:<rtspsrc0> got RTP port 39940 0:00:00.072470012 29448 0x13c2a30 DEBUG rtspsrc gstrtspsrc.c:2569:gst_rtspsrc_alloc_udp_ports:<rtspsrc0> starting RTCP on port 39941 0:00:00.072565017 29448 0x13c2a30 DEBUG rtspsrc gstrtspsrc.c:7352:gst_rtspsrc_setup_streams_start:<rtspsrc0> transport is now RTP/AVP;unicast;client_port=39940-39941 0:00:00.072609895 29448 0x13c2a30 DEBUG rtspsrc gstrtspsrc.c:528:default_before_send:<rtspsrc0> default handler 0:00:00.072623863 29448 0x13c2a30 DEBUG rtspsrc gstrtspsrc.c:528:default_before_send:<rtspsrc0> default handler 0:00:00.072634750 29448 0x13c2a30 DEBUG rtspsrc gstrtspsrc.c:6455:gst_rtspsrc_try_send:<rtspsrc0> sending message Progress: (request) SETUP stream 0 0:00:00.074462677 29448 0x13c2a30 DEBUG rtspsrc gstrtspsrc.c:6357:gst_rtsp_src_receive_response:<rtspsrc0> received response message 0:00:00.074489098 29448 0x13c2a30 DEBUG rtspsrc gstrtspsrc.c:6376:gst_rtsp_src_receive_response:<rtspsrc0> got response message 404 0:00:00.074508449 29448 0x13c2a30 WARN rtspsrc gstrtspsrc.c:6317:gst_rtspsrc_setup_auth:<rtspsrc0> error: No supported authentication protocol was found ERROR: from element /GstPipeline:pipeline0/GstRTSPSrc:rtspsrc0: Could not open resource for reading. Additional debug info: ../gst/rtsp/gstrtspsrc.c(6317): gst_rtspsrc_setup_auth (): /GstPipeline:pipeline0/GstRTSPSrc:rtspsrc0: No supported authentication protocol was found -- Sent from: http://gstreamer-devel.966125.n4.nabble.com/ _______________________________________________ gstreamer-devel mailing list [hidden email] https://lists.freedesktop.org/mailman/listinfo/gstreamer-devel |
Free forum by Nabble | Edit this page |