Problems with rtspsrc
Marc Murphy
marcmltd at marcm.co.uk
Fri Feb 25 07:12:17 PST 2011
Hi,
I am using gst-plugins-good-0.10.25 and trying to add functionality to rtspsrc plugin so I can react to a server MULTIPLE_CHOICES (300) response.
I am successfully trapping the message and extracting the redirect uri.
It is all going wrong when I attempt to force the redirect...... from what I understand the following is happening;
The pipeline is constructed using gst-launch e.g.
gst-launch -v rtspsrc location=rtsp://x.x.x.x/video ! \
rtpmp2tdepay ! playpump ! decode (playpump and decode are just symbolic of the rest of the pipeline)
The startup works OK and I get the negotiation as should happen, except for when there is a 300 response.
I have traced the path through the rtspsrc and I see the following path
gst_rtspsrc_change_state -> gst_rtspsrc_open -> gst_rtspsrc_retrieve_sdp
gst_rtspsrc_open_from_sdp -> gst_rtspsrc_send
I have modified gst_rtspsrc_send to trap the error code
case GST_RTSP_STS_MULTIPLE_CHOICES:
{
gchar *new_location;
GST_INFO("got redirection");
/* if we don't have a Location Header, we must error */
if (gst_rtsp_message_get_header (response, GST_RTSP_HDR_LOCATION,
&new_location, 0) < 0)
{
GST_INFO("Ooops no Location Header");
}
GST_INFO("Source Address %p", src);
src->addr = g_strdup (new_location);
GST_INFO("Redirect to [ %s ]", src->addr);
src->need_redirect = TRUE;
src->state = GST_RTSP_STATE_INIT;
// res = GST_RTSP_OK;
}
Then in gst_rtspsrc_open I trap the src->need_redirect
open_failed:
{
GST_WARNING_OBJECT (src, "can't setup streaming from sdp");
GST_RTSP_STATE_UNLOCK (src);
if ( src->need_redirect )
{
//Lets emit a message to the playbin.....
GstStructure *messageStruct;
GstMessage *message;
gchar * uri = g_strdup (src->addr);
messageStruct = gst_structure_new ("redirect_message",
"uri", G_TYPE_STRING, uri,
NULL);
message = gst_message_new_element (GST_OBJECT (src), messageStruct);
gst_element_post_message (GST_ELEMENT(src), message);
printf("Going to redirect to [ %s ]\n", uri);
gst_rtspsrc_close(src);
//We are being asked to redirect.
gst_rtspsrc_uri_set_uri (GST_URI_HANDLER (src),
src->addr);
goto retry;
}
The jump to retry goes back to the beginning of the gst_rtspsrc_open function. I would have thought that if I use the gst_rtspsrc_close(src); call before attempting to open with the new connection it would destroy all the current sockets to the RTSP server (RTP and RTCP ports).
Here is a session log;
Setting pipeline to PAUSED ...
gst_rtspsrc_change_state --- gst_rtspsrc_open
RTSP request message 0x7fa13918
request line:
method: 'OPTIONS'
uri: 'rtsp://x.x.x.x/video'
version: '1.0'
headers:
body:
RTSP response message 0x7fa13934
status line:
code: '200'
reason: 'OK'
version: '1.0'
headers:
key: 'CSeq', value: '1'
key: 'Public', value: 'OPTIONS'
key: 'Public', value: 'DESCRIBE'
key: 'Public', value: 'SETUP'
key: 'Public', value: 'PLAY'
key: 'Public', value: 'PAUSE'
key: 'Public', value: 'GET_PARAMETER'
key: 'Public', value: 'TEARDOWN'
key: 'Server', value: 'Kasenna MediaBase version 9.0'
key: 'Date', value: 'Fri, 25 Feb 2011 14:30:42 GMT'
body: length 0
RTSP request message 0x7fa13918
request line:
method: 'DESCRIBE'
uri: 'rtsp://x.x.x.x/video'
version: '1.0'
headers:
key: 'Accept', value: 'application/x-rtsp-mh, application/sdp'
body:
RTSP response message 0x7fa13934
status line:
code: '200'
reason: 'OK'
version: '1.0'
headers:
key: 'CSeq', value: '2'
key: 'Date', value: 'Fri, 25 Feb 2011 14:30:42 GMT'
key: 'Server', value: 'Kasenna MediaBase version 9.0'
key: 'Content-Type', value: 'application/x-rtsp-mh'
key: 'Content-Length', value: '445'
body: length 446
00000000 (0x4abb80): 3c 4d 65 64 69 61 44 65 73 63 72 69 70 74 69 6f <MediaDescriptio
00000010 (0x4abb90): 6e 3e 0a 3c 4d 65 64 69 61 43 6f 6e 74 61 69 6e n>.<MediaContain
00000020 (0x4abba0): 65 72 0a 4e 75 6d 53 74 72 65 61 6d 73 20 3d 20 er.NumStreams =
00000030 (0x4abbb0): 31 0a 3e 0a 3c 4d 65 64 69 61 53 74 72 65 61 6d 1.>.<MediaStream
00000040 (0x4abbc0): 73 3e 0a 3c 4d 65 64 69 61 53 74 72 65 61 6d 0a s>.<MediaStream.
