rtspsrc error : No supported authentication protocol was found
Sheetal D Salunkhe
sheetal.salunkhe at nortekcontrol.com
Thu Jan 14 08:27:34 UTC 2021
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
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