00000050 (0x4abbd0): 53 74 72 65 61 6d 49 44 20 3d 20 30 0a 4d 69 6d StreamID = 0.Mim
00000060 (0x4abbe0): 65 54 79 70 65 20 3d 20 61 70 70 6c 69 63 61 74 eType = applicat
00000070 (0x4abbf0): 69 6f 6e 2f 78 2d 73 67 69 6d 62 0a 4d 61 78 42 ion/x-sgimb.MaxB
00000080 (0x4abc00): 69 74 52 61 74 65 20 3d 20 33 33 36 38 32 33 39 itRate = 3368239
00000090 (0x4abc10): 0a 48 65 69 67 68 74 20 3d 20 35 37 36 0a 57 69 .Height = 576.Wi
000000a0 (0x4abc20): 64 74 68 20 3d 20 37 36 38 0a 46 72 61 6d 65 73 dth = 768.Frames
000000b0 (0x4abc30): 50 65 72 53 65 63 6f 6e 64 20 3d 20 32 35 2e 30 PerSecond = 25.0
000000c0 (0x4abc40): 30 30 30 30 30 0a 41 73 70 65 63 74 52 61 74 69 00000.AspectRati
000000d0 (0x4abc50): 6f 20 3d 20 30 2e 30 30 30 30 30 30 0a 46 72 61 o = 0.000000.Fra
000000e0 (0x4abc60): 6d 65 73 54 6f 49 46 72 61 6d 65 73 52 61 74 69 mesToIFramesRati
000000f0 (0x4abc70): 6f 20 3d 20 30 2e 30 30 30 30 30 30 0a 41 75 64 o = 0.000000.Aud
00000100 (0x4abc80): 69 6f 48 65 61 64 65 72 20 3d 20 30 0a 44 75 72 ioHeader = 0.Dur
00000110 (0x4abc90): 61 74 69 6f 6e 20 3d 20 34 39 36 35 37 39 32 30 ation = 49657920
00000120 (0x4abca0): 30 30 0a 53 74 61 72 74 54 69 6d 65 20 3d 20 30 00.StartTime = 0
00000130 (0x4abcb0): 0a 50 6c 61 79 4d 6f 64 65 73 20 3d 20 37 0a 4e .PlayModes = 7.N
00000140 (0x4abcc0): 75 6d 54 72 69 63 6b 53 70 65 65 64 20 3d 20 31 umTrickSpeed = 1
00000150 (0x4abcd0): 0a 54 72 69 63 6b 53 70 65 65 64 73 20 3d 20 31 .TrickSpeeds = 1
00000160 (0x4abce0): 32 0a 54 79 70 65 53 70 65 63 69 66 69 63 44 61 2.TypeSpecificDa
00000170 (0x4abcf0): 74 61 20 3d 20 4d 50 45 47 2d 32 0a 56 69 64 65 ta = MPEG-2.Vide
00000180 (0x4abd00): 6f 50 69 64 20 3d 20 30 0a 41 75 64 69 6f 50 69 oPid = 0.AudioPi
00000190 (0x4abd10): 64 20 3d 20 30 0a 3c 2f 4d 65 64 69 61 53 74 72 d = 0.</MediaStr
000001a0 (0x4abd20): 65 61 6d 73 3e 0a 3c 2f 4d 65 64 69 61 44 65 73 eams>.</MediaDes
000001b0 (0x4abd30): 63 72 69 70 74 69 6f 6e 3e 0d 0a 0d 0a 00 cription>.....
RTSP request message 0x7fa13768
request line:
method: 'SETUP'
uri: 'rtsp://x.x.x.x/video/'
version: '1.0'
headers:
key: 'Transport', value: 'RTP/AVP/UDP;unicast;client_port=8254-8255'
key: 'User-Agent', value: 'GSTREAMER_USER_AGENT_KA'
body:
RTSP response message 0x7fa13784
status line:
code: '300'
reason: 'Multiple Choices'
version: '1.0'
headers:
key: 'CSeq', value: '3'
key: 'Location', value: 'rtsp://x.x.x.newx/video'
key: 'Connection', value: 'close'
key: 'Date', value: 'Fri, 25 Feb 2011 14:30:42 GMT'
body: length 0
Going to redirect to [ rtsp://x.x.x.newx/video ]
make sure we're not flushing anymore
close connections
cleanup
(gst-launch-0.10:447): GStreamer-WARNING **: Element udpsrc0 is not in bin rtspsrc0
(gst-launch-0.10:447): GStreamer-WARNING **: Element udpsrc1 is not in bin rtspsrc0
RTSP request message 0x7fa13918
request line:
method: 'OPTIONS'
uri: 'rtsp://x.x.x.newx/video'
version: '1.0'
headers:
body:
RTSP response message 0x7fa13934
status line:
code: '200'
reason: 'OK'
version: '1.0'
headers:
key: 'CSeq', value: '1'
key: 'Public', value: 'OPTIONS'
key: 'Public', value: 'DESCRIBE'
key: 'Public', value: 'SETUP'
key: 'Public', value: 'PLAY'
key: 'Public', value: 'PAUSE'
key: 'Public', value: 'GET_PARAMETER'
key: 'Public', value: 'TEARDOWN'
key: 'Server', value: 'Kasenna MediaBase version 9.0'
key: 'Date', value: 'Fri, 25 Feb 2011 14:30:42 GMT'
body: length 0
RTSP request message 0x7fa13918
request line:
method: 'DESCRIBE'
uri: 'rtsp://x.x.x.newx/video'
version: '1.0'
headers:
key: 'Accept', value: 'application/x-rtsp-mh, application/sdp'
body:
RTSP response message 0x7fa13934
status line:
code: '200'
reason: 'OK'
version: '1.0'
headers:
key: 'CSeq', value: '2'
key: 'Date', value: 'Fri, 25 Feb 2011 14:30:42 GMT'
key: 'Server', value: 'Kasenna MediaBase version 9.0'
key: 'Content-Type', value: 'application/x-rtsp-mh'
key: 'Content-Length', value: '445'
body: length 446
00000000 (0x4addf8): 3c 4d 65 64 69 61 44 65 73 63 72 69 70 74 69 6f <MediaDescriptio
00000010 (0x4ade08): 6e 3e 0a 3c 4d 65 64 69 61 43 6f 6e 74 61 69 6e n>.<MediaContain
00000020 (0x4ade18): 65 72 0a 4e 75 6d 53 74 72 65 61 6d 73 20 3d 20 er.NumStreams =
00000030 (0x4ade28): 31 0a 3e 0a 3c 4d 65 64 69 61 53 74 72 65 61 6d 1.>.<MediaStream
00000040 (0x4ade38): 73 3e 0a 3c 4d 65 64 69 61 53 74 72 65 61 6d 0a s>.<MediaStream.
00000050 (0x4ade48): 53 74 72 65 61 6d 49 44 20 3d 20 30 0a 4d 69 6d StreamID = 0.Mim
00000060 (0x4ade58): 65 54 79 70 65 20 3d 20 61 70 70 6c 69 63 61 74 eType = applicat
00000070 (0x4ade68): 69 6f 6e 2f 78 2d 73 67 69 6d 62 0a 4d 61 78 42 ion/x-sgimb.MaxB
00000080 (0x4ade78): 69 74 52 61 74 65 20 3d 20 33 33 36 38 32 33 39 itRate = 3368239
00000090 (0x4ade88): 0a 48 65 69 67 68 74 20 3d 20 35 37 36 0a 57 69 .Height = 576.Wi
000000a0 (0x4ade98): 64 74 68 20 3d 20 37 36 38 0a 46 72 61 6d 65 73 dth = 768.Frames
000000b0 (0x4adea8): 50 65 72 53 65 63 6f 6e 64 20 3d 20 32 35 2e 30 PerSecond = 25.0
000000c0 (0x4adeb8): 30 30 30 30 30 0a 41 73 70 65 63 74 52 61 74 69 00000.AspectRati
000000d0 (0x4adec8): 6f 20 3d 20 30 2e 30 30 30 30 30 30 0a 46 72 61 o = 0.000000.Fra
000000e0 (0x4aded8): 6d 65 73 54 6f 49 46 72 61 6d 65 73 52 61 74 69 mesToIFramesRati
000000f0 (0x4adee8): 6f 20 3d 20 30 2e 30 30 30 30 30 30 0a 41 75 64 o = 0.000000.Aud
00000100 (0x4adef8): 69 6f 48 65 61 64 65 72 20 3d 20 30 0a 44 75 72 ioHeader = 0.Dur
00000110 (0x4adf08): 61 74 69 6f 6e 20 3d 20 34 39 36 35 37 39 32 30 ation = 49657920
00000120 (0x4adf18): 30 30 0a 53 74 61 72 74 54 69 6d 65 20 3d 20 30 00.StartTime = 0
00000130 (0x4adf28): 0a 50 6c 61 79 4d 6f 64 65 73 20 3d 20 37 0a 4e .PlayModes = 7.N
00000140 (0x4adf38): 75 6d 54 72 69 63 6b 53 70 65 65 64 20 3d 20 31 umTrickSpeed = 1
00000150 (0x4adf48): 0a 54 72 69 63 6b 53 70 65 65 64 73 20 3d 20 31 .TrickSpeeds = 1
00000160 (0x4adf58): 32 0a 54 79 70 65 53 70 65 63 69 66 69 63 44 61 2.TypeSpecificDa
00000170 (0x4adf68): 74 61 20 3d 20 4d 50 45 47 2d 32 0a 56 69 64 65 ta = MPEG-2.Vide
00000180 (0x4adf78): 6f 50 69 64 20 3d 20 30 0a 41 75 64 69 6f 50 69 oPid = 0.AudioPi
00000190 (0x4adf88): 64 20 3d 20 30 0a 3c 2f 4d 65 64 69 61 53 74 72 d = 0.</MediaStr
000001a0 (0x4adf98): 65 61 6d 73 3e 0a 3c 2f 4d 65 64 69 61 44 65 73 eams>.</MediaDes
000001b0 (0x4adfa8): 63 72 69 70 74 69 6f 6e 3e 0d 0a 0d 0a 00 cription>.....
RTSP request message 0x7fa13768
request line:
method: 'SETUP'
uri: 'rtsp://x.x.x.newx/video/'
version: '1.0'
headers:
key: 'Transport', value: 'RTP/AVP/UDP;unicast;client_port=8254-8255'
key: 'User-Agent', value: 'GSTREAMER_USER_AGENT_KA'
body:
RTSP response message 0x7fa13784
status line:
code: '200'
reason: 'OK'
version: '1.0'
headers:
key: 'CSeq', value: '3'
key: 'Transport', value: 'RTP/AVP/UDP;unicast;source=62.142.123.24;client_port=8254-8255;server_port=60398-60399;ssrc=163c081b'
key: 'Session', value: '1293466192;timeout=60'
body: length 0
/GstPipeline:pipeline0/GstRTSPSrc:rtspsrc0/GstRtpBin:rtpbin0: latency = 2000
/GstPipeline:pipeline0/GstRTSPSrc:rtspsrc0/GstRtpBin:rtpbin0: buffer-mode = slave (1)
/GstPipeline:pipeline0/GstRTSPSrc:rtspsrc0/GstUDPSrc:udpsrc2: timeout = 5000000
Pipeline is live and does not need PREROLL ...
ERROR: from element /GstPipeline:pipeline0/GstRTSPSrc:rtspsrc0: Could not write to resource.
Additional debug info:
gstrtspsrc.c(5066): gst_rtspsrc_setup_streams (): /GstPipeline:pipeline0/GstRTSPSrc:rtspsrc0:
Could not send message. (Unknown error (1))
ERROR: pipeline doesn't want to preroll.
Setting pipeline to PAUSED ...
Setting pipeline to READY ...
make sure we're not flushing anymore
construct a control url
construct a control url
try aggregate
do TEARDOWN
RTSP request message 0x7fa14208
request line:
method: 'TEARDOWN'
uri: 'rtsp://x.x.x.newx/video'
version: '1.0'
headers:
key: 'User-Agent', value: 'GSTREAMER_USER_AGENT_KA'
body:
RTSP response message 0x7fa14224
status line:
code: '200'
reason: 'OK'
version: '1.0'
headers:
key: 'CSeq', value: '4'
key: 'Date', value: 'Fri, 25 Feb 2011 14:30:43 GMT'
body: length 0
parse result?
close connections
cleanup
Setting pipeline to NULL ...
Freeing pipeline ...
Caught SIGSEGV accessing address 0x2b2f5bd8
Its almost as if the old ports havent been closed or discarded. It will successfully re-negotiate the connection but not UDP data is coming through.
If I dont do the redirect and let the pipeline go through the state change
Setting pipeline to NULL ...
Freeing pipeline ...
I can successfully use the new redirect address. I dont understand the difference between the forced close and the close applied by the pipeline going from PLAYING to READY to NULL. I thought the rtspsrc would handle its own sources and sinks and connect to the rest of the pipeline when dataflow is active.
This has been driving me mental. If anyone can point me to how I can successfully debug or find out where the blockage in the pipeline is it would be fantastic. The target is an embedded platform not a PC.
Regards
Marc
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